The Big 3 in VoIP Standards

When CommWeb set out to test vendors of so-called "IP PBX" solutions, we originally imagined we'd do some interoperability testing between different platforms. But as we talked to vendors, it became clear that standards remain very much in the air and that today's climate of multiple protocols is retarding voice-over-IP (VoIP) deployment.

We also thought it would be a good idea to take inventory of the three major standards that are presently being debated as candidates for voice-over-IP signaling: H.323; MGCP/Megaco; and SIP.

H.323

H.323 defines packet standards for terminal, equipment and services for multimedia communications over local and wide area networks communicating with systems connected to telephony networks such as ISDN. It evolved in the International Telecommunications Union from a series of videoconferencing standards. The initial version addressed communication over IP-based local area networks (LANs). Version 2 extended the protocol for wide area use and general purpose IP networks.

As it exists today, H.323 is an "umbrella" specification, covering several sub-protocols related to call setup and signaling. Chief among these are H.225, which defines the "call signaling channel", H.245, or the "call control channel", and RAS - registration, admission, status. Underlying these are the Real-Time Transport Protocol (RTP) and/or the Real Time Control Protocol (RTCP), which define the basic requirements for transporting real-time data over a packet network.

H.323 handles four major components for a network-based communication system: Terminals, Gateways, Gatekeepers and Multipoint Control Units. Gateways and gatekeepers help negotiate the PSTN interconnection, while multipoint control units (MCUs) enable multiparty audio and videoconferences.

Gateways make it possible to use standard telephones to talk over the Internet instead of multimedia computers. Gateways also handle addressing problems -- a significant issue in IP telephony. To call another multimedia PC user, you must have their Internet Protocol (IP) address. To call someone using a gateway service/product, you need only dial their phone number.

Gatekeepers are database servers that provide address translation and in some cases bandwidth management by mapping telephone numbers and IP addresses. To have a functional IP network that will handle call traffic originating or terminating at regular telephones, gatekeeper services must be used.

One of the major criticisms levied against H.323 is the time and complexity involved in setting up a call. First of all, the protocol uses multiple roundtrip messages to establish signaling and control for any call between two terminals. Moreover, H.323 requires that TCP connections be used to carry the messages, requiring an additional roundtrip exchange. The recently released version 3 is an improvement and includes a "Fast Connect" procedure that effectively consolidates the messages exchanged between terminals and a tunneling procedure that lets H.245 share a single TCP connection.

MGCP/Megaco

The Media Gateway Control Protocol (MGCP) specifies communication between call control elements and telephony gateways. It was conceived partly to address some of the perceived inadequacies of H.323 at the level of centralized network infrastructure. MGCP, in its current form, is a combination of two earlier protocols, SGCP and IPDC.

The International Engineering Task Force (IETF), through its Megaco working group, is working on a standard that uses the same architecture and baseline as MGCP, but supports ATM.

MGCP's central goal is to remain simple. It puts call signaling, control and processing intelligence in call agents or media gateway controllers. Media gateways are telephony gateways that serve as multi-service packet networks, converting audio signals and data packets. They include trunking, voice over ATM, residential, access and business gateways, network access servers and circuit switches.

The MGCP call agent performs all the same call routing functions as a gatekeeper in H.323, but has much tighter control.

Megaco, also known by its ITU designation, H.248, leapt off the starting block at the first Megaco/H.248 Interop event at the University of New Hampshire in August 2000. The event, sponsored by the Multiservice Switching Forum (MSF), the International Softswitch Consortium (ISC), and the UNH Interoperability Lab, united more than 45 representatives from 17 participating vendors, and was the first major step in validating the proposed Megaco/H.248 protocol standard.

Participating vendors tested a range of Megaco-compliant Media Gateways (MG), Media Gateway Controllers (MGCs), parsers and test boxes. The "bake-off" involved placing calls between two nodes of a single gateway as well as between nodes on separate gateways.

SIP

Session Initiation Protocol (SIP) is an application layer control protocol for creating, modifying and terminating sessions with one or more participants. Like MGCP, it is text-based. Work on SIP began in late 1996 as a way of inviting users to join Mbone sessions. It was not until 1998 when the spec was approved as an RFC by the Internet Engineering Task Force (IETF) that SIP began to gain acceptance as an IP telephony protocol.

SIP uses a "request-response" model. This is the same structural model used by HTTP. Unlike MGCP, a SIP call can be initiated and completed strictly between two clients, without the mediation of a call agent. To initiate a session, the caller sends a request to a callee's address in the form of a simple text command, and the callee responds with an acceptance or rejection of the invitation. The call will usually be mediated by a proxy server or a redirect server for routing purposes.

Look for more on VoIP next week at Solution Provider U.