AudioCodes MEDIAPACK VERSION 6.2 Product specifications

MediaPackTM Media Gateways
MediantTM Media Gateways
Multi‐Service Business Gateways
SIP CPE Release Notes Version 6.2
December 2010 Document #: LTRT‐26901 SIP Release Notes
Contents
Table of Contents
1 What's New in Release 6.2 ................................................................................. 9 1.1 Supported Hardware Platforms ..............................................................................10 1.1.1 New Hardware Platforms ........................................................................................ 10 1.1.1.1 MediaPack 1xx ........................................................................................ 10 1.1.1.2 Mediant 600 ............................................................................................. 10 1.1.1.3 Mediant 800 MSBG ................................................................................. 10 1.1.1.4 Mediant 1000 ........................................................................................... 11 1.1.1.5 Mediant 1000 MSBG ............................................................................... 11 1.1.1.6 Mediant 2000 ........................................................................................... 11 1.1.1.7 Mediant 3000 ........................................................................................... 11 1.1.2 Existing Hardware Platforms................................................................................... 12 1.1.2.1 MediaPack 1xx ........................................................................................ 12 1.1.2.2 Mediant 600 ............................................................................................. 12 1.1.2.3 Mediant 800 MSBG ................................................................................. 13 1.1.2.4 Mediant 1000 MSBG ............................................................................... 14 1.1.2.5 Mediant 1000 ........................................................................................... 16 1.1.2.6 Mediant 2000 ........................................................................................... 16 1.1.2.7 Mediant 3000 ........................................................................................... 17 1.1.3 Hardware Platforms Not Supported ........................................................................ 17 1.2 SIP New Features ..................................................................................................18 1.2.1 SIP General New Features ..................................................................................... 18 1.2.2 SIP Gateway New Features.................................................................................... 27 1.2.3 SIP SAS New Features ........................................................................................... 52 1.3 Session Border Controller New Features ...............................................................54 1.4 Media New Features ..............................................................................................63 1.5 PSTN New Features ..............................................................................................65 1.6 Networking New Features ......................................................................................67 1.7 Data-Router New Features ....................................................................................72 1.8 Security New Features ...........................................................................................75 1.9 Infrastructure New Features ...................................................................................76 1.10 Management and Provisioning New Features .......................................................77 1.10.1 Web New Features ................................................................................................. 77 1.10.2 SNMP New Features .............................................................................................. 79 1.10.3 CLI New Features ................................................................................................... 80 1.10.4 SSH New Features ................................................................................................. 81 1.10.5 Utilities New Features ............................................................................................. 83 1.11 New Parameters .....................................................................................................84 1.11.1 SIP New Parameters .............................................................................................. 84 1.11.1.1 SIP General Parameters ......................................................................... 84 1.11.1.2 SIP Gateway Parameters ........................................................................ 86 1.11.1.3 SIP SAS Parameters ............................................................................... 92 1.11.2 SBC New Parameters ............................................................................................. 93 1.11.3 Media New Parameters .......................................................................................... 95 1.11.4 PSTN New Parameters ........................................................................................... 96 1.11.5 Networking New Parameters .................................................................................. 99 1.11.6 Management and Provisioning New Parameters ................................................. 101 1.12 Modified Parameters ............................................................................................102 1.12.1 SIP Parameters ..................................................................................................... 102 1.12.1.1 SIP General Parameters ....................................................................... 102 1.12.1.2 SIP Gateway Parameters ...................................................................... 104 1.12.1.3 SIP SAS Parameters ............................................................................. 111 1.12.2 SBC Parameters ................................................................................................... 112 Version 6.2
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1.12.3 Networking Parameters ........................................................................................ 113 1.13 Obsolete Parameters ...........................................................................................114 2 Supported Features ........................................................................................ 115 2.1 SIP Gateway Features .........................................................................................115 2.1.1 Supported SIP Features ....................................................................................... 115 2.1.2 SIP Compliance Tables ........................................................................................ 119 2.1.2.1 SIP Functions ........................................................................................ 119 2.1.2.2 SIP Methods .......................................................................................... 119 2.1.2.3 SIP Headers .......................................................................................... 120 2.1.2.4 SDP Headers ......................................................................................... 122 2.1.2.5 SIP Responses ...................................................................................... 122 2.2 Supported PSTN-SIP Interworking Features .......................................................127 2.3 Supported IP Media Features ..............................................................................129 2.4 Supported Data-Router Features .........................................................................130 2.4.1 LAN Switching ....................................................................................................... 130 2.4.2 Routing .................................................................................................................. 130 2.4.3 Data Security......................................................................................................... 132 2.4.4 Supported RFC's ................................................................................................... 133 2.5 DSP Firmware Templates (Voice Codecs) ...........................................................134 2.5.1 MediaPack 1xx ...................................................................................................... 134 2.5.2 Mediant 600 and Mediant 1000 ............................................................................ 135 2.5.3 Mediant 800 MSBG ............................................................................................... 138 2.5.4 Mediant 1000 MSBG ............................................................................................. 139 2.5.5 Mediant 2000 ........................................................................................................ 142 2.5.6 Mediant 3000 ........................................................................................................ 143 2.5.6.1 DSP Template Mix Feature for Mediant 3000 ....................................... 145 3 Known Constraints......................................................................................... 147 3.1 SIP Constraints ....................................................................................................147 3.2 Media Constraints ................................................................................................149 3.3 PSTN Constraints .................................................................................................153 3.3.1 DS3 Constraints .................................................................................................... 154 3.3.2 SDH Constraints ................................................................................................... 155 3.3.3 SS7 Constraints .................................................................................................... 156 3.4 IP Media Constraints ............................................................................................158 3.5 Networking Constraints ........................................................................................160 3.6 Security Constraints .............................................................................................161 3.7 High Availability Constraints .................................................................................162 3.8 Infrastructure Constraints .....................................................................................164 3.9 Management Constraints .....................................................................................166 3.9.1 Web Constraints.................................................................................................... 166 3.9.2 SNMP Constraints................................................................................................. 171 3.9.3 CLI Constraints ..................................................................................................... 174 4 Resolved Constraints ..................................................................................... 175 4.1 SIP Resolved Constraints ....................................................................................175 4.2 Web Resolved Constraints ...................................................................................176 SIP Release Notes
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SIP Release Notes
Contents
List of Tables
Table 1-1: Available Mediant 800 MSBG Models ................................................................................. 11 Table 1-2: Mediant 1000 MSBG Voice I/O Module Hardware Revision Compatibility List ................... 15 Table 1-3: New SIP General Parameters for Release 6.2..................................................................... 84 Table 1-4: New SIP Gateway Parameters for Release 6.2 ................................................................... 86 Table 1-5: New SIP SAS Parameter for Release 6.2 ............................................................................ 92 Table 1-6: New SBC Parameters for Release 6.2 ................................................................................. 93 Table 1-7: New Media Parameters for Release 6.2 .............................................................................. 95 Table 1-8: New PSTN Parameters for Release 6.2............................................................................... 96 Table 1-9: New Networking Parameters for Release 6.2 ...................................................................... 99 Table 1-10: New Management and Provisioning Parameter for Release 6.2 ..................................... 101 Table 1-11: Modified SIP General Parameters for Release 6.2 .......................................................... 102 Table 1-12: Modified SIP Gateway Parameters for Release 6.2 ......................................................... 104 Table 1-13: Modified SIP SAS Parameters for Release 6.2 ................................................................ 111 Table 1-14: Modified SBC Parameter for Release 6.2 ........................................................................ 112 Table 1-15: Modified Networking Parameter for Release 6.2 ............................................................. 113 Table 1-16: Obsolete Parameters........................................................................................................ 114 Table 2-1: Supported IETF RFC’s ....................................................................................................... 116 Table 2-2: Supported SIP Functions.................................................................................................... 119 Table 2-3: Supported SIP Methods ..................................................................................................... 119 Table 2-4: Supported SIP Headers...................................................................................................... 120 Table 2-5: Supported SDP Headers .................................................................................................... 122 Table 2-6: Supported 1xx SIP Responses .......................................................................................... 123 Table 2-7: Supported 2xx SIP Responses .......................................................................................... 123 Table 2-8: Supported 3xx SIP Responses .......................................................................................... 123 Table 2-9: Supported 4xx SIP Responses .......................................................................................... 124 Table 2-10: Supported 5xx SIP Responses ........................................................................................ 126 Table 2-11: Supported 6xx SIP Responses ........................................................................................ 126 Table 2-12: DSP Firmware Templates for MediaPack Series ............................................................. 134 Table 2-13: DSP Firmware Templates for Mediant 600 and Mediant 1000 Analog (FXS/FXO)
Interfaces ............................................................................................................................................. 135 Table 2-14: DSP Firmware Templates for Mediant 600 and Mediant 1000 Digital Interfaces ............ 136 Table 2-15: DSP Firmware Templates for Mediant 1000 MPM Module .............................................. 137 Table 2-16: DSP Firmware Capabilities............................................................................................... 138 Table 2-17: DSP Firmware Templates for Mediant 1000 MSBG Analog (FXS/FXO) Interfaces ........ 139 Table 2-18: DSP Firmware Templates for Mediant 1000 MSBG Digital Interfaces ............................ 140 Table 2-19: DSP Firmware Templates for Mediant 1000 MSBG MPM Module .................................. 141 Table 2-20: DSP Firmware Templates for Mediant 2000 .................................................................... 142 Table 2-21: DSP Firmware Templates for Mediant 3000 .................................................................... 143 Table 2-22: DSP Firmware Templates for Mediant 3000 16E1/21T1 ................................................. 144 Table 2-23: Template Mix Feature Channel Capacity ......................................................................... 145 Version 6.2
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Reader’s Notes
SIP Release Notes
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Document #: LTRT-26901
SIP Release Notes
Notices
Notice
This document describes the Release 6.2 of AudioCodes products – MP-1xx, Mediant 600,
Mediant 800 MSBG, Mediant 1000 MSBG, Mediant 1000, Mediant 2000, and Mediant 3000.
Information contained in this document is believed to be accurate and reliable at the time of
printing. However, due to ongoing product improvements and revisions, AudioCodes cannot
guarantee accuracy of printed material after the Date Published nor can it accept responsibility
for errors or omissions. Updates to this document and other documents as well as software
files can be downloaded by registered customers at http://www.audiocodes.com/downloads.
© Copyright 2010 AudioCodes Ltd. All rights reserved.
This document is subject to change without notice.
Date Published: December-16-2010
Trademarks
AudioCodes, AC, AudioCoded, Ardito, CTI2, CTI², CTI Squared, HD VoIP, HD VoIP
Sounds Better, InTouch, IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open
Solutions Network, OSN, Stretto, TrunkPack, VMAS, VoicePacketizer, VoIPerfect,
VoIPerfectHD, What’s Inside Matters, Your Gateway To VoIP and 3GX are trademarks or
registered trademarks of AudioCodes Limited. All other products or trademarks are property
of their respective owners. Product specifications are subject to change without notice.
WEEE EU Directive
Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed
of with unsorted waste. Please contact your local recycling authority for disposal of this
product.
Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors,
Partners, and Resellers from whom the product was purchased. For Customer support for
products purchased directly from AudioCodes, contact support@audiocodes.com.
Abbreviations and Terminology
Each abbreviation, unless widely used, is spelled out in full when first used.
Version 6.2
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Related Documentation
Document Name
Product Reference Manual for SIP Gateways and Media Servers
MP-11x & MP-124 SIP Installation Manual
Mediant 600 SIP Installation Manual
Mediant 800 MSBG SIP Installation Manual
Mediant 1000 MSBG SIP Installation Manual
Mediant 1000 SIP Installation Manual
Mediant 2000 SIP Installation Manual
Mediant 3000 SIP Installation Manual
MP-11x & MP-124 SIP User’s Manual
Mediant 600 & Mediant 1000 SIP User’s Manual
Mediant 800 SIP User’s Manual
Mediant 1000 MSBG SIP User’s Manual
Mediant 2000 SIP User’s Manual
Mediant 3000 SIP User’s Manual
MSBG Series CLI Reference Guide
Note: Throughout this manual, unless otherwise specified, the term device refers to
the AudioCodes product.
SIP Release Notes
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Document #: LTRT-26901
SIP Release Notes
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1. What's New in Release 6.2
What's New in Release 6.2
This document describes the software and hardware release of Version 6.2 and is
applicable to the following AudioCodes products:
MediaPack 1xx series (MP-11x and MP-124)
Mediant 600
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 1000
Mediant 2000
Mediant 3000 series (Mediant 3000/TP-6310, Mediant 3000 HA/TP-6310, Mediant
3000/TP-8410, and Mediant 3000 HA/TP-8410)
Throughout this document, a table is used to indicate the product(s) and management
protocol(s) that are now also applicable to the feature. Only marked check boxes indicate
applicability. For example, the table below indicates that the feature is applicable to Mediant
1000 only and for the Web interface management protocol.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
-
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Note: This Release Notes describes only features that are new to the specified
devices. These features may already be supported on other devices (from
previous releases). Therefore, in the table matrix used to indicate product
applicability, only the products for which the feature is now supported are
marked. To check whether the feature is already supported on other devices,
refer to previous Release Notes or the User’s Manual.
Note: Some of the features described in this document are available on the device
only if the relevant software license key has been purchased from
AudioCodes and installed on the device. For a list of available software
license keys that can be purchased, please consult your AudioCodes sales
representative.
Note: For Mediant 800 MSBG and Mediant 1000 MSBG, open source software may
have been added and/or amended for this product. For further information,
please visit our website at: http://audiocodes.com/support or contact your
AudioCodes sales representative.
Version 6.2
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1.1
Supported Hardware Platforms
This section describes the supported hardware platforms in Release 6.2
1.1.1
New Hardware Platforms
This section describes the new hardware platforms introduced in Release 6.2.
1.1.1.1
MediaPack 1xx
No new hardware platforms.
1.1.1.2
Mediant 600
No new hardware platforms.
1.1.1.3
Mediant 800 MSBG
The Mediant 800 MSBG supports the following new hardware platforms in Release 6.2:
Optional WAN Interfaces:
•
•
Symmetric High-Speed Digital Subscriber Line (SHDSL): Single SHDSL port with
the following specifications:
♦
Four SHDSL WAN ports housed on a single R-J45 connector
♦
Conforms to ITU G.991.2 Annexes A, B, E, F and G SHDSL
♦
Up to 5,696 Kbps over a single wire pair
♦
Up to 22,784 Kbps over four wire pairs bonding, according to SHDSL.bis
(ITU G.991.2 Annexes F, G)
♦
EFM and ATM support
♦
Wetting current support on the CPE side, according to G991.2
♦
Supports both Central Office (CO) and CPE (wetting current on CO excluded)
♦
TC-PAM 16/32 Line Code
T1 WAN DSU/CSU:
IP over High-level Data Link Control (HDLC) and Point-to-Point Protocol
(PPP) encapsulations
♦
PPP password authentication protocol (PAP) and challenge-handshake
authentication protocol (CHAP) authentication
♦
Multiple IP interfaces on the T1 WAN interface
♦
E1/T1 copper wire line
SIP Release Notes
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SIP Release Notes
1. What's New in Release 6.2
The available customer-ordered hardware configurations are listed in the table below:
Table 1-1: Available Mediant 800 MSBG Models
WAN
FXO
BRI
LAN
GbE/FE
T1/E1
FXS
Power
over
Ethernet
OSN
Server
(CPU)
MP800-8S-12L-P
8
-
-
12 (4/8)
-
GbE
120W
-
MP800-1ET-2L
-
-
-
12 (4/8)
1
2 x T1
120W
-
MP800-1ET-12L-P-2T
-
-
-
12 (4/8)
1
GbE
120W
-
MP800-12S-12L-P-X1
12
-
-
12 (4/8)
-
GbE
120W
Atom
MP800-12O-12L-P-X1
-
12
-
12 (4/8)
-
GbE
120W
Atom
MP800-12S-12L-P
12
-
-
12 (4/8)
-
GbE
120W
-
MP800-4O-2L-X1
-
4
-
2 (2/0)
-
GbE
-
Atom
MP800-4S8O-2L-X1
4
8
-
2 (2/0)
-
GbE
-
Atom
MP800-4S4O-12L-PX1
4
4
-
12 (4/8)
-
GbE
50XW
Atom
MP800-4B-12L-P
-
-
4
12 (4/8)
-
GbE
120W
-
MP800-4B-12L-P-X1
-
-
4
12 (4/8)
-
GbE
120W
Atom
Model
New:
Existing:
1.1.1.4
Mediant 1000
No new hardware platforms.
1.1.1.5
Mediant 1000 MSBG
No new hardware platforms.
1.1.1.6
Mediant 2000
No new hardware platforms.
1.1.1.7
Mediant 3000
No new hardware platforms.
Version 6.2
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1.1.2
Existing Hardware Platforms
This section describes existing hardware platforms from the previous release that are also
supported in Release 6.2.
1.1.2.1
MediaPack 1xx
This release supports the following existing hardware platforms:
MP-11x combined FXS/FXO devices:
•
MP-114/FXS+FXO providing 2 FXS ports and 2 FXO ports
•
MP-118/FXS+FXO providing 4 FXS ports and 4 FXO ports
MP-11x/FXO devices:
•
MP-118/FXO providing 8 analog FXO interfaces
•
MP-114/FXO providing 4 analog FXO interfaces
MP-11x/FXS devices:
•
MP-118/FXS providing 8 analog FXS interfaces
•
MP-114/FXS providing 4 analog FXS interfaces
•
MP-112/FXS providing 2 analog FXS interfaces
MP-124/FXS providing 24 analog FXS interfaces
MP-124 with AC Power
MP-124 with DC power
1.1.2.2
Mediant 600
This release supports the following existing hardware platforms:
Up to 2 FXS modules (up to 8 FXS interfaces - RJ-11 ports)
Up to 2 FXO modules (up to 8 FXO interfaces - RJ-11 ports)
Up to 2 Digital TRUNKS modules (1 or 2 E1/T1/J1 PRI spans, including fractional
E1/T1 - RJ-48c ports)
Up to 2 BRI modules (4 to 8 ISDN Basic Rate Interfaces - RJ-45 ports, 4 ports per
module)
These interfaces are available in one of the following configurations:
1 x E1/T1 port (also Fractional E1/T1)
2 x E1/T1 ports
4 x BRI ports (supporting up to 8 voice calls)
8 x BRI ports (supporting up to 16 voice calls)
4 x BRI ports and 1 x E1/T1 port
4 x BRI ports and 4 x FXS ports
4 x BRI ports and 4 x FXO ports
4 x FXS ports and 1 x E1/T1 port
4 x FXO ports and 1 x E1/T1 port
SIP Release Notes
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SIP Release Notes
1.1.2.3
1. What's New in Release 6.2
Mediant 800 MSBG
This release supports the following existing hardware platforms:
Analog telephony interfaces, offering only FXS or only FXO interfaces, or a
combination thereof. For a list of available hardware configurations and models, refer
to Table 1-1.
Optional, analog Lifeline on FXS Port 1, maintaining PSTN connectivity upon power
failure. For the combined FXS/FXO configuration, one Lifeline is available; for the 12FXS configuration, up to three Lifelines are available.
One set of ISDN Basic Rate Interface (BRI) interfaces, providing four RJ-45 BRI S/T
ports (supporting up to 8 voice channels) for connecting ISDN terminal equipment
such as ISDN telephones. Each BRI port can be configured either as termination
equipment/user side (TE) or network termination/network side (NT). Up to eight TE
devices can be connected per BRI S/T port, by using an ISDN S-bus that provides
eight ISDN ports.
Optional, Open Solutions Network (OSN) server platform for hosting third-party
applications such as IP PBX, based on the following:
•
CPU: Intel™ Atom 1.6 GHz processor (Intel Atom N270)
•
Memory: 1 GB (or 2 GB for devices with 12 FXO ports)
•
Hard disk drive (HDD): SATA with minimum of 20 GB
•
OSN ports (on rear panel) includes the following:
♦
Three USB ports (Standard-A type) for connecting computer peripherals
(e.g., mouse and keyboard)
♦
15-pin DB-type VGA port for connecting a monitor
•
OSN LEDs (on front panel) for hard disk drive (HDD) and link activity
•
Operating System – Microsoft Windows and Linux distributions (for certified
support, please contact AudioCodes)
Single WAN port (10/100/1000Base-T copper).
Twelve Ethernet LAN ports, supporting half- and full-duplex modes, auto-negotiation,
and straight or crossover cable detection:
•
Up to four RJ-45 10/100/1000Base-T (Gigabit) ports
•
Eight RJ-45 10/100Base-TX (Fast Ethernet) ports
Power-over-Ethernet (PoE) supported on all LAN ports, complying with IEEE 802.3af2003. The maximum wattage per port is 15.4W, providing a total wattage of 120W for
all ports.
Front-panel LEDs providing operating status of FXS/FXO interfaces, LAN interfaces,
WAN interface, PoE, OSN, and power supply.
Three-prong AC supply entry for AC power (standard electrical outlet) - single,
universal 90-260 VAC.
Protective earthing screws for grounding.
Desktop or 19-inch rack mounting (using external mounting brackets).
Version 6.2
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1.1.2.4
Mediant 1000 MSBG
This release supports the following existing hardware platforms:
Mediant 1000 and Mediant 1000B chassis. The Mediant 1000B chassis is based on
the incumbent Mediant 1000 chassis, but designed to support Advanced Mezzanine
Card (AMC/AdvancedMC) form-factor modules. This chassis provides eight AMC slots
on its rear panel for housing single and mid-sized AMC modules. This chassis hosts
the CRMX module (instead of the CMX) for supporting both VoIP Gateway and MSBG
data-routing functionalities. The Mediant 1000B also provides support for the regular
Mediant 1000 telephony modules (i.e., FXS, FXO, and digital PSTN).
Note: The AMC chassis slots must only be housed with AMC modules approved
and homologated by AudioCodes – ensure not to use these slots for AMC
modules and configurations that have not been approved.
CRMX module, providing three LAN 10/100/1000Base-T ports and one WAN port in
one of the following configurations (customer ordered):
•
CRMX-C module: RJ-45 port (4-twisted pair copper cabling) providing 1 Gigabit
Ethernet (GbE) interface for connection to the Internet
•
CRMX-S module: 1000Base-SX optical fiber port (multi-mode fiber)
•
CRMS-L module: 1000Base-LX optical fiber port (single-mode fiber)
•
CRMX-T: RJ-48c (2-twisted pairs copper cabling) Data Service Unit/Channel
Service Unit (DSU/CSU) T1 WAN port, for connecting to a T1 line
•
CRMX-SD module: Symmetric High-Speed Digital Subscriber Line (SHDSL). This
module provides 3 LAN ports and 1 SHDSL port. The SHDSL port has 4 wirepairs, and therefore, supports 4 SHDSL ports on a single physical connector.
SHDSL port specifications:
♦
Conforms to ITU G.991.2 Annexes A, B, E, F and G SHDSL
♦
Up to 5,696 Kbps over a single wire pair
♦
Up to 22,784 Kbps over four wire pairs bonding, according to SHDSL.bis
(ITU G.991.2 Annexes F, G)
♦
EFM and ATM support
♦
Wetting current support on the CPE side, according to G991.2
♦
Supports both Central Office (CO) and CPE (wetting current on CO excluded)
♦
TC-PAM 16/32 Line Code
Up to 6 FXS modules, where each module provides four RJ-11 ports. Therefore, the
device can support up to 24 FXS interfaces (six modules * 4 ports per module).
Up to 6 FXO modules, where each module provides four RJ-11 ports. Therefore, the
device can support up to 24 FXO ground- or loop-start signaling interfaces (six
modules * 4 ports per module).
Up to 4 Digital E1/T1 TRUNKS modules (1 or 2 E1/T1/J1 PRI spans per module - RJ48c ports).
Up to 5 BRI modules (up to 20 ISDN Basic Rate Interface interfaces - 4 ports per
module).
Up to 3 Media Processing modules (MPM), for media processing such as
announcements and conferencing
SIP Release Notes
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Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
Open Solutions Network (OSN) server platform, available in one of the following
models:
•
OSN Ver. 1 (OSN1), based on Intel™ Celeron™ 600 MHz processor (for thirdparty applications such as IP PBX)
•
OSN Ver. 2 (OSN2), based on Intel™ Pentium™ M 1.4 GHz processor (for thirdparty applications such as IP PBX):
♦
Integrated BroadSoft PacketSmart agent
•
OSN Ver 3 (OSN3) – supported only on the Mediant 1000B chassis, offering a
high performance CPU, targeted for customer’s requiring applications:
♦
CPU: Intel® Core™ 2 Duo 1.5 GHz processors L7400 with Intel 3100
Chipset (64-bit)
♦
RAM Memory: 2 G or 4 G DDR2 with ECC
♦
Storage: Single or Dual hard-disk drive of 80 G SATA
♦
Bus/Chipset: 64 Bit
♦
L2 Cache: 2 M
♦
Interfaces: Gigabit Ethernet, supporting automatic detection and switching
between 10Base-T, 100Base-TX and 1000Base-T data transmission (AutoNegotiation). Auto-wire switching for crossed cables is also supported (AutoMDI/X); USB 2.0 via Connection Module; RS-232 COM
1.1.2.4.1 Module Hardware Revision Compatibility List
The Voice I/O module Hardware Revisions that are compatible with the Mediant 1000
MSBG chassis is listed in the table below:
Table 1-2: Mediant 1000 MSBG Voice I/O Module Hardware Revision Compatibility List
Item No.
Description
H/W Revision
FASB00334
M1K-SMX-1A1V1 Quad FXS w/ Life-Line Module
C08
FASB00335
M1K-SMX-1A1V1 Dual FXS w/ Life-Line Module
C08
FASB00398
M1K TMX-1A1V1 Dual trunks w/ life-line module
C08
FASB00399
M1K TMX-1A1V1 Single trunks w/ life-line module
C08
FASB00510
M1K-OMX-S 4 Indoor Ports With GS Rev.A1
C05
FASB00511
M1K-CFMX-1 Conference Module Rev A1v1
C06
FASB00520
M1K-OMX-S 4 Outdoor Ports W/O GS Rev. A1
C04
FASB00582
M1K-BMX-4A1
C03
GTPM00046
M1K-VM-2FXS
P03
GTPM00050
M1K-VM-1SPAN
P03
GTPM00052
M1K-VM-2SPAN
P03
GTPM00056
M1K-VM-4FXS
P03
GTPM00125
M1K-VM-4FXO-LS
P02
GTPM00126
M1K-VM-4FXO-GS/LS
P02
GTPM00127
M1K-M-CONF
P02
GTPM00174
M1K-VM-4BRI
P02
Version 6.2
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CPE SIP Products
Item No.
Description
H/W Revision
FASU00587
CRMX-T (LAN 1, 2, 3, Dual T1 WAN DSU/CSU Trunk)
P1.2
FASU00557
CRMX-L (LAN 1, 2, 3, GE WAN 1000Base-LX)
P1.5
FASU00556
CRMX-C (LAN 1, 2, 3, GE WAN copper Ethernet)
FASU00557
CRMX-S (LAN 1, 2, 3, GE WAN 1000Base-SX)
FASU00636
CRMX-SD (LAN 1, 2, 3, SHDSL WAN)
1.1.2.5
P2
P1.5
No Revision
Constraint
Mediant 1000
This release supports the following existing hardware platforms:
Up to 6 FXS modules (up to 24 FXS interfaces - RJ-11 ports, 4 ports per module)
Up to 6 FXO modules (up to 24 FXO interfaces, Ground-Start or Loop-Start signaling RJ-11 ports, 4 ports per module)
Up to 4 Digital TRUNKS modules (1, 2, or 4 E1/T1/J1 PRI spans - RJ-48c ports)
Up to 5 BRI modules (up to 20 ISDN Basic Rate Interface channels - 4 ports per
module)
Up to 3 Media Processing modules (MPM - for media processing such as
announcements and conferencing)
OSN1 - Open Solutions Network Ver. 1, Intel™ Celeron™ 600 MHz, for third-party
applications such as IP PBX
OSN2 - Open Solutions Network Ver. 2, Intel™ Pentium™ M 1.4 GHz, for third-party
applications such as IP PBX
The Mediant 1000 can be deployed with a combination of the modules listed above.
1.1.2.6
Mediant 2000
Mediant 2000 1U-chassis, hosting a TP-1610 cPCI blade supporting up to 16 E1/T1 spans.
SIP Release Notes
16
Document #: LTRT-26901
SIP Release Notes
1.1.2.7
1. What's New in Release 6.2
Mediant 3000
This release supports the following existing hardware platforms:
Mediant 3000 hosting a single TP-6310 blade, providing SONET/SDH or T3 PSTN
interfaces.
Mediant 3000 hosting two TP-6310 blades for 1+1 HA, providing SONET/SDH or T3
PSTN interfaces.
Mediant 3000 hosting a single TP-8410 blade, providing 16 E1/ 21 T1 PSTN
interfaces.
Mediant 3000 hosting a single TP-8410 blade, providing up to 63 E1/ 84 T1 PSTN
interfaces.
Mediant 3000 hosting two TP-8410 blades for 1+1 High Availability (HA), providing up
to 16 E1/ 21 T1 PSTN interfaces.
Mediant 3000 hosting two TP-8410 blades for 1+1 HA, providing up to 63 E1/ 84 T1
PSTN interfaces.
Mediant 3000 hosting a single TP-8410 blade providing 16 E1/ 21 T1 PSTN interfaces
with an integrated CPU (Intel Pentium) blade (M3K-ICPU-1), hosting third-party
applications (such as SS7 GWC).
Mediant 3000 hosting a single TP-8410 blade providing up to 63 E1/ 84 T1 PSTN
interfaces with an integrated CPU (Intel Pentium) blade (M3K-ICPU-1), hosting thirdparty applications (such as SS7 GWC).
1.1.3
Hardware Platforms Not Supported
The following hardware platforms are not supported in Release 6.2:
IPmedia 2000
IPmedia 3000
Version 6.2
17
December 2010
CPE SIP Products
1.2
SIP New Features
This section lists the new SIP features in Release 6.2.
1.2.1
SIP General New Features
The device supports the following new SIP general (Gateway and SBC applications)
features:
1.
Maximum SIP Message Size:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports the configuration of the maximum size for each SIP message.
