Application Notes for Configuring SIP Trunking between Cincinnati

Application Notes for Configuring SIP Trunking between Cincinnati
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring SIP Trunking between
Cincinnati Bell Any Distance eVantage with an Avaya IP
Telephony Network - Issue 1.0
Abstract
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
trunking between Cincinnati Bell Any Distance eVantage and an Avaya IP Telephony
Network consisting of Avaya AuraTM SIP Enablement Services and Avaya AuraTM
Communication Manager. Avaya IP, digital and analog endpoints were used to originate and
terminate calls.
Cincinnati Bell, Inc. is a member of the Avaya DevConnect Service Provider program.
Information in these Application Notes has been obtained through DevConnect compliance
testing and additional technical discussions. Testing was conducted via the DevConnect
Program at the Avaya Solution and Interoperability Test Lab.
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
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CBTS-CM
1. Introduction
These Application Notes describe the steps for configuring SIP trunking between the Cincinnati
Bell Any Distance (CBAD) eVantage solution and an Avaya IP Telephony Network consisting
of Avaya AuraTM SIP Enablement Services and Avaya AuraTM Communication Manager. Avaya
IP, digital and analog endpoints were used to originate and terminate calls.
SIP (Session Initiation Protocol) is a standards-based communications approach designed to
provide a common framework to support multimedia communication. RFC 3261 [4] is the
primary specification governing this protocol. SIP manages the establishment and termination
of connections and the transfer of related information such as the desired codec, calling party
identity, etc. Within these Application Notes, SIP is used as the signaling protocol between SIP
Enablement Services and the network services offered by Cincinnati Bell Any Distance
eVantage solution.
The CBAD eVantage solution is a turn-key business trunking solution for customers. eVantage
provides customers with a single IP connection that converges voice and data services to drive
optimization, reduce costs, and offer enhanced features not typically available in the traditional
PSTN network. Voice services, such as local, long distance, and toll free calling, as well a high
speed data and Internet services, are the primary applications of the eVantage solution.
1.1. Interoperability Compliance Testing
A simulated enterprise site consisting of a Communication Manager and SIP Enablement
Services solution supporting SIP trunking was connected to the public Internet using a dedicated
broadband connection. The enterprise site was configured to use the commercially available SIP
trunking solution provided by CBAD eVantage solution. This allowed the enterprise site to use
SIP trunking for calls to the PSTN.
The following features and functionality were covered during the SIP trunking interoperability
compliance test:
• Incoming calls to the enterprise site from the PSTN were routed to the DID numbers
assigned by Cincinnati Bell.
• Outgoing calls from the enterprise site were completed via CBAD eVantage solution to
PSTN destinations.
• Calls using H.323, SIP, digital and analog endpoints supported by the Avaya IP
Telephony network.
• Various call types including: local, long distance, international, and toll free calls.
• Calls using the G.729(a) and G.711 µLAW codecs.
• DTMF tone transmission using RFC 2833 with successful call vectoring application.
• Telephone features such as hold, transfer, conference, and call forwarding.
• Avaya one-X® Communicator in either Telecommuter or Softphone mode.
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
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1.2. Support
For technical support on Cincinnati Bell Any Distance eVantage solution, customers can call
1-866-914-9474.
2. Reference Configuration
Figure 1 illustrates an enterprise site with an Avaya SIP-based network, including SIP
Enablement Services, a S8500C Server with a G650 Media Gateway1 running Communication
Manager, and Avaya IP, digital, and analog endpoints. The enterprise site is connected to the
Cincinnati Bell Any Distance eVantage solution over the Internet and communicates using SIP.
The CBAD Network is accessible via a Cisco CUBE supporting a public IP address of
16.96.81.46.
Avaya Lab simulating
Enterprise Customer Site
Internet
TM
Avaya Aura
SIP Enablement Services
(2.160.183.200)
CBAD VoIP Network
SIP
Avaya S8500C Server
PSTN
Avaya G650 Media Gateway
(clan: 2.160.183.202)
Avaya Digital
Telephones
Avaya Analog
Telephones
Avaya 9600 Series
SIP Telephones
Avaya 4600 Series
H.323 IP Telephones
Figure 1: Avaya IP Telephony Network connected to CBAD eVantage Solution
1
This solution is compatible with other Avaya Server and Media Gateway platforms running Communication
Manager.
MDL; Reviewed:
SPOC 7/27/2009
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©2009 Avaya Inc. All Rights Reserved.
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2.1. SIP Call Flows
To better understand how calls are routed between the PSTN and the enterprise site shown in
Figure 1, two call flows are described in this section. The first call scenario is a PSTN call to
the enterprise site and the second call scenario is an outbound call from the enterprise site to the
PSTN. In both cases, the call transits the CBAD VoIP Network. Figure 2 illustrates the call
flow for a call originated from the PSTN and terminated at the enterprise site.
1.
2.
3.
4.
5.
A user on the PSTN dials a DID number assigned to an Avaya SIP telephone at the
enterprise site. The enterprise site subscribes to the CBAD eVantage solution so the call
is routed through the CBAD VoIP network.
Based on the DID number, CBAD routes the call to the enterprise site via SIP trunking.
CBAD sends SIP signaling messages to SIP Enablement Services at the enterprise site.
See the Appendix A for an example of a SIP INVITE message sent by CBAD.
SIP Enablement Services routes the call to the Avaya S8720 Server running
Communication Manager over a SIP trunk.
Since the call is destined for an Avaya SIP telephone, Communication Manager routes
the call back to SIP Enablement Services over a SIP trunk. If the destination of the call
was an H.323, digital or analog endpoint, Communication Manager would terminate the
call directly to the endpoint and steps 4 and 5 would not be required.
SIP Enablement Services terminates the call to the Avaya SIP telephone.
3
1
PSTN
CBAD VoIP
Network
2
Avaya AuraTM SIP
Enablement Services
4
5
Avaya S8500C Server
with G650 Media Gateway
Avaya 9600 Series
SIP Telephone
Figure 2: PSTN Call to the Avaya SIP Network
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
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CBTS-CM
Figure 3 illustrates the call flow for an outgoing call from an Avaya SIP telephone on the Avaya
SIP network at the enterprise site to the PSTN.
1.
2.
3.
4.
5.
