SIP vs H.323

Various standards organizations have considered signalling for voice and video over IP from different approaches. Two of the primary standards in use today are H.323 and SIP. The International Telecommunications Union (ITU) established H.323 as the first communications protocol for real time multimedia communication over IP. SIP is the Internet Engineering Task Force (IETF) approach to voice and video over IP.

H.323 is an umbrella standard that provides well-defined system architecture, implementation guidelines that cover the entire call set-up, call control, and the media used in the call. Whereas H.323 takes the more telecommunications-oriented approach to voice/video over IP, SIP takes an Internet-oriented approach. SIP is a text-based protocol that was designed to work hand in hand with other core Internet protocols such as HTTP. Many functions in a SIP-based network rely upon complementary protocols, including IP. 

The different entities that make up an H.323 network include gateways, terminals, and conferencing bridges, along with a gatekeeper. The H.323 architecture is peer-to-peer, supporting user-to-user communications without a centralized controlling entity. SIP entities include user agents that may operate as a client or server, depending on the role in any particular call. A SIP architecture requires a proxy server to route calls to other entities and a registrar. All other servers and parts of the network are undefined and not mandatory for every call.

H.323 call information is written in binary code, with a defined set of translations for each code. This was done to reduce the size of the transmission and save bandwidth. New codes have to have an agreed-upon definition between parties prior to a call. The standard can be updated, but any additions to the standard require backward compatibility with the existing standard. Features can only be added, not subtracted.

SIP itself only defines the initiation of a session. All other parts of the session are covered by other protocols, which may come from other applications or functions not necessarily designed for real time multimedia over IP. SIP commands are coded in text rather than binary. It's easier to add and understand these codes, but it does increase the size of messages that are sent. This text-coding scheme comes from the Web-browsing scheme, where it has been successful. Numbers don't have to be allocated to commands for each message in advance. If text commands are added, the other side automatically understands them.

SIP is less defined and more open than ITU standards like H.323, but that can result in interoperability difficulties because of different implementations of the standard. Every developer may implement their own version of SIP with unique extensions that aren't included in the basic standard. Two variations used today are SIP-T, which addresses SIP telephony, and DCS, a variation for packet cable voice over IP transmission. In addition to this, there are numerous proposals for using SIP for other applications, such as appliances and instant messaging, each of which have their own extensions that aren't in the basic standard.

While SIP's openness allows more interoperability with other protocols, this same openness can lead to interoperability problems because the lack of definition in the protocol itself means there are a number of different interpretations, each of which may have difficulty interoperating with others. In addition, to date there are more than 80 contributions to SIP, all of which add to the complexity of interoperability issues. "SIP Bake-offs" provide vendors an opportunity to test their products for interoperability. However, as the number of flavors of SIP implementations increase, together with increasing extensions, the completeness and effectiveness of such testing will decrease.

Both protocols provide comparable functionality using different mechanisms and provide similar quality of service. While SIP is more flexible and scalable, H.323 offers better network management and interoperability. The differences between the two protocols are diminishing with each new version. Although there are numerous industry debates about the merits of the two protocols, the truth is that both of them, along with other complementary protocols, are necessary to provide universal access and to support IP-based enhanced services.

Interoperability Scenarios
Both protocols have been widely deployed, so interoperability between SIP and H.323 is essential to ensure full end-to-end connectivity. Because of the inherent differences between H.323 and SIP, accommodation must be made to allow interoperability between the two protocols. In the simplest scenario where both protocols are used within the same administrative domain, call set-up messages must be translated, then RTP can be used for communication directly between a SIP phone and an H.323 phone.

The scenario becomes more complex when SIP and H.323 are operating in separate administrative domains. A gateway is required to translate messages, as well as information on how to find addresses of destination endpoints and convert those addresses so they can be interpreted by the other protocol.

Another issue is capabilities exchange. In H.323, after the call is set up, the two endpoints "announce" what capabilities they have for variables such as compression and video. Because these capabilities are known up front, if a variable -- such as available bandwidth -- changes during the call, the call set-up can be changed in mid-call. This couldnt be done in SIP without initiating a new call.  For interactive multimedia communication, the inability of SIP to allow mid-call capabilities negotiation could be significant.

H.323 defines conferencing as part of the standard, including both centralized and decentralized conferencing. SIP has no definition for conferencing, but there is a process within SIP for conferencing that is similar to H.323, but which has not been formally defined as part of the standard. Conferencing remains open to interpretation, with different approaches in use.

Here To Stay
Both SIP and H.323 are here to stay. There will very likely not be a "winner" or a "loser" in the SIP versus H.323 debate. Both protocols offer strengths and weaknesses. SIP is extremely flexible and can be adapted to a number of implementations. SIP allows for the use of established protocols from other applications, such as HTTP and HTML. Because these tools are already defined, it's easier to add applications like instant messaging or Web conferencing to SIP. For developers, SIP allows use of a variety of existing building blocks for applications that will interoperate with other Internet applications. Meanwhile, H.323 allows better interoperability, network management, and call control.

Instead of concentrating on one standard versus another, the voice/video over IP community needs to work on better ways of ensuring interoperability between standards to provide end-to-end connectivity throughout the network and to offer the value-added IP-centric services that will demonstrate the power of IP-based communications.



"New World" - a relative of Internet protocols -

simple, open and horizontal

"Old World" - complex, deterministic and vertical



Carrier-class solution addressing the wide area

Borne of the LAN - focusing on enterprise conferencing priorities

A simple toolkit upon which smart clients and applications can be built.

It re-uses Net elements (URLs, MIME and DNS)

H.323 specifies everything including the codec for the media and how you carry the packets in RTP

Leaves issues of reliability to underlying network

Assumes fallibility of network - an unnecessary overhead

SIP messages are formatted as text.

H.323 messages are ASN.1 binary encoded, adding complexity

Minimal delay - simplified signalling scheme makes it faster

Possibilities of delay (up to 7 or 8 seconds!)

Slim and Pragmatic

The suite is too cumbersome to deploy easily

Seamless interaction with other media -

services are only limited by the developers imagination

Services are nailed-down and constricted

Many vendors developing products

The majority of legacy existing IP telephony products rely on the H.323 suite