X-File - Alphabet Soup

Posted: 07/1999

Alphabet Soup
Voice over IP Brings Heaping Helping of Protocols
By Bob Gaughan

Voice over Internet protocol (VoIP). IP telephony. Internet telephony. No matter how you phrase it, there is little doubt the unification of the Internet and the public switched telephone network (PSTN) has the potential to dramatically revolutionize the ways in which people connect and share ideas.

Presently, the global PSTN is designed to accommodate voice traffic, but increasingly is being used for data. This paradigm shift is driving carriers to build new networks and evolve existing networks to accommodate packets. While they primarily transport web traffic, e-mail, work-group collaboration, e-commerce and other data, a growing portion of these packets contain voice.

As a result, there is a major initiative afoot to allow the PSTN voice traffic to traverse the emerging IP networks and visa versa.

Because this effort involves different models, mindsets and philosophies of how networks should be constructed, network protocols and the standards bodies bringing them forward are growing increasingly important. Protocols such as media gateway control protocol (MGCP), H.323 and session initiation protocol (SIP), some of which may not have been specifically designed for the convergence of data and telephony, will be challenged to ensure that tomorrow's voice network delivers on today's promises.

For example, the "media" in the PSTN (the digital voice contained in the time-division multiplexing [TDM] channel, for instance) travels through a media gateway (MG), where it is translated into the appropriate bits for the IP network. These transmissions between MGs utilize real-time transport protocol (RTP) along with real-time transport control protocol (RTCP). Because these "media gateways" can be numerous, they will need to be kept at a reasonable cost and be optimized for specific purposes.

To this end, the media gateway control working group in the Internet Engineering Task Force (IETF) is in the process of decomposing the media gateway concept so the call control and call level feature intelligence (call, transfer, hold, conference, park, etc.) can be separate from the actual gateway. This creates the notion of a "media gateway controller" (MGC) and separate media gateways.

Essentially, this effort combines proprietary protocols such as simple gateway control protocol (SGCP) and Internet protocol device control (IPDC), as well as MGCP. The naming of the final standard protocol has caused some discussion. Some want to label it MGCP, while others are concerned about confusion and compatibility with existing MGCP implementations. Therefore, alternative names have been discussed, such as MGCP 2 and Megaco.

The scenario above simply addresses the separation of the control function from the actual gateway. There also needs to be a protocol that allows MGC-to-MGC communication. This likely will utilize a modified version of SIP, H.323, or integrated services digital network user part (ISUP).

The International Telecommunications Union (ITU) has established a series of "recommendations" called H.323, which defines protocols and procedures for multimedia communications on, among other things, a nonquality of service IP network such as the Internet. H.323 started out as a protocol for multimedia communication on a local area network (LAN) segment without quality of service (QoS) guarantees, and then was adapted to meet the needs of Internet telephony.

The conventional wisdom is that while H.323 works, it is complex, does not interoperate, and was not designed with Internet telephony in mind. Still, H.323 exists today and, at least for the near term, is the predominant protocol used for Internet telephony. As other protocols are developed, they will either be complementary to H.323 and/or be present in hybrid networks consisting of H.323 and the newer protocols.

One new entrant is SIP, which is being developed by the multiparty multimedia session control IETF working group. SIP takes the approach of signaling by reusing many of the header fields, encoding rules, error codes and authentication mechanisms of hypertext transfer protocol (HTTP).

While SIP is not available yet, it has the potential of adding "voice" to the suite of applications accessible on evolving IP-accessed networks around the world. SIP potentially can take advantage of the evolution of the web and HTTP. For example, one user could attempt to establish a connection to another using SIP and, based on a script that accesses the user's calendar, may be informed the user can be reached at a different number or at a later time.

Both SIP and H.323 are intended to be used between "user interfaces" (what we call telephones today) for things such as connection establishment, capabilities exchange and conference control. So from an instrument designed to be an "IP telephone," one could use SIP or H.323 to set up the connection.

MGCP, H.323 and SIP are but a subset of the many protocols being developed to allow the interconnection of the PSTN and the Internet and to enable the transport of voice across IP networks. Underlying these protocols are different views as to how telephony should be modeled going forward. One is an environment where an "intelligent" network provides services to relatively "dumb" instruments, similar to today's telephone network. The other model is one of "intelligent" devices creating their own services over a "dumb" network, analogous to the Internet today. In such an environment, voice then becomes an application similar to file transfer protocol (FTP), e-mail or web traffic. For the foreseeable future, a mix of these models and protocols will be necessary as the PSTN and the Internet continue on a path toward convergence.

Bob Gaughan is director of technology marketing at Richardson, Texas-based Nortel Networks. He can be reached at (978) 288-3678.

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