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The Best for Your VoIP: SIP versus H.323
3/6/2008, Pavlína Králová
Today the classic telecommunication technologies like analog services or ISDN are being replaced by Internet technologies more and more. The Voice over IP can already replace these technologies fully today, even though for a certain price.
It offers significant simplification of infrastructure, where no telephone or data cables have to be distributed throughout the buildings and also increasing the amount of wire lines can be done much easier because mostly there is no need to perform hardware intervention into the telephone exchange or to increase the number of wires physically.
If we skip the proprietary (closed) protocols, the world of Voice over IP is controlled by two core protocols. The solution is based on the Session Initiation Protocol (SIP) and on the H.323 protocol set. In the following text, we will introduce both technologies concisely and then we will perform their comparison.
The first version of H.323 protocol set was published by International Telecommunication Union in 1996. It has not been designed entirely for supporting voice services only; it also enables image transmission or videoconferencing. Simply said, H.323 has been designed by the professionals in the field of classic telecommunication services, and the underlying design corresponds to that. All signalization, which, by the way, is very similar to ISDN signalization, is performed in binary format only; therefore, the human being cannot read it easily. The analyzer must be used for tuning. Moreover, it is a closed protocol with problematic options of expanding. The reliable TCP protocol is used for transmission of signalization.
Architecture of the system based on H.323 consists of several parts. The most important is the H.323 gatekeeper, which provides registration of the client terminals and communication control. It is not the essential component of the system; nevertheless, if present, all terminals must utilize its services. The basic function of the gatekeeper is conversion of network addresses into understandable format for H.323 and vice versa. It is also capable of managing transfer capacity. That can be used for example when the maximum number of simultaneous connections in LAN are specified within the network management – in such case the gatekeeper can refuse to deploy multiple connections if the value is exceeded. The H.323 gateway is used for interconnecting the H.323 domain with another type of network. It secures conversion of signalization and multimedia information. The Multi Control Unit (MCU) must be incorporated into the system whenever a conferencing discussion among three or more participants must be performed. The biggest advantage is the conversion of the multicast broadcast to the unicast broadcast because mixing of audio and video data is not done at the end-user devices but actually at the MCU. The only truly necessary component in telecommunication networks utilizing H.323 is the terminal, which is usually a telephone set. Each terminal must be capable of providing voice services, the support for video transmission is not obligatory.
H.323 is used for providing services under the "classic" approach, which counts on "dumb" telecommunication terminals and "smart" telephone exchanges. This means that most of the system logic as well as support for services are concentrated in the telephone exchange. This type of approach makes the end-user devices cheaper but it requires higher investment into the telephone exchange. In addition, deployment of new services is somewhat troublesome because in addition to the replacement of terminals, it is often necessary to perform at least an upgrade of the telephone exchange.
With H.323, the Real-Time Transport Protocol (RTP) is used for transmission of multimedia information, which, as opposed to signalization, utilizes the less-reliable transmission via UPD.
The Session Initiation Protocol (SIP) was created in 1998 under patronage of IETF (Internet Engineering Task Force). As opposed to H.323, the implementation of which is more similar to classic telecommunication services, the text-oriented SIP is very similar to most-widely used Internet protocols like HTTP or SMTP. Thanks to that, it is quite easily readable by humans and usually no protocol analyzers are necessary. SIP, as opposed to H.323, is not a self-contained platform; it is fundamentally the signalization protocol only. Same as with H.323, the RTP protocol is used for transmission of multimedia information. Despite the option of utilizing TCP, the UDP is being used in most of the cases, which has a positive effect on network load.