An Avaya SIP telephone originates a call to a user on the PSTN. The call request is
delivered to SIP Enablement Services. If the originator were an H.323, digital or analog
endpoint, the call request would be sent to SIP Enablement Services from the S8720
Servers running Communication Manager.
SIP Enablement Services routes the call over the SIP trunk to the Avaya S8720 Servers
running Communication Manager for origination services. This allows Communication
Manager to apply the appropriate call restrictions to the endpoint, handle call routing, and
track the status of the SIP telephone, which is an off-PBX station.
After applying the origination services, Communication Manager routes the call back to
Avaya SIP Enablement Services over a SIP trunk.
SIP Enablement Services routes the call to the CBAD VoIP Network. See Appendix A
for an example of a SIP INVITE message sent by the Avaya SIP-based network.
CBAD routes the call to the PSTN.
3
4
2
Avaya AuraTM SIP
Enablement Services
CBAD VoIP
Network
5
PSTN
1
Avaya S8500C Server with
G650 Media Gateway
Avaya 9600 Series
SIP Telephone
Figure 3: Avaya SIP Call to the PSTN
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
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3. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Hardware Component
Avaya S8500C Server
Avaya G650 Media Gateway
TN799DP C-LAN Board
TN2302AP Media Processor Board
Version
Avaya Aura Communication Manager 5.2
(R015x.002.0.947.3) with
Service Pack 1 (Update 17294)
HW13 FW032
HW21 FW120
Avaya Aura SIP Enablement Services
5.2 (SES05.2-02.0.947.3b) with Service
Pack 1(SES-02.0.947.3-SP1)
Avaya 1600 Series IP Telephone
1.1 (H.323)
Avaya 4600 Series IP Telephones
2.9 (H.323); R2.2.2 (SIP)
Avaya 9600 Series IP Telephones
3.0.2 (H.323) 2.4.1.0 (SIP)
Avaya Digital Telephones
--
Avaya Analog Telephones
--
Avaya one-X Communicator
Cisco Cube
MDL; Reviewed:
SPOC 7/27/2009
R1.030-SP3-16918
12.4 (24)T1
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©2009 Avaya Inc. All Rights Reserved.
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CBTS-CM
4. Configure Communication Manager
This section describes the steps for configuring a SIP trunk and off-PBX stations (OPS) on
Communication Manager. The SIP trunk is established between Communication Manager and
SIP Enablement Services. An off-PBX station (OPS) is configured for each Avaya SIP
telephone registered with SIP Enablement Services. Refer to [2] for additional information on
configuring an off-PBX station. All incoming calls from CBAD are received by SIP Enablement
Services and routed to Communication Manager over a SIP trunk for termination services. All
outbound calls to the PSTN are routed through Communication Manager for origination services.
Communication Manager then routes the call to SIP Enablement Services, which in turn routes
the call to the PSTN through the CBAD eVantage solution. Note that SIP Enablement Services
provides the SIP interface to the CBAD eVantage solution.
The dial plan for the configuration described in these Application Notes consisted of 10-digit
dialing for local and long-distance calls over the PSTN. In addition, Directory Assistance calls
(411), International calls (011 Country Code), Toll-Free calls, and Operator calls were also
supported. Communication Manager routed all calls using Auto Route Selection (ARS), except
for intra-switch calls. Configuring ARS is beyond the scope of these Application Notes and the
reader should refer to Error! Reference source not found. for additional information.
Communication Manager configuration was performed using the System Access Terminal
(SAT). The IP network parameters of the Avaya S8500C Server were configured via the
Maintenance web interface using an Internet browser (not shown here). Using the SAT, verify
that the Off-PBX Telephones (OPS) and SIP Trunks features are enabled on the systemparameters customer-options form. The license file installed on the system controls these
options. If a required feature is not enabled, contact an authorized Avaya sales representative.
On Page 1, verify that the number of OPS stations allowed in the system is sufficient.
display system-parameters customer-options
OPTIONAL FEATURES
G3 Version: V15
Location: 1
Platform: 12
Page
1 of
11
Software Package: Standard
RFA System ID (SID): 1
RFA Module ID (MID): 1
USED
Platform Maximum Ports: 44000 226
Maximum Stations: 36000 80
Maximum XMOBILE Stations: 0
0
Maximum Off-PBX Telephones - EC500: 10
1
Maximum Off-PBX Telephones OPS: 300
55
Maximum Off-PBX Telephones - PBFMC: 0
0
Maximum Off-PBX Telephones - PVFMC: 0
0
Maximum Off-PBX Telephones - SCCAN: 0
0
(NOTE: You must logoff & login to effect the permission changes.)
Figure 4: System-Parameters Customer-Options Form – Page 1
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
On Page 2 of the system-parameters customer-options form, verify that the number of SIP
trunks supported by the system is sufficient.
display system-parameters customer-options
OPTIONAL FEATURES
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable H.323 Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
Maximum Administered Ad-hoc Video Conferencing Ports:
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum Media Gateway VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
Page
2 of
11
USED
100
6
18000 1
0
0
0
0
0
0
0
0
0
0
0
0
600 130
0
0
0
0
10
0
0
0
128
0
128
0
0
0
(NOTE: You must logoff & login to effect the permission changes.)
Figure 5: System-Parameters Customer-Options Form – Page 2
On the system-parameters features form, set the Trunk-to-Trunk Transfer field to all to
allow calls to be transferred from the enterprise site to an endpoint on the PSTN. Otherwise,
leave the field set to none. The SIP call flows described in Section 2.1 did not require trunk-totrunk transfer to be enabled.
change system-parameters features
Page
1 of
FEATURE-RELATED SYSTEM PARAMETERS
Self Station Display Enabled? n
Trunk-to-Trunk Transfer: all
Automatic Callback - No Answer Timeout Interval (rings): 3
Call Park Timeout Interval (minutes): 10
Off-Premises Tone Detect Timeout Interval (seconds): 20
AAR/ARS Dial Tone Required? y
Music/Tone on Hold: none
Music (or Silence) on Transferred Trunk Calls? no
DID/Tie/ISDN/SIP Intercept Treatment: attd
Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred
Automatic Circuit Assurance (ACA) Enabled? n
Abbreviated Dial Programming by Assigned Lists?
