ITU-T Study Groups 11 and 16 have been actively working on IP telephony signaling. SG 16 developed Recommendation H.323 (Packet-based Multimedia Communications Systems). More recently, Study Group 11 has developed another approach to supporting telephony over packet networks: namely, that of introducing a core packet network into existing circuit-switched networks. The protocol developed for communicating between telephone exchanges in this case is known as the Bearer Independent Call Control (BICC) protocol. In addition, SG 11 has also been working on SIP related matters and Recommendation Q.1912.5 defines the signaling interworking between BICC or ISUP protocols and SIP with its associated Session Description Protocol (SDP) at an Interworking Unit (IWU).

Internet Engineering Task Force (IETF) working groups such as IPTEL (IP-telephony), PINT (PSTN-Internet), SIGTRAN (signaling transport), MEGACO (Media Gateway Control) and MMUSIC (Multiparty Multimedia Session Control) have also been working on various Internet Telephony related protocols such as Session Initiation Protocol (SIP), Session Initiation Protocol for Telephony (SIP-T), Stream Control Transport Protocol (SCTP), and Media Gateway Control (MEGACO) protocol. It is worthwhile to notice that the H.248/MEGACO protocol, used for coordinating media gateways from a media gateway controller, has been developed jointly by the ITU-T Study Group 16 and the IETF MEGACO working group.

1.1   SIGTRAN (SIGnaling TRANsport)

The mandate of the IETF SIGTRAN working group is to develop protocols that address the transport of packet-based PSTN signaling over IP Networks, taking into account functional and performance requirements of the PSTN signaling. These protocols support communications between the Media Gateway Controller and the Signaling Gateway.

The SIGTRAN working group has specified SCTP (Stream Control Transmission Protocol), RFC 2960 (proposed standard), and several adaptation layers for transmission of SS7 over IP-based networks. Some adaptation layers are:

• SS7 MTP2 - User Adaptation Layer, RFC 3331 (proposed standard), which transports MTP2 user signaling information between the SG and the MGC;

• SS7 MTP3 - User Adaptation Layer, RFC 3332 (proposed standard), which transports ISUP and SCCP messages between the SG and the MGC.

• ISDN Q.921-User Adaptation Layer, RFC 4233 (proposed standard), which defines a protocol for backhauling of ISDN Q.921 User messages over IP using SCTP.

• V5.2 - User Application Layer (V5UA), RFC 3807 (proposed standard), which defines a mechanism for backhauling of V5.2 messages over IP using SCTP.

The SIGTRAN protocol is still work in progress. RFC 2719 (informational) provides the architectural framework for signaling transport and RFC 3257 (informational) describes the applicability of the Stream Control Transmission Protocol.  RFC 3868 (proposed standard) defines a protocol for the transport of any Signaling Connection Control Part-User signaling over IP using SCTP.

1.2   BICC

The Bearer Independent Call Control (BICC) protocol, being developed by ITU-T Study Group 11, provides a means for existing operators of the PSTN, based upon circuit switched technology, to evolve their networks towards support for Voice over Packet services with minimal impact on their operations.  Although there exists some overlap in functionality between SG 11’s BICC and SG 16’s H.323, the H.323 specification is focused on small and new telecommunications carriers while the BICC specification is focused on the needs of large, incumbent network operators who have installed ISUP networks and want to delay their migration to SIP / SIP-T.

The BICC protocol is based on the CCS7 ISDN User Part (ISUP) protocol and is specified in ITU-T Recommendation Q.1901. The BICC is transported using the Application Transport Mechanism (APM). The BICC protocol is an application of the ISUP protocol definition, but it is not peer to peer compatible with ISUP. This protocol used to be referred to as ISUP+.

BICC Capability Set one (CS1), which supports communications between Media Gateway Controllers, was decided (approved) by SG 11 on June 15^th, 2000.  BICC CS2 (ITU-T Recommendation Q.1902) focuses on other bearer networks including IP networks. It addresses interface and interactions between the Media Gateway Controller and the Media Gateway.  BICC CS2 was approved in July 2001.  BICC CS3, currently under development, adds support mechanisms for end-to-end QoS and interworking with SIP.

1.3   MEGACO/H.248

The IETF MEGACO (MEdia GAteway COntrol) working group and ITU-T Study Group 16 jointly worked on defining the MEGACO/H.248 protocol. The work originated in the IETF MEGACO working group, and most technical discussion and issues closure took place in that environment.

The MEGACO/H.248 is a broadly applicable gateway control protocol. It can be used for a wide range of gateway applications moving information streams from IP networks to PSTN, ATM, and others.  The standard employs a master-slave model in which the source terminal and/or the gateway is a slave of the media gateway controller. 