This allows the user to limit the size of the SIP message that is sent over the network.
The device rejects messages exceeding this user-defined size. The maximum SIP
message size is defined using the new parameter, MaxSIPMessageLength. The
maximum size can range between 1 and 50 Kbytes.
2.
Time Limit for Receipt of SIP Messages:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides a 30-second period (limit) for the receipt of a complete SIP
message on the device’s TCP socket. For example, if the device receives a SIP
message whose SIP Content-Length header is larger than the actual message body,
the TCP socket remains open and waits for the rest of the message body to arrive.
During this time, no other messages are accepted by the socket. However, if the
remaining message body does not arrive within 30 seconds, the device terminates the
session on the socket in order to allow new messages to arrive.
SIP Release Notes
18
Document #: LTRT-26901
SIP Release Notes
3.
1. What's New in Release 6.2
SDP Offer/Answer Negotiation upon Modified Session:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature enables the flexibility of ignoring a new SDP re-offer (from the media
negotiation perspective) in certain scenarios (such as session expires). According to
RFC 3264, once an SDP session is established, a new SDP offer is considered a new
offer only when the SDP origin value is incremented. In scenarios such as session
expires, SDP negotiation is irrelevant and thus, the origin field is not changed.
Even though some SIP devices don’t follow this behavior and don’t increment the
origin value even in scenarios where they want to re-negotiate, the device up till now
re-negotiated regardless of the origin field value. Since this caused problems in certain
scenarios, this feature resolves this by assuming that the remote party operates
according to RFC 3264, and in cases where the origin field is not incremented, the
device does not re-negotiate SDP capabilities.
This feature offers the following options:
•
The device negotiates any new SDP re-offer, regardless of the origin field
(default).
•
The device negotiates only an SDP re-offer with an incremented origin field.
This feature is configured using the new parameter, EnableSDPVersionNegotiation.
Version 6.2
19
December 2010
CPE SIP Products
4.
SIP Secure Connection Vulnerability:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for securing the device’s resources against SIP spam
and invalid SIP messages:
•
Securing memory resources:
Socket Resource Abuse: Connections that are established without
subsequent data transmission are released (after one minute), allowing the
establishment of new connections.
♦
Established Connection Flood: The device detects and subsequently
discards any flood of “false” connections (which typically prevents
establishment of new legitimate connections). The device effectively
manages its socket resources, releasing unused sockets for required
connections.
♦
•
•
CPU:
♦
Loop-Amplification Scenario: The device prevents routing between its
interfaces. The attacker needs to convince the device to re-write a request to
a location, which resolves to the device itself. This can be done if the routing
is according to the SIP Request-URI header and the address specified is the
device’s IP address. This results in the server over loading itself. Another
method for creating loops is through a SIP proxy to which the device routes
and this proxy routes it back to the device.
For MSBG products, the SBCMaxForwardsLimit parameter is used to limit
the SIP Max-Forwards header value.
♦
Malformed SIP Requests: Malformed SIP message requests are typically
sent to cause false, expensive SIP parsing, thereby wasting CPU resources.
The device’s parsing has been significantly improved to detect malformed
messages and to reject such messages in early parsing stages.
SIP Vulnerabilities:
General Parser Errors: Parser errors (invalid SIP messages) do not cause
loss of service.
♦
SIP Content-Length header greater than the message’s body: This can
cause delayed or no service by causing a TCP to wait for that body to arrive.
- TCP: maximum message length is dictated.
- UDP: Content-Length is validated with the packet size. If the packet size is
not as declared in the Content-Length header, only the actual body size is
validated and the Content-Length header is ignored.
♦
Invalid Content-Length header: The device ignores such messages.
♦
Null characters are allowed only in the SIP message’s body according to the
SIP ABNF. The device rejects messages that arrive with null characters in
the headers part of the message. This ensures that the device doesn’t
forward invalid messages that can be harmful to the internal network.
♦
SIP Release Notes
20
Document #: LTRT-26901
SIP Release Notes
5.
1. What's New in Release 6.2
Transcoding Mode for SBC and IP-to-IP Applications:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows separate transcoding mode configuration for the SBC and IP-to-IP
applications. This is important as the required DSP resources for transcoding between
these applications differ. Typically, SBC uses less resources than the IP-to-IP
application, which generally requires two DSP’s. Therefore, two different parameters
with the same options exist for configuring transcoding mode for these applications:
6.
•
TranscodingMode: used for the SBC application (the default–“Only if Required”–is
to not use resources)
•
IP2IPTranscodingMode: used for the IP-to-IP application (the default–“Force”–is
to use full transcoding resources)
Proxy Redundancy Mode per Proxy Set:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows the configuration of the Proxy Redundancy mode per Proxy Set.
Until now, the Proxy Redundancy mode could be configured only for the entire device
(using the existing “global” ProxyRedundancyMode parameter). This feature is
configured using the new parameter, ProxyRedundancyMode in the Proxy Set table
(ProxySet parameter):
•
Not configured - the “global” parameter ProxyRedundancyMode applies (default).
•
Parking - the device continues operating with a redundant (now active) Proxy until
the next failure, after which it operates with the next redundant Proxy.
•
Homing - the device always attempts to operate with the primary Proxy server
(i.e., returns to the primary Proxy whenever available).
Note that if this parameter is configured, then the global parameter is ignored.
Version 6.2
21
December 2010
CPE SIP Products
7.
SIP Authorization Header in Initial Registration Request:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports the inclusion of the SIP Authorization header in initial registration
(REGISTER) requests sent by the device. This feature is enabled by the new
parameter, EmptyAuthorizationHeader.
The Authorization header contains the credentials of a User Agent (UA) in a request to
a server. The sent REGISTER message populates the Authorization header with the
following parameters:
•
username - value of the private user identity
•
realm - domain name of the home network
•
uri - domain name’s SIP URI of the home network
•
nonce - empty value
•
response - empty value
This feature complies with registration procedures according to the IMS 3GPP
TS24.229 and PKT-SP-24.220 specifications.
An example of a REGISTER message with an Authorization header is shown below:
REGISTER sip:home1.net SIP/2.0
Via: SIP/2.0/UDP icscf1.home1.net;branch=z9hG4bKea1dof
Max-Forwards: 68
From: <sip:alice@home1.net>;tag=s8732n
To: sip:alice@home1.net
Authorization: Digest username=alice_private@home1.net,
realm=”home1.net”, nonce=””, response=”e56131d19580cd833064787ecc”
SIP Release Notes
22
Document #: LTRT-26901
SIP Release Notes
8.
1. What's New in Release 6.2
SIP Route Header in Initial Registration Request:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports the inclusion of the SIP Route header in initial registration
(REGISTER) requests sent by the device. This feature is enabled by the new
parameter, InitialRouteHeader (by default, this parameter is disabled).
When the device sends a REGISTER message (either for initial registration or to reregister), the Route header includes the proxy's FQDN or IP address and port
according to the configured Proxy Set, for example:
Route: <sip:10.10.10.10;lr;transport=udp>
or
Route: <sip: pcscf-gm.ims.rr.com;lr;transport=udp>
9.
CRLF Keep-Alive Mode:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for the carriage-return and line-feed sequences (CRLF)
Keep-Alive mechanism, according to RFC 5626 “Managing Client-Initiated
Connections in the Session Initiation Protocol (SIP)”. This feature prevents an
“avalanche” of keep-alive messages by multiple SIP UAs to a specific server.
The SIP UA (i.e., device) uses a simple periodic message as a keep-alive mechanism
to keep the flow to the proxy or registrar alive (used, for example, to keep NAT
bindings open). For connection-oriented transports such as TCP/TLS, this is based on
CRLF. This mechanism uses a client-to-server "ping" keep-alive and a corresponding
server-to-client "pong" message. This ping-pong sequence allows the client, and
optionally the server, to notify if its flow is still active and useful for SIP traffic. If the
client does not receive a pong in response to its ping, it declares the flow “dead” and
opens a new flow in its place. In the CRLF Keep-Alive mechanism, the client
periodically sends a double-CRLF (the "ping"), then waits to receive a single CRLF
(the "pong"). If the client does not receive a "pong" within this user-defined time, it
considers the flow failed.
Version 6.2
23
December 2010
CPE SIP Products
This feature is enabled by the new parameter, UsePingPongKeepAlive. In addition, the
maximum interval after which a CRLF Keep-Alive (“ping”) is sent after a received
“pong” is configured using the new parameter, PingPongKeepAliveTime (5 to
2,000,000 sec; default is 120). The device uses the range of 80-100% of this userdefined value as the actual interval. For example, if the parameter value is set to 200
sec, the interval used is any random time between 160 to 200 seconds.
Notes:
•
The Keep-Alive mechanism begins after a successful response from a SIP
REGISTER request.
•
The device sends a CRLF message to the Proxy Set only if the Proxy Keep-Alive
feature (EnableProxyKeepAlive parameter) is enabled and its transport type is set
to TCP or TLS. The device first sends a SIP OPTION message to establish the
TCP/TLS connection and if it receives any SIP response, it continues sending the
CRLF keep-alive sequences.
10. Increase in SIP Configuration Tables:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The following SIP protocol configuration tables have been increased to 32 indices
(rows):
•
SRD (SRD parameter) Proxy Set (ProxySet parameter)
•
Proxy IPs (ProxyIP parameter)
•
SIP Interface (SIPInterface parameter)
•
IP Group (IPGroup parameter)
•
Account (Account parameter)
11. Increase in SIP Media Realm Table:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The maximum number of row entries in the SIP Media Realm table (CpMediaRealm
parameter) has been increased from 16 to 64. This now accommodates the increase
of the Multiple Interface table (over 16 interfaces), allowing each IP interface to be
assigned to a different SIP Media Realm.
SIP Release Notes
24
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
12. Increased Characters for Request-URI Host Name per IP Group:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The maximum number of characters has been increased from 49 to 100 for defining
the SIP Request-URI host name used in INVITE and REGISTER messages sent to an
IP Group, or the host name in the From header of INVITE messages received from the
IP Group. This host name is defined using the SIP Group Name parameter
(IPGroup_SIPGroupName) in the IP Group table (IPGroup parameter).
13. Destination SRD in Tel-to-IP Routing Table:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for defining an SRD as the IP destination for Tel-to-IP
routing rules in the Outbound IP Routing table (Prefix parameter). Therefore, any one
of the following IP destinations can be configured in this table:
Version 6.2
•
IP address or FQDN (existing support).
•
IP Group - the call is routed to the Proxy Set (IP address) or SRD associated with
the IP Group (existing support).
•
SRD - the call is routed as follows (new support):
♦
Proxy Set associated with the SRD.
♦
If a Proxy Set is not defined for the SRD, then the call is sent to:
Media/RTP (voice): Media Realm (media port and media IP interface)
associated with SRD.
SIP signaling: SIP Interface (SIP port and control IP interface)
associated with SRD.
25
December 2010
CPE SIP Products
14. Multiple NATs for SIP Control and RTP Media using Static NAT Rules:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for multiple network address translation (NAT) using
static NAT IP rules. Different VoIP interfaces (SIP control and RTP media traffic) can
now be translated into different NAT IP addresses. This feature allows, for example,
the separation of VoIP traffic between different ISTP’s and topology hiding (of internal
IP addresses to the “public” network).
This feature is configured using the new configuration table, Static NAT table
(NATTranslation parameter). Each IP interface configured in the Multiple Interface
table (InterfaceTable parameter) can be associated with a NAT rule in this new table,
translating the source IP address and port of the outgoing packet into the NAT address
(IP address and port range). Up to 32 NAT rules can be defined.
Notes:
•
Due to this enhanced feature, the WANIPAddress parameter (previously required
to define the WAN IP address for the SIP application layer) is now obsolete.
However, if this new table is not configured and the WANIPAddress parameter is
set, then a NAT rule is automatically added to this table with an IP address
defined as the WANIPaddress and a port range of 0-65535.
•
The device’s priority method for performing NAT is as follows (not relevant for
SBC application):
a. Using an external STUN server (STUNServerPrimaryIP parameter) to assign
a NAT address for all interfaces.
b. Using the StaticNATIP parameter to define one NAT IP address for all
interfaces.
c. Using the new NAT Translation table to define NAT per interface
(NATTranslation parameter).
If NAT is not configured, the device sends the packet according to its IP address
defined in the Multiple Interface table.
SIP Release Notes
26
Document #: LTRT-26901
SIP Release Notes
1.2.2
1. What's New in Release 6.2
SIP Gateway New Features
The device supports the following new SIP Gateway application features:
1.
Microsoft’s Wideband Real-Time RTAudio Codec Support:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for Microsoft’s Real-Time RTAudio (MS-RTA) wideband
codec. The MS-RTA codec can be used in transcoding sessions with the G.722
wideband codec. In addition, wideband Linear PCM files can be played on an MS-RTA
channel, preserving the file’s wideband quality. The codec is available when DSP
Version Template Number 1 is used.
•
Valid packet intervals (range): 20 ms (intervals of multiples of 20 ms using m
factor)
•
Maximal m factor is 10
•
Silence suppression is not supported
•
Coder rates: 18000 - 29000
•
Payload type (default): 114
This codec is configured using the existing CodersGroup parameter.
2.
Transparent Codec Packetization Time:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for packetization time (“ptime”) of 10 ms for the
Transparent codec. This codec is configured using the existing CodersGroup
parameter.
Version 6.2
27
December 2010
CPE SIP Products
3.
Adapting Comfort Noise to Remote Party:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature enhances the device’s negotiation of comfort noise generation (CNG)
between SIP UAs. The device always attempts to adapt to the remote UA’s request for
CNG, even if silence suppression (SCE) is disabled for the voice codec.
CNG is indicated in the SIP INVITE’s SDP media description line as “CN”, i.e., payload
type 13 (relevant only for G.711 and G.726 codecs).
To determine CNG support, the device uses the parameters ComfortNoiseNegotiation
and EnableStandardSIDPayloadType, and the codec’s SCE setting (enabled or
disabled). If the ComfortNoiseNegotiation and EnableStandardSIDPayloadType
parameters are enabled, then the following occurs:
•
If the device is the initiator, it sends a “CN” in the SDP only if the SCE of the
codec is enabled. If the remote UA responds with a “CN” in the SDP, then CNG
occurs; otherwise, CNG does not occur.
•
If the device is the receiver and the remote SIP UA does not send a “CN” in the
SDP, then no CNG occurs. If the remote side sends a “CN”, then CNG occurs
even if the codec’s SCE is disabled.
If the ComfortNoiseNegotiation parameter is disabled, then the device does not send
“CN” in the SDP, unless the EnableStandardSIDPayloadType parameter and the
codec’s SCE are enabled, in which case, CNG occurs.
4.
Hotline Duration per Port for Automatic Dialing:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows the configuration of a timeout per port (channel) for automatic
dialing when the Hotline feature is enabled. If the phone (connected to the specific
port) is off-hooked and no digit is pressed for this user-defined duration (timeout), the
device automatically initiates a call to the user-defined destination phone number. This
timeout is configured using the new parameter, HotLineToneDuration in the existing
Automatic Dialing table (TargetOfChannel parameter). This parameter has a value
range of 0 to 60 seconds and default of 16.
SIP Release Notes
28
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
Notes:
5.
•
The Hotline duration can be set for all ports using the existing, “global” parameter,
HotLineToneDuration.
•
This feature is applicable only to FXS and FXO interfaces.
Distinctive Ringing per FXS Based on Destination Number:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports distinctive ringing patterns and call waiting tones based on
destination (called) number (or prefix) for IP-to-Tel calls (FXS interfaces). This is in
addition to the existing support based on calling number. For example, if multiple
destination numbers are used to route incoming SIP INVITE requests to a specific FXS
port, a different ringing pattern can be configured per destination number. This feature
is configured using the existing ToneIndex parameter.
6.
Channel Select Mode for Ringing Hunt Group:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature offers a new channel select method–Ring to Hunt Group (9)–for allocating
IP-to-Tel calls to FXS ports (channels) pertaining to a specific Hunt Group. This
method routes calls to all the FXS ports pertaining to the Hunt Group. Therefore, when
an IP-to-Tel call is received by the device for a specific Hunt Group, all telephones
connected to the FXS ports belonging to the Hunt Group start ringing. The call is
received by whichever telephone answers the call first (after which the other phones
stop ringing).
The new Ring to Hunt Group channel select option is supported by the “global”
ChannelSelectMode parameter as well as by the ChannelSelectMode parameter for
the Trunk Group Settings table (TrunkGroupSettings parameter).
Version 6.2
29
December 2010
CPE SIP Products
7.
Increase in Characters for Endpoint Phone Number:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The maximum number of characters that can be defined for an endpoint phone
number has been increased from 25 to 50. The phone numbers are configured using
the TrunkGroup parameter (Endpoint Phone Number table for MP-1xx and Trunk
Group table for digital devices).
8.
Forwarding IP-to-Tel Calls to Alternative SIP Request-URI upon Busy Trunk:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature enhances the existing Forward On Busy Trunk Destination feature
(configured using the ForwardOnBusyTrunkDest parameter), which until now
supported the configuration of only an IP address for call redirection. This feature now
allows the redirect destination to be configured as a SIP Request-URI user name and
host part (i.e., user@host), or host name (i.e., IP address). For example, the below
configuration forwards IP-to-Tel calls to destination user “112” at host IP address
10.13.4.12, port 5060, using transport protocol TCP, if Trunk Group ID 2 is
unavailable:
ForwardOnBusyTrunkDest 1 = 2, 112@10.13.4.12:5060;transport=tcp;
When configured with user@host, the original destination number is replaced by the
user part.
SIP Release Notes
30
Document #: LTRT-26901
SIP Release Notes
9.
1. What's New in Release 6.2
Flash-Hook with Digit Sequence:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides enhanced support for flash-hook digit sequences for FXS
interfaces. The existing FlashKeysSequenceStyle parameter includes an additional
option (2) that provides the following flash-hook digit sequences:
•
Flash Hook key only (no other digit): Places an active call on hold.
•
Flash Hook and then digit 2: Answers the call-waiting call (the first call is put on
hold).
•
Flash Hook and then digit 2: Toggles between active call and on-hold call.
•
Flash Hook and then digit 3: Enters a three-way conference. Note that the
settings of the existing ConferenceCode parameter are ignored.
•
Flash Hook and then digit 4: Transfers a call (instead of going on-hook).
10. On-Board, Three-Way Conferencing:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for on-board, three-way conferencing for FXS interfaces.
The call conference is established by the device without the need of an external
conference server. The device sets up the conference call using its IP media channels
from the IP media module (i.e., MPM module). This feature is configured using the new
option, On Board (2) for the existing 3WayConferenceMode parameter.
The maximum number of simultaneous on-board, three-way conference calls
supported by the device is:
•
2 for Mediant 800
•
5 for Mediant 1000 and Mediant 1000 MSBG
(Note that this feature is already supported by MP-11x.)
Version 6.2
31
December 2010
CPE SIP Products
11. High Definition Three-Way Conferencing:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports high definition, three-way conferencing using wideband codecs
(e.g., G.722 and AMR-WB). Therefore, this feature allows conference participants to
experience wideband voice quality. Call conferences can also include narrowband and
wideband participants.
12. SIP Proprietary Header for Call QoS Statistics:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature enables the inclusion of call quality of service (QoS) statistics in SIP BYE
and SIP 200 OK responses to BYE, using a proprietary SIP header, X-RTP-Stat. This
proprietary header is enabled using the new parameter, QoSStatistics. By default, this
parameter is disabled.
The X-RTP-Stat header provides the following statistics:
•
Number of received and sent voice packets
•
Number of received and sent voice octets
•
Received packet loss, jitter (in ms), and latency (in ms)
The X-RTP-Stat header includes the following fields:
•
PS=<voice packets sent>
•
OS=<voice octets sent>
•
PR=<voice packets received>
•
OR=<voice octets received>
•
PL=<receive packet loss>
•
JI=<jitter in ms>
•
LA=<latency in ms>
SIP Release Notes
32
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
An example of the X-RTP-Stat header in a SIP BYE message is shown below:
BYE sip:302@10.33.4.125 SIP/2.0
Via: SIP/2.0/UDP 10.33.4.126;branch=z9hG4bKac2127550866
Max-Forwards: 70
From: <sip:401@10.33.4.126;user=phone>;tag=1c2113553324
To: <sip:302@company.com>;tag=1c991751121
Call-ID: 991750671245200001912@10.33.4.125
CSeq: 1 BYE
X-RTP-Stat:
PS=207;OS=49680;;PR=314;OR=50240;PL=0;JI=600;LA=40;
Supported: em,timer,replaces,path,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCR
IBE,UPDATE
User-Agent: Sip-Gateway-/v.6.2A.008.006
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0
13. GRUU Support According to RFC 5627:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for Globally Routable User Agent (UA) URI (GRUU),
according to RFC 5627. This is used for obtaining a GRUU from a registrar and for
communicating a GRUU to a peer within a dialog. This support is provided by the
existing parameter EnableGRUU.
A GRUU is a SIP URI that routes to an instance-specific UA and can be reachable
from anywhere. There are a number of contexts in which it is desirable to have an
identifier that addresses a single UA (using GRUU) rather than the group of UA’s
indicated by an Address of Record (AOR). For example, in call transfer where user A
is talking to user B, and user A wants to transfer the call to user C. User A sends a
REFER to user C:
REFER sip:C@domain.com SIP/2.0
From: sip:A@domain.com;tag=99asd
To: sip:C@domain.com
Refer-To: (URI that identifies B's UA)
The Refer-To header needs to contain a URI that user C can use to place a call to
user B. This call needs to route to the specific UA instance that user B is using to talk
to user A. User B should provide user A with a URI that has to be usable by anyone. It
needs to be a GRUU.
Version 6.2
33
December 2010
CPE SIP Products
•
Obtaining a GRUU: The mechanism for obtaining a GRUU is through
registrations. A UA can obtain a GRUU by generating a REGISTER request
containing a Supported header field with the value “gruu”. The UA includes a
“+sip.instance” Contact header parameter of each contact for which the GRUU is
desired. This Contact parameter contains a globally unique ID that identifies the
UA instance. The global unique ID is created from one of the following:
♦
If the REGISTER is per the device’s client (endpoint), it is the MAC address
concatenated with the phone number of the client.
♦
If the REGISTER is per device, it is the MAC address only.
♦
When using TP, “User Info” can be used for registering per endpoint. Thus,
each endpoint can get a unique id – its phone number. The globally unique
ID in TP is the MAC address concatenated with the phone number of the
endpoint.
If the remote server doesn’t support GRUU, it ignores the parameters of the
GRUU. Otherwise, if the remote side also supports GRUU, the REGISTER
responses contain the “gruu” parameter in each Contact header. This parameter
contains a SIP or SIPS URI that represents a GRUU corresponding to the UA
instance that registered the contact. The server provides the same GRUU for the
same AOR and instance-id when sending REGISTER again after registration
expiration. RFC 5627 specifies that the remote target is a GRUU target if its’
Contact URL has the "gr" parameter with or without a value.
•
Using GRUU: The UA can place the GRUU in any header field that can contain a
URI. It must use the GRUU in the following messages: INVITE request, its 2xx
response, SUBSCRIBE request, its 2xx response, NOTIFY request, REFER
request and its 2xx response.
14. Sending/Receiving IP Calls to/from Trusted IP Addresses:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for sending and receiving IP calls only to and from
“trusted” IP addresses respectively. Trusted IP addresses are those that appear in the
Outbound IP Routing/Tel to IP Routing table (routing table) or Proxy Set table. In
addition, if a fully-qualified domain name (FQDN) is defined in the routing table or
Proxy Set table, the call is allowed only if the resolved DNS IP address appears in one
of these tables. Calls with IP addresses or resolved IP addresses (in case of DNS
queries) that do not appear in the routing table or Proxy Set table are rejected.
This feature is configured using the new option, “Secure All Calls” (2) for the existing
SecureCallsfromIP parameter.
SIP Release Notes
34
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
15. Trunk Re-Registration upon Return to Service:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports re-registration of the device’s trunks that return to service. When
using the Account table to register a Trunk Group (to a Proxy server), if all trunks
pertaining to the Trunk Group are down, the device unregisters the trunks. If any trunk
belonging to the Trunk Group is returned to service, the device registers them again.
This ensures, for example, that the Proxy does not send INVITEs to trunks that are out
of service.
16. Maximum Simultaneous TLS Connections:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for an increase in the maximum number of simultaneous
Transport Layer Security (TLS) sessions. The device now supports up to 500 TLS
sessions. The maximum TLS sessions for other products remains the same:
Version 6.2
•
10: MP-1xx
•
50 Mediant 800 MSBG
•
100: Mediant 600, Mediant 1000, Mediant 1000 MSBG and Mediant 2000
35
December 2010
CPE SIP Products
17. Table Increase for Source Phone Number Manipulation Table for IP-to-Tel Calls:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The Source Phone Number Manipulation Table for IP-to-Tel Calls table
(SourceNumberMapIP2Tel parameter) has been increased from 20 to 120 indices
(rows).
18. Enhanced IP-to-Tel Number Manipulation:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for configuring enhanced (additional) source and
destination number manipulation rules for IP-to-Tel calls. This feature enhances the
use of the existing number manipulation tables (NumberMapIP2Tel and
SourceNumberMapIP2Tel parameters for destination and source number
manipulation, respectively). The initial and additional number manipulations are both
done in these tables. The additional manipulation is performed on the initially
manipulated source and destination numbers. Therefore, additional manipulation can
be implemented to perform complex manipulation rules that can now be
accommodated by the limiting size of the manipulation tables.
This feature is enabled using the following new parameters:
•
PerformAdditionalIP2TELSourceManipulation for source number manipulation
•
PerformAdditionalIP2TELDestinationManipulation for destination number
manipulation
SIP Release Notes
36
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
19. RTCP XR Voice Quality Monitoring:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for RTP Control Protocol Extended Reports (RTCP XR)
according to RFC 3611. RTCP XR is a VoIP management control that defines a set of
metrics containing information for assessing VoIP call quality and diagnosis. RTCP XR
measures call quality such as packet loss, delay, signal/noise/echo levels, estimated
R-factor, and mean opinion score (MOS). RTCP XR extends the RTCP reports defined
in RFC 3550 by providing additional VoIP metrics.
RTCP XR information publishing is implemented in the device according to <draftjohnstonsipping-rtcp-summary-07>. This draft defines how a SIP UA publishes the
detailed information to a defined collector. RTCP XR messages containing key callquality-related metrics are exchanged periodically (user-defined) between the device
and the SIP UA. This allows an analyzer to monitor these metrics midstream or a
device to retrieve them using SNMP. The device can send RTCP XR reports to an
Event State Compositor (ESC) server using SIP PUBLISH messages.
These reports can be sent at the end of each call (configured using
RTCPXRReportMode) and according to a user-defined interval (RTCPInterval or
DisableRTCPRandomize) between consecutive reports. To enable RTCP XR
reporting, the VQMonEnable parameter is used. This feature requires that the device
be installed with the relevant Software Upgrade Key.
Additional RTCP XR parameters include VQMonGMin, VQMonBurstHR,
VQMonDelayTHR, VQMonEOCRValTHR, RTCPXRESCTransportType, and
RTCPXREscIP.
20. T.38 SDP T38MaxBitRate Negotiation:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports the negotiation of the T.38 maximum supported fax data rate
provided in SIP’s SDP T38MaxBitRate parameter. The negotiated T38MaxBitRate is
the minimum rate supported between the local and remote endpoints. The maximum
rate supported by the device is configured using the existing parameter,
FaxRelayMaxRate.
Version 6.2
37
December 2010
CPE SIP Products
21. New Range and Defaults for T.38 Fax Maximum Buffer Size:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides a modified value range and default for the T38FaxMaxBufferSize
parameter, which is used in SIP T.38 negotiation in SDP.
•
Value range: 500 to 3000 bytes
•
Default value:
♦
1,024 bytes: MP-1xx, Mediant 1000, Mediant 1000 MSBG, Mediant 2000
♦
3000 bytes: Mediant 800 MSBG, Mediant 3000
22. New Range and Defaults for T.38 Fax Maximum Datagram Size:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides a modified value range and default for the T38MaxDatagramSize
parameter, which is used in SIP T.38 negotiation in SDP.
•
Value range: 120 to 600 bytes
•
Default value:
♦
238 bytes: MP-1xx, Mediant 1000, Mediant 1000 MSBG, Mediant 2000
♦
560 bytes: Mediant 800 MSBG, Mediant 3000
SIP Release Notes
38
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
23. Fax Transmission behind NAT:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for transmission from fax machines (connected to the
device) located inside a Network Address Translation (NAT). Generally, the firewall
blocks T.38 (and other) packets received from the WAN, unless the device behind the
NAT sends at least one IP packet from the LAN to the WAN through the firewall. If the
firewall blocks T.38 packets sent from the termination IP fax, the fax fails.
To overcome this, the device sends No-Op (“no-signal”) packets to open a pinhole in
the NAT for the answering fax machine. The originating fax does not wait for an
answer, but immediately starts sending T.38 packets to the terminating fax machine.
This feature is enabled using the new parameter, T38FaxSessionImmediateStart. The
No-Op packets are enabled using the existing NoOpEnable and NoOpInterval
parameters.
24. Euro-ISDN Message Waiting Indication (MWI) for IP-to-Tel Calls:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for Euro-ISDN MWI for IP-to-Tel calls. The device
interworks SIP MWI NOTIFY messages to Euro-ISDN Facility information element (IE)
MWI messages. This is supported by the existing VoiceMailInterface parameter (set to
8).
Version 6.2
39
December 2010
CPE SIP Products
25. B-Channel Off-Hook Cut-Through:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature enables PSTN CAS channels/endpoints to receive incoming IP calls even
if the B-channel is in off-hook state. This feature is enabled using the new parameter,
DigitalCutThrough.
The following scenario describes the mechanism of this feature:
a. A Tel-to-IP call is established (connected) by the device for a B-channel.
b. The device receives a SIP BYE (i.e., IP side ends the call) and plays a reorder
tone to the PSTN side for the duration set by the new parameter,
CutThroughTimeForReOrderTone. The device releases the call towards the IP
side (sends SIP 200 OK).
c. The PSTN side, for whatever reason, doesn't go on-hook. However, the device
regards the state as on-hook.
d. If a new IP call is received for this B-channel after the reorder tone has ended, the
device “cuts through” the channel and connects the call immediately (even though
the B-channel is in physical off-hook state) without playing a ring tone. If an IP call
is received while the reorder tone is played, then the device rejects the call.