Auto Abbreviated/Delayed Transition Interval (rings):
Protocol for Caller ID Analog Terminals:
Display Calling Number for Room to Room Caller ID Calls?
18
n
2
Bellcore
n
Figure 6: System-Parameters Features Form
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
In the IP Node Names form, assign an IP address and host name for the C-LAN board in the
Avaya G650 Media Gateway and for SIP Enablement Services at the enterprise site. The host
names will be used throughout the other configuration screens of Communication Manager.
change node-names ip
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1 of
2
IP NODE NAMES
Name
clan1
medpro1
ses
default
IP Address
2.160.183.202
2.160.183.203
2.160.183.200
0.0.0.0
( 4 of 12
administered node-names were displayed )
Use 'list node-names' command to see all the administered node-names
Use 'change node-names ip xxx' to change a node-name 'xxx' or add a node-name
Figure 7: IP Nodes Names Form
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
In the IP Network Region form, the Authoritative Domain field is configured to match the
domain name configured on SIP Enablement Services. In this configuration, the domain name is
avremote.com. By default, IP-IP Direct Audio (shuffling) is enabled to allow audio traffic to be
sent directly between IP endpoints without using media resources in the Avaya G650 Media
Gateway. In addition, DTMF transmission using RFC 2833 (described later) is also required for
shuffling among IP devices as shown in Figure 10. The IP Network Region form also specifies
the IP Codec Set to be used for local calls and calls routed over the SIP trunk to SIP Enablement
Services. This codec set is used when its corresponding network region (i.e., IP Network Region
‘1’) is specified in the Far-end Network Region field of the SIP signaling group as shown in
Figure 10.
change ip-network-region 1
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19
IP NETWORK REGION
Region: 1
Location: 1
Authoritative Domain: avremote.com
Name: Main
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 1
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048
IP Audio Hairpinning? n
UDP Port Max: 60001
DIFFSERV/TOS PARAMETERS
RTCP Reporting Enabled? y
Call Control PHB Value: 46
RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46
Use Default Server Parameters? y
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5
AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS
RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
Figure 8: IP Network Region Form
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
In the IP Codec Set form, select the audio codec type supported for calls routed over the SIP
trunk. The form is accessed via the change ip-codec-set 1 command. Note that IP codec set ‘1’
was specified in IP Network Region ‘1’ shown in Figure 8. The default settings of the IP Codec
Set form are shown below. However, the IP Codec Set form may specify multiple codecs,
including G.711 and G.729 to allow the codec for the call to be negotiated during call
establishment.
change ip-codec-set 1
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2
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2: G.729A
3:
4:
5:
6:
7:
Silence
Suppression
n
n
Frames
Per Pkt
2
2
Packet
Size(ms)
20
20
Figure 9: IP Codec Set – Page 1
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
Prior to configuring a SIP trunk group for communication with SIP Enablement Services, a SIP
signaling group must be configured. The following signaling group is used for outgoing calls to
the PSTN through the CBAD eVantage solution. Configure the Signaling Group form shown in
Figure 10 as follows:
Set the Group Type field to sip.
The Transport Method field will default to tls (Transport Layer Security).
Specify the C-LAN board in the G650 Media Gateway and the SIP Enablement Services
Server as the two ends of the signaling group in the Near-end Node Name field and the
Far-end Node Name field, respectively. These field values are taken from the IP Node
Names form shown in Figure 7.
Ensure that the recommended TLS port value of 5061 is configured in the Near-end
Listen Port and the Far-end Listen Port fields.
The preferred codec for the call will be selected from the IP codec set assigned to the IP
network region specified in the Far-end Network Region field. Although the same
network region (Network Region 1) was used for local and PSTN calls in this
configuration, a different network region for PSTN calls could have been specified.
Enter the domain name of SIP Enablement Services in the Far-end Domain field. The
Far-end Domain field was filled with the IP address of the Cisco CUBE SIP proxy/SBC,
16.96.81.46. The Far-end Domain field is specified in the Uniform Resource Identifier
(URI) of the “SIP To Address” in the INVITE message. Mis-configuring this field may
prevent calls from being successfully established to other SIP endpoints or to the PSTN.
If calls to/from SIP endpoints are to be shuffled, then the Direct IP-IP Audio
Connections field must be set to y.
The DTMF over IP field should be set to the default value of rtp-payload.
Communication Manager supports DTMF transmission using RFC 2833. The default
values for the other fields may be used.
add signaling-group 11
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1
SIGNALING GROUP
Group Number: 11
Group Type: sip
Transport Method: tls
Near-end Node Name: clan1
Near-end Listen Port: 5061
Far-end Node Name: ses
Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain: 16.96.81.46
Bypass If IP Threshold Exceeded? n
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? n
H.323 Station Outgoing Direct Media? n
Direct IP-IP Audio Connections? Y
IP Audio Hairpinning? n
Alternate Route Timer(sec): 6
Figure 10: Signaling Group for Outgoing Calls to PSTN
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
The following signaling group is used for incoming calls from the PSTN. A different signaling
group is required because CBAD specifies a different domain in the FROM header of the SIP
INVITE message than what was configured in the far-end domain name field of the signaling
group shown in. The Far-end Domain field was filled with as.voip.fuse.net, which would match
the domain sent by CBAD. Follow the instructions described for the signaling group configured
above for the other fields.
add signaling-group 10
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1
SIGNALING GROUP
Group Number: 10
Group Type: sip
Transport Method: tls
Near-end Node Name: clan1
Near-end Listen Port: 5061
Far-end Node Name: ses
Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain: as.voip.fuse.net
Bypass If IP Threshold Exceeded? n
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? n
H.323 Station Outgoing Direct Media? n
Direct IP-IP Audio Connections? Y
IP Audio Hairpinning? n
Alternate Route Timer(sec): 6
Figure 11: Signaling Group for Incoming Calls from the PSTN
Configure the Trunk Group form as shown in Figure 12. This trunk group is used for outgoing
calls to the PSTN. Set the Group Type field to sip, set the Service Type field to public-ntwrk,
specify the signaling group associated with this trunk group in the Signaling Group field, and
specify the Number of Members supported by this SIP trunk group. For a call between the
PSTN and a SIP endpoint, two trunk members are used for the duration of the call. For a call
between the PSTN and a non-SIP endpoint, one trunk member is used for the duration of the call.