ITU-T approved Recommendation H.248 June 15, 2000 and the IETF issued a MEGACO protocol RFC 2885 shortly after. RFC 2886 (Errata) records the errors found in the MEGACO/H.248 protocol document [RFC 2885], along with the changes proposed in the text of that document to resolve them. RFC 3015 (proposed standard) is the result of applying the changes in RFC 2886 to the text of RFC 2885.  RFC 3015 obsoletes RFC 2885 and RFC 2886. The ITU-T H.248 Packages Guide Release 1 summarizes packages that have been standardized in the time frame from 06/2000 to 06/2001. 

RFC 3525 - Gateway Control Protocol Version 1 replaces RFC 3015. RFC 3525 incorporates the original text of RFC 3015, modified by corrections and clarifications discussed on the MEGACO E-mail list. H.248 version 2 was completed at the SG 16 meeting of Feb. 2002.  RFC 3525/H.248v2 includes correction updates to RFC 3015/H.248v1 which were in the Implementors’ Guide plus other changes such as: Deprecation of the Modem Descriptor; Clarification of Audit text, and addition of directed audits; Enhancement of Digit Collection; and Addition of Nx64 multiplexing to the Multiplex Descriptor.

H.248 was renumbered when revised on 2002-03-29. H.248 main body, Annexes A to E and Appendix I were included in H.248.1 - “Gateway Control Protocol Version 1”. Subsequent annexes were sequentially numbered in the series, e.g. H.248 Annex F became H.248.2. 

In May 22, 2002, H.248.1 – “Gateway Control Protocol Version 2” was approved. Version 2 include several enhancements to Version 1, such as individual property, signal, event and statistic auditing; improved multiplex handling; topology for streams; improved description of profiles; and ServiceChange capability change. Currently the last Annex added is H.248.45 (“MGC Information Package”). H.248.1 – Gateway Control Protocol Version 3” was approved in September 2005. Version 3 includes several enhancements, clarifications and corrections.

1.4   SIP

The Session Initiation Protocol (SIP), [3], developed by the IETF Multiparty Multimedia Session Control (MMUSIC) working group and currently specified as a Proposed Standard (RFC 3261), builds on a simple text-based request-response architecture similar to other Internet protocols such as HTTP. The proposed standard was published in June 2002. The SIP work drew enough attention to create a separate SIP working group to continue the development. Since its publication, it has been recognized that SIP needs extensions in order to offer carrier-grade telephony applications. The SIP working group is considering proposals to achieve this. These proposals add new functionality to the base SIP protocol, such as the use of MCUs for multiparty conferences; call transfer functionality, reliable provisional responses ("Call proceeding") and pre-call media cut-through.

The Session Description Protocol (SDP), RFC 2327 (proposed standard), is used in SIP to convey session parameters such as media encoding. The MMUSIC group is working to enhance the functionality of SDP.

PacketCable, a project conducted by CableLabs and its members, is the most significant early adopter of SIP.  In the PacketCable initiative it was also recognized that SIP has to be extended to offer carrier-grade services.  This resulted in the PacketCable Distributed Call Signaling (DCS) specification.  Some of the ideas put forward in the DCS specification have been published as Internet Drafts. The PacketCable Call Management Server to CMS protocol uses the Session Initiation Protocol 2.0 (SIP) specification with extensions and usage rules that support commonly available local and CLASS services. This protocol is referred to as the Call Management Server Signaling (CMSS) protocol.

SIP is a peer-to-peer signaling control protocol for creating, modifying and terminating sessions (e.g., conferences, telephone calls and multimedia distribution) with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these. SIP invitations used to create sessions carry session descriptions, which allow participants to agree on a set of compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can register their current location. SIP is not tied to any particular conference control protocol. SIP is designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities.

RFC 3261 covers basic functionality and there are several related Internet drafts covering services.  SIP has rapidly growing industry momentum at the system and device level. There have been several bake-offs with different vendors demonstrating interoperability of basic calls. Currently there are several proposals for extensions waiting for discussion in the working group.

1.5   SIP-T

The IETF Session Initiation Protocol Project INvestiGation (SIPPING) working group is chartered to document the use of SIP for several applications related to telephony and multimedia, and to develop requirements for any extensions to SIP needed for those applications. One of those extensions is to support call/session control. SIP-T (SIP-Telephony) formerly known as SIP-BCP-T (SIP Best Current Practice for Telephony interworking) is a mechanism that uses SIP to facilitate the interconnection of the PSTN with SIP networks. SIP-T is more of an interface agreement on a collection of standards as opposed to a separate protocol.  SIP-T messages carry further sub-messages, such as the complete PSTN User Part message for signaling information and SDP (Session Description Protocol) messages for conveying connectivity end-point information and media-path characteristics. 