This new parameter has also been added to the Tel Profile table (TelProfile
parameter), allowing the Digital Cut-Through feature to be applied to specific Bchannels using specific CAS tables (by assigning this Tel Profile to the required Bchannels only).
26. Sending and Receiving RTP Packets without SIP Messages:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature enables the device to start sending and/or receiving RTP (media) packets
to and from remote endpoints without the need to establish a SIP session. The remote
IP address is determined according to the Outbound IP Routing table (Prefix
parameter). The port used is the same port as the local RTP port (configured by the
existing BaseUDPPort parameter and the channel on which the call is received).
This feature can be configured per trunk using the new RTPOnlyModeForTrunk_ID
parameter or for all trunks using the new RTPOnlyMode parameter.
SIP Release Notes
40
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
27. Pre-emption of IP-to-Tel E911 Emergency Calls:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports the pre-emption of IP-to-Tel E911 emergency calls. If the device
receives an E911 call and there are unavailable channels, the device terminates one
of the calls and sends the E911 call to that channel. This feature is enabled by the new
value option, “Emergency” (3) of the existing CallPriorityMode parameter.
Note that this feature is applicable to FXO, CAS, and ISDN interfaces.
28. Combined Support for Digit Map and External Dial Plan File:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for implementing both the Digit Map (DigitMapping
parameter) and external Dial Plan (DialPlanIndex parameter). For example, the Digit
Map can be used for complex digit patterns (which are not supported by the Dial Plan)
and the Dial Plan can be used for long lists of relatively simple digit patterns. In
addition, as timeout between digits is not supported by the Dial Plan, the Digit Map can
be used to define digit patterns (MaxDigits parameter) that are shorter than those
defined in the Dial Plan or left at default. For example, “xx.T” instructs the device to
use the Dial Plan and if no matching digit pattern, it waits for two more digits and then
after timeout (TimeBetweenDigits parameter) it sends the collected digits. Therefore,
this ensures that calls are not rejected as a result of their digit pattern not been
completed in the Dial Plan.
Note: By default, the device first attempts to locate a matching digit pattern in the Dial
Plan, and if not found, then uses the Digit Map.
Version 6.2
41
December 2010
CPE SIP Products
29. Multi-AMD Sensitivity Table:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for multi-language/country Answering Machine Detection
(AMD). This is achieved by the new Multi-AMD Sensitivity table. The table provides up
to 4 different AMD Sensitivity suites (selected by the new
AMDSensitivityParameterSuit parameter), where each suite comprises up to 16
different sensitivity permutations (selected by the new AMDSensitivityLevel
parameter).
The AMD Sensitivity table is loaded to the device as a binary (*.dat) file) auxiliary file.
The DConvert utility can be used to convert the file from XML before loading to the
device. The file name or URL is configured using the new parameters,
AMDSensitivityFileUrl and AMDSensitivityFileName respectively.
30. Answer Machine Detector (AMD) per Call:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for configuring AMD per call based on the called number
or Trunk Group. This is achieved by the inclusion of the following AMD parameters in
the IP Profile table (IPProfile parameter): AMDSensitivityParameterSuit,
AMDSensitivityLevel, AMDMaxGreetingTime, and AMDMaxPostSilenceGreetingTime.
This IP Profile can then be assigned to a Trunk Group in the Inbound IP Routing table
(PSTNPrefix parameter).
SIP Release Notes
42
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
31. Additional VXML Script Attributes for Invoking VXML Scripts:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for the following additional VXML session variables and
events:
•
session.connection.local.uri
•
session.connection.remote.uri
•
session.connection.protocol.name
•
session.connection.protocol.version (version is ‘2’ instead of ‘2.0’)
•
session.connection.redirect (redirect reason and screening information contains
‘_’ instead of whitespace between words)
•
session.connection.aai
32. Activating VXML Script on Receipt of Regular INVITE:
Applicable Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for activating a Voice XML (VXML) script using the
VXMLID parameter in the Request-URI user part only, upon receipt of a regular
INVITE message. For example:
Request-URI = <VXMLID>http://mydomain.com/myscript.cgi@host;
Up till now, VXML was invoked on the receipt of SIP Request-URIs such as:
<VXMLID>@host;voicexml= http://...
This feature is supported for IP-to-Tel and Tel-to-IP calls. For specified dialed phone
numbers, the user part can be manipulated by adding a VXML script path. For
example, upon receipt of the INVITE request, INVITE sip:100@myhost the device
can be configured to manipulate (using the IP to Tel Manipulation table) the RequestURI user part to voicexml=http://myhost.com/script.cgi@myhost.
Version 6.2
43
December 2010
CPE SIP Products
33. Enhanced ISDN BRI Support:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides enhanced support for Integrated Services Digital Network (ISDN)
Basic Rate Interface (BRI) phones connected to the device. This feature enables the
device to route IP-to-Tel calls (including voice and fax) to specific BRI ports (channels).
This is achieved by the following configuration elements:
•
New ISDN Supplementary Services table (ISDNSuppServ parameter): This table
allows you to define BRI phone extension numbers per BRI port pertaining to a
specific BRI module. Therefore, this offers support for point-to-multipoint
configuration of several phone numbers per BRI channel. Up to eight phone
numbers can be defined per BRI trunk. In addition, for each BRI endpoint, the
following optional configurations can be defined:
♦
User ID and password - for registering the BRI endpoint to a third-party
softswitch for authentication and/or billing
♦
Caller ID name - for displaying the BRI endpoint’s caller ID to a dialed
destination, if enabled (i.e., “Presentation” is not restricted)
♦
Caller ID presentation/restriction
♦
Enable/disable sending caller ID to BRI phone
•
Existing Channel Select Mode (ChannelSelectMode parameter): The new option,
“Select Trunk by ISDN Supplementary Services Table” (10) in the Trunk Group
Settings table (TrunkGroupSettings parameter), allows the routing of IP-to-Tel
calls to specific BRI channels according to the settings in the ISDN
Supplementary Services table.
34. Registration Status Display for BRI Endpoints:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for displaying the registration status of BRI endpoints in
the Web interface. This information is displayed in the Registration Status page
(Status & Diagnostics tab > VoIP Status menu > Registration Status).
SIP Release Notes
44
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
35. BRI Call Forwarding Supplementary Service:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for call forwarding (CF) services initiated by ISDN Basic
BRI phones connected to the device. Upon receipt of an ISDN Facility message for call
forward from the BRI phone, the device sends a SIP INVITE to the softswitch with a
user-defined code in the SIP To header, representing the reason for the call forward.
The codes for the call forward can be defined using the following new parameters:
•
SuppServCodeCFU - Call Forward Unconditional
•
SuppServCodeCFUDeact - Call Forward Unconditional Deactivation
•
SuppServCodeCFB - Call Forward on Busy
•
SuppServCodeCFBDeact - Call Forward on Busy Deactivation
•
SuppServCodeCFNR - Call Forward on No Reply
•
SuppServCodeCFNRDeact - Call Forward on No Reply Deactivation
Note that these codes must be defined according to the softswitch (i.e., the softswitch
must recognize them).
Below is an example of an INVITE message sent by the device indicating
unconditional call forward (“*72”) to extension number 100. This code is defined using
the SuppServCodeCFU parameter.
INVITE sip:*72100@10.33.8.53;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.33.2.5:5060;branch=z9hG4bKWDSUKUHWFEXQSVOUVJGM
From: <sip:400@10.33.2.5;user=phone>;tag=DUOROSXSOYJJLNBFRQTG
To: <sip:*72100@10.33.8.53;user=phone>
Call-ID: GMNOVQRRXUUCYCQSFAHS@10.33.2.5
CSeq: 1 INVITE
Contact: <sip:400@10.33.2.5:5060>
Supported: em,100rel,timer,replaces
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCR
IBE
User-Agent: Sip Message Generator V1.0.0.5
User-to-User: 31323334;pd=4
Content-Type: application/sdp
Content-Length: 155
Version 6.2
45
December 2010
CPE SIP Products
36. BRI Suspend-Resume Supplementary Service:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for call suspend and resume services initiated by ISDN
BRI phones connected to the device. During an ongoing call, the BRI phone user can
suspend the call by typically pressing the phone’s “P” button or a sequence of buttons
(depending on the phone) and then on-hooking the handset. To resume the call, the
phone user typically presses the same button(s) again and then off-hooks the phone.
During the suspended state, the device plays a Howler tone to the remote party.
Note that instead of pressing the call park button(s), the phone cable can also be
disconnected (suspending the call) and then reconnected again (resuming the call).
If the phone user does not resume the call within a user-defined interval (configured
using the existing HeldTimeout parameter), the device releases the call.
37. Attended Call Transfer for BRI Interfaces:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for attended (consultation) call transfer for BRI phones
(user side) connected to the device and using the Euro ISDN protocol. BRI call transfer
is according to ETSI TS 183 036, Section G.2 (Explicit Communication Transfer –
ECT). Call transfer is enabled using the existing parameters, EnableTransfer and
EnableHoldtoISDN.
Note: Currently, the BRI interface does not support blind call transfer.
SIP Release Notes
46
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
38. MWI for BRI Interfaces:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for MWI on BRI phones connected to the device and
using the Euro ISDN BRI variant. When this feature is activated and a voice mail
message is recorded to the mail box of a BRI extension, the softswitch sends a
notification to the device. In turn, the device notifies the BRI extension and a red light
flashes on the BRI extension’s phone. Once the voice message is retrieved, the MWI
light on the BRI extension turns off.
This feature is configured by the new option, “ETSI” (8) of the existing
VoiceMailInterface parameter and enabled by the existing EnableMWI parameter.
39. Keypad Protocol for DTMF Digits According to ETS 300 122-1:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for ISDN Information messages received from BRI
phones using Information messages. During a call, if the BRI phone user presses
digits (for example, for interactive voice response/IVR applications such as retrieving
voice mail messages), ISDN Information messages received by the device for each
digit are sent in the voice channel to the IP network as DTMF signals. The method by
which the device “injects” these DTMF signals into the IP is defined by the existing
parameter, TxDTMFOption.
Version 6.2
47
December 2010
CPE SIP Products
40. Interworking Date/Time between SIP and Euro-ISDN:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for interworking date and time between SIP and EuroISDN:
•
SIP-to-ISDN interworking: The date and time from the received SIP 200 OK
message is included in the ISDN Connect message by the addition of the
Date/Time Information Element (IE).
•
ISDN-to-SIP interworking: The date and time in the Date/Time IE received from
the ISDN Connect message is interworked to the SIP Date header and sent to the
IP network.
This feature is essential for synchronization between the PSTN (for example, PBX)
and the IP network.
Note: The Date/Time IE can only be sent by an ISDN network-side trunk.
41. Interworking ISDN Offhook IE to SIP INVITE:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for interworking the ISDN Setup message with an
offhook indicator IE to SIP INVITE’s Request-URI header (in addition to the existing
support for the Contact header). Upon receipt of an ISDN offhook indicator IE from the
PSTN side, the device sends a SIP INVITE to the IP side with the parameters
“tgrp=hotline” or “tgrp=hotline-ccdata” included in the Request-URI and Contact
headers.
This feature is configured using the new option, “Hotline Extended” (4) of the existing
UseSIPTgrp parameter.
Note: This feature is applicable to Department of Defense (DoD) applications,
according to the UCR 2008 Change 1 specification.
SIP Release Notes
48
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
42. Redirect Number Overwritten by Destination Number for IP-to-Tel Calls:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows the device to overwrite the redirect number with the destination
number from the received SIP INVITE message in the outgoing ISDN call. This is
achieved by assigning an IP Profile (IPProfile parameter) defined with the
CopyDest2RedirectNumber parameter set to 1, to the IP-to-Tel Routing table
(PSTNPrefix parameter). Even if there is no SIP Diversion or History header in the
incoming INVITE message, the outgoing Q.931 Setup message will contain a redirect
number.
43. Replacing Calling Number with Redirect Number in Incoming ISDN-to-IP Call:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows the device to replace the calling number with the redirect number in
ISDN-to-IP calls. When such a replacement occurs, the calling name is deleted and
left blank. The outgoing INVITE message does not include the redirect number that
was used to replace the calling number. The replacement is done only if a redirect
number is present in the incoming call. This feature is enabled using the new
parameter, ReplaceCallingWithRedirectNumber.
Version 6.2
49
December 2010
CPE SIP Products
44. Manipulation of Redirect Reason for Tel-to IP Calls:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for configuring a default redirect reason for Tel-to-IP
calls. This is configured by the new parameter, Tel2IPDefaultRedirectReason.
The device includes this default redirect reason in the SIP History-Info header of the
outgoing INVITE if the received Q931 ISDN Setup message does not include a
redirect reason or includes only an “unknown” reason. If a redirect reason exists in the
received Setup message, the new parameter is ignored and the device sends the
INVITE message with the reason according to the received Setup message. If the
parameter is not configured, the outgoing INVITE is sent with the redirect reason (if
none or “unknown” then no reason) as received in the Setup message.
45. UC Namespace for MLPP Calls:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for configuring a new default namespace, “UC” for MultiLevel Precedence & Preemption (MLPP) calls received from the ISDN side and
destined for the Application server. This is configured by setting the existing parameter
MLPPDefaultNamespace to the new option, “UC” (5). The namespace value is not
present in the Precedence IE of the PRI Setup message. Therefore, this value is used
in the Resource-Priority header of the outgoing SIP INVITE request.
SIP Release Notes
50
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
46. Response to SIP OPTIONS only if Trunk Group 1 is Available:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for responding to received SIP OPTIONS (e.g., from a
softswitch) only if Trunk Group #1 has available trunks. If all the trunks pertaining to
Trunk Group #1 are down or busy, the device does not respond to received SIP
OPTIONS. This feature is enabled using the new parameter,
TrunkStatusReportingMode.
47. ISDN Facility IE Trace:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for enabling ISDN traces of Facility Information Elements
(IE) for ISDN call diagnostics. This allows you to trace all the parameters contained in
the Facility IE and view them in the Syslog. This feature is enabled using the new
parameter, FacilityTrace.
Note: For this feature to be functional, the GWDebugLevel parameter must be enabled
(set to at least level 1).
Version 6.2
51
December 2010
CPE SIP Products
1.2.3
SIP SAS New Features
The device supports the following new SIP Standalone Survivability (SAS) application
features:
1.
Blocking Incoming Calls from Unregistered SAS Users:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature rejects SIP INVITE messages received from unregistered SAS users, in
SAS Normal and Emergency modes. This feature is enabled by the new parameter
SASBlockUnRegUsers.
2.
Maximum Registered SAS Users:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The maximum number of SAS users that can be registered to the device has been
increased to 600.
SIP Release Notes
52
Document #: LTRT-26901
SIP Release Notes
3.
1. What's New in Release 6.2
SAS Normal Mode Routing Enhancement:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature enhances routing capabilities for SAS Normal mode by using the IP-to-IP
Routing table to route SAS IP-to-IP calls to different destinations (and not only to the
SAS Proxy Set). This is configured by the new option, “Use Routing Table only in
normal mode (before forward to proxy)” (4) of the existing SASSurvivabilityMode
parameter. Note that when this parameter is set to this option, the IP-to-IP Routing
table (IP2IPRouting parameter) is used only when SAS is in Normal mode and is
unavailable when SAS is in Emergency mode.
4.
SAS Manipulation on INVITE Requests:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for SAS manipulation on the SIP Request-URI user part
of incoming INVITE requests (in addition to the existing support for REGISTER
requests). Once manipulated, the SAS application searches for the user in the
registration database. Manipulation is configured in the existing SAS Registration
Manipulation Table (SASRegistrationManipulation).
Note: This feature does not change the Request-URI of the outgoing INVITE
message.
Version 6.2
53
December 2010
CPE SIP Products
1.3
Session Border Controller New Features
The device supports the following new Session Border Controller (SBC) features:
1.
SBC Support by Mediant 3000:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The SBC application is now fully supported by the Mediant 3000 (including Mediant
3000 High Availability). The SBC application now replaces the previously supported
B2BUA application and includes all the SBC features supported by the MSBG series
products, for example SIP header manipulations and USER-type IP Groups (i.e.,
registration of users in internal database).
2.
Maximum Registered SBC Users:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The maximum number of SBC users that can be registered to the device’s database
has been increased to 200 for Mediant 800 MSBG and 600 for Mediant 1000 MSBG.
3.
Maximum Registered SBC Users:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The maximum number of SBC users that can be registered to the database of a fully
populated device has been increased to 2,000.
SIP Release Notes
54
Document #: LTRT-26901
SIP Release Notes
4.
1. What's New in Release 6.2
SBC Registration Timers:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature now enables the device to retain the original value of the SIP Expires
header received from the user or proxy. This feature also applies when the SBC is in
“survivability” state (i.e., REGISTER requests cannot be forwarded to the proxy and is
terminated by the SBC).
This feature is configured using the following new parameters:
5.
•
SBCUserRegistrationTime: This parameter replaces the now obsolete
SBCRegistrationTime parameter. This parameter defines the duration of the
periodic registrations between the user and the SBC. When set to 0, the SBC
does not change the Expires value received in the user’s REGISTER request. If
no Expires header is received in the REGISTER and SBCUserRegistrationTime is
set to 0, the Expires value is set to 180 seconds (by default).
•
SBCProxyRegistrationTime: This parameter replaces the RegistrationTime
parameter. This parameter defines the duration for which the user is registered in
the proxy database. When set to 0, the SBC sends the Expires value as received
from the user to the proxy. (Note: The RegistrationTime parameter is still
applicable to non-SBC applications.)
•
SBCSurvivabilityRegistrationTime: This parameter defines the duration of the
periodic registrations between the user and SBC when the SBC is in survivability
state. When set to 0, the SBC uses the value of the SBCUserRegistrationTime
parameter for the SBC response.
IP Alert Timeout to SBC Outgoing INVITE Message:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows the configuration of a timeout (in seconds) for SBC outgoing
(outbound IP routing) SIP INVITE messages. If the called IP party does not answer the
call within this user-defined interval, the device disconnects the session. The device
starts the timeout count upon receipt of a SIP 180 Ringing response from the called
party. If no other SIP response (for example, 200 OK) is received thereafter within this
timeout, the call is released. This feature is configured using the new parameter,
SBCAlertTimeout.
Version 6.2
55
December 2010
CPE SIP Products
6.
Max-Forwards SIP Header Size:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides control over the SIP Max-Forwards header value. The MaxForwards header is used to limit the number of servers (such as proxies) that can
forward the SIP request. The Max-Forwards value indicates the remaining number of
times this request message is allowed to be forwarded. This count is decremented by
each server that forwards the request.
The Max-Forwards SIP header value is configured using the new parameter,
SBCMaxForwardsLimit. The value range can be between 1 and 70 (the default is 10).
This parameter affects the Max-Forwards header in the received message, as follows:
7.
•
If the received header’s original value is 0, the message is not passed on and is
rejected.
•
If the received header’s original value is less than the parameter’s value, the
header’s value is decremented before being sent.
•
If the received header’s original value is greater than the parameter’s value, the
header’s value is replaced by the parameter’s value.
Active SBC Call Continuity during High Availability Blade Switchover:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for maintaining active SBC call sessions during a blade
switchover. Upon blade failure, the standby blade takes over from the previously active
blade and current SBC calls are preserved (and not disconnected as in the previous
release).
SIP Release Notes
56
Document #: LTRT-26901
SIP Release Notes
8.
1. What's New in Release 6.2
Classifying IP Call to Proxy Set based on IP Address, Port, and Transport Type:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides the ability to classify an IP call to a Proxy Set based on either its
IP address, or its IP address, port, and transport type (e.g., UDP). As a Proxy Set can
be defined with up to five Proxy IP addresses with or without associated ports and/or
transport type, this provides greater flexibility in classifying a call with a Proxy Set. This
feature is configured using the new parameter, ClassificationInput in the existing Proxy
Set table (ProxySet parameter).
9.
SIP-Dialog Rate Control:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for SIP-dialog rate control, using the “token bucket”
mechanism. The token bucket is a control mechanism that dictates the rate of SIPdialog setups based on the presence of tokens in the bucket – a logical container that
holds aggregate SIP dialogs to be accepted or transmitted. Tokens in the bucket are
removed ("cashed in") for the ability to setup a dialog. Therefore, a flow can setup
dialogs up to its peak burst rate if there are adequate tokens in the bucket and if the
burst threshold is configured appropriately:
•
Every SIP dialog setup request must attempt to take a token from the bucket.
•
If there are no tokens, the request is dropped.
•
New tokens are added to the bucket at a user-defined rate (token rate).
•
If the bucket contains the maximum number of tokens, tokens to be added at that
moment are dropped.
A token bucket is configured in the SBC Admission Control table
(SBCAdmissionControl parameter), using the following new parameters:
•
Rate = Rate at which tokens are added to the bucket (i.e., token rate). One token
is added to the bucket every 1000/Rate milliseconds. The rate of dialog setups
per second, or unlimited if set to 0 (default).
•
Max Burst = Maximum tokens that can fill the bucket. At any given time, the
bucket cannot contain more than this amount of tokens. The maximum burst size
for the dialog setup rate, unlimited if set to 0 (default).
Dropped requests are replied with the 486 “Busy Here” SIP response. Dropped
requests are not counted in the bucket.
Version 6.2
57
December 2010
CPE SIP Products
10. Mediation without Transcoding:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for mediation without transcoding. The device
transparently relays RTP traffic (through Digital Signal Processing/DSP) and the
following features are supported:
•
SRTP-RTP, SRTP tunneling
•
Remote media NAT traversal (RFC 4961)
•
Broken connection notification
•
RTCP XR statistics
•
Rogue RTP filtering
11. Interworking SIP Diversion and History-Info Headers:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for interworking between the SIP Diversion and HistoryInfo headers. This is important, for example, to networks that support the Diversion
header but not the History-Info header, or vice versa. Therefore, mapping between
these headers is crucial for preserving the information in the SIP dialog regarding how
and why (e.g., call redirection) the call arrived at a certain SIP UA.
This feature is configured in the IP Profile table (IPProfile parameter) using the
following new parameters:
•
SBCDiversionMode - defines the device’s handling of the Diversion header:
•
SBCHistoryInfoMode - defines the device’s handling of the History-Info header:
The handling of the SIP Diversion and History-Info headers is described in the table
below:
SIP Header Present in Received SIP Message
Parameter Value
HistoryInfoMode = Add
DiversionMode =
Remove
SIP Release Notes
Diversion
Diversion
converted to
History-Info.
Diversion
removed.
58
History-Info
Not present.
Diversion and
History-Info
Diversion added to
History-Info.
Diversion removed.
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
SIP Header Present in Received SIP Message
Parameter Value
Diversion and
History-Info
History-Info
Diversion
Not present.
History-Info
converted to
Diversion.
History-Info
removed.
History-Info added
to Diversion.
History-Info
removed.
HistoryInfoMode =
Disable
DiversionMode = Add
Diversion
converted to
History-Info.
Not present.
Diversion added to
History-Info.
HistoryInfoMode =
Disable
DiversionMode = Add
Not present.
History-Info
converted to
Diversion.
History-Info added
to Diversion.
HistoryInfoMode = Add
DiversionMode = Add
Diversion
converted to
History-Info.
History-Info
converted to
Diversion.
Headers are
synced and sent.
HistoryInfoMode =
Remove
DiversionMode =
Remove
Diversion
removed.
History-Info
removed.
Both removed.
HistoryInfoMode =
Remove
DiversionMode = Add
12. GRUU Support According to RFC 5627:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for Globally Routable User Agent (UA) URI (GRUU),
according to RFC 5627. This is used for obtaining a GRUU from a registrar and for
communicating a GRUU to a peer within a dialog. This support is provided by the
existing parameter, EnableGRUU. In addition, this feature allows the device to act as a
GRUU server for its SIP UA clients, providing them with public GRUU’s, according to
RFC 5627. The public GRUU provided to the client is depicted in the SIP Contact
header parameters, "pub-gruu". Public GRUU remains the same over registration
expirations. On the other SBC leg communicating with the Proxy/Registrar, the device
acts as a GRUU client. The GRUU server feature is enabled by the new parameter,
SBCGruuMode.
The device creates a GRUU value for each of its registered clients, which is mapped to
the GRUU value received from the Proxy server. In other words, the created GRUU
value is only used between the device and its clients (endpoints).
Public-GRUU: sip:userA@domain.com;gr=unique-id
•
Version 6.2
Obtaining a GRUU: The mechanism for obtaining a GRUU is through
registrations. A UA can obtain a GRUU by generating a REGISTER request
containing a Supported header field with the value “gruu”. The UA includes a
“+sip.instance” Contact header parameter of each contact for which the GRUU is
desired. This Contact parameter contains a globally unique ID that identifies the
UA instance. The global unique ID is created from one of the following:
59
December 2010
CPE SIP Products
If the REGISTER is per the device’s client (endpoint), it is the MAC address
concatenated with the phone number of the client.
♦
If the REGISTER is per device, it is the MAC address only.
If the remote server doesn’t support GRUU, it ignores the parameters of the
GRUU. Otherwise, if the remote side also supports GRUU, the REGISTER
responses contain the “gruu” parameter in each Contact header. This parameter
contains a SIP or SIPS URI that represents a GRUU corresponding to the UA
instance that registered the contact. The server provides the same GRUU for the
same AOR and instance-id when sending REGISTER again after registration
expiration. RFC 5627 specifies that the remote target is a GRUU target if its’
Contact URL has the "gr" parameter with or without a value.
♦
•
Using GRUU: The UA can place the GRUU in any header field which can contain
a URI. It must use the GRUU in the following messages: INVITE request, its 2xx
response, SUBSCRIBE request, its 2xx response, NOTIFY request, REFER
request and its 2xx response.
13. New IP Group Type “GATEWAY”:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for a new IP Group type–“GATEWAY”–defined in the IP
Group table (IPGroup parameter). This is used for scenarios in which the SBC device
receives requests to and from a gateway that represents multiple users, as shown
below:
In previous releases, the IP Group could be defined as one of the following types:
•
SERVER: Used when the destination address (configured by the Proxy Set –
ProxySet parameter) of the IP Group such as an ITSP, a Proxy, an IP-PBX, or an
Application server is known.
•
USER: Represents a group of users (such as IP phones and softphones) where
their location is dynamically obtained by the device, using the SIP Contact
header, when REGISTER requests and responses traverse (or are terminated) by
the device. These users are considered remote (far-end) users.
SIP Release Notes
60
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
The “GATEWAY” type is necessary for requests representing multiple users, as the
existing options above are not suitable:
•
The IP Group cannot be defined as a SERVER since its destination address is
unknown during configuration.
•
The IP Group cannot be defined as a USER since the SIP Contact header of the
incoming REGISTER does not represent a specific user. The Request-URI user
part can change and therefore, the device is unable to identify an already
registered user and therefore, adds an additional record to the database.
The IP address of the “GATEWAY” IP Group is obtained dynamically from the host
part of the Contact header in the REGISTER request received from the IP Group.
Therefore, routing to this IP Group is possible only once a REGISTER request is
received. If a REGISTER refresh request arrives, the device updates the new location
(i.e., IP address) of the IP Group. If the REGISTER fails, no update is performed. If an
UN-REGISTER request arrives, the IP address associated with the IP Group is deleted
and therefore, no routing to the IP Group is done.
The IP Group table also has a new read-only field–“Gateway Group Location”– which
displays the current destination IP address of the IP Group.
14. Minimum Session Expires (SIP Min-SE Header):
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for configuring the minimum amount of time (in seconds)
between session refresh requests in a dialog before the session is considered timed
out. This value is sent in the SIP Min-SE header. This feature is configured using the
new parameter, SBCMinSE. When set to zero (default), the SBC does not limit
Session-Expires.
15. Multiple RTP Media Streams per Call Session:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for multiple RTP media streams per SBC call session.
Up to five different media types can be included in a session - audio (m=audio), video
(m=video), text, (m=text), and fax (m=image). Therefore, this also allows the device to
provide transcoding of various attributes in the SDP offer/answer (e.g., codec, port,
and packetization time) per media type. If the device is unable to perform transcoding
(for example, does not support the codec), it relays the SBC dialog transparently.
Version 6.2
61
December 2010
CPE SIP Products
16. General SBC Configuration Enghancements:
Product
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This release provides the following general SBC configuration enhancements:
•
In Release 6.0, the IP address of the WAN interface had to be defined in the Data
configuration section, and then once defined, it had to be assigned to the SIP
application layer for the SBC configuration (using the WANIPAddress parameter).
Now, in Release 6.2, there is no need to assign the WAN IP address in the SBC
configuration section; it is acquired automatically from the Data section.
•
In Release 6.0, the user had to configure port forwarding in the Data section for all
SIP interfaces and Media Realms relating to the WAN interface. Now, in Release
6.2, port forwarding for the SIP interfaces and Media Realms are configured
automatically.
•
In Release 6.0, a “Static NAT” had to be configured in the Data section for all SIP
interfaces and Media Realms relating to the WAN interface. Now, in Release 6.2,
the Static NAT for SIP interfaces and Media Realms are configured automatically.
•
In Release 6.0, it was forbidden to use the same port range for the WAN and LAN
interfaces. For instance, the user couldn’t assign two SIP interfaces, one for LAN
and one for WAN, using the same port (i.e. 5060). Now, in Release 6.2, this
limitation no longer applies and these interfaces can be assigned the same ports.
•
In Release 6.2, more than one VLAN is available on the WAN interface - one for
media and control, and several VLANs for data. The user is required to choose
the VLAN assigned to media and control. Note that it's not backward compatiable
and the user MUST specify the WAN VLAN in the Multiple Interface page.
•
To facilitate SBC configuration, Release 6.2 now provides a new “master” page the new SRD page. This page allows the user to configure a new SRD and its
parameters as well as assign SIP interfaces to the SRD. In addition, this page
displays (read-only) all the IP Groups and Proxy Sets associated with the SRD.
SIP Release Notes
62
Document #: LTRT-26901
SIP Release Notes
1.4
1. What's New in Release 6.2
Media New Features
The device supports the following new media features:
1.
