the signaling path and proxies all signaling messages. This allows the gatekeeper to know the exact call state and be able to invoke features.

11.4.3 Versions

There are six versions of H.323, which reflect the evolution of this protocol. H.323 is fully backwards compatible, so gatekeepers and terminals must support flows and mechanisms defined in all previous versions. Version 1 was approved in 1996 and was titled “Visual Telephone Systems over Networks with Non-Guaranteed Quality of Service.” Version 2 included alternative call setup schemes to speed up the call setup. Two schemes were added to H.323, called FastStart and H.245 tunneling. Versions 3 and 4 added more features to H.323 and additional annexes. Of interest to Internet devices is the support for H.323 URLs [12], full UDP support instead of TCP, and also the standardization of the use of DNS by H.323 in Annex O. Version 6 is the current version of H.323.

H.323 has carved out two niche areas in current deployed systems—it is widely deployed in small PSTN replacement networks for handling simple phone calls, and it dominates the IP videoconferencing market. Simple PSTN replacement networks that only originate and terminate phone calls without even basic features do not use most of the key advantages of SIP. As a result, they have little incentive to upgrade to SIP until the widespread adoption of SIP eventually makes SIP gateway ports much cheaper than H.323 ports. However, since SIP is earlier in its development cycle than H.323, this is not likely to happen for a number of years. New implementations of these types of systems will likely deploy SIP from the start, seeking to “future proof” the investment, but there is little incentive for deployed systems to upgrade. Also, the availability of commercial SIP to H.323 signaling gateways based on [13] allows both networks to work together and complete calls. The videoconferencing market is dominated by PSTN ISDN devices, which have been deployed since 1990. Since H.323 shares the same heritage, it was designed to easily interwork with these existing systems.

References

[1]Anttalainen, T., Introduction to Telecommunications Network Engineering, Norwood, MA: Artech House, 1999.

[2]Vemuri, A., and J. Peterson, “Session Initiation Protocol for Telephones (SIP-T): Context and Architectures,” BCP 63, RFC 3372, September 2002.

[3]Camarillo, G., et al., “Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping,” RFC 3398, December 2002.

[4]Elwell, J., et al., “Interworking Between the Session Initiation Protocol (SIP) and QSIG,” BCP 117, RFC 4497, May 2006.

[5]Johnston, A., et al., “Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows,” BCP 76, RFC 3666, December 2003.

[6]Zimmerer, E., et al., “MIME Media Types for ISUP and QSIG Objects,” RFC 3204, December 2001.

[7]Arango, M., et al., “Media Gateway Control Protocol (MGCP) Version 1.0,” RFC 2705, October 1999.

[8]International Telecommunication Union, “Implementors’ Guide for Recommendation H.248.1 Version 1 (03/2002) (‘Media Gateway Control Protocol’),” ITU-T Recommenda tion H.248.1, April 2006.

[9]Cuervo, F., et al., “Megaco Protocol Version 1.0,” RFC 3015, November 2000.

[10]Taylor, T., “Reclassification of RFC 3525 to Historic,” RFC 5125, February 2008.

[11]“Packet-Based Multimedia Communications Systems,” ITU Recommendation H.323, 2006.

[12]Levin, O., “H.323 Uniform Resource Locator (URL) Scheme Registration,” RFC 3508, April 2003.

[13]Schulzrinne, H., and C. Agboh, “Session Initiation Protocol (SIP)-H.323 Interworking Requirements,” RFC 4123, July 2005.