Configure the other fields in bold and accept the default values for the remaining fields.
add trunk-group 11
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21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
11
CBTS Outgoing
two-way
n
0
public-ntwrk
Group Type: sip
CDR Reports: y
COR: 1
TN: 1
TAC: 111
Outgoing Display? n
Night Service:
Auth Code? n
Signaling Group: 11
Number of Members: 30
Figure 12: Trunk Group for Outgoing Calls to PSTN – Page 1
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
On Page 2 of the trunk group form, set the Preferred Minimum Session Refresh Interval(sec)
to the agreed upon rate discussed with CBAD. The value of 900 was chosen for testing purposes.
add trunk-group 11
Group Type: sip
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21
TRUNK PARAMETERS
Unciode Name? y
Redirect on OPTIM Failure: 5000
SCCAN? N
Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 900
Figure 13: Trunk Group for Outgoing Calls to PSTN – Page 2
On Page 3 of the trunk group form, set the Numbering Format field to public. This field
specifies the format of the calling party number sent to the far-end.
add trunk-group 11
TRUNK FEATURES
ACA Assignment? n
Page
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21
Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? y
Replace Unavailable Numbers? y
Show ANSWERED BY on Display? y
Figure 14: Trunk Group for Outgoing Calls to PSTN – Page 3
On Page 4 of the trunk group form, set the Telephone Event Payload Type to 101 for the
proper exchange of RFC 2833 DTMF events between Communication Manager and the CBAD
eVantage solution. Set the fields Send Transferring Party Information and Overwrite Calling
Identity to y to permit trunk to trunk transfer to work correctly.
add trunk-group 11
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21
PROTOCOL VARIATIONS
Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? y
Send Diversion Header? n
Support Request History? n
Telephone Event Payload Type: 101
Overwrite Calling Identity? y
Figure 15: Trunk Group for Outgoing Calls to PSTN – Page 4
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
Repeat the trunk group configuration in Figure 12 through Figure 15 for the trunk group used
for incoming calls from the PSTN. The only difference would be to specify the signaling group
configured in Figure 11 for this trunk group. All other fields may be entered as shown.
Note: To call an endpoint on the Avaya SIP-based network from the PSTN, a 10-digit DID
number is dialed. This 10-digit dialed number is received by Communication Manager and
converted to the appropriate 5-digit extension in the Incoming Call Handling Table (not
shown) for trunk group ‘10’.
add trunk-group 10
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21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
10
CBTS incoming
two-way
n
0
public-ntwrk
Group Type: sip
CDR Reports: y
COR: 1
TN: 1
TAC: 110
Outgoing Display? n
Night Service:
Auth Code? n
Signaling Group: 10
Number of Members: 30
Figure 16: Trunk Group for Incoming Calls from PSTN
Configure the Public/Unknown Numbering Format form to send the calling party number to
the far-end. Add an entry so that the local stations 68010 and 68020 who have calls routed over
the SIP trunk group 11 will send the number expected by the CBAD eVantage solution for call
authentication and for display purposes to the far-end. In the example shown in Figure 17, a
CPN prefix is added to the 5-digit extension so that a 10-digit calling party number (e.g.,
extension 68010 is converted to 5135551234) is sent to the far-end.
Note: The 10-digit CPN must be recognized by the CBAD VoIP network or the call will be
denied.
change public-unknown-numbering 0
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2
NUMBERING - PUBLIC/UNKNOWN FORMAT
Total
Ext Ext
Trk
CPN
CPN
Len Code
Grp(s)
Prefix
Len
Total Administered: 5
5 68010
11
5135551234
10
Maximum Entries: 9999
5 68020
11
5135551235
10
Figure 17: Public Unknown Format Form
MDL; Reviewed:
SPOC 7/27/2009
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CBTS-CM
The first step in configuring an off-PBX station (OPS) for the Avaya SIP telephones registered
with SIP Enablement Services is to add a station with the appropriate station type as shown in
Figure 18. A descriptive name may also be provided. The Class of Restriction (COR) and Class
of Service (COS) assigned to the station should be configured with the appropriate call
restrictions. Repeat this step for each SIP endpoint at the enterprise site.
add station 68020
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6
STATION
Extension:
Type:
Port:
Name:
68020
9600SIP
S00009
Johnny SIP
Lock Messages? n
Security Code:
Coverage Path 1:
Coverage Path 2:
Hunt-to Station:
BCC:
TN:
COR:
COS:
0
1
1
1
STATION OPTIONS
Loss Group: 19
Speakerphone:
Display Language:
Survivable GK Node Name:
Survivable COR:
Survivable Trunk Dest?
2-way
english
Time of Day Lock Table:
Personalized Ringing Pattern:
Message Lamp Ext:
Mute Button Enabled?
Expansion Module?
internal
y
1
68020
y
0
Media Complex Ext:
IP SoftPhone? n
Customizable Labels? y
Figure 18: SIP Station – Page 1
On Page 2 of the station form, verify that the Per Station CPN – Send Calling Number field is
set to ‘y’ or blank to allow calling party number information to be sent to the far-end when
placing outgoing calls from this station. The default value for this field is blank.
add station 68020
Page
2 of
6
STATION
FEATURE OPTIONS
LWC Reception:
LWC Activation?
LWC Log External Calls?
CDR Privacy?
Redirect Notification?
Per Button Ring Control?
Bridged Call Alerting?
Active Station Ringing:
spe
y
n
n
y
n
n
single
H.320 Conversion? n
Service Link Mode: as-needed
Multimedia Mode: enhanced
MWI Served User Type:
AUDIX Name:
Emergency Location Ext: 20003
Auto Select Any Idle Appearance?
Coverage Msg Retrieval?
Auto Answer:
Data Restriction?
Idle Appearance Preference?
Bridged Idle Line Preference?
Restrict Last Appearance?
n
y
none
n
n
n
y
EMU Login Allowed? n
Per Station CPN - Send Calling Number?
