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Feature Article
August 2003


H.323 Perseveres

BY PAUL JONES & LARRY SCHESSEL

H.323, an industry-standard protocol suite for converging audio, video, and data communication over packet-switched networks, is widely deployed over both private and public IP networks. It forms the network foundation for most of the world�s VoIP services, and installations are growing.

A recent study by an independent testing company, for example, revealed that of 23 IP PBXs from 16 vendors, 20 supported H.323 for call control, 21 supported the protocol for call signaling, and 15 supported it for delivering features to endpoints. By contrast, the closest competing protocol to H.323 was supported by just one IP PBX for call control and by two products for signaling. On the Service Provider side, the H.323 Forum (www.h323forum.org) identifies over 70 Service Providers worldwide using the H.323 protocol to carry public voice traffic as well as nine carriers each carrying over 1 billion voice minutes per month.

Since its inception in 1996 as a private, LAN-based conferencing protocol, H.323, an International Telecommunications Union -- Telecommunications Sector (ITU-T) standard, has evolved to fuel carrier-grade public Internet telephony and conferencing services. Significant scalability enhancements to the protocol over the years have made these wide-ranging implementations possible.

In May of this year, the ITU-T approved H.323 version 5, which concentrated on �protocol hardening,� or stabilizing the core feature set. This effort began with version 4, but is the primary focus of version 5. Going forward, new H.323-standard features will generally be developed through extensions to the base protocol, with few changes expected for the core protocol.

Together, the scalability improvements and stabilized code have rendered H.323 both extensible and mature, traits that have contributed to H.323�s market leadership. Initially having borrowed heavily from its H.320 predecessor for ISDN-based video conferencing systems, H.323 is now used in commercially available network services that include toll bypass, residential voice and video, wholesale voice transit, PC-to-phone, and videoconferencing services.

More than 90 percent of VoIP traffic is carried using H.323, and it is supported on 80 percent of new videoconferencing systems. On public networks alone, billions of minutes of traffic per month worldwide run on H.323, according to the H.323 Forum.


A �Carrier Grade� Protocol
Scalability enhancements to H.323 that have enabled the large-scale deployment of the protocol include the following:
� Use of direct H.225.0 call signaling between endpoints such as gateways, user terminals, and multipoint conferencing units (MCUs).
� A reduction in the number of messages that must be exchanged between network elements to set up and connect calls.
� Call capacity advertisements from gateway to gatekeeper.
� A well-defined means of performing address resolution across a large carrier backbone.
� A means to recover from a network device failure.


Let�s take a brief look at each of these characteristics.

Direct Endpoint Signaling

It is not necessary for an H.323 gatekeeper to route the call signaling for every call. In fact, this is not typical in most large deployments.

H.323 gatekeepers are the network elements that determine how to route calls. They can become network bottlenecks when routing call signaling on a very large scale. Instead, in large networks, gatekeepers usually just resolve the IP address of where a call should be routed and send the address directly to the calling endpoint. Signaling takes place directly between the two endpoints, rather than bogging down the gatekeeper.

If desired, gatekeepers can route the call signaling as a way to provide network-based services to users. Such network architectures have been successfully deployed. However, key to building large H.323 networks has been the large-scale use of direct signaling and applying routed signaling only where network-based services are needed.

Reduced Messaging To Set Up And Connect Calls
Another scalability enhancement to H.323 networks is a feature called Fast Connect. Depending on the transport layer protocol in use, such as Transmission Control Protocol (TCP) or User Datagram Protocol (UDP), it is possible to establish an H.323 call in as few as 1.5 roundtrip messages. Calls to check an IP voice mailbox could be set up and connected in just two messages: Setup and Connect.

With Fast Connect, an H.323 endpoint sends a Setup message proposing an open call with another endpoint using one of three popular codecs: G.711, G.729, or G.723. The called endpoint selects one of those codecs and returns it in the reply. For most endpoints, this works, and calls are established with a minimum of messages. Also, with the use of a new feature called Extended Fast Connect, the mechanisms of Fast Connect may be re-used to re-negotiate capabilities on-the-fly while the call is established or being established.

In a worst-case scenario, the receiving endpoint doesn�t support one of the proposed codecs. So the endpoints open an H.245 multimedia control protocol session and exchange several handshaking messages to establish media channels.

For negotiating more complex capability sets, the H.245 control channel is often necessary. The reason is that, as the number of options and parameters within an endpoint�s capability set increases, so does the need for a well-defined negotiating mechanism. H.323 endpoints can establish calls and media channels quickly and also have the luxury of exchanging and using richer capability sets through the use of H.245.