As with SIP, SIP-T directly negotiates a media connection between gateways. Endpoint information is carried in SDP from which can describe both IP and ATM endpoints. SIP-T is still a work in progress in the IETF. RFC 3372 (best current practice) provides a description of the uses of PSTN-SIP gateways, uses cases, and identifies mechanisms necessary for interworking.  RFC 3398 (proposed standard) describes a way to perform the mapping between the Session Initiation Protocol (SIP) and the ISDN User Part (ISUP) of Signaling System No. 7 (SS7). In addition, RFC 3578 (proposed standard) describes a way to map ISUP overlap signaling to Session Initiation Protocol (SIP).

1.6   Q.1912.5

ITU-T Recommendation Q.1912.5 (Interworking between Session Initiation Protocol (SIP) and Bearer Independent Call Control or ISDN User Part) defines the signalling interworking between BICC or ISUP protocols and SIP with its associated Session Description Protocol (SDP) at an Interworking Unit (IWU). TRQ.2815 (Requirements for Interworking BICC/ISUP Network with Originating/Destination Networks based on SIP and SDP) specifies the set of common capabilities required to interwork between SIP and BICC.ISUP for three different profiles.

Profile A was defined to satisfy the demand represented by 3GPP in TA 24.229 V5.1.0 (2002-06). The work on this protocol was driven by mobile operators and vendors.  Profile B complements Profile A, and both of them are intended to support traffic that terminates within the SIP network.  Profile C supports the trunking of traffic via transit SIP networks using MIME encoded encapsulated ISUP (SIP-I).  Figure 3 describes the main scope of each profile defined in TRQ.2815.

An agreement between the IETF and ITU-T on how to align these efforts (SIP-T and Q.1912.5) has not been reached. Q.1912.5 was approved on March 2004.

Figure  – Use of SIP/SIP-I Interworking

1.7   H.323

ITU-T Recommendation H.323 (Packet-based Multimedia Communications Systems), [4], addresses how PC telephones or existing telephones via adaptors can be connected to packet networks and inter-work with circuit-switched public telephony networks through gateways. H.323 is part of a larger series of standards that enable videoconferencing across a range of networks. Known as H.32X, this series includes H.320 for narrowband ISDN (N-ISDN), H.321 for broadband ISDN (B-ISDN) and H.324 for General Switched Telephone Network (GSTN) communications. The following diagram illustrates the H.323 protocol stack.

Figure  – H.323 Protocol Stack

Communications under H.323 are a mix of audio, video, data and control signals.  H.323 includes:

• H.245 for control,

• H.225.0 for connection establishment,

• H.332 for large conferences,

• H.450.1, H450.2 and H.450.3 for supplementary services,

•  H.235 for security, and

• H.246 for interoperability with circuit-switched services. 

Audio capabilities, RTP/RTCP media transport, call setup, Registration, Admission and Status (RAS) control and H.245 signaling are required components; all other capabilities, including video and data are optional.

The H.323 standard employs a peer-to-peer model in which the source terminal and/or the gateway is a peer of the destination terminal and/or gateway.  It optionally requires a gatekeeper function analogous to a connection manager.  H.323 is most applicable for implementation in end-points that have integrated processing power.  These include PC-based Internet Telephony clients and VoIP gateways integrated with PBX and key systems with inherent call-processing power. H.323 is the most widely employed standard among first-generation Internet Telephony solutions.

H.323 is a relatively mature set of standards. The first version was approved in 1996 by the ITU-T Study Group 16. Version 2 was approved in January 1998 and version 3 in September 1999.  SG 16 approved version 4 in November 2000, Version 5, in July 2003 and Version 6 in June 2006. The Implementer’s Guide for H.323 Systems document contains a compilation of reported defects identified in the versions of ITU-T Recommendation H.323 and its related Recommendations currently in force.

One of the advantages of using a more mature protocol such as H.323 is that there has been significant interoperability testing leading to multi-vendor interworking. The International Multimedia Teleconferencing Consortium (IMTC) holds several H.323 interoperability events each year since October 1996. These vendor-only events allow developers to do pair-wise testing with other vendors and lead to multi-vendor inter-working as well help to fix inconsistencies in the standard. More than 50 vendors have participated in these events. One of the biggest advantages of H.323 is the maturity and the large degree of multi-vendor interoperability.

Further enhancing the domain of H.323, both the IMTC and iNOW! Working Group have been developing profiles of the H.323 standard for specific applications. Interoperability testing of iNOW! profiles is currently being included in the IMTC interoperability events.

This text is part of the Technical Notebook on NGN of PCC.I

Coordinator: Mr. Wayne Zeuch (Rapporteur of the Standards Coordination Group)