Display Client Redirection? n
Select Last Used Appearance? n
Coverage After Forwarding? s
Direct IP-IP Audio Connections? y
Always Use? n IP Audio Hairpinning? n
Figure 19: SIP Station – Page 2
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On Page 4 of the station form, configure the appropriate number of call appearances for the SIP
telephone. For example, the Avaya 9630 SIP Telephone was configured to support three call
appearances as shown in Figure 20.
add station 68020
Page
4 of
6
STATION
SITE DATA
Room:
Jack:
Cable:
Floor:
Building:
Headset?
Speaker?
Mounting:
Cord Length:
Set Color:
ABBREVIATED DIALING
List1:
List2:
BUTTON ASSIGNMENTS
1: call-appr
2: call-appr
3: call-appr
4:
n
n
d
0
List3:
5:
6:
7:
8:
Figure 20: SIP Station – Page 4
The second step of configuring an off-PBX station is to configure the Stations with Off-PBX
Telephone Integration form so that calls destined for a SIP telephone at the enterprise site are
routed to SIP Enablement Services, which will then terminate the call to the SIP telephone. On
this form, specify the extension of the SIP endpoint and set the Application field to OPS. The
Phone Number field is set to the digits to be sent over the SIP trunk. In this case, the SIP
telephone extensions configured on SIP Enablement Services also match the extensions of the
corresponding stations on Communication Manager. However, this is not a requirement.
Finally, the Trunk Selection field is set to 1, the SIP trunk group number. This field specifies
the trunk group used to route the outgoing call. Another option for routing a call over a SIP
trunk group is to use Auto Alternate Routing (AAR) or Auto Route Selection (ARS) routing
instead. If either option is preferred, the Trunk Selection field would be set to aar or ars.
Configuration of other AAR or ARS forms would also be required. Refer to Error! Reference
source not found. for information on routing calls using AAR or ARS. Repeat this step for each
SIP endpoint at the enterprise site.
change off-pbx-telephone station-mapping 68020
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
68020
Application Dial
CC
Prefix
OPS
-
Phone Number
68020
Page
Trunk
Selection
1
1 of
3
Config
Set
1
Figure 21: Stations with Off-PBX Telephone Integration – Page 1
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On Page 2, set the Call Limit field to the maximum number of calls that may be active
simultaneously at the station. In this example, the call limit is set to 3, which corresponds to the
number of call appearances configured on the station form. Accept the default values for the
other fields.
change off-pbx-telephone station-mapping 68020
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
68020
Appl
Name
OPS
Call
Limit
3
Mapping
Mode
both
Calls
Allowed
all
Page
Bridged
Calls
none
2 of
3
Location
Figure 22: Stations with Off-PBX Telephone Integration – Page 2
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5. Configure SIP Enablement Services
This section covers the administration of SIP Enablement Services (SES). SIP Enablement
Services is configured via an Internet browser using the Administration web interface. To access
the Administration web interface, enter http://<ip-addr>/admin as the URL in an Internet
browser, where <ip-addr> is the IP address of SIP Enablement Services. Log in with the
appropriate credentials and then select the SIP Enablement Services from the Administration pull
down menu from the title bar. The main screen shown in Figure 23 is displayed.
Figure 23: Main Screen
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From the left pane of the Administration web interface, expand the Server Configuration option
and select System Properties. In the View System Properties screen, enter the domain name
assigned to the Avaya SIP-based network and the SIP License Host. For the SIP License Host
field, enter the fully qualified domain name or the IP address of the SES server that is running
the WebLM application and has the associated license file installed. This entry should always
correspond to the localhost unless the WebLM server is not co-resident with this server. After
configuring the View System Properties screen, click the Update button.
Figure 24: System Properties
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After setting up the domain in the View System Properties screen, create a host entry for SIP
Enablement Services. The following example shows the Edit Host screen since the host had
already been configured. Enter the IP address of SIP Enablement Services in the Host IP
Address field. The Profile Service Password was specified during the system installation.
Next, configure the Host Type field. In this example, the host server was configured as an SES
combined home-edge. The default values for the other fields may be used as shown in Figure
25. Click the Update button.
Figure 25: Host
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Under the Communication Manager Servers option in the Administration web interface, select
Add to add the Avaya S8500C Server in the enterprise site since a SIP trunk is required between
Communication Manager and SIP Enablement Services. In the Add Communication Manager
Server Interface screen, enter the following information:
•
•
•
•
A descriptive name in the Communication Manager Server Interface Name field (e.g.,
CLAN1).
Select the home server in the Host field.
Select TLS (Transport Link Security) for the Link Type. TLS provides encryption at the
transport layer.
Enter the IP address of the C-LAN board in the Avaya G650 Media gateway in the SIP
Trunk IP Address field.
After completing the Add Communication Manager Server Interface screen, click the Add
button. Refer to [3] for additional information on configuring the remaining fields.
Figure 26: Add Communication Manager Server Interface
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Incoming calls originated from the PSTN and arriving at SIP Enablement Services are routed to
Communication Manager for termination services. Calls to be routed to Communication
Manager are specified in a Communication Manager Server Address Map. The Uniform
Resource Identifier (URI) of an incoming INVITE message is compared to the pattern
configured in the address map, and if there is a match, the call is routed to Communication
Manager. The URI usually takes the form of sip:user@domain, where domain can be a
domain name or an IP address. In this example, user is actually the telephone number of the
phone. An example of a URI would be sip:5135551234@2.160.183.202. Only
incoming calls from the PSTN require a Communication Manager address map. By default, all
calls originated from an Avaya SIP telephone are routed through Communication Manager for
origination services because the Avaya SIP telephones are assigned a media server extension.
To configure a Communication Manager Server Address Map, select Communication
Manager Servers in the left pane of the Administration web interface. This will display the List
Communication Manager Servers screen. Click on the Map link associated with the
appropriate server to display the List Communication Manager Server Address Map screen
and click on the Add Map In New Group link. The screen shown in Figure 27 is displayed.
Provide a descriptive name in the Name field and enter the regular expression to be used for the
pattern matching in the Pattern field. In this configuration, the pattern specification matches a
URI that begins with sip:513 followed by seven digits. Note that DID numbers beginning
with area code 513 were assigned to endpoints at the enterprise site. See Appendix B for a more
detailed description of the syntax for address map patterns. Click the Add button. Repeat this
procedure to add an address map for routing incoming toll-free calls, if necessary.