Another scalability improvement is the ability to tunnel H.245 messages within the call signaling channel, rather than a separate TCP connection. Using H.245 tunneling, normal call signaling messages also carry H.245 messages. This consolidated-message approach removes the latency of signaling and reduces the number of messages that the endpoints must process. Further, when TCP is used as a transport, it reduces the number of socket connections required for a single call.

Call Capacity Advertisement And Load Balancing
H.323 gateways can advertise their call capacity to gatekeepers for efficient load balancing among available gateway resources. The gatekeeper serves as a �traffic cop,� forwarding traffic to a destination gateway that it knows has the resources to terminate a call. Without the gatekeeper�s ability to monitor gateway resources, calls might frequently fail.

Gateways can report their availability to gatekeepers on a full-system basis or more granularly, based on the capacity of a group of DS0 (64 Kbps) circuits.

To overcome gateway failures, gatekeepers can choose primary and alternate gateways for terminating a call. If the most preferred gateway fails, the gatekeeper can route the call via the alternate gateway.

Address Resolution
H.323 gatekeepers play a vital role in resolving addresses within H.323 networks. Typically, an endpoint does not have the wherewithal to resolve the address of the destination endpoint, because it cannot account for variables such a remote gateway�s current load or the least-cost route. The technical and business rules that drive the address resolution functions -- especially within the service provider network -- support the notion that such functions should be left to network elements such as the gatekeeper.

A single gatekeeper often does not know the address of a remote endpoint, either. In very large public networks, there are generally many routes, rules, and interconnection points. Gatekeepers generally communicate with other gatekeepers to resolve addresses, which might account for gateway resource availability, least-cost routing requirements, necessary quality-of-service (QoS), bandwidth requirements, and so forth.

Recovering From Device Failure
In order for H.323 to succeed in the service provider market, it is essential to prevent the loss of calls due to network device failures. H.323 has the wherewithal to recover from a failed network connection without dropping the call. Of course, there is little that can be done from the H.323 signaling standpoint if the physical connection to a DS0 is lost, but it is certainly possible to �route around� failed H.323 entities. Work currently underway within the ITU will also allow intermediate call signaling entities (such as Gatekeepers that route call signaling, proxies, etc.) to safely remove themselves from the call path. This leads to improved scalability and reduces chances of call failure.

STILL FOCUSED ON MULTIMEDIA
With so much attention given to scalability, performance, and robustness issues, one might be led to believe that H.323 is less useful for multimedia and is primarily a �voice� protocol. Certainly, that is not the case. In fact, H.323 still retains all of the strong multimedia capabilities that it has had from the beginning. Along the way, additional capabilities have been added to truly support a rich multimedia environment for the user, including the use of text messages, audio, video, electronic whiteboarding, and application sharing. While some of the multimedia components are not part of the �core� H.323 protocol, those capabilities are tightly and smoothly integrated for seamless operation.

What About SIP?
Interestingly, H.323 development began at the same time as that of the Session Initiation Protocol (SIP), an emerging Internet Engineering Task Force (IETF) standard. Whereas H.323�s focus is on voice, video, and data conferencing, SIP is currently more oriented toward VoIP-only implementations, in that it doesn�t support video control capabilities. Even so, SIP has targeted a range of enhanced applications for which H.323 was not designed, including instant messaging, and multiplayer video games.

The choice of H.323 versus SIP often comes down to business and application requirements. For example, if one wanted to build a client/server multimedia application, SIP would be a clear winner as it has the basic capabilities required for a closed client/server environment. However, if one wanted to deploy a carrier network that supports video and far-end camera control, SIP would not be the right choice, as it is still missing some necessary features to perform that function. Specifically, SIP does not include a video control channel nor a far end camera control protocol, two features vital for video services.

In some cases, SIP and H.323 can complement each other. For example, a call agent might use SIP when directing a call to voicemail and H.323 for the trunking interface. This is an example where users can leverage the strengths of each protocol and choose the protocol that is most suitable for the task.

As it has matured, H.323 has established itself as the current market-leading protocol for running voice and video over IP networks. In the future however, as other protocols are introduced, network operators will likely look at establishing interoperability between them to gain the benefits of each protocol.

Paul Jones is rapporteur of the ITU-T Q.2/16 committee, which is responsible for standardization of H.323, and is a voice systems architect at Cisco Systems. Larry Schessel is president of the H.323 Forum and manager of product marketing, protocols, and session applications at Cisco.

[ Return To The August 2003 Table Of Contents ]


Internet Telephony Protocols: Is Any One The Winner?

BY MICHAEL O'HARA

Engage any telecom expert in a discussion of packet telephony, and your conversation will likely be peppered with references to a wide variety of networking protocols. While there are many protocols important in deploying a converged voice and data network, four protocols stand out as being most popular in implementing voice over packet networks: H.323, Megaco/H.248, Media Gateway Control Protocol (MGCP), and Session Initiation Protocol (SIP). Each protocol has its strengths and weaknesses with respect to issues such as ease of implementation, extensibility, and suitability for various network applications, Quality of Service (QoS), and security. The typical next-generation network may include one or more of these protocols.