Media Packet Forwarding:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
This feature provides support for UDP and RTP “aware” packet media streaming
through a Media Forwarding session for the SBC application. This feature enhances
the IP-to-IP forwarding capability by supporting UDP/RTP awareness. The following
capabilities are included:
2.
•
Incoming Media latching
•
NAT Traversal handling
•
SRTP-to-RTP translation
•
RTP statistics per endpoint in the SBC session
•
Unknown RTP Payload Types handling
•
Unknown codecs handling
•
SRTP tunneling
•
T.38 forwarding
V.34 Fax Relay over T.38 Version 3:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
This feature provides support for ITU-T T.38 Version 3. This allows the transmission of
V.34 fax over T.38, providing high fax transmission speeds and high fax transmission
reliability. T.38 version 3 is enabled using the new parameter, SIPT38Version.
Version 6.2
63
December 2010
CPE SIP Products
Note: To use V.34, the relevant DSP template must be used.
3.
G.722 Codec Support:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
This feature provides support for the G.722 speech codec, according to RFC 3551.
G.722 was developed by ITU-T as a wideband codec with a bit rate of 64 kbps. This
codec is configured using the CodersGroup parameter.
4.
Wideband AMR Codec Support:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
This feature provides support for the Adaptive Multi-Rate Wideband (AMR-WB)
speech codec, according to RFC 3267. The AMR-WB speech codec was originally
developed by 3GPP for GSM and 3G cellular systems. Similar to AMR, the AMR-WB
codec is a multi-mode speech codec. AMR-WB supports nine wideband speech
coding modes with bit rates ranging from 6.6 to 23.85 kbps. This codec is configured
using the CodersGroup parameter.
SIP Release Notes
64
Document #: LTRT-26901
SIP Release Notes
1.5
1. What's New in Release 6.2
PSTN New Features
The device supports the following new PSTN features:
1.
E1/T1 PSTN Interface:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports a single E1/T1 PSTN port interface. This E1/T1 interface can
be configured to any one of the following PSTN protocols: Transparent, CAS, or ISDN.
2.
Implicit Mode for Explicit Call Transfer (ECT):
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports Implicit ECT transfer for Euro ISDN BRI protocol (in addition
to the existing Explicit ECT support). In Implicit ECT, the transfer is done between two
calls on the same trunk. Typically, the Implicit mechanism is used for BRI and the
Explicit mechanism for PRI.
Version 6.2
65
December 2010
CPE SIP Products
3.
T310 Timer Configuration for ISDN Variant NI2:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports configuration of the T310 timer for the ISDN variant National
ISDN with sub variant 2 (NI2). This solution is similar to the T310 timer for DMS and
Euro ISDN. The T310 timer is the timer that is set when the ISDN interface receives a
Call Proceeding message. If no Alerting, Progress, or Connect message is received
within the duration of T310, the call clears.
The T310 timer can be configured using the existing parameter, ISDNTimerT310.
When set to 0, the timer uses the default timer according to the protocol’s
specifications.
4.
Removing Validation Check of UUIE Data:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports a bit-field that determines the behavior of the Q.931 protocol.
This is configured per trunk, using the new parameter, ISDNNSBehaviour2. The
relevant bit is NS_BEHAVIOUR2_ANY_UUI (0x0008). When this bit is set, any User to
User Information Element (UUIE) is accepted for any protocol discriminator. This
feature is useful for interoperability with non-standard switches.
SIP Release Notes
66
Document #: LTRT-26901
SIP Release Notes
1.6
1. What's New in Release 6.2
Networking New Features
The device supports the following new networking features:
1.
ARIA Encryption Algorithm for TLS:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports ARIA algorithm cipher encryption for Transport Layer
Security (TLS) connections. ARIA is a symmetric key block cipher algorithm standard
developed by the Korean National Security Research Institute. To support this
encryption, the existing HTTPSCipherString parameter must be set to a value which
includes ARIA or ALL.
2.
VoIP Firewall:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports the configuration of inbound access (block or allow) firewall
rules for the VoIP interface. This feature is configured using the existing Firewall
Settings table (AccessList parameter).
Version 6.2
67
December 2010
CPE SIP Products
3.
VoIP Firewall per Network Interface:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports the configuration of firewall access rules (block or allow) per
VoIP network interface. These are the VoIP interfaces as defined in the Multiple
Interface table (InterfaceTable parameter). The firewall rules are configured in the
existing Firewall Settings table (AccessList parameter).
4.
LAN Port Monitoring:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports the monitoring of traffic traversing its LAN ports (i.e., Port
Mirroring). This includes monitoring of egress and/or ingress traffic. This feature is
useful for analyzing traffic or debugging network problems. The new CLI commands,
port monitor and show data port-monitoring are used for displaying this
information.
SIP Release Notes
68
Document #: LTRT-26901
SIP Release Notes
5.
1. What's New in Release 6.2
Default Gateway per IP Interface:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
This feature provides enhanced routing with its support for multiple default gateways.
Each IP network interface (whether it’s Control, Media, or OAMP) can now be defined
with a default gateway. This is configured in the existing Multiple Interface table
(InterfaceTable parameter).
6.
Static Routing Table Enhancements:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device’s IP Routing table (StaticRouteTable parameter) has been enhanced to
provide offline configuration of IP routing rules. The rules in this table can now be
associated with an IP interface defined in the Multiple Interface table (rather than the
previously used index). Therefore, the routing decision is now based on the source
subnet/VLAN. In the previous release, the routing decision was based on the
destination IP address only and therefore, for example, packets from the Media subnet
could be routed to the Control subnet and vice versa. This enhanced routing behavior
prevents cases where static routes are lost and simplifies the static routes
configuration for Control and Media subnets.
In addition, a new table column has been added to provide the current status of a
configured route (“Active” or “Inactive”). After adding a new route, the device attempts
to apply the route. If the operation fails, the route is marked as “Inactive” until it can
become active (following a change in the routing/interface tables).
Version 6.2
69
December 2010
CPE SIP Products
7.
Increase in Maximum Call Control/Media Interfaces:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports the configuration of up to 31 Control/Media interfaces (as
compared to 15 in the previous release). Together with the single OAMP interface, a
total of 32 IP interfaces can be configured. This is done in the Multiple Interface table
(InterfaceTable parameter).
8.
Remote GARP Detection of Media Streams (VoIP)
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports automatic change of the remote Ethernet MAC address in
sent (Tx) packets of all relevant active media channels upon receiving Gratuitous
Address Resolution Protocol (GARP) packet. This is configured in the existing
parameter, EnableDetectRemoteMACChange (options 2 and 3).
SIP Release Notes
70
Document #: LTRT-26901
SIP Release Notes
9.
1. What's New in Release 6.2
Internet Protocol Version 6 (IPv6) Support:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
Application
Gateway/GW
SBC
SAS
The device now supports the configuration of IPv6 Internet Layer protocol IP interfaces
(defined using the InterfaceTable parameter), which is based on the definition of a
128-bit address (as opposed to 32 bits for IPv4). The following table lists IPv6-related
RFCs that are currently supported by the device (some only partially supported).
•
RFC 1981 Path MTU Discovery for IP version 6
•
RFC 2373 IP Version 6 Addressing Architecture
•
RFC 2374 IPv6 Aggregatable Global Unicast Address Format
•
RFC 2375 IPv6 Multicast Address Assignments
•
RFC 2460 Internet Protocol, Version 6 (IPv6) Specification
•
RFC 2461 Neighbor Discovery for IP Version 6 (IPv6)
•
RFC 2463 Internet Control Message Protocol (ICMPv6)
•
RFC 2464 Transmission of IPv6 Packets over Ethernet
•
RFC 2474 Definition of the Differentiated Services Field (DSField) in the IPv4 &
IPv6 Headers
•
RFC 2553 Basic Socket Interface Extensions for IPv6
•
RFC 2710 Multicast Listener Discovery (MLD) for IPv6
•
RFC 2893 Transition Mechanisms for IPv6 Hosts and Routers.
•
Note: Tunneling mechanisms are not supported.
•
RFC 3266 Support for IPv6 in Session Description Protocol (SDP)
•
RFC 3484 Default Address Selection for Internet Protocol version 6 (IPv6)
•
RFC 4193 Unique Local IPv6 Unicast Addresses
For more information on these RFC’s, contact AudioCodes.
Version 6.2
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December 2010
CPE SIP Products
1.7
Data-Router New Features
The device supports the following new data-router features:
1.
Dual WAN over T1 Interface Ports:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports two physical WAN T1 links, thereby allowing a bandwidth of
up to 3 Mbps. The T1 WAN connection is through a Dual T1 line interface (according
to ANSI T1.403-1999). The device uses its Dual T1 WAN Data Service Unit/Channel
Service Unit (DSU/CSU) port interface to transmit and receive data using IP over
Point-to-Point Protocol (PPP) framing (up to two separate links), IP over High-Level
Data Link Control (HDLC) framing (up to two separate links), or bundling both physical
links into a single logical link using IP over Multilink Point-to-Point Protocol (MLP)
framing (RFC 1717).
2.
BER Test for T1 WAN Interface:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports bit error rate (BER) testing on the T1 WAN interface for the
far-end transmission link. The BER Test is done by generating and detecting both
pseudorandom and repeating bit patterns. The test is enabled using the ber-test
CLI command.
SIP Release Notes
72
Document #: LTRT-26901
SIP Release Notes
3.
1. What's New in Release 6.2
Local-line Loopback on T1 WAN Interface:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports local-line loopback on the T1 WAN interface. In this
loopback, packets from the Tx line interface connect to the Rx line interface. The
maximum time of this loopback is user-defined. This featured is enabled and
configured using the loopback CLI command.
4.
WAN Ethernet Port Settings (Speed and Duplex Mode):
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports the configuration of link speed and duplex mode of the WAN
Ethernet port:
5.
•
Port speed: 10BaseT, 100Base-T, 1000Base-T, or autonegotiation
•
Duplex mode: half-duplex, full-duplex, or autonegotiation
IPSec Crypto CLI:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports the configuration of IPSec tunneling using "crypto" CLI
commands. The main CLI commands include the following:
•
Version 6.2
crypto isakmp key: defines a preshared authentication key
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December 2010
CPE SIP Products
6.
•
crypto isakmp policy: defines an Internet Key Exchange (IKE) policy
•
crypto ipsec transform-set: defines a transform set of acceptable
combination of security protocols and algorithms
•
crypto map: creates or modifies a crypto map entry and enters the crypto map
configuration mode
Ethernet in the First Mile (EFM) for SHDSL WAN Interface:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device’s SHDSL WAN interface now supports the EFM standard. EFM provides a
single (end-to-end) approach for transmitting Ethernet traffic between the WAN and
LAN over different topologies such as copper and fiber optic point-to-point networks.
This typically results in greater broadband access and reduces the bottleneck in the
network.
7.
Virtual Routing and Forwarding (VRF) Lite:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports Virtual Routing and Forwarding (VRF). This technology
allows multiple instances of a routing table to co-exist within the same router at the
same time. The device’s VRF feature allows interfaces to be clustered into a VRF to
provide segregated routing domains. The VRF feature uses the device’s single
physical router as multiple logical routers (up to 32). Each VRF is associated with its
own routing table. When creating fully separated logical routers on the same physical
router, every interface can be mapped to a specified VRF and static routes can be
added to it. The main CLI command for configuring VRF is ip vrf.
Note: Some features are available only on the default, unnamed, VRF. These include,
amongst others, BGP, OSPF, RIP, Management interfaces (Web, CLI and SNMP),
and SIP (when using the device’s VoIP component). For a complete list of features
supported only on the default VRF, please contact AudioCodes.
SIP Release Notes
74
Document #: LTRT-26901
SIP Release Notes
1.8
1. What's New in Release 6.2
Security New Features
The device supports the following new security feature:
1.
ARIA Encryption Algorithm for SRTP:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for the ARIA algorithm cipher encryption for SRTP. This
is an alternative option to the existing support for the AES algorithm. ARIA is a
symmetric key block cipher algorithm standard developed by the Korean National
Security Research Institute. The ARIA offered suite supports 128-bit and 192-bit key
encryption sizes with HMAC SHA-1 cryptographic hash function.
ARIA encryption is enabled by the new parameter, AriaProtocolSupport. The required
ARIA crypto suite for SRTP is according to the existing parameter, SRTPofferedSuites.
For ARIA encryption of SRTP, the device must be installed with the relevant Software
Upgrade Feature Key.
Version 6.2
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December 2010
CPE SIP Products
1.9
Infrastructure New Features
The device supports the following new infrastructure features:
1.
Enhanced Syslog Message Management using Facility Levels:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature supports the configuration of the Facility level (0 through 7) for the
device’s Syslog messages, according to RFC 3164. This feature is useful, for example,
to customers who collect the device’s and other equipments’ Syslog messages at one
single server. The device’s Syslog messages can easily be identified and distinguished
from other Syslog messages by its Facility level. Therefore, in addition to filtering
Syslog messages according to IP address, the messages can be filtered according to
Facility level.
The Facility level is configured by the new parameter, SyslogFacility, which provides
the following levels:
2.
•
16 - local use 0 (local0) - default
•
17 - local use 1 (local1)
•
18 - local use 2 (local2)
•
19 - local use 3 (local3)
•
20 - local use 4 (local4)
•
21 - local use 5 (local5)
•
22 - local use 6 (local6)
•
23 - local use 7 (local7)
Power over Ethernet (PoE) Port Status:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides Power over Ethernet (PoE) status indication when an IP Phone
is connected to one of the device’s LAN ports. This status is provided by the CLI
GetPOEPortStatusCmd command, which when run, displays the status.
SIP Release Notes
76
Document #: LTRT-26901
SIP Release Notes
1.10
1. What's New in Release 6.2
Management and Provisioning New Features
1.10.1 Web New Features
The device supports the following new Web interface features:
1.
New Web Navigation Tree:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides an improved hierarchical structure of the Configuration tab’s
menus in the Navigation tree of the Web interface. The Configuration tab is now
organized according to the following main folders:
2.
•
System: contains the system-related configuration items
•
VoIP: contains the VoIP-related configuration items
•
Data: contains the data-routing related configuration items – applicable only to
Mediant 800 MSBG and Mediant 1000 MSBG
Management Web Pages Re-design:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The Web interface’s management pages have been re-designed as follows:
Version 6.2
•
The configuration of management elements and applications (such as RADIUS,
Telnet, and SSH) are now located in the Navigation tree under the Configuration
tab in the new System menu.
•
The active alarms are now located in the Navigation tree under the Status tab in
a new folder Carrier Grade Alarms under the System menu.
•
The maintenance and software upgrade pages are now located under the
Maintenance tab.
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December 2010
CPE SIP Products
3.
Reorganization of SIP Menus in Web Navigation Tree:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides an improved hierarchical structure of the SIP-related menus and
items in the Navigation tree of the Web interface. The SIP menus and items are now
organized according to functionality and feature and therefore, can now easily be
found due to their improved logical grouping and location in the Navigation tree.
4.
Home Page Display of Device State:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides a new status field–“Gateway Operational State”–in the General
Information pane on the Web interface’s Home page. This status field indicates the
device’s operational state as follows:
•
"LOCKED" – the device is locked (i.e. no new calls are accepted)
•
"UNLOCKED" – the device is not locked
•
“SHUTTING DOWN” – the device is currently shutting down
SIP Release Notes
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Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
1.10.2 SNMP New Features
The device supports the following new Simple Network Management Protocol (SNMP)
features:
1.
ini File Configuration through SNMP:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows the configuration of ini file parameters that are not supported by the
Element Management System (EMS) tool, by using the existing acSysGenericINILine
SNMP parameter. This feature is useful for parameters that are currently not
supported by regular SNMP and where the user does not want to use the external ini
file.
To enter more than one line, the character set “$@$” must be added between lines of
ini file parameter settings.
2.
Engine ID Configuration:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature allows the configuration of the SNMP Engine ID for SNMP v2/v3 agents,
by using the new ini file parameter, SNMPEngineIDString. This parameter is
configured as a string value of up to 36 characters and needs a device reset to take
effect. The Engine ID is used for authenticating a user attempting to access the SNMP
agent on the device.
The default value is 00:00:00:00:00:00:00:00:00:00:00:00 12 hex characters. The
provided key must be set with 12 hex values delimited by a colon (“:”).
If the supplied key does not pass validation of the 12 hex values or if it is set with the
default value, the engine ID will be generated according to RFC 3411.
Note: Before setting this parameter, all SNMPv3 users must be deleted, otherwise the
configuration is ignored.
Version 6.2
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December 2010
CPE SIP Products
1.10.3 CLI New Features
The device supports the following new command-line interface (CLI) features:
1.
Main CLI Configuration Modes:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides support for three main CLI configuration modes in which the
device can be configured. These include VoIP, system, and data. To enter one of
these modes, the following commands must be run:
2.
•
configure voip (or conf v) – accesses the VoIP configuration mode
•
configure system (or conf s) – accesses the System configuration mode
•
configure data (or conf d) – accesses the Data configuration mode
CLI show Command:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides enhanced functioning of the CLI show command for displaying
the status of the device according to the three main CLI configuration modes - VoIP,
system, and data.
To view the relevant status, the following commands must be run:
•
show voip <command> - displays VoIP status according to the available
<command> command (for an explanation, run show voip ?)
•
show system <command> - displays system status according to the available
<command> command (for an explanation, run show system ?)
•
show data <command> - displays data status according to the available
<command> command (for an explanation, run show data ?)
Note: The show commands can be accessed from any configuration mode, by using
the do command, for example, do show voip <command>.
SIP Release Notes
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Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
1.10.4 SSH New Features
The device supports the following new Secure Shell (SSH) features:
1.
RSA-2048 Public-Key Encryption for SSH:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports the generation and handling of 2048-bit RSA public-key
encryption algorithms for Secure Shell (SSH). This feature is configured using the new
parameter, TLSPkeySize. Once this parameter is set to 2048, new self-signed
certificates are generated with 2048-bit keys.
2.
Maximum SSH Login Attempts:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports the configuration of the maximum allowed SSH login
attempts when entering an incorrect password by an administrator. If this maximum is
attained, the SSH session is rejected. This feature is configured by the new parameter,
SSHMaxLoginAttempts.
Version 6.2
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December 2010
CPE SIP Products
3.
SSH Last Login:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports the display in SSH sessions of the time and date of the last
SSH login. The SSH login message displays the number of unsuccessful login
attempts since the last successful login. To enable or disable this message the new
parameter, SSHEnableLastLoginMessage is used. Note that the last login information
is cleared when the device is reset.
4.
Maximum Simultaneous SSH Sessions:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
The device now supports the configuration of the maximum number of allowed
simultaneous SSH sessions (up to two). This feature is configured by the new
parameter, SSHMaxSessions.
SIP Release Notes
82
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
1.10.5 Utilities New Features
The device supports the following new utilities feature:
1.
Running DConvert from Command Line:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Management Protocol
Web
INI
SNMP
EMS
CLI
This feature provides a new software version of the TrunkPack Downloadable
Conversion (DConvert) utility which can be run from a command line. This is a
separate executable, called 'dccmd'. The 'dccmd' supports only the handling of
secured ini file and dial plan file.
Version 6.2
83
December 2010
CPE SIP Products
1.11
New Parameters
This section describes the new parameters for Release 6.2. The ini file parameter is
enclosed in square brackets and if a corresponding Web interface parameter exists, then
this parameter appears above it.
1.11.1 SIP New Parameters
This section describes the new SIP parameters for Release 6.2.
1.11.1.1 SIP General Parameters
The table below describes the new general SIP parameters for Release 6.2. These
parameters are applicable to the device’s Gateway and SBC applications.
Table 1-3: New SIP General Parameters for Release 6.2
Parameter
Description
Max SIP Message Length
[KB]
[MaxSIPMessageLength]
Defines the maximum size (in Kbytes) allowed for each SIP message
that can be sent over the network. Messages exceeding this userdefined size are rejected by the device.
The valid value range is 1 to 50. The default is 50.
Transcoding Mode
[IP2IPTranscodingMode]
Defines the voice transcoding mode (media negotiation) between two
user agents for the IP-to-IP application.
[0] Only if Required = Do not force. Many of the media settings
(such as gain control) are not implemented on the voice stream.
The device passes packets RTP to RTP without any processing.
[1] Force = Force transcoding on outgoing IP leg. The device
interworks the media by implementing DSP transcoding. (default)
Aria Protocol Support
[AriaProtocolSupport]
Enables ARIA crypto suite support.
[0] Disable = ARIA support is disabled (default).
[1] Enable = ARIA support is enabled.
For configuring the ARIA bit-key size (128 or 192 bit), use the
parameter SRTPofferedSuites.
SIP Release Notes
84
Document #: LTRT-26901
SIP Release Notes
1. What's New in Release 6.2
Parameter
Description
Static NAT Table
[NATTranslation]
Version 6.2
This ini file table parameter defines multiple NATs for SIP control and
RTP media using Static NAT Rules. This table creates NAT rules for
translating source IP address per VoIP interface (SIP control and RTP
media traffic) into NAT IP addresses. This allows, for example, the
separation of VoIP traffic between different ISTP’s, and topology
hiding (of internal IP addresses to the “public” network). Each IP
interface (configured in the Multiple Interface table - InterfaceTable
parameter) can be associated with a NAT rule in this table, translating
the source IP address and port of the outgoing packet into the NAT
address (IP address and port range).
The format of this parameter is as follows:
[ NATTranslation ]
FORMAT NATTranslation_Index =
NATTranslation_SourceIPInterfaceName,
NATTranslation_TargetIPAddress, NATTranslation_SourceStartPort,
NATTranslation_SourceEndPort, NATTranslation_TargetStartPort,
NATTranslation_TargetEndPort;
[ \NATTranslation ]
Where:
• SourceIPInterfaceName is the name of the IP interface as defined
in the Multiple Interface table.
• TargetIPAddress is the global IP address.
• TargetStartPort and TargetEndPort is the port range (1-66535) of
the global address.
• SourceStartPort and SourceEndPort is the port range (1-66535)of
the IP interface.
Note:
This table can include up to 32 indices.
The device’s priority method for performing NAT is as follows (not
relevant for SBC application):
Uses an external STUN server (STUNServerPrimaryIP parameter)
to assign a NAT address for all interfaces.
Uses the StaticNATIP parameter to define one NAT IP address for
all interfaces.
Uses the NATTranslation parameter to define NAT per interface.
If NAT is not configured (by any of the above-mentioned methods),
the device sends the packet according to its IP address defined in
the Multiple Interface table.
If the StaticNATIP parameter is not configured and the
WANIPAddress parameter is configured, then a NAT rule is
automatically added to this table with an IP address defined for
WANIPaddress and a port range of 0-65535.
85
December 2010
CPE SIP Products
1.11.1.2 SIP Gateway Parameters
The table below describes the new SIP Gateway application parameters for Release 6.2.
Table 1-4: New SIP Gateway Parameters for Release 6.2
Parameter
Description
Add Empty Authorization
Header
[EmptyAuthorizationHead
er]
Determines whether the SIP Authorization header is included in initial
registration (REGISTER) requests sent by the device.
[0] Disable (default)
[1] Enable
The Authorization header carries the credentials of a user agent (UA)
in a request to a server. The sent REGISTER message populates the
Authorization header with the following parameters:
username - set to the value of the private user identity
realm - set to the domain name of the home network
uri - set to the SIP URI of the domain name of the home network
nonce - set to an empty value
response - set to an empty value
For example:
Authorization: Digest username=alice_private@home1.net,
realm=”home1.net”, nonce=””,
response=”e56131d19580cd833064787ecc”
Note: This registration header is according to the IMS 3GPP
TS24.229 and PKT-SP-24.220 specifications.
Add initial Route Header
[InitialRouteHeader]
Determines whether the SIP Route header is included in initial
registration or re-registration (REGISTER) requests sent by the
device.
[0] Disable (default)
[1] Enable
When the device sends a REGISTER message, the Route header
includes either the Proxy's FQDN or the IP address and port according
to the configured Proxy Set, for example:
Route: <sip:10.10.10.10;lr;transport=udp>
or
Route: <sip: pcscf-gm.ims.rr.com;lr;transport=udp>
[TrunkStatusReportingMo
de]
SIP Release Notes
Determines whether the device responds to SIP OPTIONS for Trunk
Group #1.
[0] Disable (default)
[1] Enable = If all the trunks pertaining to Trunk Group #1 are down
or busy, the device does not respond to received SIP OPTIONS
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Parameter
Description
[EnableSDPVersionNegoti
ation]
This feature enables the flexibility of ignoring a new SDP re-offer (from
the media negotiation perspective) in certain scenarios (such as
session expires). According to RFC 3264, once an SDP session is
established, a new SDP offer is considered a new offer only when the
SDP origin value is incremented. In scenarios such as session
expires, SDP negotiation is irrelevant and thus, the origin field is not
changed.
Even though some SIP devices don’t follow this behavior and don’t
increment the origin value even in scenarios where they want to renegotiate, the device can now assume that the remote party operates
according to RFC 3264, and in cases where the origin field is not
incremented, the device does not re-negotiate SDP capabilities.
[0] Disable = The device negotiates any new SDP re-offer,
regardless of the origin field (default).
[1] Enable = The device negotiates only an SDP re-offer with an
incremented origin field.
[FacilityTrace]
Enables ISDN traces of Facility Information Elements (IE) for ISDN
call diagnostics. This allows you to trace all the parameters contained
in the Facility IE and view them in the Syslog.
[0] Disable (default)
[1] Enable
Note: For this feature to be functional, the GWDebugLevel parameter
must be enabled.
[PerformAdditionalIP2TEL
DestinationManipulation]
Enables additional destination number manipulation for IP-to-Tel calls.
[0] Disable (default)
[1] Enable
The additional manipulation is done on the initially manipulated
destination number, and this additional rule is also configured in the
manipulation table (NumberMapIP2Tel parameter). This enables you
to configure only a few manipulation rules for complex manipulation
(that generally require many rules).
[PerformAdditionalIP2TEL
SourceManipulation]
Enables additional source number manipulation for IP-to-Tel calls.
[0] Disable (default)
[1] Enable
The additional manipulation is done on the initially manipulated source
number, and this additional rule is also configured in the manipulation
table (SourceNumberMapIP2Tel parameter). This enables you to
configure only a few manipulation rules for complex manipulation (that
generally require many rules).
[ReplaceCallingWithRedir
ectNumber]
Enables replacing the calling number with the redirect number in
ISDN-to-IP calls. When such a replacement occurs, the calling name
is deleted and left blank. The outgoing INVITE message does not
include the redirect number that was used to replace the calling
number. The replacement is done only if a redirect number is present
in the incoming call.
[0] = Disable (default)
[1] = Enable
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Parameter
QoS statistics in SIP
Release Call
[QoSStatistics]
Description
Determines whether the device includes call quality of service (QoS)
statistics in SIP BYE and SIP 200 OK response to BYE, using the
proprietary SIP header, X-RTP-Stat.
[0] = Disable (default)
[1] = Enable
The X-RTP-Stat header provides the following statistics:
Number of received and sent voice packets
Number of received and sent voice octets
Received packet loss, jitter (in ms), and latency (in ms)
The X-RTP-Stat header contains the following fields:
PS=<voice packets sent>
OS=<voice octets sent>
PR=<voice packets received>
OR=<voice octets received>
PL=<receive packet loss>
JI=<jitter in ms>
LA=<latency in ms>
Below is an example of the X-RTP-Stat header in a SIP BYE
message:
BYE sip:302@10.33.4.125 SIP/2.0
Via: SIP/2.0/UDP 10.33.4.126;branch=z9hG4bKac2127550866
Max-Forwards: 70
From: <sip:401@10.33.4.126;user=phone>;tag=1c2113553324
To: <sip:302@company.com>;tag=1c991751121
Call-ID: 991750671245200001912@10.33.4.125
CSeq: 1 BYE
X-RTP-Stat:
PS=207;OS=49680;;PR=314;OR=50240;PL=0;JI=600;LA=40;
Supported: em,timer,replaces,path,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REF
ER,INFO,SUBSCRIBE,UPDATE
User-Agent: Sip-Gateway-/v.6.2A.008.006
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0
RTP-Only Mode
RTP Only Mode
[RTPOnlyMode]
SIP Release Notes
Enables the device to start sending and/or receiving RTP packets to
and from remote endpoints without the need to establish a SIP
session. The remote IP address is determined according to the
Outbound IP Routing table. The port is the same port as the local RTP
port (configured by the parameter BaseUDPPort and the channel on
which the call is received).
[0] Disable (default)
[1] Transmit & Receive = Send and receive RTP
[2] Transmit Only= Send RTP only
[3] Receive Only= Receive RTP only
Notes:
To configure the RTP Only mode per trunk, use the
RTPOnlyModeForTrunk_ID.
If per trunk configuration (using RTPOnlyModeForTrunk) is set to a
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Parameter
Description
value other than the default, the RTPOnlyMode parameter value is
ignored.
[RTPOnlyModeForTrunk_I
D]
Enables the device to start sending and/or receiving RTP packets to
and from remote endpoints without the need to establish a SIP
session. This is configured per trunk.
The remote IP address is determined according to the 'Outbound IP
Routing' table. The port is the same port as the local RTP port
(configured by the parameter BaseUDPPort and the channel on which
the call is received).
[-1] Not Configured = use the global parameter (RTPOnlyMode)
value for all channels (default)
[0] Disable
[1] Transmit & Receive = send and receive RTP packets
[2] Transmit Only = send RTP packets only
[3] Receive Only = receive RTP packets only
Note: The ID in the ini file parameter depicts the trunk number, where
0 is the first trunk.
AMD Parameters
Web: Answer Machine
Detector Sensitivity
Parameter Suit
[AMDSensitivityParameter
Suit]
Determines the AMD parameter suit that you want the device to use.
[0] = USA Parameter suite with normal detection sensitivity
resolution (8 sensitivity levels, from 0 to 7). The level is configured
by the AMDSensitivityLevel parameter. (default)
[1] = USA Parameter Suite with high detection sensitivity resolution
(16 sensitivity levels, from 0 to 15). The level is configured by the
AMDSensitivityLevel parameter.
[2]-[3] = Other countries parameter suites with up to 16 sensitivity
levels.