Figure 27: Communication Manager Server Address Map
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After the Communication Manager Server Address Map is added, the first Communication
Manager Server Contact is created automatically. For the address map added in Figure 28, the
following contact was created:
sip:$(user)@2.160.183.202:5061;transport=tls
The contact specifies the IP address of the C-LAN board in the Avaya G650 Media Gateway and
the transport protocol used to send SIP signaling messages. The user in the original request URI
is substituted for $(user). After configuring the media server address map, the List
Communication Manager Server Address Map screen appears as shown in Figure 28.
Figure 28: List Media Server Address Map
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Add a user for each Avaya SIP telephone registering with SIP Enablement Services. In the Add
User screen, enter the extension of the SIP endpoint in the Primary Handle field. Enter a user
password in the Password and Confirm Password fields. In the Host field, select the SIP
Enablement Services server hosting the domain (sipsp.avaya.com) for this user. Enter the First
Name and Last Name of the user. To associate a Communication Manager server extension
with this user, select the Add Communication Manager Extension checkbox. Calls from this
user will always be routed through Avaya Communication Manager over the SIP trunk for
origination services. The Add Communication Manager Extension screen shown in Figure
30 will be displayed after adding this user profile by clicking on the Add button.
Figure 29: Add User
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In the Add Communication Manager Extension screen, enter the Extension configured on the
media server, shown in Figure 30, for the previously added user. Usually, the media server
extension and the user extension are the same (recommended). Select the Communication
Manager Server assigned to this extension. Click the Add button.
Figure 30: Add Media Server Extension
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The last step is to configure the CBAD eVantage SIP proxy/SBC as a trusted host on SIP
Enablement Services. As a trusted host, SIP Enablement Services will not issue SIP
authentication challenges for incoming requests from the CBAD eVantage solution. Specify the
IP address of the SIP proxy/SBC in the IP Address field and set the Host field to the IP address
of SIP Enablement Services. A descriptive comment can be provided in the Comment field.
Figure 31: Add Trusted Host
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6. Configure the Cincinnati Bell Any Distance eVantage
Solution
To use Cincinnati Bell Any Distance (CBAD) eVantage solution, a customer must request
service from Cincinnati Bell using their sales processes. Sales information for Cincinnati,
Dayton, Ohio and Northern Kentucky can be reached at 1-888-CIN-BELL (246-2355). All other
areas should call 1-317-816-5100, Option 1.
The following table contains the configuration information, coordinated with CBAD, which was
used during the interoperability compliance testing to verify the CBAD eVantage Solution.
Feature
Test Configuration
Specify Codec(s) Required:
G.711mu-law
G.729A
RFC2833 DTMF (required)
The network configuration described in these Application
Notes was tested with the codecs (payload types) listed in
the left column.
Note: RFC2833 is required for shuffling SIP calls.
Define Dial Plan
10-digit dialing, directory assistance, toll-free,
international, operator, and collect calls were supported by
the test configuration.
Listed Directory Numbers
provided by CBAD
Listed directory numbers should be assigned to the
endpoints at the enterprise site. This allows calls to be
delivered from the PSTN. In this configuration, listed
directory numbers beginning with area code 513 were
assigned to the SIP, H.323, digital, and analog endpoints
in the enterprise network. In addition, these DID numbers
will be sent as the CPN to the CBAD VoIP network for
authentication.
CBAD provides Proxy IP
Address
The IP address of the Cisco CUBE SIP proxy/SBC to
reach the CBAD eVantage solution was 16.96.81.46.
Customer provides IP Address of
SIP Enablement Services
The IP address of SIP Enablement Services in the
enterprise network was 2.160.183.200. CBAD used this
IP address for routing calls destined to the listed directory
numbers assigned to the enterprise site.
SIP Transport Protocol and Port
SIP signaling was transported between SIP Enablement
Services and CBAD eVantage Solution using UDP and
port 5060.
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7. General Test Approach and Test Results
This section describes the interoperability compliance testing used to verify SIP trunking
interoperability between Cincinnati Bell Any Distance eVantage solution and the Avaya SIP
based network. This section covers the general test approach and the test results.
An enterprise site containing an Avaya SIP based network was connected using SIP trunking (via
general purpose Internet services) to the CBAD eVantage solution. The SIP trunk was
established between SIP Enablement Services and a Cisco CUBE. This allowed the enterprise
site to access the PSTN network through the CBAD eVantage solution. The general test
approach included the following:
•
•
•
•
•
•
•
•
Incoming calls to the Avaya IP network from the PSTN routed through the CBAD VoIP
network.
Outgoing calls from the Avaya IP network to the PSTN routed through the CBAD VoIP
network.
Calls originated and terminated on SIP, H.323, digital and analog endpoints in the Avaya
enterprise network.
Various call types including: local, long distance, international, toll-free, operator, and
directory assistance calls.
Voice calls using G.711 and G.729 codecs, including codec negotiation. For codec
negotiation, the CBAD VoIP network will select its configured preferred codec for the
call.
DTMF transmission using RFC 2833.
Direct IP-to-IP media (also known as “Shuffling” which allows IP endpoints to send
audio (RTP) packets directly to each other without using media resources on the Avaya
Media Gateway).
Telephony features including call transfers, conferencing, call forwarding, call hold, and
EC500. These features were initiated for PSTN calls. See EC500 and call forwarding
issues identified in the observations list.
Interoperability testing of the sample configuration was completed with successful results.
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The following observations were noted:
1. Cincinnati Bell does not support 0+ dialing from their services. Users will hear a
recorded message that the number dialed is out of service. Users may dial 0 to reach an
automated attendant.
2. eVantage does not support FAX capabilities across the SIP trunk to the PBX, but can
provide FAX lines via traditional FXS connections.
3. Cincinnati Bell supports incoming toll free numbers by routing the toll free number to a
specified DID number.
4. EC500: The EC500 feature (i.e., Extension to Cellular) applies to a user who can be
reached at their Avaya desk phone or a cellular phone over the PSTN by dialing a single
DID number. When a call is made to this DID number from the PSTN, the desk phone
and cellular phone should ring simultaneously allowing the user to answer the call on
either phone depending on their location. However, in this configuration, when an
incoming PSTN call arrives to an Avaya desk phone with EC500 enabled, the outgoing
EC500 call to the user's cellular phone over the PSTN is denied by the CBAD VoIP
network. The outgoing call is denied because Avaya sends out the calling number of the
PSTN user, which is unknown to the CBAD VoIP network and can’t be authenticated. In
this case, only the Avaya desk phone will ring since the outgoing EC500 call was denied.
