IP Telephony Intergateway Protocols

H.323, MGCP, and SIP are common protocols that address specific aspects of the technology needed to develop complex IP telephony systems.

As IP telephony equipment manufacturers move their technologies from the laboratory into the real world, it is clear that the technical challenge of building a scaleable network of end devices and gateways is greater than expected. The technologies needed to encode and transmit voice and fax traffic have been perfected, but the art of call control and address management for large corporate or service provider platforms still needs to evolve.

As a result, a number of protocols that allow IP telephony systems to intercommunicate have been defined. This article examines three common IP telephony protocols currently used in systems: H.323, MGCP, and SIP. These protocols lead an intertwined existence, and are often combined in many applications.

In 1996, the International Telecommunications Union (ITU) adopted the oldest of the three intergateway protocols, H.323, which was followed by an updated Version 2 in January 1998. H.323 is considered to be an umbrella standard, encompassing a number of subordinate standards. Because of this, the ITU can define H.323 by reusing a number of other existing data and telecommunications standards such as Q.931, G.711, and G.723.1.

Originally proposed by Intel and PictureTel, H.323 defined a flexible means for multimedia teleconferencing equipment to communicate and provide application-sharing features over an IP stack. The designers proposed a standard that could be used on a variety of devices including videophones, desktop PCs, and large multiport gateways. As a result, H.323 is comprehensive and offers a variety of media types and compression techniques to be used in various devices.

Version 2 of the standard addresses a number of efficiency and scaleability issues that the original standard did not cover. The result is an H.323 standard that is much better suited to real-world deployment. H.323 continues to be enhanced to better address issues such as scaleability and support for additional applications such as IP fax.

The key strength of H.323 is its maturity, which has allowed a number of software vendors to develop robust implementations. The standard’s maturity has also allowed the various vendors to eliminate interoperability issues, permitting the deployment of a wide range of H.323-capable devices into the market. Since the H.323 standard includes an adaptation of the Q.931 protocol for call-control, many developers with experience in existing ISDN telephony are familiar with the call control model. In fact, the events and parameters can often be directly passed from H.323 into applications that previously operated with ISDN.

The original H.323 Version 1 recommendation suffered from slow call setup because many messages were interchanged between end devices before the voice path was established. The fast call setup features that are allowed in Version 2 have overcome this problem. Because of the complexity of the standard, many products that require basic “quick and dirty” intergateway call control find H.323 too complex or expensive.

When defining H.323, designers work from the perspective of an end device, not a device that resides within the existing PSTN. As a result, H.323 cannot integrate with SS7 or leverage the powerful capabilities that SS7 has to offer. Additionally, H.323’s scaleability has proven to be an issue in very large applications. Designers using deployed gateways that included thousands of ports found the centralized state management to be a bottleneck.

The cost of implementation has been an issue when an inexpensive end device is required. The complexity of the standard requires reasonable processing capability at the end device, which has prevented implementation on devices such as set-top cable boxes and handheld wireless devices.

Based on the market’s reaction to H.323, it looks as if the sweet spot for H.323 is in the 1- to 200-port system, located at or near the end points. H.323 works well in environments where there is enough processing capability to implement the call control and media processing. H.323 has gained its strongest support as an IP telephony solution for enterprises.

Media Gateway Control Protocol (MGCP) provides a means to interconnect a large number of IP telephony gateways, allowing them to work together as one. MGCP assumes that a call agent (CA) performs the intelligence of all call-control operations and that a media gateway controller (MGC) carries out all media processing and conversion.

The MGCP specification was developed by various companies including Telecordia and Lucent and was published as an informative request for comment (RFC 2705) by the IETF. It is the result of a merger between the Simple Gateway Control Protocol (SGCP) and Internet Protocol Device Control (IPDC) protocols, but MGCP is not a recognized standard. The Megaco working group of the IETF and the ITUA are working together to develop a recommendation based on MGCP under the name H.248 (formerly H.gcp). This core document and its related specifications are targeted for completion in February 2000 and will be published as IETF standards track RFCs.

When the H.323 gateway provides the media conversion and the SS7 gateway translates the call-control information, MGCP can be used with an array of H.323 gateways and SS7 gateways. In this case, MGCP relays all call-control information from the end-point device to the network. Using this mechanism empowers the developer to leverage the capabilities provided in the SS7 network and allows much larger IP telephony systems to be deployed than by using H.323 alone.

To negotiate the media path and capabilities for individual calls, MGCP depends on Session Description Protocol (SDP), which is part of the MGCP specification. SDP allows the negotiation of the Real Time Protocol (RTP) port and IP address for the end points, the voice-coding method (such as G.711 and G.723.1), the packetization period, and other connection-type parameters.

MGCP strengths include:

• It is particularly suited to large deployed applications because it was defined to solve a specific problem with large deployed systems.

• Use of MGCP allows for good integration into the SS7 network, which gives greater control and throughput in handling calls.

• MGCP splits the media handling and signaling functions, thus providing a simpler implementation that can be developed by multiple vendors.

The following is a list of some of the protocol’s weaknesses:

• MGCP is too complex for smaller applications.

• MGCP will compete with the H.248/Megaco standards that will be endorsed by the IETF and the ITU in early 2000. Thus, carriers that require MGC may elect to use either MGCP or H.248. Therefore, H.248 implementations may ultimately replace earlier MGCP versions.

Clearly the home for MGCP is in the carrier space — deliver- ing thousands of lines of IP telephony.

The Session Initiated Protocol (SIP) provides a means to communicate call-control information from end devices or proxy servers to each other or to gateway devices. This protocol is the result of the MMUSIC working group of the IETF. As such, SIP is similar to many of the existing Internet protocols, including the popular HTTP.

SIP is considered a lightweight protocol because it uses simple text commands that are easily created and parsed by end devices. SIP only uses six directives to manage the call-control information. This simplicity is key to the SIP protocol being selected for very low-cost applications.

SIP does not define a media-transport mechanism, which allows it to be used in applications where the media transport might be proprietary, thus providing increased efficiency and possibly reduced cost. SIP also allows the call-control messages to be carried over any datagram protocol, making it useful for environments that are not TCP/IP-based (Novell or other proprietary protocols).

Some SIP strengths are:

• The expandable nature of the protocol allows future capabilities to be easily defined and quickly implemented.

• It is simple and easy to embed into inexpensive end-user devices.

• The protocol was designed to ensure interoperability and enable different devices to communicate.

• Non-telephony developers find the protocol easy to understand.

SIP weaknesses include:

• SIP is very new, so most applications are in the prototype stage.

• The protocol has a narrow scope and thus has limited applications by itself; however, it gains flexibility when used with other protocols.

• SIP is only a small piece of a complete solution. Numerous other software components are required to build a complete IP telephony product.

Low-cost end devices are natural applications for SIP. Devices such as wireless phones, set-top cable boxes, Ethernet phones, and other devices with limited computing and memory resources are suited to this protocol.

Because SIP is a stand-alone call-control protocol, it is currently being looked at as the leading candidate to substitute for the call-control portions of MGCP.