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VoIP Standards Compared

“The best thing about standards is there are so many to choose from”, or so the quote says. This is true of the VoIP standardisation world, and from the mid 90s the standardisation track for media control protocols has been following largely parallel tracks in the ITU-T and IETF...

In the mid-90s the IETF started work on media control protocols for use in the multimedia backbone (MBONE), an experimental IP overlay network for developing multicast applications. At about the same time the ITU-T were developing an evolution of the H.320 video conferencing standards to include packet networks rather than ISDN based transport. H.323 (a collection of protocols referenced from the H.323 umbrella specification) for a while became the de-facto standard for voice over IP, and it was incorporated into a large number of products including the Windows NetMeeting™ application. The work of the IETF continued to gather pace throughout the late 90s resulting in the Session Initiation Protocol (SIP) and Session Description Protocol (SDP). SIP and SDP have now almost universally replaced the H.323 protocol suite for Internet voice applications and corporate VoIP.

During the same time-frame the ITU-T, ETSI and IETF groups were all looking at means of moving the PSTN network onto a fourth generation switching infrastructure based on a packet core (IP). This wasn’t the first attempt, in the late 80s the ITU-T worked on specifications for the enhancement to the Integrated Service Digital Network (ISDN) called Broadband ISDN (B-ISDN). The B-ISDN framework had ATM at its core as a means of delivering a multi-service network, and an architecture of centralised call control and off-switch call processing logic in Intelligent Network (IN) nodes called Service Control Points (SCP). This first attempt was not built in any real-world networks, although the work on ATM standardisation proved valuable in other applications. However standardisation effort continued into the mid 90s.

During the late 90s standardisation efforts in the both the ITU-T/ETSI organisations and in the IETF concentrated on delivering an IP core and a separation of: media bearers, signalling and common control logic across the IP core. This work resulted in two control protocols, Media Gateway Control Protocol (MGCP) and MeGaCo. MeGaCo was originally specified within the IETF, but later adopted with some minor changes as the ITU-T specification H.248. It also resulted in an approach for carrying ISDN signalling traffic (SS7) across an IP network, this was the work of the SIGnalling TRANsport (SIGTRAN) working group in the IETF, resulting in a number of standards for:

  • Stream Control Transmission Protocol SCTP
  • MTP3 User Adaption layer M3UA
  • MTP2 User Adaption - M2UA
  • MTP2 peer-to-peer adaption - M2PA
  • SCCP User Adaption layer – SUA

Also latterly for non-SS7 signalling two other adaptation layers have been defined (the later one, mainly for use in the UK networks), together with V5.2 Adaption layers for Access network signalling.

  • ISDN (Q.931) User Adaption Layer - IUA
  • DPNSS User Adaption layers – DUA

The final approach of both the IETF and ITU-T/ETSI was to harmonise their efforts with the use of the IETF-specified Sigtran protocols and the ITU-T issuing H.248 based upon the IETF Megaco protocol. Further work has been carried out in the ITU-T (Q.1912.5) for interworking between SIP and ISUP (called SIP-I) and Bearer Independent Call Control (BICC) for the control of bearers on either ATM or IP. The BICC Protocol allows the harmonisation of call control irrespective of the underlying transport infrastructure whether it is ATM or IP. It main goal is to allow an integrated approach and migration path from 3G networks utilising ATM as their main core transport mechanism to an IP core.

Meanwhile the IETF have been busy defining SIP-T which allows the encapsulation of ISUP messages as SIP payloads, for interworking between Softswitches without having to map every ISUP feature to SIP methods.

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