If the call originates from a local Avaya telephone, this issue does not occur because the
CBAD VoIP network can authenticate the local Avaya user, if a DID number has been
assigned to the user.
5. Call Forwarding Off-Net: This issue is similar to the EC500 issue described above in
that an incoming PSTN call delivered to an Avaya station with Call Forwarding enabled
to an off-net PSTN phone will be denied by the CBAD VoIP network because it won't be
able to authenticate the calling number of the PSTN user sent by Avaya. In this case, the
call will not be forwarded and the PSTN caller will hear “busy” tone. If the call
originates from a local Avaya telephone, this issue does not occur because the CBAD
VoIP network can authenticate the local Avaya user, if a DID number has been assigned
to the user.
Workaround: Enable the Special Application (SA8972) – Overwrite Calling Identity in
the system-parameters special-applications form to overwrite the incoming calling
party number (CPN) from the PSTN to the DID number of the local station. The
Overwrite Calling Identity field on Page 4 of the outgoing SIP trunk group (e.g., trunk
group 100 in this configuration) should also be set to (y)es.
6. DTMF Tones to Avaya IP Phones: When shuffling is enabled, IP phones will hear
clicks. If the call is not shuffled, tones will be played out by the media processor board.
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8. Verification Steps
This section provides verification steps that may be performed to verify that the H.323, digital
and analog endpoints can place outbound and receive inbound calls through Cincinnati Bell Any
Distance eVantage solution.
1. Verify that endpoints at the enterprise site can place calls to the PSTN and that the call
remains active for more than 1 minute. This time period is included to verify that proper
routing of the SIP messaging has satisfied SIP protocol timers.
2. Verify that endpoints at the enterprise site can receive calls from the PSTN and that the call
can remain active for more than 1 minute.
3. Verify that the user on the PSTN can terminate an active call by hanging up.
4. Verify that an endpoint at the enterprise site can terminate an active call by hanging up.
5. If Shuffling is enabled, verify that a call originated or terminated on an Avaya IP telephone is
shuffled. To determine if the call is shuffled, identify the trunk member active on the call by
running the status trunk <group> command on the SAT of Communication Manager. Next,
run the status trunk group/member command and check the Audio Connection field. If the
call is shuffled, the field should be set to ip-direct; otherwise, the field would be set to ip-tdm.
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9. Conclusion
These Application Notes describe the configuration steps required to connect an enterprise site
consisting of an Avaya SIP-based Network to the PSTN via the Cincinnati Bell Any Distance
eVantage solution. The CBAD eVantage solution is a SIP-based Voice over IP solution for
customers ranging from small businesses to large enterprises. The CBAD eVantage solution
provides businesses a flexible, cost-saving alternative to traditional hardwired telephony trunk
lines.
10. Additional References
This section references the Avaya documentation relevant to these Application Notes. The
following Avaya product documentation is available at http://support.avaya.com.
[1] Administering Avaya AuraTM Communication Manager, Document 03-300509, Issue 5.0,
Release 5.2, May 2009,
[2] SIP Support in Avaya AuraTM Communication Manager Running on the Avaya S8xxx
Servers, May 2009, Issue 9, Document Number 555-245-206.
[3] Installing, Administering, Maintaining, and Upgrading Avaya AuraTM SIP Enablement
Services, May 2009, Issue 7, Document Number 03-600768.
[4] RFC 3261 SIP: Session Initiation Protocol http://www.ietf.org/
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APPENDIX A: Sample SIP INVITE Messages
This section displays the format of the SIP INVITE messages sent by the CBAD VoIP Network
and the Avaya SIP Network at the enterprise site. Customers may use these INVITE messages
for comparison and troubleshooting purposes. Differences in these messages may indicate
different configuration options selected.
Sample SIP INVITE Message from CBAD eVantage Solution:
No.
Time
Source
Destination
21 9.844315
16.96.81.46
2.160.183.200
sip:5135551234@2.160.183.200:5060, with session description
Protocol Info
SIP/SDP
Request: INVITE
Frame 21 (1165 bytes on wire, 1165 bytes captured)
Ethernet
II,
Src:
Cisco_6a:ef:e0
(00:07:eb:6a:ef:e0),
Dst:
Ibm_84:59:d3
(00:14:5e:84:59:d3)
Internet Protocol, Src: 16.96.81.46 (16.96.81.46), Dst: 2.160.183.200 (2.160.183.200)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:5135551234@2.160.183.200:5060 SIP/2.0
Method: INVITE
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 16.96.81.46:5060;branch=z9hG4bK19E1571
Transport: UDP
Sent-by Address: 16.96.81.46
Sent-by port: 5060
Branch: z9hG4bK19E1571
From: "REDBANK,NJ" <sip:7328523042@as.voip.fuse.net>;tag=91A4ABC-E88
SIP Display info: "REDBANK,NJ"
SIP from address: sip:7328523042@as.voip.fuse.net
SIP tag: 91A4ABC-E88
To: <sip:5135551234@2.160.183.200>
SIP to address: sip:5135551234@2.160.183.200
Date: Thu, 11 Jun 2009 14:22:22 GMT
Call-ID: 1D392B02-55CA11DE-81B3D6BD-1647EF06@as.voip.fuse.net
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 490125922-1439306206-2175719101-373812998
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY,
INFO, REGISTER
CSeq: 101 INVITE
Sequence Number: 101
Method: INVITE
Timestamp: 1244730142
Contact: <sip:7328523042@16.96.81.46:5060>
Contact Binding: <sip:7328523042@16.96.81.46:5060>
URI: <sip:7328523042@16.96.81.46:5060>
SIP contact address: sip:7328523042@16.96.81.46:5060
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 8
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 291
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 3383 8819 IN IP4
16.96.81.46
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Owner Username: CiscoSystemsSIP-GW-UserAgent
Session ID: 3383
Session Version: 8819
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 16.96.81.46
Session Name (s): SIP Call
Connection Information (c): IN IP4 16.96.81.46
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 16.96.81.46
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 16874 RTP/AVP 0 18 101
Media Type: audio
Media Port: 16874
Media Proto: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: ITU-T G.729
Media Format: 101
Connection Information (c): IN IP4 16.96.81.46
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 16.96.81.46
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute Fieldname: rtpmap
Media Format: 18
MIME Type: G729
Media Attribute (a): fmtp:18 annexb=no
Media Attribute Fieldname: fmtp
Media Format: 18 [G729]
Media format specific parameters: annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
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Sample SIP INVITE Message from SIP Enablement Services to CBAD eVantage Solution:
No.
