VoIP Module
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VoIP Module
Oxygen provides the Elements VoIP hardware components in a separate module that connects to Hydrogen as a clip-on, enabling
Internet telephony. Oxygen contains two RJ11 telephony ports for connecting standard analogue telephones, DECT/cordless base
stations or fax.
Oxygen does not contain software;; this is managed and run
using Hydrogen´s integrated processor, avoiding additional
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Oxygen´s voice stacks and codecs are enhanced by a
packetloss and jitter reducing VoiceEngine. This dramatically
reduces customer complaints and support costs for the
operator, a common challenge as VoIP often originates from
Smart phones and Wi-Fi hotspots.
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maintained.
In addition to the advanced VoIP client, Hydrogen monitors call
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reporting system.
When combined with Fusion management software, this
allows the monitoring of all calls deployment wide and
generates statistics, diagrams and reports displaying service
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47 mm
175 mm
47 mm
Oxygen VoIP module front view
Oxygen VoIP module back view
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Highlights
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however it only activates, when the Oxygen module is
connected to the Hydrogen router:
‡ Integrated voice engine. Will recover from 20% to
30% incoming packet loss, reduce jitter and delay to
a level that is not detected by the customer
Once connected:
‡ Industry standard SIP (RFC3261) enabled for both
IPv4 & IPv6 operations
Step 1:
‡ A wide variety of audio codecs, both narrow band
and wide band
Step 2:
Step 3:
Oxygen is detected and the voice
applic ation is initiated.
The TR-104 parameters are enabled
as part of the TR-069 provisioning.
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becomes visible in the web interface.
‡ Hydrogen sends SYSLOG messages notifying the
MOS levels (5 to 10 second internals)
‡ Monitors call connect, disconnect and SIP
registration
‡ High availability VoIP with ICE and multiple,
concurrent outbound proxies
Oxygen
Hydrogen
LAN4
LAN3
LAN2
LAN1
WAN
USB
eSATA
47 mm
reset
175 mm
205 mm
12V DC
on/off
PHONE 1
PHONE 2
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Oxygen Features
Signalling and Media negotiation for VoIP
‡ RFC 3261 Session Initiation Protocol, supporting
transports UDP/TCP/TLS over IPv4 and IPv6.
‡ RFC 3262 Reliability of provisional responses in
Session Initiation Protocol.
‡ RFC 3263 Locating SIP servers. The implementation
and usage of DNS NAPTR, SRV and A (IPv4) or AAAA
(IPv6) records.
‡ RFC 4566 SDP: Session Description Protocol
(obsoletes RFC 2327).
‡ RFC 3266 Support for IPv6 in Session Description
Protocol (SDP).
‡ RFC 3264 An Offer / Answer model with Session
Description Protocol (SDP)
‡ RFC 4317 Session Description Protocol (SDP) Offer/
Answer Examples.
‡ 5)&$FWLRQV$GGUHVVLQJ,GHQWLÀHG,VVXHVZLWK
the Session Initiation Protocol’s (SIP) Non-INVITE
Transaction.
‡ 5)&3UREOHPV,GHQWLÀHG$VVRFLDWHGZLWK
the Session Initiation Protocol’s (SIP) Non-INVITE
Transaction.
NAT traversal and keep-alive
‡ RFC 3489 STUN – Simple Traversal of User
Datagram Protocol (UDP) Through Network Address
Translators (NATs). (For backwards compatibility).
‡ RFC 5626 Managing Client-Initiated Connections in
the Session Initiation Protocol (SIP)
‡ RFC 5627 Obtaining and Using Globally Routable
User Agent URIs (GRUUs) in the Session Initiation
Protocol (SIP).
‡ RFC 3327 Session Initiation Protocol (SIP) Extension
Header Field for Registering Non-Adjacent Contacts
‡ RFC 3581 An Extension to the Session Initiation
Protocol (SIP) for Symmetric Response Routing.
‡ RFC 5245 Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator
(NAT) Traversal for Offer/Answer Protocols.
Media transport
‡ RFC 3550 A Transport Protocol for Real-Time
Applications.
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Conferences with Minimal Control.
‡ RFC 5057 Multiple Dialog Usages in the Session
Initiation Protocol.
‡ RFC 3605 Real Time Control Protocol (RTCP)
attribute in Session Description Protocol (SDP).
‡ RFC 2976 The SIP INFO Method.
‡ RFC 3711 The Secure Real-Time Transport Protocol
(SRTP).