Note: This parameter replaces the existing AMDSensitivityResolution
parameter.
Web: AMD Sensitivity Level
[AMDSensitivityLevel]
Determines the AMD detection sensitivity level of the selected AMD
parameter suite.
The valid value range is 0 (for best detection of an answering
machine) to 15 (for best detection of a live call). The default value is 8.
Note: This parameter replaces the existing
AMDDetectionSensitivityHighResolution parameter.
[AMDMaxGreetingTime]
Maximum greeting time
[AMDMaxPostGreetingSile
nceTime]
Maximum duration of silence from after the greeting time is over
(defined by AMDMaxGreetingTime) until the AMD decision.
[AMDSensitivityFileName]
Name of the AMD Sensitivity auxiliary file.
[AMDSensitivityFileUrl]
URL path to the AMD Sensitivity file for download from a remote
server.
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Parameter
Description
CRLF Keep-Alive Mode Parameters
[UsePingPongKeepAlive]
Determines whether the carriage-return and line-feed sequences
(CRLF) Keep-Alive mechanism, according to RFC 5626 “Managing
Client-Initiated Connections in the Session Initiation Protocol (SIP)” is
used for reliable, connection-orientated transport types such as TCP.
[0] Disable (default)
[1] Enable
The SIP user agent/client (i.e., device) uses a simple periodic
message as a keep-alive mechanism to keep their flow to the proxy or
registrar alive (used for example, to keep NAT bindings open). For
connection-oriented transports such as TCP/TLS this is based on
CRLF. This mechanism uses a client-to-server "ping" keep-alive and a
corresponding server-to-client "pong" message. This ping-pong
sequence allows the client, and optionally the server, to tell if its flow is
still active and useful for SIP traffic. If the client does not receive a
pong in response to its ping, it declares the flow “dead” and opens a
new flow in its place. In the CRLF Keep-Alive mechanism the client
periodically sends a double-CRLF (the "ping") then waits to receive a
single CRLF (the "pong"). If the client does not receive a "pong" within
an appropriate amount of time, it considers the flow failed.
Note: The device sends a CRLF message to the Proxy Set only if the
Proxy Keep-Alive feature (EnableProxyKeepAlive parameter) is
enabled and its transport type is set to TCP or TLS. The device first
sends a SIP OPTION message to establish the TCP/TLS connection
and if it receives any SIP response, it continues sending the CRLF
keep-alive sequences.
Ping-Pong Keep-Alive Time
[PingPongKeepAliveTime]
Defines the periodic interval (in seconds) after which a “ping” (doubleCRLF) keep-alive is sent to a proxy/registrar, using the CRLF KeepAlive mechanism.
The default range is 5 to 2,000,000. The default is 120.
The device uses the range of 80-100% of this user-defined value as
the actual interval. For example, if the parameter value is set to 200
sec, the interval used is any random time between 160 to 200
seconds. This prevents an “avalanche” of keep-alive by multiple SIP
UAs to a specific server.
RTCP XR Parameters
Enable RTCP XR
[VQMonEnable]
Enables voice quality monitoring and RTCP XR, according to draft-ietfsipping-rtcp-summary-13.
[0] Disable = Disable (default)
[1] Enable = Enables
Note: For this parameter to take effect, a device reset is required.
Minimum Gap Size
[VQMonGMin]
Voice quality monitoring - minimum gap size (number of frames). The
default is 16.
Burst Threshold
[VQMonBurstHR]
Voice quality monitoring - excessive burst alert threshold. if set to -1
(default), no alerts are issued.
Delay Threshold
[VQMonDelayTHR]
Voice quality monitoring - excessive delay alert threshold. if set to -1
(default), no alerts are issued.
SIP Release Notes
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Parameter
Description
R-Value Delay Threshold
[VQMonEOCRValTHR]
Voice quality monitoring - end of call low quality alert threshold. if set
to -1 (default), no alerts are issued.
RTCP Packet Interval
[RTCPInterval]
Defines the time interval (in msec) between adjacent RTCP reports.
The interval range is 0 to 65,535. The default interval is 5,000.
Disable RTCP Interval
Randomization
[DisableRTCPRandomize]
Controls whether RTCP report intervals are randomized or whether
each report interval accords exactly to the parameter RTCPInterval.
[0] Disable = Randomize (default)
[1] Enable = No Randomize
Esc Transport Type
[RTCPXRESCTransportTy
pe]
Determines the transport layer used for outgoing SIP dialogs initiated
by the device to the RTCP XR Collection Server.
[-1] Not Configured (default)
[0] UDP
[1] TCP
[2] TLS
Note: When set to ‘Not Configured’, the value of the parameter
SIPTransportType is used.
RTCP XR Collection Server
[RTCPXREscIP]
IP address of the Event State Compositor (ESC). The device sends
RTCP XR reports to this server, using SIP PUBLISH messages. The
address can be configured as a numerical IP address or as a domain
name.
RTCP XR Report Mode
[RTCPXRReportMode]
Determines whether RTCP XR reports are sent to the Event State
Compositor (ESC), and if so, defines the interval in which they are
sent.
[0] Disable = RTCP XR reports are not sent to the ESC (default).
[1] End Call = RTCP XR reports are sent to the ESC at the end of
each call.
[2] End Call & Periodic = RTCP XR reports are sent to the ESC at
the end of each call and periodically according to the parameter
RTCPInterval.
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Parameter
Description
Cut Through Parameters
[DigitalCutThrough]
Enables PSTN CAS channels/endpoints to receive incoming IP calls
even if the B-channels are in off-hook state.
[0] Disabled (default)
[1] Enabled
When enabled, this feature operates as follows:
1 A Tel-to-IP call is established (connected) by the device for a Bchannel.
2 The device receives a SIP BYE (i.e., IP side ends the call) and
plays a reorder tone to the PSTN side for the duration set by the
CutThroughTimeForReOrderTone parameter. The device releases
the call towards the IP side (sends SIP 200 OK).
3 The PSTN side, for whatever reason, remains off-hook.
4 If a new IP call is received for this B-channel after the reorder tone
has ended, the device “cuts through” the channel and connects the
call immediately (even though the B-channel is in physical off-hook
state) without playing a ring tone. If an IP call is received while the
reorder tone is played, then the device rejects the call.
Notes:
If this parameter is disabled and the PSTN side remains in off-hook
state after the IP call ends the call, the device releases the call after
60 seconds.
A special CAS table can be used to report call status events
(Active/Idle) to the PSTN side during Cut Through mode.
[CutThroughTimeForReOr
derTone]
Defines the duration (in seconds) of the reorder tone played to the
PSTN side after the IP call party releases the call, for the Cut Through
feature. After the tone has ended, incoming call is answered
immediately if the FXS is off-hook (analog) / PSTN is connected
(digital).
The valid values are 0 to 30. The default is 0 (i.e., no reorder tone is
played).
For enabling the Cut Through feature, use the DigitalCutThrough
(CAS channels) or CutThrough (FXS channels) parameter.
1.11.1.3 SIP SAS Parameters
The table below describes the new SIP SAS application parameter for Release 6.2.
Table 1-5: New SIP SAS Parameter for Release 6.2
Parameter
SAS Block Unregistered
Users
[SASBlockUnRegUsers]
SIP Release Notes
Description
Determines whether the device rejects SIP INVITE requests received
from unregistered SAS users. This applies to SAS Normal and
Emergency modes.
[0] Un-Block =Allow INVITE from unregistered SAS users (default)
[1] Block = Reject dialog-establishment requests from un-registered
SAS users
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1.11.2 SBC New Parameters
The table below describes the new SBC application parameters for Release 6.2.
Table 1-6: New SBC Parameters for Release 6.2
Parameter
Description
SBC No Answer Timeout
[SBCAlertTimeout]
Defines the timeout (in seconds) to SBC outgoing (outbound
IP routing) SIP INVITE messages. If the called IP party does
not answer the call within this user-defined interval, the
device disconnects the session. The device starts the
timeout count upon receipt of a SIP 180 Ringing response
from the called party. If no other SIP response (for example,
200 OK) is received thereafter within this timeout, the call is
released.
The valid range is 0 to 3600 seconds. the default is 600.
SBC GRUU Mode
[SBCGruuMode]
Determines the Globally Routable User Agent (UA) URI
(GRUU) support (according to RFC 5627).
[0] None = No GRUU is supplied to users.
[1] As Proxy = The device provides same GRUU types as
the proxy provided the device’s GRUU clients. (default)
[2] Temporary only = Supply only temporary GRUU to
users. (Currently not supported.)
[3] Public only = The device provides only public GRUU to
users.
[4] Both = The device provides temporary and public
GRUU to users. (Currently not supported.)
SBC Max Forwards Limit
[SBCMaxForwardsLimit]
Defines the value of the Max-Forwards SIP header.
The valid value range is 1-70. The default is 10.
This parameter affects the Max-Forwards header in the
received message, as follows:
If the received header’s original value is 0, the message is
not passed on and is rejected.
If the received header’s original value is less than the
user-defined parameter, the header’s value is
decremented before being sent on.
If the received header’s original value is greater than the
user-defined parameter, the header’s value is replaced by
the user-defined parameter’s value
Minimum Session-Expires
[SBCMinSE]
Defines the minimum amount of time (in seconds) between
session refresh requests in a dialog before the session is
considered timed out. This value is conveyed in the SIP MinSE header.
The valid range is 0 (default) to 1,000,000 (where 0 means
that the device does not limit Session-Expires).
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Parameter
Description
SBC Registration Timers
SBC User Registration Time
[SBCUserRegistrationTime]
Defines the duration of the periodic registrations between the
user and the SBC (SBC responds with this value to user).
When set to 0, the SBC does not change the Expires value
received in the user’s REGISTER request. If no Expires
header is received in the REGISTER and
SBCUserRegistrationTime = 0, the Expires value is set to
180 seconds (by default).
The valid range is 0 to 2,000,000 seconds. The default is 0.
SBC Proxy Registration Time
[SBCProxyRegistrationTime]
Defines the duration for which the user is registered in the
proxy database (after the device forwarded the REGISTER).
When set to 0, the SBC sends the Expires value as received
from the user to the proxy.
The valid range is 0 to 2,000,000 seconds. The default is 0.
SBC Survivability Registration Time
[SBCSurvivabilityRegistrationTime]
Defines the duration of the periodic registrations between the
user and SBC, when the SBC is in survivability state (i.e.,
when REGISTER requests cannot be forwarded to the
proxy, and is terminated by the SBC). When set to 0, the
SBC uses the value of the parameter
SBCUserRegistrationTime for the SBC response.
The valid range is 0 to 2,000,000 seconds. The default is 0.
SIP Release Notes
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1.11.3 Media New Parameters
The table below describes the new media parameters for Release 6.2.
Notes: These parameters are applicable only to the Gateway application.
Table 1-7: New Media Parameters for Release 6.2
Parameter
Description
[T38FaxSessionImmediateStart]
Enables fax transmission of T.38 “no-signal” packets to the
terminating fax machine.
[0] Disable (default)
[1] Enable
This is used for transmission from fax machines (connected to
the device) located inside a Network Address Translation
(NAT). Generally, the firewall blocks T.38 (and other) packets
received from the WAN, unless the device behind NAT sends at
least one IP packet from the LAN to the WAN through the
firewall. If the firewall blocks T.38 packets sent from the
termination IP fax, the fax fails.
To overcome this, the device sends No-Op (“no-signal”) packets
to open a pinhole in the NAT for the answering fax machine.
The originating fax does not wait for an answer, but immediately
starts sending T.38 packets to the terminating fax machine.
Note: To enable No-Op packet transmission, use the
NoOpEnable and NoOpInterval parameters.
T38 Version
[SIPT38Version]
Selects the T.38 fax relay version.
[-1] Not Configured = No T.38 (default)
[0] T.38 version 0 (default)
[3] T.38 version 3 = T.38 Version 3 (V.34 over T.38 support)
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1.11.4 PSTN New Parameters
The table below describes the new PSTN parameters for Release 6.2.
Notes: These parameters are applicable only to the Gateway application.
Table 1-8: New PSTN Parameters for Release 6.2
Parameter
[ISDNNSBehaviour2]
Description
Bit-field to determine several behavior options, which influences
the behavior of the Q.931 protocol. When this bit is set to
NS_BEHAVIOUR2_ANY_UUI (0x0008), any User to User
Information Element (UUIE) is accepted for any protocol
discriminator. This feature is useful for interoperability with nonstandard switches.
Tel2IP Default Redirect Reason
[Tel2IPDefaultRedirectReason]
Default redirect reason for Tel-to-IP calls when no redirect
reason (or “unknown”) exists in the received Q931 ISDN Setup
message. The device includes this default redirect reason in the
SIP History-Info header of the outgoing INVITE.
If a redirect reason exists in the received Setup message, this
parameter is ignored and the device sends the INVITE message
with the reason according to the received Setup message. If this
parameter is not configured (-1), the outgoing INVITE is sent
with the redirect reason as received in the Setup message (if
none or “unknown” reason, then without a reason).
[-1] Not Configured (default) = Received redirect reason is
not changed
[1] Busy = Call forwarding busy
[2] No Reply = Call forwarding no reply
[9] DTE Out of Order = Call forwarding DTE out of order
[10] Deflection = Call deflection
[15] Systematic/Unconditional = Call forward unconditional
BRI-to-SIP Supplementary Services Codes for Call Forward Note: Upon receipt of an ISDN Facility
message for call forward from the BRI phone, the device sends a SIP INVITE to the softswitch with a
user-defined code in the SIP To header, representing the reason for the call forward.
Call Forward Unconditional
[SuppServCodeCFU]
Prefix code for activating Call Forward Unconditional sent to the
softswitch.
The valid value is a string. The default is an empty string.
Note: The string must be enclosed in single apostrophe (e.g.,
‘*72’).
Call Forward Unconditional
Deactivation
[SuppServCodeCFUDeact]
Prefix code for deactivating Call Forward Unconditional
Deactivation sent to the softswitch.
The valid value is a string. The default is an empty string.
Note: The string must be enclosed in single apostrophe (e.g.,
‘*72’).
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Parameter
Description
Call Forward on Busy
[SuppServCodeCFB]
Prefix code for activating Call Forward on Busy sent to the
softswitch.
The valid value is a string. The default is an empty string.
Note: The string must be enclosed in single apostrophe (e.g.,
‘*72’).
Call Forward on Busy Deactivation
[SuppServCodeCFBDeact]
Prefix code for deactivating Call Forward on Busy Deactivation
sent to the softswitch.
The valid value is a string. The default is an empty string.
Note: The string must be enclosed in single apostrophe (e.g.,
‘*72’).
Call Forward on No Reply
[SuppServCodeCFNR]
Prefix code for activating Call Forward on No Reply sent to the
softswitch.
The valid value is a string. The default is an empty string.
Note: The string must be enclosed in single apostrophe (e.g.,
‘*72’).
Call Forward on No Reply
Deactivation
[SuppServCodeCFNRDeact]
Prefix code for deactivating Call Forward on No Reply
Deactivation sent to the softswitch.
The valid value is a string. The default is an empty string.
Note: The string must be enclosed in single apostrophe (e.g.,
‘*72’).
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Parameter
Description
BRI Supplementary Services Table
[ISDNSuppServ]
SIP Release Notes
This table configures the BRI endpoints and their authentication
and caller ID settings.
[ ISDNSuppServ ]
FORMAT ISDNSuppServ_Index =
ISDNSuppServ_PhoneNumber, ISDNSuppServ_Module,
ISDNSuppServ_Port, ISDNSuppServ_UserId,
ISDNSuppServ_UserPassword, ISDNSuppServ_CallerID,
ISDNSuppServ_IsPresentationRestricted,
ISDNSuppServ_IsCallerIDEnabled;
[ \ISDNSuppServ ]
Where:
PhoneNumber = The telephone extension number for the BRI
endpoint.
Module = The BRI module number to which the BRI
extension pertains.
Port = The port number (on the BRI module) to which the BRI
extension is connected.
UserID = User ID for registering the BRI endpoint to a thirdparty softswitch for authentication and/or billing.
UserPassword = User password for registering the BRI
endpoint to a third-party softswitch for authentication and/or
billing. Note: For security, the password is displayed as an
asterisk (*).
CallerID = Caller ID name of the BRI extension (sent to the IP
side). The valid value is a string of up to 18 characters.
IsPresentationRestricted = Determines whether the BRI
extension sends its Caller ID information to the IP when a call
is made.
[0] Allowed = The device sends the string defined in the
'Caller ID' field when this BRI extension makes a Tel-toIP call.
[1] Restricted = The string defined in the 'Caller ID' field
is not sent.
IsCallerIDEnabled = Enables the receipt of Caller ID.
[0] Disabled = The device does not send Caller ID
information to the BRI extension.
[1] Enabled = The device sends Caller ID information to
the BRI extension
For example:
ISDNSuppServ 0 = 400, 1, 1, user, pass, callerid, 0, 1;
ISDNSuppServ 1 = 401, 1, 1, user, pass, callerid, 0, 1;
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1.11.5 Networking New Parameters
The table below describes the new networking parameters for Release 6.2.
Notes: These parameters are applicable to the Gateway and SBC applications.
Table 1-9: New Networking Parameters for Release 6.2
Parameter
Description
[TLSPkeySize]
Defines key size (in bits) for RSA public-key encryption for newly
self-signed generated keys.
[512]
[768]
[1024] (default)
[2048]
[SSHMaxLoginAttempts]
Maximum SSH login attempts allowed for entering an incorrect
password by an administrator before the SSH session is
rejected.
The valid range is 1 to 3. the default is 3.
[SSHEnableLastLoginMessage]
Enables or disables the message display in SSH sessions of the
time and date of the last SSH login. The SSH login message
displays the number of unsuccessful login attempts since the last
successful login.
[0] Disable
[1] Enable (default)
Note: The last SSH login information is cleared when the device
is reset.
[SSHMaxSessions]
Maximum number of simultaneous SSH sessions.
The valid range is 1 to 2. The default is 2 sessions.
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Parameter
Description
[StaticRouteTable]
This ini file table parameter configures the IP Routing table for
defining static IP routing rules. These rules can be associated
with IP interfaces defined in the Multiple Interface table. The
routing decision is thus based on the source subnet/VLAN. If not
associated with an IP interface, the static IP rule is based on
destination IP address.
The format of this table is as follows:
[ StaticRouteTable ]
FORMAT StaticRouteTable_Index =
StaticRouteTable_InterfaceName, StaticRouteTable_Destination,
StaticRouteTable_PrefixLength, StaticRouteTable_Gateway,
StaticRouteTable_Description;
[ \StaticRouteTable ]
Where:
InterfaceName = IP network interface associated with this
routing rule (index of the network interface as defined in the
Multiple Interface table)
Destination = IP address of the destination host/network
PrefixLength = Subnet mask of the destination host/network
Gateway = IP address of the router (next hop) to which the
packets are sent if their destination matches the rules in the
adjacent columns
Description = Arbitrary description of this rule
Syslog Facility Number
[SyslogFacility]
Facility level (0 through 7) for the device’s Syslog messages,
according to RFC 3164. This allows the user to identify Syslog
messages generated by the device.
This is useful, for example, if you collect the device’s and other
equipments’ Syslog messages, at one single server. The
device’s Syslog messages can easily be identified and
distinguished from other Syslog messages by its Facility level.
Therefore, in addition to filtering Syslog messages according to
IP address, the messages can be filtered according to Facility
level.
16 - local use 0 (local0) - default
17 - local use 1 (local1)
18 - local use 2 (local2)
19 - local use 3 (local3)
20 - local use 4 (local4)
21 - local use 5 (local5)
22 - local use 6 (local6)
23 - local use 7 (local7)
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1. What's New in Release 6.2
1.11.6 Management and Provisioning New Parameters
The table below describes the new management and provisioning parameter for Release
6.2.
Notes: These parameters are applicable to the Gateway and SBC applications.
Table 1-10: New Management and Provisioning Parameter for Release 6.2
Parameter
[SNMPEngineIDString]-
Version 6.2
Description
Defines the SNMP engine ID for SNMPv3/SNMPv2 agents. This is
used for authenticating a user attempting to access the agent on the
device.
The ID can be a string of up to 36 characters. The default value is
00:00:00:00:00:00:00:00:00:00:00:00 (12 Hex octets characters). The
provided key must be set with 12 Hex values delimited by a colon (":")
in the format xx:xx:...:xx. For example,
00:11:22:33:44:55:66:77:88:99:aa:bb
Notes:
For this parameter to take effect, a device reset is required.
Before setting this parameter, all SNMPv3 users must be deleted;
otherwise, the parameter setting is ignored.
If the supplied key does not pass validation of the 12 Hex values
input or it is set with the default value, the engine ID is generated
according to RFC 3411.
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1.12
Modified Parameters
This section describes parameters from the previous release that have been modified in
Release 6.2.
1.12.1 SIP Parameters
This section describes SIP parameters from the previous release that have been modified
in Release 6.2.
1.12.1.1 SIP General Parameters
The table below describes the SIP general parameters from the previous release that have
been modified in Release 6.2.
Table 1-11: Modified SIP General Parameters for Release 6.2
Parameter
Coders Group Table
[CodersGroup0]
[CodersGroup1]
[CodersGroup2]
[CodersGroup3]
[CodersGroup4]
Description
Modifications:
WB-RTA (Mediant 3000)
G.722 (Mediant 800 MSBG) AMR-WB (Mediant 800 MSBG)
Transparent ptime (Mediant 600, Mediant 1000, Mediant 1000
MSBG, Mediant 2000)
The format of this parameter is as follows:
[ CodersGroup0]
FORMAT CodersGroup0_Index = CodersGroup0_Name,
CodersGroup0_pTime, CodersGroup0_rate,
CodersGroup0_PayloadType, CodersGroup0_Sce;
[ \CodersGroup0 ]
Coder Name
SIP Release Notes
Packetization
Time (msec)
Rate (kbps)
Payload
Type
Silence
Suppression
WB-MS-RTA
[ms_rta_wb]
20 ms
29000
114
n/a
G.722
[g722]
20 (default),
40, 60, 80, 100,
120
64 (default)
Always 9
N/A
AMR-WB
[Amr-WB]
20 (default)
6.6 [0],
8.85 [1],
12.65 [2],
14.25 [3],
15.85 [4],
18.25 [5],
19.85 [6],
23.05 [7],
23.85 [8]
(default)
Dynamic (0127)
Disable [0]
Transparent
[Transparent]
10, 20
(default), 40,
60, 80, 100,
120
Always 64
Dynamic (0127)
Disable [0]
102
Enable [1]
Enable [1]
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SIP Release Notes
1. What's New in Release 6.2
Parameter
Description
Proxy Set Table
[ProxySet]
Modifications:
ClassificationInput (Mediant 800 MSBG, Mediant 1000 MSBG,
Mediant 3000)
ProxyRedundancyMode (All Products)
The format of this parameter is as follows:
[ProxySet]
FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive,
ProxySet_ProxyKeepAliveTime,
ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap,
ProxySet_SRD, ProxySet_ClassificationInput,
ProxySet_ProxyRedundancyMode;
[\ProxySet]
Where:
ClassificationInput = classifies an IP call to a Proxy Set based on
either its IP address, or its IP address, port, and transport type
(e.g., UDP:
[0] Compare only IP (default)
[1] Compare IP, port and transport type
ProxyRedundancyMode = defines the Proxy Redundancy feature
per Proxy Set:
[-1] = Not configured – the “global” parameter
ProxyRedundancyMode applies (default).
[0] Parking = The device continues operating with a redundant
(now active) Proxy until the next failure, after which it operates
with the next redundant Proxy.
[1] Homing = The device always attempts to operate with the
primary Proxy server (i.e., returns to the primary Proxy
whenever available).
Note: If Proxy Redundancy mode is configured per Proxy Set, then
the global ProxyRedundancyMode parameter is ignored.
IP Security
[SecureCallsFromIP]
Modification: Option [2] “Secure All Calls” added (all products).
Determines the device’s handling (allowing or rejecting) of the receipt
of IP calls. This feature is useful for preventing unwanted SIP calls,
SIP messages, and/or VoIP spam.
[0] Disable = The device accepts all SIP calls (default).
[1] Secure Incoming Calls = The device accepts SIP calls only
from IP addresses defined in the 'Outbound IP Routing’ table/’Tel
to IP Routing' or Proxy Set table, or IP addresses resolved from
DNS servers from FQDN which appear in the Proxy Set table. All
other incoming calls are rejected.
[2] Secure All Calls = The device accepts SIP calls only from IP
addresses (in dotted-decimal notation format) defined in the
'Outbound IP Routing’ table or Proxy Set table, and rejects all other
incoming calls. In addition, if a fully-qualified domain name (FQDN)
is defined in the routing table or Proxy Set table, the call is allowed
to be sent only if the resolved DNS IP address appears in one of
these table; otherwise it is rejected.
Note: If the parameter is set to [2], when using Proxies or Proxy Sets,
it’s unnecessary to configure the Proxy IP addresses in the routing
table. The device allows SIP calls received from the Proxy IP
addresses even if these are not configured in the routing table.
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Parameter
Description
Outbound IP Routing Table
[Prefix]
Modification: PREFIX_DestSRD added.
This ini file table parameter configures the Outbound IP Routing table
for routing Tel-to-IP and IP-to-IP calls. The format of this parameter is
as follows:
[PREFIX]
FORMAT PREFIX_Index = PREFIX_DestinationPrefix,
PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId,
PREFIX_MeteringCode, PREFIX_DestPort, PREFIX_SrcIPGroupID,
PREFIX_DestHostPrefix, PREFIX_DestIPGroupID,
PREFIX_SrcHostPrefix, PREFIX_TransportType,
PREFIX_SrcTrunkGroupID, PREFIX_DestSRD;
[\PREFIX]
1.12.1.2 SIP Gateway Parameters
The table below describes the SIP Gateway parameters from the previous release that
have been modified in Release 6.2.
Table 1-12: Modified SIP Gateway Parameters for Release 6.2
Parameter
Description
Automatic Dialing Table
[TargetOfChannel]
Modification: HotLineToneDuration parameter added (MP-1xx,
Mediant 600, Mediant 1000, Mediant 1000 MSBG).
[TargetOfChannel]
FORMAT TargetOfChannel_Index = TargetOfChannel_Destination,
TargetOfChannel_Type, TargetOfChannel_HotLineToneDuration,
TargetOfChannel_Module, TargetOfChannel_Port;
[\TargetOfChannel]
Where:
HotLineToneDuration = defines the timeout per port (channel) for
automatic dialing when the Hotline feature is enabled. If the phone
(connected to the specific port) is off-hooked and no digit is pressed
for this user-defined duration (timeout), the device automatically
initiates a call to the user-defined destination phone number. This
parameter has a valid value range of 0 to 60 seconds, with default
as 16.
Tone Index Table
[ToneIndex]
Modification: ToneIndex_DestinationPrefix parameter added (MP1xx, Mediant 600, Mediant 800 MSBG, Mediant 1000 MSBG, Mediant
1000).
[ToneIndex]
FORMAT ToneIndex_Index = ToneIndex_FXSPort_First,
ToneIndex_FXSPort_Last, ToneIndex_SourcePrefix,
ToneIndex_DestinationPrefix, ToneIndex_PriorityIndex;
[\ToneIndex]
Where:
DestinationPrefix = destination (called) number (or prefix) for
associating a distinctive ringing pattern and call waiting tone for IPto-Tel calls (FXS interfaces).
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Parameter
Description
Trunk Group Settings Table
[TrunkGroupSettings]
Modifications: New ChannelSelectMode options:
[9] Ring to Hunt Group (MP-1xx, Mediant 600, Mediant 800 MSBG,
Mediant 1000 MSBG, Mediant 1000) [10] Select Trunk by ISDN
Supplementary Services Table (Mediant 600, Mediant 800 MSBG,
Mediant 1000 MSBG, Mediant 1000)
[TrunkGroupSettings]
FORMAT TrunkGroupSettings_Index =
TrunkGroupSettings_TrunkGroupId,
TrunkGroupSettings_ChannelSelectMode,
TrunkGroupSettings_RegistrationMode,
TrunkGroupSettings_GatewayName,
TrunkGroupSettings_ContactUser,
TrunkGroupSettings_ServingIPGroup,
TrunkGroupSettings_MWIInterrogationType;
[\TrunkGroupSettings]
Where ChannelSelectMode defines the method for allocating
incoming IP-to-Tel calls to a channel (port):
[0] By Dest Phone Number = Selects the device's channel
according to the called number (default.)
[1] Cyclic Ascending = Selects the next available channel in an
ascending cyclic order. Always selects the next higher channel
number in the Trunk Group. When the device reaches the highest
channel number in the Trunk Group, it selects the lowest channel
number in the Trunk Group and then starts ascending again.
[2] Ascending = Selects the lowest available channel. It always
starts at the lowest channel number in the Trunk Group and if that
channel is unavailable, selects the next higher channel.
[3] Cyclic Descending = Selects the next available channel in
descending cyclic order. It always selects the next lower channel
number in the Trunk Group. When the device reaches the lowest
channel number in the Trunk Group, it selects the highest channel
number in the Trunk Group and then starts descending again.
[4] Descending = Selects the highest available channel. It always
starts at the highest channel number in the Trunk Group and if that
channel is unavailable, selects the next lower channel.
[5] Dest Number + Cyclic Ascending = The device first selects the
channel according to the called number. If the called number isn't
found, it then selects the next available channel in ascending cyclic
order. Note that if the called number is found but the port
associated with this number is busy, the call is released.
[6] By Source Phone Number = The device selects the channel
according to the calling number.
[7] Trunk Cyclic Ascending = The device selects the channel from
the first channel of the next trunk (adjacent to the trunk from which
the previous channel was allocated). This option is applicable only
to digital interfaces.
[8] Trunk & Channel Cyclic Ascending = The device implements the
Trunk Cyclic Ascending and Cyclic Ascending methods to select the
channel. This method selects the next physical trunk (pertaining to
the Trunk Group) and then selects the B-channel of this trunk
according to the cyclic ascending method (i.e., selects the channel
after the last allocated channel). This option is applicable only to
digital interfaces.