Time
Source
Destination
1 0.000000
2.160.183.200
16.96.81.46
sip:7328523042@16.96.81.46, with session description
Protocol Info
SIP/SDP Request: INVITE
Frame 1 (1484 bytes on wire, 1484 bytes captured)
Ethernet II, Src: Ibm_84:59:d3 (00:14:5e:84:59:d3), Dst: Cisco_6a:ef:e0
(00:07:eb:6a:ef:e0)
Internet Protocol, Src: 2.160.183.200 (2.160.183.200), Dst: 16.96.81.46 (16.96.81.46)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:7328523042@16.96.81.46 SIP/2.0
Method: INVITE
[Resent Packet: False]
Message Header
Accept-Language: en
Call-ID: 80c25eb69d64de1b2174a4c839a00
CSeq: 1 INVITE
Sequence Number: 1
Method: INVITE
From: "Lange Public"
<sip:5135551234@avremote.com:5061>;tag=80c25eb69d64de1b1174a4c839a00
SIP Display info: "Lange Public"
SIP from address: sip:5135551234@avremote.com:5061
SIP tag: 80c25eb69d64de1b1174a4c839a00
Record-Route:
<sip:2.160.183.200:5060;lr>,<sip:12.160.183.202:5061;lr;transport=tls>
To: "7328523042" <sip:7328523042@16.96.81.46>
SIP Display info: "7328523042"
SIP to address: sip:7328523042@16.96.81.46
Via: SIP/2.0/UDP
2.160.183.200:5060;branch=z9hG4bK8383830303033636367461.0,SIP/2.0/TLS
12.160.183.202;psrrposn=2;received=12.160.183.202;branch=z9hG4bK80c25eb69d64de1b3174a4
c839a00
Transport: UDP
Sent-by Address: 2.160.183.200
Sent-by port: 5060
Branch: z9hG4bK8383830303033636367461.0,SIP/2.0/TLS
Content-Length: 214
Content-Type: application/sdp
Contact: "Lange Public" <sip:5135551234@12.160.183.202:5061;transport=tls>
Contact Binding: "Lange Public"
<sip:5135551234@12.160.183.202:5061;transport=tls>
URI: "Lange Public" <sip:5135551234@12.160.183.202:5061;transport=tls>
SIP Display info: "Lange Public"
SIP contact address: sip:5135551234@12.160.183.202:5061
Max-Forwards: 68
User-Agent: Avaya CM/R015x.02.0.947.3
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
Supported: timer,replaces,join,histinfo,100rel
Alert-Info: <cid:internal@16.96.81.46>;avaya-cm-alert-type=internal
Min-SE: 1800
Session-Expires: 1800;refresher=uac
P-Asserted-Identity: "Lange Public" <sip:5135551234@avremote.com:5061>
P-Charging-Vector: icid-value="AAS:342-b65ec2801de649d4c4a17b09a83"
History-Info: <sip:7328523042@16.96.81.46>;index=1,"7328523042"
<sip:7328523042@16.96.81.46>;index=1.1
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
35 of 38
CBTS-CM
Owner/Creator, Session Id (o): - 1 1 IN IP4 12.160.183.202
Owner Username: Session ID: 1
Session Version: 1
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 12.160.183.202
Session Name (s): Connection Information (c): IN IP4 12.160.183.203
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 12.160.183.203
Bandwidth Information (b): AS:64
Bandwidth Modifier: AS [Application Specific (RTP session bandwidth)]
Bandwidth Value: 64 kb/s
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 2456 RTP/AVP 0 18 101
Media Type: audio
Media Port: 2456
Media Proto: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: ITU-T G.729
Media Format: 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute Fieldname: rtpmap
Media Format: 18
MIME Type: G729
Media Attribute (a): fmtp:18 annexb=no
Media Attribute Fieldname: fmtp
Media Format: 18 [G729]
Media format specific parameters: annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
36 of 38
CBTS-CM
APPENDIX B: Specifying Pattern Strings in Address Maps
The syntax for the pattern matching used within SES is a Linux regular expression used to match
against the URI string found in the SIP INVITE message. Regular expressions are a way to
describe text through pattern matching. The regular expression is a string containing a
combination of normal text characters, which match themselves, and special metacharacters,
which may represent items like quantity, location or types of characters.
The pattern matching string used in Avaya SES may use any of the following metacharacters:
•
•
Normal text characters and numbers match themselves.
Common metacharacters used are:
− A period . matches any character once (and only once).
− An asterisk * matches zero or more of the preceding characters.
− Square brackets enclose a list of any character to be matched. Ranges are
designated by using a hyphen. Thus the expression [12345] or [1-5] both
describe a pattern that will match any single digit between 1 and 5.
− Curly brackets containing an integer ‘n’ indicate that the preceding character must
be matched exactly ‘n’ times. Thus 5{3} matches ‘555’ and [0-9]{10} indicates
any 10 digit number.
− The circumflex character ^ as the first character in the pattern indicates that the
string must begin with the character following the circumflex.
Putting these constructs together as used in this document, the pattern to match the SIP INVITE
string for any valid “1+ 10 digit” number in the North American Dial Plan would be:
^sip:1[0-9]{10}
This reads as: “Strings that begin with exactly “sip:1” and having any 10 digits following will
match.
A typical INVITE request below uses the shaded portion to illustrate the matching pattern.
INVITE sip:17325551638@20.1.1.10:5060;transport=udp SIP/2.0
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
37 of 38
CBTS-CM
©2009
Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at devconnect@avaya.com.
MDL; Reviewed:
SPOC 7/27/2009
Solution & Interoperability Test Lab Application Notes
©2009 Avaya Inc. All Rights Reserved.
38 of 38
CBTS-CM
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