‡ RFC 3311 The Session Initiation Protocol (SIP)
UPDATE Method.
‡ RFC 3515 The Session Initiation Protocol (SIP)
REFER method.
‡ RFC 3891 The Session Initiation Protocol (SIP)
“Replaces” header.
‡ RFC 3892 The Session Initiation Protocol (SIP)
Referred-By Mechanism.
‡ RFC 5589 Session Initiation Protocol (SIP) Call
Control – Transfer.
‡ RFC 4579 Session Initiation Protocol (SIP) Call
Control – Conferencing for User Agents.
‡ RFC 3420 Internet Media Type message/sipfrag.
‡ RFC 3966 The tel URI for Telephone Numbers
‡ RFC 4028 Timers in the Session Initiation
Protocol (SIP)
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‡ RFC 3842 A Message Summary and Message
Waiting Indication Event Package for the Session
Initiation Protocol (SIP).
Signalling for IM
‡ RFC 3856 A Presence Event Package for the Session
Initiation Protocol (SIP).
‡ RFC 3863 Presence Information Data Format (PIDF)
‡ RFC 4568 Session Description Protocol (SDP)
Security Descriptions for Media Streams.
‡ RFC 2833 RTP Payload for DTMF Digits, Telephony
Tones, and Telephony Signals.
‡ RFC 4733 RTP Payload for DTMF Digits, Telephony
Tones, and Telephony Signals. (Obsoletes RFC 2833).
‡ RFC 3611 RTP Control Protocol Extended Reports
(RTCP XR).
‡ RFC 3952 Real-Time Transport Protocol (RTP)
Payload Format for internet Low Bit Rate Codec
(iLBC) Speech.
Description Protocol (SDP)
‡ RFC 3711 The Secure Real-Time Transport Protocol
(SRTP).
‡ RFC 4568 Session Description Protocol (SDP)
Security Descriptions for Media Streams.
‡ RFC 2833 RTP Payload for DTMF Digits, Telephony
Tones, and Telephony Signals.
‡ RFC 4733 RTP Payload for DTMF Digits, Telephony
Tones, and Telephony Signals. (Obsoletes RFC 2833).
‡ RFC 3611 RTP Control Protocol Extended Reports
(RTCP XR).
‡ RFC 3952 Real-Time Transport Protocol (RTP)
Payload Format for internet Low Bit Rate Codec
(iLBC) Speech.
‡ RFC 3903 Session Initiation Protocol (SIP) Extension
for Event State Publication.
‡ RFC 3428 Session Initiation Protocol (SIP) Extension
for Instance Messaging.
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Codecs and audio processing for voice
Calling features
‡ G.711 u-Law and G.711 A-Law codecs
‡ Call transfer (attended, unattended)
‡ G.729AB codec
‡ Call forwarding (unconditional, busy, no-answer)
‡ G.723.1 codec
‡ Call hold/release
‡ G.726 codec
‡ 3-way local conferencing
‡ G.722.1 wideband codec.
‡ Redial
‡ AMR-NB codec.
‡ Call return (call the last caller)
‡ RFC 3951 Internet Low Bit-Rate Codec (iLBC).
‡ Caller ID display (FSK, DTMF modulation)
‡ Voice Activity Detection (VAD) for codecs that do not
have this built-in.
‡ Call Waiting indication (tone and visual)
‡ Message Waiting Indication (tone and visual).
‡ Comfort Noise Generation (CNG) for codecs that do
not have this built-in.
‡ Three-way calling - local conferencing
‡ G.168 Line Echo Cancellation.
‡ Message waiting indication
‡ Tone generator.
‡ Dial plan support (as described in RFC 3435,
section 2.1.5)
‡ DTMF tone detector and automatic suppression.
‡ Flexible media handling for local conferencing.
Supports participants with different sample-rates,
as well as participants with different packet size /
packet time.
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‡ Caller ID presentation (DTMF, FSK-Belcore/ETSI, BT,
JAPAN-CLIP/JCLIP)
‡ Fixed and Adaptive jitter buffer, capable of triggering
Packet Loss Concealment (PLC) functionality.
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‡ Gain control
Codecs and processing for FAX
‡ Support for T.38 with V.17 (or slower) Fax
modulation.
‡ Support for T.38 with UDPTL network transport.
‡ Support for T.38 with RTP network transport.
FXS
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‡ RJ11 sockets
‡ Voice & fax capabilities
‡ Auto-detection of Fax tones, disable / enable media
and FXS features accordingly.
‡ G711 pass-through FAX.
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