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Parameter
Description
[9] Ring to Hunt Group = The device allocates IP-to-Tel calls to all
the FXS ports (channels) belonging to a specific Hunt Group. When
an IP-to-Tel call is received by the device for a specific Hunt Group,
all telephones connected to the FXS ports belonging to the Hunt
Group start ringing. The call is eventually received by whichever
telephone answers the call first (and the other phones will stop
ringing).
[10] Select Trunk by ISDN SuppServ Table = The device selects
the BRI port/module according to the settings in the ISDN
Supplementary Services table.
IP Profile Table
[IPProfile]
Modifications:
AMD parameters added: AMDSensitivityParameterSuit,
AMDSensitivityLevel, AMDMaxGreetingTime,
AMDMaxPostGreetingSilenceTime (Mediant 600, Mediant 1000
MSBG, Mediant 1000, Mediant 2000, Mediant 3000)
SBCDiversionMode, SBCHistoryInfoMode (Mediant 800 MSBG,
Mediant 1000 MSBG, Mediant 3000)
[IPProfile]
FORMAT IPProfile_Index = IPProfile_ProfileName,
IPProfile_IpPreference, IPProfile_CodersGroupID,
IPProfile_IsFaxUsed, IPProfile_JitterBufMinDelay,
IPProfile_JitterBufOptFactor, IPProfile_IPDiffServ,
IPProfile_SigIPDiffServ, IpProfile_SCE,
IPProfile_RTPRedundancyDepth, IPProfile_RemoteBaseUDPPort,
IPProfile_CNGmode, IPProfile_VxxTransportType,
IPProfile_NSEMode, IpProfile_IsDTMFUsed,
IPProfile_PlayRBTone2IP, IPProfile_EnableEarlyMedia,
IPProfile_ProgressIndicator2IP, IPProfile_EnableEchoCanceller,
IPProfile_CopyDest2RedirectNumber,
IPProfile_MediaSecurityBehaviour, IPProfile_CallLimit, IPProfile_
DisconnectOnBrokenConnection, IPProfile_FirstTxDtmfOption,
IPProfile_SecondTxDtmfOption, IPProfile_RxDTMFOption,
IpProfile_EnableHold, IpProfile_InputGain, IpProfile_VoiceVolume,
IpProfile_AddIEInSetup, IpProfile_SBCExtensionCodersGroupID,
IPProfile_MediaIPVersionPreference, IPProfile_TranscodingMode,
IpProfile_SBCAllowedCodersGroupID,
IpProfile_SBCAllowedCodersMode,
IpProfile_SBCMediaSecurityBehaviour,
IpProfile_SBCRFC2833Behavior,
IpProfile_SBCAlternativeDTMFMethod, IpProfile_SBCAssertIdentity,
IpProfile_AMDSensitivityParameterSuit,
IpProfile_AMDSensitivityLevel, IpProfile_AMDMaxGreetingTime,
IpProfile_AMDMaxPostSilenceGreetingTime,
IpProfile_SBCDiversionMode, IpProfile_SBCHistoryInfoMode;
[\IPProfile]
Call Priority Mode
[CallPriorityMode]
Modification: Option [3] “Emergency” added (all products).
Enables priority call handling.
[0] Disable = Disable (default).
[1] MLPP = MLPP Priority Calls handling is enabled.
[3] Emergency = Pre-emption of IP-to-Tel E911 emergency calls. If
the device receives an E911 call and there are currently no
available channels, the device terminates one of the channel calls
and sends the E911 call.
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Parameter
Description
Flash Keys Sequence Style
[FlashKeysSequenceStyle
]
Modification: Option [2] added (MP-1xx, Mediant 600, Mediant 800
MSBG, Mediant 1000 MSBG, Mediant 1000).
Hook flash keys sequence style.
[0] 0 = Flash hook (default) - only the phone's Flash button is used,
according to the following scenarios:
During an existing call, if the user presses Flash, the call is put
on hold; a dial tone is heard and the user is able to initiate a
second call. Once the second call is established, on-hooking
transfers the first (held) call to the second call.
During an existing call, if a call comes in (call waiting), pressing
Flash places the active call on hold and answers the waiting
call; pressing Flash again toggles between these two calls.
[1] 1 = Sequence of Flash Hook and digit:
Flash + 1: holds a call or toggles between two existing calls
Flash + 2: makes a call transfer.
Flash + 3: makes a three-way conference call (if the ThreeWay Conference feature is enabled, i.e., the parameter
Enable3WayConference is set to 1 and the parameter
3WayConferenceMode is set to 2).
[2] 2 = Sequence of Flash Hook and digit:
Flash Hook only: places a call on hold.
Flash + 2: places a call on hold and answers a call-waiting call,
or toggles between active and on-hold calls.
Flash + 3: makes a three-way conference call (if the parameter
Enable3WayConference is set to 1 and the parameter
3WayConferenceMode is set to 2). Note that the settings of the
ConferenceCode parameter are ignored.
Flash + 4: makes a call transfer.
3-Way Conference Mode
[3WayConferenceMode]
Modification: Option [2] On-Board added (Mediant 800 MSBG,
Mediant 1000 MSBG, Mediant 1000).
Defines the mode of operation when the 3-Way Conference feature is
used.
[0] AudioCodes Media Server = The Conference-initiating INVITE
(sent by the device) uses the ConferenceID concatenated with a
unique identifier as the Request-URI. This same Request-URI is set
as the Refer-To header value in the REFER messages that are sent
to the two remote parties. This conference mode is used when
operating with AudioCodes IPMedia conferencing server. (Default)
[1] Non-AudioCodes Media Server = The Conference-initiating
INVITE (sent by the device) uses only the ConferenceID as the
Request-URI. The conference server sets the Contact header of the
200 OK response to the actual unique identifier (Conference URI) to
be used by the participants. This Conference URI is then included
(by the device) in the Refer-To header value in the REFER
messages sent by the device to the remote parties. The remote
parties join the conference by sending INVITE messages to the
conference using this conference URI.
[2] On Board = On-board 3-way conference. The conference is
established on the device without the need for an external
Conference server. The device sets up the call conference using its
IP media channels. These channels are obtained from the IP media
module (i.e., MPM module). Note that the MPM module(s) must be
installed to support three-way conferencing. You can limit the
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Parameter
Description
number of simultaneous, on-board 3-way conference calls, by using
the parameter MaxInBoardConferenceCalls.
Notes:
This parameter is applicable only to FXS interfaces.
When using an external conference server (i.e., [0] or [1]), up to six
three-way conference calls is supported.
T38 Fax Max Buffer
[T38FaxMaxBufferSize]
Modification: Value range and default (all products).
Defines the maximum size (in bytes) of the device's T.38 buffer. This
value is included in the outgoing SDP when T.38 is used for fax relay
over IP.
The valid range is 500 to 3,000.
The default value is:
1024 - MP-1xx, Mediant 1000 MSBG, Mediant 1000, Mediant 2000
3000 - Mediant 800 MSBG, Mediant 3000
T.38 Max Datagram Size
[T38MaxDatagramSize]
Modification: Value range and default (all products).
Defines the maximum size (in bytes) of a T.38 datagram that the
device can receive. This value is included in the outgoing SDP when
T.38 is used.
The valid range is 120 to 600.
The default value is:
238 - MP-1xx, Mediant 1000 MSBG, Mediant 1000, Mediant 2000
560 - Mediant 800 MSBG, Mediant 3000
SRTP offered Suites
[SRTPofferedSuites]
Modification: Options [4] and [8] added (Mediant 3000).
Defines the offered SRTP crypto suites.
[0] All = All available crypto suites (default)
[1] AES_CM_128_HMAC_SHA1_80 = device uses AES-CM
encryption with a 128-bit key and HMAC-SHA1 message
authentication with a 80-bit tag.
[2] AES_CM_128_HMAC_SHA1_32 = device uses AES-CM
encryption with a 128-bit key and HMAC-SHA1 message
authentication with a 32-bit tag.
[4] ARIA_128_BIT = device uses ARIA_128 encryption with a 128bit key and HMAC-SHA1 message authentication with a 32-bit tag.
[8] ARIA_192_BIT = device uses ARIA_192 encryption with a 192bit key and HMAC-SHA1 message authentication with a 32-bit tag.
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1. What's New in Release 6.2
Parameter
Description
Voice Mail Interface
[VoiceMailInterface]
Modification: Option [8] ETS added (Mediant 600, Mediant 800
MSBG, Mediant 1000 MSBG, Mediant 1000).
Enables the device's Voice Mail application and determines the
communication method used between the PBX and the device.
[0] None (default)
[1] DTMF
[2] SMDI (N/A)
[3] QSIG
[4] SETUP Only = For ISDN
[5] MATRA/AASTRA QSIG
[6] QSIG SIEMENS = QSIG MWI activate and deactivate
messages include Siemens Manufacturer Specific Information
(MSI)
[7] IP2IP = The device's IP2IP application is used for interworking
between an IP Voice Mail server and the device. This is
implemented for sending unsolicited SIP NOTIFY messages
received from the Voice Mail server to an IP Group (configured
using the parameter NotificationIPGroupID).
[8] ETSI = Euro ISDN, according to ETS 300 745-1 V1.2.4, section
9.5.1.1. Enables MWI interworking from IP to Tel, typically used for
BRI phones.
MLPP Default Namespace
[MLPPDefaultNamespace]
Modification: Option [5] UC added (all except MP-1xx).
Determines the Namespace used for MLPP calls received from the
ISDN side and destined for the Application server. The Namespace
value is not present in the Precedence IE of the PRI Setup message.
Therefore, the value is used in the Resource-Priority header of the
outgoing SIP INVITE request.
[1] DSN = DSN (default)
[2] DOD = DOD
[3] DRSN = DRSN
[5] UC = UC
Use Tgrp Information
[UseSIPTgrp]
Modification: Option [4] Hotline Extended added (all except MP-1xx).
Determines whether the SIP 'tgrp' parameter is used. This SIP
parameter specifies the Trunk Group to which the call belongs
(according to RFC 4904). For example, the SIP message below
indicates that the call belongs to Trunk Group ID 1:
INVITE sip::+16305550100;tgrp=1;trunkcontext=example.com@10.1.0.3;user=phone SIP/2.0
[0] Disable (default) = The 'tgrp' parameter isn't used.
[1] Send Only = The Trunk Group number is added to the 'tgrp'
parameter value in the Contact header of outgoing SIP messages.
If a Trunk Group number is not associated with the call, the 'tgrp'
parameter isn't included. If a 'tgrp' value is specified in incoming
messages, it is ignored.
[2] Send and Receive = The functionality of outgoing SIP
messages is identical to the functionality described in option 1. In
addition, for incoming SIP INVITEs, if the Request-URI includes a
'tgrp' parameter, the device routes the call according to that value
(if possible). The Contact header in the outgoing SIP INVITE (Telto-IP call) contains “tgrp=<source trunk group ID>;trunk-
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Parameter
Description
context=<gateway IP address>”. The <source trunk group ID> is
the Trunk Group ID where incoming calls from Tel is received. For
IP-Tel calls, the SIP 200 OK device's response contains
“tgrp=<destination trunk group ID>;trunk-context=<gateway IP
address>”. The <destination trunk group ID> is the Trunk Group ID
used for outgoing Tel calls. The <gateway IP address> in “trunkcontext” can be configured using the parameter SIPGatewayName.
[3] UCR 2008 = Interworks the hotline "Off Hook Indicator"
parameter between SIP and ISDN:
For IP-to-ISDN calls:
- The device interworks the SIP tgrp=hotline parameter
(received in INVITE) to ISDN Setup with the Off Hook Indicator
IE of “Voice”, and “Speech” Bearer Capability IE. Note that the
Off Hook Indicator IE is described in UCR 2008 specifications.
- The device interworks the SIP tgrp=hotline-ccdata parameter
(received in INVITE) to ISDN Setup with an Off Hook Indicator
IE of “Data”, and with “Unrestricted 64k” Bearer Capability IE.
The following is an example of the INVITE with tgrp=hotlineccdata:
INVITE sip:1234567;tgrp=hotline-ccdata;trunkcontext=dsn.mil@example.com
For ISDN-to-IP calls:
- The device interworks ISDN Setup with an Off Hook
Indicator of “Voice” to SIP INVITE with “tgrp=hotline;trunkcontext=dsn.mil” in the Contact header.
- The device interworks ISDN Setup with an Off Hook indicator
of “Data” to SIP INVITE with “tgrp=hotline-ccdata;trunkcontext=dsn.mil” in the Contact header.
- If ISDN Setup does not contain an Off Hook Indicator IE and
the Bearer Capability IE contains “Unrestricted 64k”, the
outgoing INVITE includes “tgrp=ccdata;trunk-context=dsn.mil”.
If the Bearer Capability IE contains “Speech”, the INVITE in
this case does not contain tgrp and trunk-context parameters.
[4] Hotline Extended = Interworks the ISDN Setup message’s
hotline "OffHook Indicator" Information Element (IE) to SIP
INVITE’s Request-URI and Contact headers. (Note: For IP-toISDN calls, the device handles the call as described for option [3].)
The device interworks ISDN Setup with an Off Hook Indicator
of “Voice” to SIP INVITE Request-URI and Contact header
with “tgrp=hotline;trunk-context=dsn.mil”.
The device interworks ISDN Setup with an Off Hook indicator
of “Data” to SIP INVITE Request-URI and Contact header with
“tgrp=hotline-ccdata;trunk-context=dsn.mil”.
If ISDN Setup does not contain an Off Hook Indicator IE and
the Bearer Capability IE contains “Unrestricted 64k”, the
outgoing INVITE Request-URI and Contact header includes
“tgrp=ccdata;trunk-context=dsn.mil”. If the Bearer Capability IE
contains “Speech”, the INVITE in this case does not contain
tgrp and trunk-context parameters.
Note: IP-to-Tel configuration (using the parameter PSTNPrefix)
overrides the 'tgrp' parameter in incoming INVITE messages.
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1.12.1.3 SIP SAS Parameters
The table below describes the SIP SAS parameters from the previous release that have
been modified in Release 6.2.
Table 1-13: Modified SIP SAS Parameters for Release 6.2
Parameter
SAS Survivability Mode
[SASSurvivabilityMode]
Version 6.2
Description
Modification: Option [4] Use IP-to-IP Routing Table added (all
products).
Determines the Survivability mode used by the SAS application.
[0] Standard = Incoming INVITE and REGISTER requests are
forwarded to the defined Proxy list of SASProxySet in Normal mode
and handled by the SAS application in Emergency mode (default).
[1] Always Emergency = The SAS application does not use KeepAlive messages towards the SASProxySet, instead it always
operates in Emergency mode (as if no Proxy in the SASProxySet is
available).
[2] Ignore Register = Use regular SAS Normal/Emergency logic
(same as option [0]), but when in Normal mode incoming
REGISTER requests are ignored.
[3] Auto-answer REGISTER = When in Normal mode, the device
responds to received REGISTER requests by sending a SIP 200
OK (instead of relaying the registration requests to a Proxy), and
enters the registrations in its SAS database.
[4] Use Routing Table only in Normal mode = The device uses the
IP-to-IP Routing table to route IP-to-IP SAS calls only when in SAS
Normal mode (and is unavailable when SAS is in Emergency
mode).
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1.12.2 SBC Parameters
The table below describes SBC parameters from the previous release that have been
modified in Release 6.2.
Table 1-14: Modified SBC Parameter for Release 6.2
Parameter
Description
IP Group Table
[IPGroup]
Modification: Option [2] for IPGroup_Type (Mediant 800 MSBG,
Mediant 1000 MSBG, Mediant 3000); Option [2] for
IPGroup_EnableSurvivability.
[IPGroup]
FORMAT IPGroup_Index = IPGroup_Type, IPGroup_Description,
IPGroup_ProxySetId, IPGroup_SIPGroupName,
IPGroup_ContactUser, IPGroup_EnableSurvivability,
IPGroup_ServingIPGroup, IPGroup_SipReRoutingMode,
IPGroup_AlwaysUseRouteTable, IPGroup_RoutingMode,
IPGroup_SRD, IPGroup_MediaRealm, IPGroup_ClassifyByProxySet,
IPGroup_ProfileId, IPGroup_MaxNumOfRegUsers,
IPGroup_InboundManSet, IPGroup_OutboundManSet;
[\IPGroup]
Where:
IPGroup_Type:
[0] SERVER
[1] USER
[2] GATEWAY
IPGroup_EnableSurvivability:
[0] Disable (default)
[1] Enable if Necessary
[2] Always Enable = Survivability mode is always enabled. The
communication with the Serving IP Group (e.g., IP-PBX) is
always considered as failed. The device uses its database for
routing calls between the clients (e.g., IP phones) of the USERtype IP Group.
Admission Control Table
[SBCAdmissionControl]
Modification: Rate and MaxBurst parameters added (Mediant 800
MSBG, Mediant 1000 MSBG).
[SBCAdmissionControl]
FORMAT SBCAdmissionControl_Index =
SBCAdmissionControl_LimitType, SBCAdmissionControl_IPGroupID,
SBCAdmissionControl_SRDID, SBCAdmissionControl_RequestType,
SBCAdmissionControl_RequestDirection,
SBCAdmissionControl_Limit, SBCAdmissionControl_LimitPerUser,
SBCAdmissionControl_Rate, SBCAdmissionControl_MaxBurst;
[\SBCAdmissionControl]
Where:
Rate = rate at which tokens are added to the bucket (i.e., token
rate). One token is added to the bucket every 1000/Rate
milliseconds. In other words, this is the rate of dialog setups per
second, or unlimited if set to 0 (default).
MaxBurst = Maximum tokens that can fill the bucket. At any given
time, the bucket cannot contain more than this amount of tokens. In
other words, this is the maximum burst size for the dialog setup
rate, or unlimited if set to 0 (default).
SIP Release Notes
112
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SIP Release Notes
1. What's New in Release 6.2
1.12.3 Networking Parameters
The table below describes networking parameters from the previous release that have been
modified in Release 6.2.
Notes: These parameters are applicable to the Gateway and SBC applications.
Table 1-15: Modified Networking Parameter for Release 6.2
Parameter
Firewall Settings
[AccessList]
Version 6.2
Description
Modification: Use_Specific_Interface and AccessList_Interface_ID
parameters added (all products).
[AccessList]
FORMAT AccessList_Index = AccessList_Source_IP,
AccessList_PrefixLen, AccessList_Start_Port, AccessList_End_Port,
AccessList_Protocol, AccessList_Use_Specific_Interface,
AccessList_Interface_ID, AccessList_Packet_Size, AccessList_Byte_Rate,
AccessList_Byte_Burst, AccessList_Allow_Type;
[\AccessList]
Where:
Use_Specific_Interface = determines whether you want to apply the rule
to a specific network interface (defined in the Multiple Interface table):
[0] Disable (default)
[1] Enable
AccessList_Interface_ID = network interface (as defined in the Multiple
Interface table) to which you want to apply the rule. This is applicable if
you enabled the 'Use Specific Interface' field.
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1.13
Obsolete Parameters
The table below lists parameters from the previous release that are now obsolete in
Release 6.2.
Table 1-16: Obsolete Parameters
Obsolete Parameter
Description
[AMDSensitivityResolution]
This parameter has been replaced by the new
parameter, AMDSensitivityParameterSuit.
[AMDDetectionSensitivityHighResolution]
This parameter has been replaced by the new
parameter, AMDSensitivityLevel.
[RoutingTableDestinationsColumn]
This parameter has been replaced by the new ini file
table parameter, StaticRouteTable.
[RoutingTableDestinationMasksColumn]
This parameter has been replaced by the new ini file
table parameter, StaticRouteTable.
[RoutingTableGatewaysColumn]
This parameter has been replaced by the new ini file
table parameter, StaticRouteTable.
[RoutingTableHopsCountColumn]
This parameter has been replaced by the new ini file
table parameter, StaticRouteTable.
[RoutingTableInterfacesColumn]
This parameter has been replaced by the new ini file
table parameter, StaticRouteTable.
[VLANControlVLANID]
This parameter has been replaced by the Multiple
Interface Table.
[VLANOamVLANID]
This parameter has been replaced by the Multiple
Interface Table.
[VLANMediaVLANID]
This parameter has been replaced by the Multiple
Interface Table.
[SBCRegistrationTime]
This parameter has been replaced by the new ini file
parameter SBCUserRegistrationTime.
[T38Version]
This parameter has been replaced by the new ini file
parameter SIPT38Version.
[WANIPAddress]
This parameter has been replaced by the new ini file
table parameter NATTranslation.
[EnableStandardSIDPayloadType]
This parameter is now obsolete.
[EnableNoiseReduction]
This parameter is now obsolete.
[L1L1ComplexTxUDPPort]
This parameter is now obsolete.
[L1L1ComplexRxUDPPort]
This parameter is now obsolete.
SIP Release Notes
114
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SIP Release Notes
2
2. Supported Features
Supported Features
This chapter describes various features supported by the device.
2.1
SIP Gateway Features
This section lists the SIP Gateway application’s main features.
2.1.1
Supported SIP Features
The device supports the following main SIP Gateway features:
Reliable User Datagram Protocol (UDP) transport, with retransmissions.
Transmission Control Protocol (TCP) Transport layer.
SIPS using TLS.
T.38 real-time fax using SIP (including V.34 T.38 fax relay for Mediant 800 MSBG and
Mediant 3000). Note that if the remote side includes the fax maximum rate parameter
in the SDP body of the INVITE message, the device returns the same rate in the
response SDP.
Operates with proxy server, or without using an internal routing table.
Fallback to internal routing table if proxy server does not respond.
Up to 32 IP Groups and 32 Proxy Sets. Each Proxy Set can comprise up to 15 IP
addresses. If the primary proxy server fails, the device automatically switches to a
redundant proxy server.
Up to 32 Registration Accounts, where each account can be used for operation with
different ITSP/SIP Trunks.
Domain name resolving using DNS NAPTR and SRV records for Proxy, Registrar and
domain names that appear in the SIP Contact and Record-Route headers.
Load balancing of proxy servers using round robin or random weights.
Proxy and registrar authentication (handling 401 and 407 responses) using MD5
Digest method. Accepted challenges are kept for future requests to reduce the network
traffic.
Single device registration, multiple registration of all device endpoints, or registration
per Trunk Group (using Accounts).
Supported SIP Methods (messages): INVITE, CANCEL, BYE, ACK, REGISTER,
OPTIONS, INFO, REFER, UPDATE, NOTIFY, PRACK, SUBSCRIBE and PUBLISH.
Modifying connection parameters for an already established call (re-INVITE).
Operating with Redirect server and handling 3xx responses.
Early media (supporting 183 Session Progress).
Receipt of SIP REFER messages outside of a dialog.
Responds to SIP OPTIONS messages outside a SIP dialog and in mid-call. Generates
OPTIONS messages as Proxy keep-alive mechanism.
Publishes the total number of free Tel channels in a 200 OK response to an OPTIONS
requests.
Receipt and DNS resolution of FQDNs received in SDP.
Negotiation of coder from a list of given coders.
Negotiation of dynamic payload types.
Multiple ptime values per coder.
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DTMF out-of-band transfer using:
•
INFO method <draft-choudhuri-sip-info-digit-00.txt>
•
INFO method, compatible with Cisco gateways
•
NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt>
•
INFO method, compatible with Korea Telecom format
Handling forking proxy multiple responses.
Supports ITU V.152 - Procedures for supporting Voice-Band Data over IP Networks.
Call Hold and Transfer Supplementary services using SIP REFER, Refer-To,
Referred-By, Replaces and NOTIFY messages.
Three-way conferencing.
Table 2-1: Supported IETF RFC’s
RFC Number
RFC Title
RFC 2327
SDP
RFC 5079
Rejecting Anonymous Requests in SIP
RFC 4904
Representing Trunk Groups in tel/sip Uniform Resource Identifiers (URIs)
RFC 2617
HTTP Authentication: Basic and Digest Access Authentication
RFC 2782
A DNS RR for specifying the location of services
RFC 2833
Telephone event
RFC 3261
SIP
RFC 3262
Reliability of Provisional Responses in SIP
RFC 3263
Locating SIP Servers
RFC 3264
Offer/Answer Model
RFC 3265
(SIP)-Specific Event Notification
RFC 3310
Hypertext Transfer Protocol (HTTP)
Authentication and Key Agreement (AKA)
RFC 3311
UPDATE Method
RFC 3323
Privacy Mechanism
RFC 3325
Private Extensions to the SIP for Asserted Identity within Trusted Networks
RFC 3326
Reason header
RFC 3327
SIP Extension Header Field for Registering Non-Adjacent Contacts
RFC 3361
Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session
Initiation Protocol (SIP) Servers (Not applicable to Mediant 800 MSBG and
Mediant 1000 MSBG.)
RFC 3372
SIP-T
RFC 3389
RTP Payload for Comfort Noise
RFC 3420
Internet Media Type message/sipfrag
RFC 3455
Private Header (P-Header) Extensions to SIP for the 3rd-Generation
Partnership Project (3GPP)
RFC 3489
STUN - Simple Traversal of UDP Through NATs
RFC 3515
Refer Method
SIP Release Notes
116
Digest
Authentication
Using
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SIP Release Notes
2. Supported Features
RFC Number
RFC Title
RFC 3578
Interworking of ISDN overlap signalling to SIP
RFC 3581
Symmetric Response Routing - rport
RFC 3605
RTCP attribute in SDP
RFC 3608
SIP Extension Header Field for Service Route Discovery During Registration
RFC 3611
RTP Control Protocol Extended Reports (RTCP XR)
RFC 3665
SIP Basic Call Flow Examples
RFC 3666
SIP to PSTN Call Flows
RFC 3680
A SIP Event Package for Registration (IMS)
RFC 3711
The Secure Real-time Transport Protocol (SRTP)
RFC 3725
Third Party Call Control
RFC 3824
Using E.164 numbers with SIP (ENUM)
RFC 3842
MWI
RFC 3891
"Replaces" Header
RFC 3892
The SIP Referred-By Mechanism
RFC 3903
SIP Extension for Event State Publication
RFC 3911
The SIP Join Header [Partial]
RFC 3959
The Early Disposition Type for SIP
RFC 3960
Early Media and Ringing Tone Generation in SIP (partial compliance)
RFC 3966
The tel URI for Telephone Numbers
RFC 4028
Session Timers in the Session Initiation Protocol
RFC 4040
RTP Payload Format for a 64 kbit/s Transparent Call - Clearmode
RFC 4117
Transcoding Services Invocation
RFC 4235
An INVITE-Initiated Dialog Event Package for SIP (partial)
RFC 4240
Basic Network Media Services with SIP - NetAnn
RFC 4244
An Extension to SIP for Request History Information
RFC 4320
Actions Addressing Identified Issues with SIP Non-INVITE Transaction
RFC 4321
Problems Identified Associated with SIP Non-INVITE Transaction
RFC 4411
Extending SIP Reason Header for Preemption Events
RFC 4412
Communications Resource Priority for SIP
RFC 4458
SIP URIs for Applications such as Voicemail and Interactive Voice Response
RFC 4475
SIP Torture Test Messages
RFC 4497 or
ISO/IEC 17343
Interworking between SIP and QSIG
RFC 4568
SDP Security Descriptions for Media Streams for SRTP
RFC 4715
Interworking of ISDN Sub Address to sip isub parameter
RFC 4730
A SIP Event Package for Key Press Stimulus (KPML) [Partial]
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RFC Number
RFC Title
RFC 4733
RTP Payload for DTMF Digits
RFC 4961
Symmetric RTP and RTCP for NAT
RFC 5022
Media Server Control Markup Language (MSCML)
RFC 5627
Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in SIP
RFC 5628
Registration Event Package Extension for GRUU
ECMA-355, ISO/IEC
22535
QSIG tunneling
draft-ietf-sipprivacy-04.txt
SIP Extensions for Network-Asserted Caller Identity using Remote-Party-ID
header
draft-levy-sipdiversion-08
Diversion Indication in SIP
draft-ietf-sipping-cctransfer-05
Call Transfer
draft-ietf-sippingrealtimefax-01
SIP Support for Real-time Fax: Call Flow Examples
draft-choudhuri-sipinfo-digit-00
SIP INFO method for DTMF digit transport and collection
draft-mahy-sippingsignaled-digits-01
Signaled Telephony Events in the Session Initiation Protocol
draft-ietf-sipconnect-reuse-06
Connection Reuse in SIP
draft-ietf-sippingrtcp-summary-10
SIP Package for Voice Quality Reporting Event, using SIP PUBLISH
draft-johnstonsipping-cc-uui-04
Transporting User to User Information for Call Centers using SIP
draft-mahy-iptelcpc-06
The Calling Party's Category tel URI Parameter
Note: The following SIP features are not supported:
SIP Release Notes
•
Preconditions (RFC 3312)
•
SDP - Simple Capability Declaration (RFC 3407)
•
S/MIME
•
Outbound, Managing Client-Initiated Connections (RFC 5626)
•
SNMP SIP MIB (RFC 4780)
•
SIP Compression – RFC 5049 (SigComp)
•
ICE (RFC 5245)
•
Connected Identity (RFC 4474)
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2.1.2
2. Supported Features
SIP Compliance Tables
The SIP device complies with RFC 3261, as shown in the following subsections.