Protocol Overview
In comparing these protocols, one main differentiator is the model in which they distribute intelligence. H.323 and SIP operate between peer clients, while MGCP and Megaco operate between �master and slave� entities. With master/slave devices such as MGCP and Megaco-based gateways, IADs, and telephones, the control model is quite similar to traditional telephony equipment: The call agent supplies all instructions to the �dumb� end device, directing it to wait for signals, collect digits, play tones, open ports, and release connections. This offers simple implementation, low-cost end devices, and few interoperability issues. On the flip side, intelligent SIP-based endpoints trade lower cost and ease of network implementation in favor of a model that delivers much richer services.

H.323
Developed by the ITU, H.323 actually encompasses several protocols, including H.245 for media control, H.225 for connection establishment between endpoints, H.332 for large conferences, H.450 for supplementary services and Real Time Protocol (RTP) for transport (SIP also uses RTP for transport, which simplifies interworking between the two.) H.323 was initially developed for multimedia conferencing over local-area networks (LANs). Although it is the most widely deployed packet telephony standard today, H.323 is rapidly losing ground to SIP, due to the perceived inflexibility of the H.323 protocol suite. However, the standards continue to evolve to try and accommodate the needs of Internet telephony, including increasing efficiency and supporting additional services.

SIP
Developed by the IETF, SIP is a lightweight, text-based signaling protocol used for establishing sessions in an IP network. It uses many of the constructs and concepts of Internet protocols such as HTTP and SMTP. Based on principles gained from the Internet community, SIP is an application-independent protocol, which was designed at the outset to be extremely flexible and extensible. As the name implies, SIP deals generically with sessions, which can include voice, video, or data. The sessions are described using a separate protocol called Session Description Protocol (SDP). SDP is transported in the message body of a SIP message. The media that is actually exchanged in a session is transparent to SIP. SIP has been enthusiastically embraced for next-generation applications such as telephony over packet services, voice-enabled e-commerce applications, presence management, instant messaging services, and voice-controlled Web browsing.


MGCP and Megaco
While SIP and H.323 have distinct differences, MCGP and Megaco share many similarities. Both operate in a master/slave configuration where a media gateway controller instructs the media gateway to establish, control, and release connections between one or more media streams. The similarities between the two can be traced to the roots of the protocols. Megaco was the final derivative of several draft standards, including MGCP. In general, MGCP and Megaco have constructs to accomplish the same types of tasks. However, because MGCP was a precursor to Megaco, Megaco has refined and extended many of the functions. As such, Megaco is a more complex protocol.

The Megaco model allows for more flexibility and finer control by the media gateway controller. In terms of resource reservation and control, media processing and stream management, Megaco has greater capabilities as well, which makes it a better protocol for applications such as multimedia conferencing. Megaco is much more flexible when it comes to the underlying transport type. While MGCP defines only UDP as a transport layer for signaling messages, Megaco allows TCP, UDP, SCTP, and ATM. Megaco also has better resource allocation and stream management mechanisms.

Because the functions of MGCP and Megaco concentrate only on the media stream, both can enable the same types of telephony applications, although the procedure for implementation may be much simpler in one protocol or the other. Conversely, a simple implementation is often offset by added flexibility.

Which Protocol Comes Out On Top?
In reality, there is no single protocol that wins out over the others. H.323, SIP, MGCP, and Megaco are all integral protocols in the world of Internet telephony. Similar services can be offered using various combinations of these protocols. For example, a multimedia conferencing application could be offered using H.323 or using a combination of SIP between the media gateway controllers and MGCP between the media gateway controller and media gateway.

For carriers and vendors, the decision as to which protocols to embrace must begin with the decision as to where to locate the network intelligence and control. Some carriers may be most comfortable with the �dumb� terminal model of H.323, which allows for easier network management and upgrades. Some may want the easer implementation for a rich class of services offered by SIP endpoints. The MGCP/Megaco model is particularly well suited for low-cost media gateways used for access such as IADs and IP telephones. What is clear, however, is that despite all the debate, the consensus is that H.323, SIP, MGCP, and Megaco will all be a part of the next-generation voice network.

Michael O�Hara is the vice president of marketing at Sonus Networks. Sonus is a leading provider of voice infrastructure products for the new public network. The company�s solutions are designed to enable service providers to quickly and effectively deploy an integrated network capable of carrying both voice and data traffic, and to deliver a range of innovative, new services.

[ Return To The August 2003 Table Of Contents ]



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