2.1.2.1
SIP Functions
The device supports the following SIP Functions:
Table 2-2: Supported SIP Functions
Function
Supported
User Agent Client (UAC)
Yes
User Agent Server (UAS)
Yes
Proxy Server
The device supports working with third-party Proxy Servers such as
Nortel CS1K/CS2K, Avaya, Microsoft OCS, Alcatel, 3Com, BroadSoft,
Snom, Cisco and many others
Redirect Server
The device supports working with third-party Redirection servers
Registrar Server
The device supports working with third-party Registration servers
2.1.2.2
SIP Methods
The device supports the following SIP Methods:
Table 2-3: Supported SIP Methods
Method
Supported
Comments
INVITE
Yes
ACK
Yes
BYE
Yes
CANCEL
Yes
REGISTER
Yes
Send only
REFER
Yes
Inside and outside of a dialog
NOTIFY
Yes
INFO
Yes
OPTIONS
Yes
PRACK
Yes
UPDATE
Yes
PUBLISH
Yes
SUBSCRIBE
Yes
Version 6.2
Send only
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2.1.2.3
SIP Headers
The device supports the following SIP Headers:
Table 2-4: Supported SIP Headers
Header Field
Supported
Accept
Yes
Accept–Encoding
Yes
Alert-Info
Yes
Allow
Yes
Also
Yes
Asserted-Identity
Yes
Authorization
Yes
Call-ID
Yes
Call-Info
Yes
Contact
Yes
Content-Disposition
Yes
Content-Encoding
Yes
Content-Length
Yes
Content-Type
Yes
Cseq
Yes
Date
Yes
Diversion
Yes
Encryption
No
Expires
Yes
Fax
Yes
From
Yes
History-Info
Yes
Join
Yes
Max-Forwards
Yes
Messages-Waiting
Yes
MIN-SE
Yes
Organization
No
P-Associated-URI
Yes
P-Asserted-Identity
Yes
P-Charging-Vector
Yes
P-Preferred-Identity
Yes
Priority
Yes
SIP Release Notes
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SIP Release Notes
2. Supported Features
Header Field
Supported
Proxy- Authenticate
Yes
Proxy- Authorization
Yes
Proxy- Require
Yes
Prack
Yes
Reason
Yes
Record- Route
Yes
Refer-To
Yes
Referred-By
Yes
Replaces
Yes
Require
Yes
Remote-Party-ID
Yes
Response- Key
Yes
Retry-After
Yes
Route
Yes
Rseq
Yes
Session-Expires
Yes
Server
Yes
Service-Route
Yes
SIP-If-Match
Yes
Subject
Yes
Supported
Yes
Target-Dialog
Yes
Timestamp
Yes
To
Yes
Unsupported
Yes
User- Agent
Yes
Via
Yes
Voicemail
Yes
Warning
Yes
WWW- Authenticate
Yes
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2.1.2.4
SDP Headers
The device supports the following SDP Headers:
Table 2-5: Supported SDP Headers
SDP Header Element
Supported
v - Protocol version
Yes
o - Owner/ creator and session identifier
Yes
a - Attribute information
Yes
c - Connection information
Yes
d - Digit
Yes
m - Media name and transport address
Yes
s - Session information
Yes
t - Time alive header
Yes
b - Bandwidth header
Yes
u - Uri Description Header
Yes
e - Email Address header
Yes
i - Session Info Header
Yes
p - Phone number header
Yes
y - Year
Yes
2.1.2.5
SIP Responses
The device supports the following SIP responses:
1xx Response - Information Responses
2xx Response - Successful Responses
3xx Response - Redirection Responses
4xx Response - Client Failure Responses
5xx Response - Server Failure Responses
6xx Response - Global Responses
SIP Release Notes
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SIP Release Notes
2. Supported Features
2.1.2.5.1 1xx Response – Information Responses
Table 2-6: Supported 1xx SIP Responses
1xx Response
Supported
Comments
100
Trying
Yes
The device generates this response upon receiving a Proceeding
message from ISDN or immediately after placing a call for CAS
signaling.
180
Ringing
Yes
The device generates this response for an incoming INVITE
message. Upon receiving this response, the device waits for a
200 OK response.
181
Call is
Being
Forwarded
Yes
The device doesn't generate these responses. However, the
device does receive them. The device processes these
responses the same way that it processes the 100 Trying
response.
182
Queued
Yes
The device generates this response in Call Waiting service.
When the SIP device receives a 182 response, it plays a special
waiting Ringback tone to the telephone side.
183
Session
Progress
Yes
The device generates this response if the Early Media feature is
enabled and if the device plays a Ringback tone to IP
2.1.2.5.2 2xx Response – Successful Responses
Table 2-7: Supported 2xx SIP Responses
2xx Response
Supported
Comments
200
OK
Yes
-
202
Accepted
Yes
-
2.1.2.5.3 3xx Response – Redirection Responses
Table 2-8: Supported 3xx SIP Responses
3xx Response
Supported
Comments
300
Multiple
Choice
Yes
The device responds with an ACK, and then resends the
request to the first new address in the contact list.
301
Moved
Permanently
Yes
The device responds with an ACK, and then resends the
request to the new address.
302
Moved
Temporarily
Yes
The device generates this response when call forward is
used to redirect the call to another destination. If such a
response is received, the calling device initiates an INVITE
message to the new destination.
305
Use Proxy
Yes
The device responds with an ACK, and then resends the
request to a new address.
380
Alternate
Service
Yes
The device responds with an ACK, and then resends the
request to a new address.
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2.1.2.5.4 4xx Response – Client Failure Responses
Table 2-9: Supported 4xx SIP Responses
4xx Response
Supported
Comments
400
Bad Request
Yes
The device doesn't generate this response. Upon receipt of
this message, and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
401
Unauthorized
Yes
Authentication support for Basic and Digest. Upon receiving
this message, the device issues a new request according to
the scheme received on this response.
402
Payment
Required
Yes
The device doesn't generate this response. Upon receipt of
this message, and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
403
Forbidden
Yes
The device doesn't generate this response. Upon receipt of
this message, and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
404
Not Found
Yes
The device generates this response if it is unable to locate
the callee. Upon receiving this response, the device notifies
the User with a Reorder Tone.
405
Method Not
Allowed
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
406
Not Acceptable
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
407
Proxy
Authentication
Required
Yes
Authentication support for Basic and Digest. Upon receiving
this message, the device issues a new request according to
the scheme received on this response.
408
Request
Timeout
Yes
The device generates this response if the no-answer timer
expires. Upon receipt of this message and before a 200 OK
has been received, the device responds with an ACK and
disconnects the call.
409
Conflict
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
410
Gone
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
411
Length
Required
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
413
Request Entity
Too Large
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
415
Unsupported
Media
Yes
If the device receives a 415 Unsupported Media response, it
notifies the User with a Reorder Tone.
The device generates this response in case of SDP
mismatch.
SIP Release Notes
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SIP Release Notes
4xx Response
2. Supported Features
Supported
Comments
420
Bad Extension
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
423
Interval Too
Brief
Yes
The device does not generate this response. On reception
of this message the device uses the value received in the
Min-Expires header as the registration time.
433
Anonymity
Disallowed
Yes
If the device receives a 433 Anonymity Disallowed, it sends
a DISCONNECT message to the PSTN with a cause value
of 21 (Call Rejected). In addition, the device can be
configured, using the Release Reason Mapping, to generate
a 433 response when any cause is received from the PSTN
side.
480
Temporarily
Unavailable
Yes
If the device receives a 480 Temporarily Unavailable
response, it notifies the User with a Reorder Tone.
This response is issued if there is no response from remote.
481
Call
Leg/Transaction
Does Not Exist
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
482
Loop Detected
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
483
Too Many Hops
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
484
Address
Incomplete
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
485
Ambiguous
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
486
Busy Here
Yes
The SIP device generates this response if the called party is
off-hook and the call cannot be presented as a call waiting
call. Upon receipt of this response, the device notifies the
User and generates a busy tone.
487
Request
Canceled
Yes
This response indicates that the initial request is terminated
with a BYE or CANCEL request.
488
Not Acceptable
Yes
The device doesn't generate this response. Upon receipt of
this message and before a 200 OK has been received, the
device responds with an ACK and disconnects the call.
491
Request
Pending
Yes
When acting as a UAS: the device sent a re-INVITE on an
established session and is still in progress. If it receives a
re-INVITE on the same dialog, it returns a 491 response to
the received INVITE.
When acting as a UAC: If the device receives a 491
response to a re-INVITE, it starts a timer. After the timer
expires, the UAC tries to send the re-INVITE again.
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2.1.2.5.5 5xx Response – Server Failure Responses
Table 2-10: Supported 5xx SIP Responses
5xx Response
500
Internal Server Error
501
Not Implemented
502
Bad gateway
503
Service Unavailable
504
Gateway Timeout
505
Version Not Supported
Comments
Upon receipt of any of these Responses, the
device releases the call, sending an appropriate
release cause to the PSTN side.
The device generates a 5xx response according
to the PSTN release cause coming from the
PSTN.
2.1.2.5.6 6xx Response – Global Responses
Table 2-11: Supported 6xx SIP Responses
6xx Response
600
Busy Everywhere
603
Decline
604
Does Not Exist Anywhere
606
Not Acceptable
SIP Release Notes
Comments
Upon receipt of any of these Responses, the
device releases the call, sending an appropriate
release cause to the PSTN side.
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2.2
2. Supported Features
Supported PSTN-SIP Interworking Features
Note: This section is applicable only to digital PSTN products (i.e., Mediant 600,
Mediant 800 MSBG, Mediant 1000 MSBG, Mediant 1000, Mediant 2000, and
Mediant 3000).
The SIP device supports various ISDN PRI protocols such as EuroISDN, North American
NI2, Lucent 4/5ESS, Nortel DMS100, Meridian1 DMS100, Japan J1, as well as QSIG. PRI
support includes User Termination or Network Termination side. ISDN-PRI protocols can be
defined on an E1/T1 basis (i.e., different variants of PRI are allowed on different E1/T1
spans).
The device also supports numerous variants of CAS protocols for E1 and T1 spans,
including MFCR2, E&M wink start, E&M immediate start, E&M delay dial/start, loop-start,
and ground start. CAS protocols can be defined on an E1/T1 basis (i.e., different variants of
CAS are allowed on different E1/T1 spans).
The device simultaneously supports different variants of CAS and PRI protocols on different
E1/T1 spans (no more than four simultaneous PRI variants).
PSTN-to-SIP and SIP-to-PSTN Called and Calling numbers can be optionally modified
according to rules defined using the device's configuration parameters.
Mediant 600, Mediant 800 MSBG, Mediant 1000 MSBG, and Mediant 1000 also support
various ISDN BRI protocols such as Euro-ISDN and QSIG. BRI support includes User
Termination or Network Termination side.
The device supports the following PSTN-SIP interworking features:
Definition and use of Trunk Groups for routing IP-to-PSTN calls.
B-channel negotiation for PRI spans.
ISDN Non Facility Associated Signaling (NFAS). (Not applicable to Mediant 800
MSBG.)
PRI-to-SIP interworking according to RFC 4497 or ISO/IEC 17343.
PRI-to-SIP Interworking of Q.931 Display (Calling Name) information element (IE).
PRI (NI-2, 5ESS)/EURO ISDN/QSIG to SIP interworking of Calling Name using Facility
IE in Setup and Facility messages.
Configuration of Numbering Plan and Type for IP-to-ISDN calls.
Interworking and flexible mapping of PSTN-to-SIP release causes.
Interworking of ISDN redirect number to SIP Diversion header (according to IETF
<draft-levy-sip-diversion-05.txt>) or to History-Info header (RFC 4244).
Optional change of redirect number to called number for ISDN-to-IP calls.
Interworking of ISDN calling line Presentation & Screening indicators using RPID
header <draft-ietf-sip-privacy-04.txt>.
Interworking of Q.931 Called and Calling Number Type and Number Plan values using
the RPID header or using the phone-context parameter (RFC 4904).
Interworking of Calling and Called Subaddress values for SIP-to-ISDN calls (RFC
4715).
ISDN en-block or overlap dialing for incoming Tel-to-IP calls.
ISDN to SIP interworking of ISDN Overlap Dialing according to RFC 3578.
Digit map pattern to reduce the dialing period when Overlap dialing is used.
Routing of IP-to-Tel calls to predefined trunk groups.
Version 6.2
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CPE SIP Products
Configurable channel-select mode per trunk group.
Various number manipulation rules for IP-to-Tel and Tel-to-IP, called and calling
numbers.
Option to configure ISDN Transfer Capability per trunk.
Transfer of User-to-User Information Element (UUIE) between SIP and PRI.
ISDN Transfer using TBCT / RLT / ECT.
QSIG messages tunneling over SIP according to ECMA-355(ISO/IEC 22535).
ISDN PRI messages tunneling over SIP messages according to RFC 3372 - SIP-T.
Interworking of Hold/Retrieve supplementary services from QSIG/EuroISDN to SIP
and vice versa.
QSIG and EURO ISDN Deflection (Rerouting), interworking of Call Rerouting Facility
to SIP 302 response in both IP-to-Tel and Tel-to IP directions.
QSIG MWI Notifications and QSIG MWI Interrogation.
QSIG Single Step Transfer from QSIG to SIP, and vice versa.
QSIG Path Replacement from SIP to QSIG.
QSIG BRI Suspend/Resume.
Euro-ISDN MWI
BRI Call Forwarding (supported by the softswitch that does the call forward operation;
the device simply provides the softswitch the information (by DTMF) about the forward
numbers.
Comply with DoD UCR 2008 Change 1 specifications, including Multi-Level
Precedence & Preemption (MLPP) using NI-2. (Not applicable to Mediant 800 MSBG.)
Calling Party Category interworking between PRI and SIP according to <draft-mahyiptel-cpc-05>.
E911 Generic Information IE (according to Telcordia GR-2968-CORE) from SIP to NI2. (Not applicable to Mediant 800 MSBG.)
SIP Release Notes
128
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SIP Release Notes
2.3
2. Supported Features
Supported IP Media Features
Note: This section is applicable only to Mediant 1000 and Mediant 1000 MSBG.
The Mediant 1000 and Mediant 1000 MSBG support the following IP media features:
VoIP Conference Bridge, supporting up to 60 channels.
Up to 20 three-way conferences.
Up to 60 participants in a single conference. Up to 3 active talkers; the rest being
listeners only.
VoIP Announcement media server supporting up to 120 VoIP channels.
Playing via HTTP/NFS streaming.
Recording via NFS streaming.
Play/Record using Low Bit Rate (LBR) coders.
Repeat playing announcement.
Playing announcements using Early Media SIP session.
Detection and relay of DTMF digits using INFO messages.
Conferencing and Network Announcements (NetAnn) according to RFC 4240.
The following features are supported via MSCML (RFC 5022):
•
Play request for playing of complex, dynamic announcements.
•
PlayCollect request for playing announcements and collecting digits.
•
PlayRecord request for playing announcements and then recording the voice
stream received from the user.
•
Applying IVR services on PSTN-to-IP calls.
•
Enhanced conferencing according to the MSCML framework.
•
Signal Events Notifications, allowing a client to be notified of various call progress
signals.
Bearer Channel Tandeming (BCT) for Lawful Intercept purposes.
Voice Extensible Markup Language (VXML) Version 2.0.
Transcoding according to RFC 4240 and RFC 4117.
Version 6.2
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December 2010
CPE SIP Products
2.4
Supported Data-Router Features
The supported data-router features are listed in the subsections below.
Note: This section is applicable only to Mediant 800 MSBG and Mediant 1000
MSBG.
2.4.1
LAN Switching
The device supports the following LAN switching features:
Mediant 800 MSBG:
1.
2.
3.
4.
5.
6.
LAN switch ports with non-blocking switching performance.
Up to four RJ-45 10/100/1000Base-T (Gigabit Ethernet) LAN ports.
Eight RJ-45 10/100Base-TX (Fast Ethernet) LAN ports.
802.1Q for VLAN tagging.
Power-over-Ethernet (PoE) supported on all LAN ports, complying with IEEE
802.3af-2003 (15.4W maximum wattage per port; 120 total wattage over all ports).
LAN ports support half- and full-duplex modes, auto-negotiation, and straight or
crossover cable detection.
Mediant 1000 MSBG:
1.
2.
3.
2.4.2
CRMX module supporting three on-board 10/100/1000Base-T, auto-negotiated
LAN switch ports with non-blocking switching performance.
802.1Q for VLAN tagging.
High performance lookup engine with support for up to 1024 MAC addresses.
Routing
The device supports the following new routing features:
1.
Dynamic Host Configuration Protocol (DHCP):
•
DHCP server, DHCP relay and DHCP client
•
DHCP server supports fixed binding of IP-to-MAC address
•
DHCP server supports user-defined DNS server allocation
2.
Multiple IP interfaces for LAN routing: IP interfaces assignment to different VLANs.
3.
Assignment of different VLAN IDs to VoIP and Data traffic.
4.
Symmetric High-Speed Digital Subscriber Line (SHDSL):
•
SIP Release Notes
ATM:
♦
RFC 2684 over AAL5 with LLC-SNAP and VC-MUX over AAL5.
♦
ATM UNI 4.1
♦
UBR, CBR, VBR-RT and VBR-NRT
♦
IP QoS
♦
Multiple IP interfaces
♦
RFC 2684 Routed Mode
♦
RFC 2364 PPPoA
♦
RFC 2516 PPPoE over ATM
♦
Up to 8 PVCs
130
Document #: LTRT-26901
SIP Release Notes
•
5.
6.
2. Supported Features
EFM:
♦
ITU G.991.2 Annex E for Ethernet, also known as EFM or 2Base-TL, as
defined in IEEE 802.3ah
♦
Up to 8 IP interfaces for EFM/2Base-TL
♦
IP QoS classification, Marking, scheduling and shaping
♦
802.1Q VLANs
Routing protocols:
•
Static routing
•
RIPv1 - RFC 1058
•
RIPv2 - RFC 2453
•
OSPFv2 - RFC 2328
•
BGPv4 - RFC 1771 and RFC 2858
•
BGP Extended Community Attribute for BGP/MPLS VPNs
•
Policy-based routing (e.g. DSCP-based and BGP policy routing)
Network Address Translation (NAT/NAPT):
•
ACL-like classification with ALG support
•
Source and destination-based IP addresses ACLs
•
Multiple NAT and NAPT WAN addresses
7.
WAN access via PPPoE, PPTP, L2TP, DHCP.
8.
Quality of service (QoS):
•
Traffic Classification and Marking:
Connection-based (with SPI) or packet-based classification.
♦
VoIP classification for both SIP signaling and media traffic: tracking UDP
ports selected by SDP offer-answer negotiations.
♦
Explicit classification criteria:
Source/destination MAC and IP addresses
Protocol (ALG-based)
L4 port numbers
DSCP/802.1p
Length of packets or their data portion only
♦
DSCP and 802.1p marking
♦
•
Version 6.2
Traffic scheduling and shaping:
♦
Ingress traffic policing
♦
Traffic reservation - when not utilized, other classes are served with extra
bandwidth
♦
Maximum egress traffic shaping
♦
Scheduling: Strict Priority, Fair, Weighted Round-Robin Queuing, and ClassBased WRR Queuing
♦
Queue management:
RED
TCP Serialization Reduction to minimize jitter in VoIP environments
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CPE SIP Products
2.4.3
Data Security
The device supports the following new data security features:
1.
Zero Configuration Firewall wizard with three security levels:
•
Minimum (Inbound and Outbound policies set to ‘Accept’)
•
Typical (Inbound policy set to ‘Reject’; Outbound Policy to ‘Accept’)
•
Maximum (only selected applications are allowed in Outbound policy)
2.
Access Control for pinpoint security policy.
3.
Extensive list of ALG-modules combined with SPI for error-free configuration and
maximum security.
4.
Port-forwarding and DMZ support for local applications and hosts.
5.
Website Restriction allows static URL-based blocking of public/extranet websites.
6.
Advanced Filtering allows full control on Inbound/Outbound Rules per interface/device.
7.
Site-to-Site VPN:
•
Supports two IPSec use-cases:
Site-to-Site (Gateway-to-Gateway) VPN
♦
Teleworker (User-to-Gateway) VPN
♦
•
Fully compliant with IPSec RFCs:
RFC 2401 - Security Architecture for IP
♦
RFC 2402 - IP Authentication Header
♦
RFC 2406 – ESP
♦
RFC 2403 and RFC 2404 for Authentication
♦
8.
9.
PPTP/L2TP Client-Server VPN:
•
Supports two VPN use-cases:
♦
Server support for remote Teleworker VPN access
♦
Client-to-Gateway support with PPTP/L2TP
•
Point-to-Point Tunneling Protocol - RFC 2637
•
Layer Two Tunneling Protocol - RFC 2661 (with L2TP/IPSec)
•
Support all WiN OS versions as well as Linux
DoS and IDS/IPS:
•
Denial of Service (DoS) protection: TCP RST, Ping Flood, ICMP Echo storm,
UDP snork attack, ICMP Smurf, UDP fraggle and more
•
IP spoofing attacks: FTP bounce, Broadcast/multicast source IP attack
•
Intrusion and scanning attacks:
♦
IP source route, ICMP Echo reply without request, ICMP Ping sweep, TCP
Stealth
♦
Scan (FIN, XMAS, NULL), UDP port, FTP passive attack, loopback/Echo
chargen, Block security hazard ICMP messages
•
IP fragment overlap, Ping of Death, Fragmentation attacks and more
SIP Release Notes
132
Document #: LTRT-26901
SIP Release Notes
2.4.4
2. Supported Features
Supported RFC's
The device materially supports the following RFC's:
IP/Routing:
•
RFC 768 UDP
•
RFC 791 IP
•
RFC 792 ICMP
•
RFC 793 TCP
•
RFC 959 FTP
•
RFC 2822 Internet Message Format
•
RFC 1305 NTP
•
RFC 826 ARP
•
RFC 2131 DHCP
•
RFC 1918 IP Addressing
•
RFC 2516 PPPoE
•
RFC 2663 NAT
QoS:
•
IEEE 802.1p Priority Tagging
•
RFC 1349
•
RFC 2475
•
RFC 2597
•
RFC 3246
IPSec:
Version 6.2
•
RFC 2401 Security Architecture for IP
•
RFC 2402 AH - IP Authentication Header
•
RFC 2403 IPsec Authentication - MD5
•
RFC 2404 IPsec Authentication - SHA-1
•
RFC 2405 IPsec Encryption - DES
•
RFC 2406 ESP - IPsec Encryption
•
RFC 2407 IPsec DOI
•
RFC 2408 ISAKMP
•
RFC 2409 IKE
•
RFC 2410 IPsec Encryption – NULL
•
RFC 2411
•
RFC 2412 OAKLEY
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CPE SIP Products
2.5
DSP Firmware Templates (Voice Codecs)
This section lists the DSP templates and corresponding voice codecs support.
2.5.1
MediaPack 1xx
The device supports the following DSP firmware templates:
Table 2-12: DSP Firmware Templates for MediaPack Series
DSP Template
0
1
Number of Channels
Default
SRTP Enabled
Default
SRTP Enabled
MP-112 FXS/FXO
2
2
2
2
MP-114 FXS/FXO
4
3
3
3
MP-118 FXS/FXO
8
6
6
6
MP-124
24
20
20
20
Voice Coder
G.711 A/Mu-law PCM
Yes
Yes
G.726 ADPCM
Yes
Yes
G.727 ADPCM
Yes
Yes
G.723.1
Yes
Yes
G.729 A, B
Yes
Yes
EG.711
Yes
-
-
Yes
G.722
Notes:
SIP Release Notes
•
Installation and use of vocoders is subject to obtaining the appropriate
license and to royalty payments.
•
The number of channels refers to the device's maximum channel
capacity.
•
For other DSP template configurations, please contact AudioCodes.
134
Document #: LTRT-26901
SIP Release Notes
2.5.2
2. Supported Features
Mediant 600 and Mediant 1000
The device supports the following DSP firmware templates:
Notes:
•
Installation and use of vocoders is subject to obtaining the appropriate
license and to royalty payments.
•
The number of channels refers to the device's maximum channel
capacity.
•
The maximum number of channels on any form of analog, digital and
MPM modules assembly is 120.
•
Assembly of the MPM module in Slot 6 enables DSP conferencing
capabilities.
•
For other DSP template configurations, please contact AudioCodes.
Table 2-13: DSP Firmware Templates for Mediant 600 and Mediant 1000 Analog (FXS/FXO)
Interfaces
DSP Template
0, 1, 2, 4, 5
10, 11, 12, 14,15
Number of Channels
Default Settings
4
3
With SRTP
3
3
Voice Coder
G.711 A/Mu-law PCM
Yes
Yes
G.726 ADPCM
Yes
Yes
G.727 ADPCM
Yes
Yes
G.723.1
Yes
Yes
G.729 A, B
Yes
Yes
EG.711
Yes
-
-
Yes
G.722
Version 6.2
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December 2010
CPE SIP Products
Table 2-14: DSP Firmware Templates for Mediant 600 and Mediant 1000 Digital Interfaces
DSP Template
0 or 10
1 or 11
2 or 12
4 or 14
5 or 15
6 or 16
Number of Spans
1
2
4
1
2
4
1
2
4
1
2
4
1
2
4
1
2
4
Number of Channels
Default
Settings
31
62
120
31
48
80
24
36
60
31
60
100
24
36
60
31
60
100
128-ms
EC
31
60
100
31
48
80
26
36
60
31
60
100
24
36
60
31
60
100
SRTP
31
60
100
-
-
-
-
-
-
31
48
80
24
36
60
31
48
80
IPM
Features*
31
60
100
-
-
-
-
-
-
-
-
-
-
-
-
31
60
100
IPM
Features
& SRTP
31
48
80
-
-
-
-
-
-
-
-
-
-
-
-
31
48
80
Voice Coder
G.711 A/
Mu-law
PCM
Yes
Yes
Yes
Yes
Yes
Yes
G.726
ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.727
ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.723.1
Yes
-
-
-
-
-
G.729 A, B
Yes
Yes
Yes
Yes
Yes
Yes
GSM FR
Yes
Yes
-
-
-
-
MS GSM
Yes
Yes
-
-
-
-
iLBC
-
-
-
-
Yes
-
EG.711
-
-
-
Yes
Yes
-
EVRC
-
-
Yes
-
-
-
QCELP
-
-
Yes
-
-
-
AMR
-
Yes
-
-
-
-
GSM EFR
-
Yes
-
-
-
-
Transpare
nt
Yes
Yes
Yes
Yes
Yes
Yes
* IPM Features refers to configuration that includes at least one of the following:
Mounted MPM module in Slot 6 for conferencing applications.
Enabled IPM Detectors (e.g. Answer Detector).
Mounted MPM modules with the IP Media Channels feature enabled.
SIP Release Notes
136
Document #: LTRT-26901
SIP Release Notes
2. Supported Features
Table 2-15: DSP Firmware Templates for Mediant 1000 MPM Module
DSP Template
0 or 10
1 or 11
2 or 12
4 or 14
5 or 15
6 or 16
Assembly Slot Number
1-5
6
1-5
6
1-5
6
1-5
6
1-5
6
1-5
6
Number of Channels
Default Settings
40
20
32
16
24
12
40
20
24
12
40
20
SRTP
32
16
-
-
-
-
32
16
24
12
32
16
Voice Coder
G.711 A/Mu-law
PCM
Yes
Yes
Yes
Yes
Yes
Yes
G.726 ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.727 ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.723.1
Yes
-
-
-
-
-
G.729 A, B
Yes
Yes
Yes
Yes
Yes
Yes
GSM FR
Yes
Yes
-
-
-
-
MS GSM
Yes
Yes
-
-
-
-
iLBC
-
-
-
-
Yes
-
EG.711
-
-
-
Yes
Yes
-
EVRC
-
-
Yes
-
-
-
QCELP
-
-
Yes
-
-
-
AMR
-
Yes
-
-
-
-
GSM EFR
-
Yes
-
-
-
-
Yes
Yes
Yes
Yes
Yes
Yes
Transparent
Note: Only DSP Templates 0, 6, 10, and 16 support IPM detectors.
Version 6.2
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December 2010
CPE SIP Products
2.5.3
Mediant 800 MSBG
The device supports the following DSP firmware capabilities:
Table 2-16: DSP Firmware Capabilities
Number of
Basic DSP
Channels
Telephony Ports
Advanced DSP Capabilities
IPM
AMR
Transcoding Conference
Detectors (WB/NB)
Sessions
(WB/NB)
31 / 24
Yes
-
-
-
30 / 24
Yes
Yes
-
-
12
-
-
-
-
11
Yes
-
-
-
10
Yes
Yes
-
-
FXO/FXS Combination x 8
8
Yes
Yes
-
-
4 x BRI
8
Yes
Yes
-
-
4 x FXS
4
-
-
2
Yes
Yes
1
4 x FXO
4
Yes
Yes
1
1 conference
bridge with up
to 4
participants
1 x E1/T1
12 x FXS
FXS/FXO Combination x 12
Notes:
SIP Release Notes
•
Basic includes the following functionality: G.711 A/Mu-law, G.722, G.726,
G.729 A/B, G.723.1, T38 Version 3, Fax/Modem VBD, IBS (CPT, UDT,
DTMF, MF), Caller ID, and Echo Canceller.
•
Transcoding capability may require an appropriate Software Upgrade
Feature Key.
•
IPM Detectors currently supports only the Answer Detector.
•
Installation and use of vocoders is subject to obtaining the appropriate
license in advance of use, and royalty payments.
•
The Number of Channels represents maximum capacity. On Analog/BRI
depopulated hardware interface assemblies, remaining DSP channels
may be used as IP-only channels for three-way conferencing and
transcoding.
•
Digital hardware assemblies do not support IP-only channels.
138
Document #: LTRT-26901
SIP Release Notes
2.5.4
2. Supported Features
Mediant 1000 MSBG
The device supports the following DSP firmware templates:
Notes:
•
Installation and use of vocoders is subject to obtaining the appropriate
license and to royalty payments.
•
The number of channels refers to the device's maximum channel
capacity.
•
The maximum number of channels on any form of analog, digital and
MPM modules assembly is 120.
•
Assembly of the MPM module in Slot 6 enables DSP conferencing
capabilities.
•
For other DSP template configurations, please contact AudioCodes.
Table 2-17: DSP Firmware Templates for Mediant 1000 MSBG Analog (FXS/FXO) Interfaces
DSP Template
0, 1, 2, 4, 5, 6
10, 11, 12, 14,15, 16
Number of Channels
Default Settings
4
3
With SRTP
3
3
Voice Coder
G.711 A/Mu-law PCM
Yes
Yes
G.726 ADPCM
Yes
Yes
G.727 ADPCM
Yes
Yes
G.723.1
Yes
Yes
G.729 A, B
Yes
Yes
EG.711
Yes
-
-
Yes
G.722
Version 6.2
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December 2010
CPE SIP Products
Table 2-18: DSP Firmware Templates for Mediant 1000 MSBG Digital Interfaces
DSP Template
0 or 10
1 or 11
2 or 12
4 or 14
5 or 15
6 or 16
Number of Spans
1
2
4
1
2
4
1
2
4
1
2
4
1
2
4
1
2
4
Number of Channels
Default
Settings
31
72
120
31
48
80
24
36
60
31
60
100
24
36
60
31
60
100
With 128-ms
EC
31
60
100
31
48
80
26
36
60
31
60
100
24
36
60
31
60
100
With SRTP
31
60
100
-
-
-
-
-
-
31
48
80
24
36
60
31
48
80
With IPM
Features (*)
31
60
100
-
-
-
-
-
-
-
-
-
-
-
-
31
60
100
With IPM
Features &
SRTP
31
48
80
-
-
-
-
-
-
-
-
-
-
-
-
31
48
80
Voice Coder
G.711 A/Mulaw PCM
Yes
Yes
Yes
Yes
Yes
Yes
G.726 ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.727 ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.723.1
Yes
-
-
-
-
-
G.729 A, B
Yes
Yes
Yes
Yes
Yes
Yes
GSM FR
Yes
Yes
-
-
-
-
MS GSM
Yes
Yes
-
-
-
-
iLBC
-
-
-
-
Yes
-
EG.711
-
-
-
Yes
Yes
-
EVRC
-
-
Yes
-
-
-
QCELP
-
-
Yes
-
-
-
AMR
-
Yes
-
-
-
-
GSM EFR
-
Yes
-
-
-
-
G.722
-
-
-
-
-
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Transparent
* IPM Features refers to configuration that includes at least one of the following:
Mounted MPM module in Slot 6 for conference applications.
Enabled IPM Detectors (e.g. Answer Detector).
Mounted MPM modules with the IP Media Channels feature enabled.
SIP Release Notes
140
Document #: LTRT-26901
SIP Release Notes
2. Supported Features
Table 2-19: DSP Firmware Templates for Mediant 1000 MSBG MPM Module
DSP Template
0 or 10
1 or 11
2 or 12
4 or 14
5 or 15
6 or 16
Assembly Slot Number
1-5
6
1-5
6
1-5
6
1-5
6
1-5
6
1-5
6
Number of Channels
Default Settings
40
20
32
16
24
12
40
20
24
12
40
20
With SRTP
32
16
-
-
-
-
32
16
24
12
32
16
Voice Coder
G.711 A/Mu-law
PCM
Yes
Yes
Yes
Yes
Yes
Yes
G.726 ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.727 ADPCM
Yes
Yes
Yes
Yes
Yes
-
G.723.1
Yes
-
-
-
-
-
G.729 A, B
Yes
Yes
Yes
Yes
Yes
Yes
GSM FR
Yes
Yes
-
-
-
-
MS GSM
Yes
Yes
-
-
-
-
iLBC
-
-
-
-
Yes
-
EG.711
-
-
-
Yes
Yes
-
EVRC
-
-
Yes
-
-
-
QCELP
-
-
Yes
-
-
-
AMR
-
Yes
-
-
-
-
GSM EFR
-
Yes
-
-
-
-
G.722
-
-
-
-
-
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Transparent
Note: Only DSP Templates 0, 6, 10, and 16 support IPM detectors.
Version 6.2
141
December 2010
CPE SIP Products
2.5.5
Mediant 2000
The device supports the following DSP firmware templates:
Notes:
•
Installation and use of vocoders is subject to obtaining the appropriate
license and to royalty payments.
•
The number of channels refers to the device's maximum channel
capacity.
•
DSP templates 1 and 2 are not supported on reduced hardware
assemblies (i.e., one or two trunks).
•
For other DSP template configurations, please contact AudioCodes.
Table 2-20: DSP Firmware Templates for Mediant 2000
DSP Template
0
1
2
4
5
Number of Channels
Default Setting
480
320
240
400
240
With 128 ms EC
400
320
240
400
240
With SRTP
400
-
240
240
240
With IPM Detectors
400
320
240
400
240
With IPM Detectors &
SRTP
320
-
240
240
240
Voice Coder
Transparent
Yes
Yes
Yes
Yes
Yes
G.711 A/Mu-law PCM
Yes
Yes
Yes
Yes
Yes
G.726 ADPCM
Yes
Yes
Yes
Yes
Yes
G.727 ADPCM
Yes
Yes
Yes
Yes
Yes
G.723.1
Yes
-
-
-
-
G.729 A, B
Yes
Yes
Yes
-
-
GSM FR
Yes
Yes
-
-
-
MS GSM
Yes
Yes
-
-
-
EVRC
-
-
Yes
-
-
QCELP
-
-
Yes
-
-
AMR
-
Yes
-
-
-
GSM EFR
-
Yes
-
-
-
iLBC
-
-
-
-
Yes
EG.711
-
-
-
Yes
Yes
SIP Release Notes
142
Document #: LTRT-26901
SIP Release Notes
2.5.6
2. Supported Features
Mediant 3000
The device supports the following DSP firmware templates:
Notes:
•
Installation and use of vocoders is subject to obtaining the appropriate
license and to royalty payments.
Number of channels refers to the device's maximum channel capacity.
For other DSP template configurations, please contact AudioCodes.
•
•
Table 2-21: DSP Firmware Templates for Mediant 3000
DSP Template
0
1
2
Supplementary Capabilities
4
5
7
9
10
11
Number of Channels
SRTP
ARIA
RTCP XR
-
-
-
2016
2016
1764
1260
1260
1638
1008
1512
630
-
-
Yes
1890
1890
1638
1134
1134
1638
1008
1386
630
Yes
-
-
1764
-
-
-
-
1638
1008
-
630
Yes
-
Yes
1638
-
-
-
-
1512
1008
-
630
Yes
Yes
-
1638
-
-
-
-
1638
1008
-
504
Yes
Yes
Yes
1638
-
-
-
-
1638
1008
-
504
Voice Coders
AMR
-
Yes
-
Yes
-
-
-
-
-
AMR-WB
-
-
-
Yes
-
-
-
-
-
EVRC
-
-
Yes
-
Yes
-
-
-
-
EVRC-B
-
-
-
-
Yes
-
-
-
-
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
-
-
-
Yes
-
-
-
Yes
Yes
G.723.1
Yes
-
-
-
-
-
-
-
-
G.726 ADPCM
Yes
Yes
Yes
Yes
Yes
Yes
-
-
-
G.727 ADPCM
Yes
Yes
Yes
Yes
Yes
Yes
-
-
-
G.729 A, B
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
GSM EFR
-
Yes
-
Yes
-
-
-
-
-
Yes
Yes
-
Yes
-
-
-
-
-
-
-
-
-
-
Yes
-
-
-
Yes
Yes
-
Yes
-
-
-
-
-
EG.711
-
-
-
-
-
Yes
-
-
-
MS RTA (NB)
-
-
-
-
-
-
Yes
-
Yes
MS RTA (WB)
-
-
-
-
-
-
-
-
Yes
T.38 Version 3
-
-
-
-
-
-
-
Yes
-
G.711 A/Mu-law PCM
G.722
GSM FR
iLBC
MS GSM
Version 6.2
143
December 2010
CPE SIP Products
Table 2-22: DSP Firmware Templates for Mediant 3000 16E1/21T1
DSP Template
0
1
2
Supplementary Capabilities
4
5
7
9
10
11
Number of Channels
SRTP
ARIA
RTCP XR
-
-
-
480
480
480
360
360
468
288
432
150
-
-
Yes
480
480
468
324
324
468
288
396
150
Yes
-
-
480
-
-
-
-
468
288
-
150
Yes
-
Yes
468
-
-
-
-
432
288
-
150
Yes
Yes
-
468
-
-
-
-
396
288
-
120
Yes
Yes
Yes
468
-
-
-
-
396
288
-
120
Voice Coders
AMR
-
Yes
-
Yes
-
-
-
-
-
AMR-WB
-
-
-
Yes
-
-
-
-
-
EVRC
-
-
Yes
-
Yes
-
-
-
-
EVRC-B
-
-
-
-
Yes
-
-
-
-
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
-
-
-
Yes
-
-
-
Yes
Yes
G.723.1
Yes
-
-
-
-
-
-
-
-
G.726 ADPCM
Yes
Yes
Yes
Yes
Yes
Yes
-
-
-
G.727 ADPCM
Yes
Yes
Yes
Yes
Yes
Yes
-
-
-
G.729 A, B
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
GSM EFR
-
Yes
-
Yes
-
-
-
-
-
Yes
Yes
-
Yes
-
-
-
-
-
-
-
-
-
-
Yes
-
-
-
Yes
Yes
-
Yes
-
-
-
-
-
EG.711
-
-
-
-
-
Yes
-
-
-
MS RTA (NB)
-
-
-
-
-
-
Yes
-
Yes
MS RTA (WB)
-
-
-
-
-
-
-
-
Yes
T.38 Version 3
-
-
-
-
-
-
-
Yes
-
G.711 A/Mu-Law PCM
G.722
GSM FR
iLBC
MS GSM
SIP Release Notes
144
Document #: LTRT-26901
SIP Release Notes
2.5.6.1
2. Supported Features
DSP Template Mix Feature for Mediant 3000
Mediant 3000 can operate (and be loaded) with up to two DSP templates. The channel
capacity per DSP template is approximately 50%, with alignment to the number of DSP's
present in the device.
Table 2-23: Template Mix Feature Channel Capacity
Number of Channels
DSP Template Mix
Mediant 3000
Version 6.2
1 (AMR) / 2 (EVRC)
960/924
1 (AMR) / 5 (EVRCB)
768/780
1 (AMR) / 7 (iLBC)
864/936
145
December 2010
CPE SIP Products
Reader’s Notes
SIP Release Notes
146
Document #: LTRT-26901
SIP Release Notes
3
3. Known Constraints
Known Constraints
This section lists known constraints in Release 6.2.
Note for Mediant 1000 MSBG:
When upgrading from Version 5.8 to Version 6.0, ensure that you use firmware file
version F6.00AL.003.015 or later. Previous 6.0 firmware versions cause loss of IP
connectivity when DHCP is enabled (default). If you use a previous 6.0 firmware
version, ensure that you disable DHCP or that a DHCP serve exists.
Note for Mediant 2000:
Software version 4.6 and later cannot be loaded to Mediant 2000 with an 8-span TPM
operating with a 64-MByte RAM. In addition, regardless of the amount of spans,
Mediant 2000 with 64-MByte on-board memory can't be upgraded to 5.8 or later.
3.1
SIP Constraints
This release includes the following known SIP constraints:
1.
IP media features such as play and/or record of announcements, and conferencing are
not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The SBC application does not support termination of REFER\3xx\Hold\Re-INVITE.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
147
December 2010
CPE SIP Products
3.
The ICMP protocol is not supported. The device is unable to generate ICMP PING
packets. The SIP Gateway Alternative Routing feature using PING packets is not
supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
4.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
T.38 fax relay using the same port as RTP (T38UseRTPport = 1) is not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
5.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
For the IP-to-IP application, since the back-to-back user agent (B2BUA) mode is
based on full termination at each leg, some requests, headers and URI parameters
and message bodies are omitted or changed while traversing the device. Responses
to requests within a SIP dialog are always sent independently at each leg regardless of
the other leg’s response.
•
The following SIP Methods are omitted by the IP-to-IP application:
MESSAGE
♦
PUBLISH
♦
SUBSCRIBE
♦
NOTIFY
♦
Out-of-dialog REFER
♦
Any other proprietary Method
♦
•
The following SIP message components are omitted by the IP-to-IP application:
♦
Message body (other than SDP)
♦
Specific parameters in the SIP headers handled by the device (such as To,
From, P-Asserted, Diversion, Remote Party ID, and Contact)
♦
Specific parameters in the SDP – these parameters may affect the RTP flow
at each leg independently
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
148
Document #: LTRT-26901
SIP Release Notes
6.
3. Known Constraints
High Transcoding probability (devices other than MSBG always perform transcoding).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Media Constraints
This release includes the following known voice, RTP and RTCP constraints:
1.
The RTP payload size on RTP forwarding in the SBC application cannot exceed 1,000
bytes. A workaround for this constraint is to reduce the MTU to less than 1,000 bytes
on remote endpoints.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
If the initial transcoding session has one side using a narrowband coder (e.g. G.711),
modifying the transcoding connection to wideband coders still results in narrowband
voice quality. A workaround for this constraint is to ensure that the entire session uses
wideband coders.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
149
December 2010
CPE SIP Products
3.
The Transparent coder (RFC 4040) poses the following limitations:
•
The coder can be used only when using physical terminations
•
No detection of IBS (e.g., DTMF)
•
Generation of IBS is only toward the network
•
No fax/modem detection or generation (i.e., no support for T.38 and Bypass)
A workaround for this constraint is to use the G.711 coder instead.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
4.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When the device is assembled with eight trunks (240 channels) and using the G.729 or
G.711 coder with sample interval of 10 ms, the channel capacity is reduced to seven
trunks (or 210 channels).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
5.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When performing an IP-to-IP call with a wideband (WB) coder on each leg, if the
Fax/Modem Transport type for one of the legs is not Transparent, the interconnection
is made using a narrowband coder, therefore, the wideband quality of the call is not
maintained. The user should avoid setting any Fax/Modem enhanced capabilities on
WB IP-to-IP calls for which the user wants to maintain wideband quality.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
150
Document #: LTRT-26901
SIP Release Notes
6.
3. Known Constraints
Announcements and streaming cannot be performed on IP-to-IP wideband calls.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
7.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The RFC 2198 Redundancy mode with RFC 2833 is not supported (i.e., if a complete
DTMF digit is lost, it is not reconstructed). The current RFC 2833 implementation
supports redundancy for lost inter-digit information. Since the channel can construct
the entire digit from a single RFC 2833 end packet, the probability of such inter-digit
information loss is very low.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
8.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The duration resolution of the On and Off time digits when dialing to the network using
RFC 2833 relay is dependent on the basic frame size of the coder being used.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
9.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The Calling Tone (CNG) detector must be set to Transparent mode to detect a fax
CNG tone received from the PSTN, using the Call Progress Tone detector.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
151
December 2010
CPE SIP Products
10. EVRC Interleaving according to RFC 3558 is supported only on the receiving side.
Supporting this mode on the transmitting side is not mandatory according to this RFC.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
11. To change the DSP template, either the Mixed Template table or the DSP Template
single values can be used.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
12. 30 msec RTP frames using the EG.711 coder is not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
13. Playback with duration set to less than 20 msec is not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
152
Document #: LTRT-26901
SIP Release Notes
3. Known Constraints
14. RTCP XR is not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
15.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When using IP-to-IP mediation, the channel capacity may be reduced.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.3
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
PSTN Constraints
This release includes the following known PSTN constraints:
1.
All the device's trunks must belong to the same Protocol Type (i.e., either E1 or T1).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
After changing the trunk configurations from the initial factory default (i.e., trunks are of
Protocol Type 'None'), a device reset is required (i.e., the change cannot be made onthe-fly).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
153
December 2010
CPE SIP Products
3.
When configuring the framing method to 'Extended Super Frame' (0) or 'Super Frame'
(1), the framing method is converted to another framing method. The correct value that
is updated in the device is displayed in the Web interface:
•
For E1: 'Extended Super Frame' (0) and 'Super Frame' (1) are converted to 'E1
FRAMING MFF CRC4 EXT' (c).
•
For T1: 'Extended Super Frame' (0) is converted to 'T1 FRAMING ESF CRC6'
(D). In addition, 'Super Frame' (1) is converted to 'T1 FRAMING F12' (B).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
4.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When configuring the device with E1 trunks, negotiation of CRC4 (for either
EXTENDED_SUPER_FRAME or E1_FRAMING_MFF_CRC4_EXT framing methods)
should not be used. A framing method other than EXTENDED_SUPER_FRAME and
E1_FRAMING_MFF_CRC4_EXT must be selected.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.3.1
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
DS3 Constraints
This release includes the following known DS3 constraints:
1.
The BIT voice path can fail when using the DS3 interface.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
154
Document #: LTRT-26901
SIP Release Notes
2.
3. Known Constraints
When the DS3 interface is not connected, a trunk under this DS3 interface can appear
in either LOF or AIS alarm state.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The DS3 External clock is not relevant for Asynchronous mapping of DS3 in OC3.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.3.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
SDH Constraints
This release includes the following known SDH constraints:
1.
TU-11 Byte Synchronous mapping is not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
For SDH/SONET and DS3 interfaces, if a trunk was in LOF alarm and the alarm was
then cleared, the trunk tends to revert to the RAI alarm for a short period before
moving to “no alarm” state.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
155
December 2010
CPE SIP Products
3.
In STM-1 and OC3 configurations, path alarms do not show the correct state if the
higher level is not synchronized. For example, if there is no LOS on both PSTN Port A
and Port B, the path level displays "No Alarm".
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.3.3
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
SS7 Constraints
This release includes the following known SDH constraints:
1.
SS7 capacity limitations:
•
Up to 2 signaling nodes can be configured per device.
•
Up to 2 Alias Point Codes can be configured per signaling node, but only 1
signaling node can be configured with Alias Point Codes.
•
Up to 64 signaling links can be configured per device.
•
Up to 32 link-sets can be configured per signaling node.
•
Up to 8 links can be configured per link-set.
•
Up to 30 route-sets can be configured per signaling node.
•
Up to 4 link-sets can be configured per route-set (routes per route-set).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
SS7 MTP3 Signaling Node (SN) Redundancy Limitations:
•
Up to two blades can be configured as Shared Point Code.
•
Blade redundancy parameters cannot be changed on-the-fly (except for
SS7MTP3RdcyTblSyncInterval).
•
The Add & Delete commands can be performed on the SN-Blade Table
Configuration when the SN is set to Offline.
•
ini file configurations after reset are not relayed to remote devices.
•
SN operations (InService/Start/Stop) are local to the device they are performed
on.
•
On-the-fly configurations that are performed on one device are passed only to the
remote devices that established a TCP connection with that device. (For this
reason, redundancy SN-Blade table configurations are not passed between the
new device that is configured and the other device.)
•
SS7 operations that are performed on one device are passed only to the remote
devices that have an InService x-link to that device.
SIP Release Notes
156
Document #: LTRT-26901
SIP Release Notes
3. Known Constraints
•
SS7 and UAL configuration must be the same on devices that share the same
Point Code.
•
On-the-fly configuration of a link (Create/InService/Offline/Delete) must be
performed from the device that the link physically belongs to.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The following tables cannot be configured in offline configuration mode: SN_Timers,
LinkSet_Timers and MTP2_Parms. (Offline configuration mode is used in configuring
SS7 using ini file via Web interface - restore all defaults, load ini to the device and
reset with burn).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
157
December 2010
CPE SIP Products
3.4
IP Media Constraints
This release includes the following known IP media constraints:
1.
Playback to the IP side of LBR Voice Prompts:
•
Sending DTMF signals present in the file as RFC 2833 is not supported during
playback, i.e., if the file/voice prompt contains digits, they are passed as voice and
not as RFC 2833.
•
Generation of signals to the IP during playback is not possible.
•
If the user wishes to pass DTMF signals present in the file over RFC 2833, or
generate in-band signals towards the network during playback, the user must
convert the LBR file into an HBR file (G.711 Alaw or G.711 uLaw).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Voice Prompts files larger than 1 Mbyte cannot be permanently stored on flash
memory. Therefore, they are loaded directly to the RAM and must be loaded again
after the device is reset.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When playing or recording an announcement when using a variable rate coder, the
configured MSCML offset must be set to zero.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
158
Document #: LTRT-26901
SIP Release Notes
4.
3. Known Constraints
No option to detect the beginning and end of speech and therefore, the signal is
unable to start or stop recording accordingly. This means that the MSCML play/record
function ("endsilence" attribute) is supported only when PRT (pre-recording time) and
PST (post-recording time) value equals 0.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
5.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The number of simultaneous recorded voice channels is limited by the HTTP server’s
capability. This capacity can be less than the capacity supported by the device.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
6.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The "Regular Expression Digitmaps" MSCML feature is not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
159
December 2010
CPE SIP Products
3.5
Networking Constraints
This release includes the following known networking constraints:
1.
Enabling the UDP checksum calculation is not applied to CALEA and IP-to-IP calls
with UDP connections. The UDP checksum field is set to zero in these cases.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
In certain cases, when the Spanning-Tree algorithm is enabled on the external
Ethernet switch port that is connected to the device, the external switch blocks all
traffic from entering and leaving the device for some time after the device is reset. This
may result in the loss of important packets such as BootP and TFTP requests, which in
turn, may cause a failure in device start-up. A possible workaround is to set the ini file
parameter BootPRetries to 5, causing the device to issue 20 BootP requests for 60
seconds. Another workaround is to disable the spanning tree on the port of the
external switch that is connected to the device.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Configuring the device to auto-negotiate mode while the opposite port is set manually
to full-duplex (either 10BaseT or 100BaseTX) is invalid. It is also invalid to set the
device to one of the manual modes while the opposite port is configured differently.
The user is encouraged to always prefer full-duplex connections over half-duplex, and
100BaseTX over 10BaseT (due to the larger bandwidth).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
160
Document #: LTRT-26901
SIP Release Notes
4.
3. Known Constraints
Debug Recording:
•
Only one IP target is allowed.
•
Maximum of 50 trace rules are allowed simultaneously.
•
Maximum of 5 media stream recordings are allowed simultaneously.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.6
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Security Constraints
This release includes the following known security constraint:
1.
IPSec tunnels work with pre-shared secrets but not with certificates. The option for
certificate authentication exists on the IPSec configuration Web page, but the IKE
negotiation does not proceed beyond ISAKMP Main Mode. A workaround for this
constraint is to use pre-shared key authentication.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The following CLI commands don’t function with SHDSL-ATM interfaces:
•
show data interfaces
•
show data ip interface
•
show data qos match-map
•
show data hosts
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
161
December 2010
CPE SIP Products
3.
Once SecureStartup mode is enabled, it can’t be disabled correctly thereafter.
Attempting to revert to non-Secure startup causes all parameters to return to defaults.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
4.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The SSH session closes when issuing the cf get CLI command to write the entire
configuration file to the SSH session. A workaround for this constraint is to use the cf
view CLI command to view the configuration file with page breaks.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.7
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
High Availability Constraints
This release includes the following known Mediant 3000 High Availability (HA) constraints:
1.
The Graceful Lock feature does not function when HA is enabled. Attempting to do so
causes errors in the Syslog.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When using IPSec for control protocol transport, the device may experience a large
bulk of Syslog error messages during switchover. These messages can be ignored as
the switchover should succeed and the connection with the softswitch is restored.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
162
Document #: LTRT-26901
SIP Release Notes
3.
3. Known Constraints
During HA switchover, the APS active interface status (e.g., PSTN-B is currently
“Active” and PSTN-A is “Inactive”) is not transferred to the redundant blade. As a
result, if the PSTN-B interface was active before switchover, PSTN-A can be active
after switchover. The information regarding which interface is active is not maintained
after switchover.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
4.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The Voice Prompt file needs be reloaded to the device after the Hitless software
upgrade has completed.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
163
December 2010
CPE SIP Products
3.8
Infrastructure Constraints
This release includes the following known infrastructure constraints:
1.
When configuring the Syslog parameters through the WAN interface (i.e., Syslog
server IP address and enable/disable Syslog messages), error, notice, or debug
messages may appear in the log (e.g., syslog/rs232). These messages should be
ignored.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When using BITS with line-synch mode, only APS protected mode is supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The following parameters do not return to their default values when attempting to
restore them to defaults using the Web interface or SNMP, or when loading a new ini
file using BootP/TFTP:
•
VLANMode
•
VLANNativeVLANID
•
RoutingTableDestinationsColumn
•
RoutingTableDestinationPrefixLensColumn
•
RoutingTableInterfacesColumn
•
RoutingTableGatewaysColumn
•
RoutingTableHopsCountColumn
•
RoutingTableDestinationMasksColumn
•
EnableDHCPLeaseRenewal
•
RoutingTableDestinationMasksColumn
•
IPSecMode
•
CASProtocolEnable
•
EnableSecureStartup
•
UseRProductName
•
LogoWidth
SIP Release Notes
164
Document #: LTRT-26901
SIP Release Notes
3. Known Constraints
•
WebLogoText
•
UseWeblogo
•
UseProductName
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
4.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The Multiple Interface table does not return to default values when attempting to
restore it to defaults using the Web or SNMP interfaces, or when loading a new ini file
using BootP/TFTP.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
5.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Files loaded to the device must not contain spaces in their file name. Including spaces
in the name prevents the file from being saved to the device's flash memory (or copied
to the redundant blade – for Mediant 3000 HA).
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
165
December 2010
CPE SIP Products
3.9
Management Constraints
3.9.1
Web Constraints
This release includes the following known Web constraints:
1.
Changing the RADIUS state (from Online to Offline and vice versa) does not function
correctly. The RADIUS enable/disable is an offline feature. As such, when changing it
through the Web interface, the message should indicate that the effect will take place
after a reset. However, trying to do so causes a prompt for user/password to appear,
and it must be the administrator.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Caller ID types that are not supported appear in the list. The DTMF Caller ID types
appear in the list of possible caller IDs even though they are not supported for these
products. A workaround for this constraint is to ensure that the selected caller ID is
indeed supported
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The Trunk Settings page has some graphic issues. For example, the Submit button
appears in the middle of the page.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
166
Document #: LTRT-26901
SIP Release Notes
4.
3. Known Constraints
The Web Search feature may produce incorrect search results. For example, a search
result for the TLS version parameter directs the user to the incorrect page instead of
the Security Settings page under the System menu.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
5.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When performing a software upgrade using the Software Upgrade wizard, if the user
selects the check box for using the existing file, the Send File button should become
unavailable. However, it remains active. A workaround for this constraint is not to click
this button.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
6.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
On the Home page, when only the digital module is assembled, the analog legend
describing the icon color codes also appears.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
7.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The Help icon on the toolbar is applicable only for the non-data pages. Clicking it when
a data page is displayed will show the last help topic that was opened.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
167
December 2010
CPE SIP Products
8.
A monitoring-only Web user (“User Monitor”) is a read-only user that cannot perform
any configuration or actions. However, some of the action buttons such as Burn and
Submit are not disabled when in this mode. Note that clicking these buttons results in
an error message, and the action will not be performed.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
9.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The horizontal scroll bar is missing in the Connection Status page (Status &
Diagnostics tab > Data Status menu > Connection Statistics). This results in some
loss of information at the end of the line.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
10. When creating a scenario, some of the Web pages cannot be added to it.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
11. When using the Software Upgrade Wizard, if the Voice Prompt (VP) file is loaded and
the Next button is clicked while the progress bar is displayed, the file is not loaded to
the device. Despite this failure, the user receives a message that the file has been
successfully downloaded.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
168
Document #: LTRT-26901
SIP Release Notes
3. Known Constraints
12. On the 'Software Upgrade Wizard' page, the software upgrade process must be
completed prior to clicking the Back button. Clicking the Back button before the wizard
completes causes a display distortion.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
13. On the Configuration > SS7 Configuration > SN > Linkset > Linkset-link > View link, an
incorrect link appears. Sometimes a non-configured link is displayed or a link that is
connected to another linkset is displayed.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
14. On the 'IP Interface Status' page (under the Status & Diagnostics menu), the IP
addresses may not be fully displayed if the address is greater than 25 characters.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
15. When using the Trunk Scroll Bar on the 'Trunk Settings' page, some trunks may not be
displayed on the Trunks panel when scrolling too fast.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
169
December 2010
CPE SIP Products
16. On some SS7 Web pages, all available buttons become visible for a split second while
the page is loaded, even though they should not appear.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
17. On the 'IP Settings' page, adding an interface with invalid characters (e.g., <, >, ", and
') may result in a corrupted Web page. Submitting the corrupted Web page may result
in unexpected behavior such as no response from the device.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
18. The fax counters (Attempted Fax Calls Counter and Successful Fax Calls Counter) in
the 'Status & Diagnostics' page do not function correctly.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
170
Document #: LTRT-26901
SIP Release Notes
3.9.2
3. Known Constraints
SNMP Constraints
This release includes the following known Simple Network Management Protocol (SNMP)
constraints:
1.
When configuring acSysInterfaceTable using SNMP or the Web interface, validation is
performed only after device reset.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The DS3 ifAdmin-State field cannot be changed in the IF-Table, using SNMP.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
3.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
When setting snmpTargetAddrTagList to NULL (by removing a row in the
snmpTargetAddrTable) without changing the corresponding entry in the
snmpCommunityTable, an inconsistency occurs when a switchover occurs between
the Active and Redundant blades. In the Active blade, the object appears with a NULL
value; in the Redundant blade, a non-NULL value appears.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
171
December 2010
CPE SIP Products
4.
In the DS3/E3 Current Table, the objects dsx3CurrentSEFSs and dsx3CurrentUASs
are not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
5.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
In the DS3/E3 Interval Table, the following
dsx3IntervalPSESs and dsx3IntervalSEFSs.
objects
are
not
supported:
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
6.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The dsx3Total Table is not supported.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
172
Document #: LTRT-26901
SIP Release Notes
7.
3. Known Constraints
When defining or deleting SNMPv3 users, the v3 trap user must not be the first to be
defined or the last to be deleted. If there are no non-default v2c users, this results in a
loss of SNMP contact with the device.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
8.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The Admin State does not change to “Redundant”.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
173
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CPE SIP Products
3.9.3
CLI Constraints
This release includes the following known command-line interface (CLI) constraints:
1.
When connecting to the device using Telnet (CLI), Syslog messages do not appear by
default. The Show Log command can be used to enable this feature.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
174
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SIP Release Notes
4
4. Resolved Constraints
Resolved Constraints
This section lists constraints from previous releases that have been resolved.
4.1
SIP Resolved Constraints
The following SIP constraints from previous releases have been resolved:
1.
The On-Board, Three-Way Conferencing feature is not supported when using SRTP.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
The device does not support On-Board Three-Way Conferencing and three-way
conferencing using a third-party Conference server, simultaneously.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
Version 6.2
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
175
December 2010
CPE SIP Products
4.2
Web Resolved Constraints
The following Web constraints from previous releases have been resolved:
1.
The AnalogSignalTransportType parameter appears.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
2.
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
Clicking the Submit button in the 'CAS State Machines' page sometimes causes a
“Page Not Found" error.
Product
MP-11x
MP-124
Mediant 600
Mediant 1000
Mediant 800 MSBG
Mediant 1000 MSBG
Mediant 2000
SIP Release Notes
Mediant 3000/TP-6310
Mediant 3000 HA/TP-6310
Mediant 3000/TP-8410
Mediant 3000 HA/TP-8410
176
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SIP Release Notes
4. Resolved Constraints
Reader’s Notes
.
Version 6.2
177
December 2010
Release Notes Ver. 6.2 www.audiocodes.com