SIP creates a client-server type connection, whereat, same as for example with HTTP, it is a stateless protocol. Therefore, each of the nodes must be equipped with both, the client and the server part. The most important and simultaneously the only obligatory equipment of the SIP network are the client terminals called the User Agents. In this regard, the SIP and H.323 do not differ. The other, this time the optional components, are the SIP Proxy Server, SIP Registrar Server, and SIP Redirect Server. Same as with H.323, these components are optional only in a virtual sense. If you want to have a functional telephone system, you will need them. The SIP Registrar Server is the key element of the functional telephone network; it is used for registration of the User Agents and their subsequent addressing. The SIP Proxy Server is used for routing the conversations based on information for SIP Registrar Server. Thus, all signalization at SIP Proxy Server is translated and SIP is not transparent. The SIP Redirect Server is used for similar purposes as SIP Proxy Server; however, it does not hand over the signalization but only refers to the other node. Its usage is not frequent in common circumstances.
In real-time deployment, differentiation between the SIP Registrar Server and the SIP Proxy Server is more or less rubbed out because most implementations cover these functions with single hardware. This is the reason why literature sources in configuration of various end-user devices commonly interchange or simplify these terms to a sole "SIP Server".
Differences between H.323 and SIP
Already from the aforementioned information, we can get partial idea about differences between the technologies built on protocol sets of H.323 and SIP. The SIP is being currently preferred over H.323. Mainly the fact in favor of SIP is its significantly higher simplicity, extendibility and analyzability as opposed to H.323, provided all possibilities remain intact. These favorable characteristics are accompanied by lesser effort on centralization of services as well. As mentioned earlier, the H.323, same as with all classic telephone services, presupposes all logic to be integrated in the telephone exchange and the end-user devices utilizing the services of telephone exchange only. SIP prefers the opposite approach, which in the end admittedly makes the end-user devices more expensive, but on the other hand, it significantly lowers the cost of acquiring the telephone exchange. In addition, the deployment of new functions can be done through a sole exchange of end-user devices. Moreover, large product portfolio of end-user devices is available by default for SIP in all price categories. The market also offers relatively large number of various telephone exchanges and the open-source solutions can be used for complex installations. Nevertheless, for the home-office use or for the use within small-size organizations, the best choice would be to utilize a ready-to-use solution, which could be for example Signamax 065-9101 SIP Private Branch eXchange, which integrates the function of SIP Private Branch eXchange and the function of SIP Proxy Server into a single compact device.
For the deployment of VoIP technology within your home-office or your organization, you will be most likely interested in differences affecting practical design or run of the telephone system. These systems are shown in the table below:
Investment into SIP devices is much more beneficial for smaller and simple installations suitable for home-offices and small and mid-size companies due to lower prices and wider portfolio of the end-user devices. Nevertheless, H.323 is still being used in professional product portfolios of the big players offering complete systems of all telephone network components. H.323 is also used for videoconferencing more often because of current technological insufficiency of SIP. The choice is dependent on the technology your telecommunication operator uses. The conditions in the Czech Republic call for usage of SIP, it is therefore more beneficial for the customers to build their systems on SIP platform.
If we skip the proprietary (closed) protocols, the world of Voice over IP is controlled by two core protocols. The solution is based on the Session Initiation Protocol (SIP) and on the H.323 protocol set. In the following text, we will introduce both technologies concisely and then we will perform their comparison.
The first version of H.323 protocol set was published by International Telecommunication Union in 1996. It has not been designed entirely for supporting voice services only; it also enables image transmission or videoconferencing. Simply said, H.323 has been designed by the professionals in the field of classic telecommunication services, and the underlying design corresponds to that. All signalization, which, by the way, is very similar to ISDN signalization, is performed in binary format only; therefore, the human being cannot read it easily. The analyzer must be used for tuning. Moreover, it is a closed protocol with problematic options of expanding. The reliable TCP protocol is used for transmission of signalization.
Architecture of the system based on H.323 consists of several parts. The most important is the H.323 gatekeeper, which provides registration of the client terminals and communication control. It is not the essential component of the system; nevertheless, if present, all terminals must utilize its services. The basic function of the gatekeeper is conversion of network addresses into understandable format for H.323 and vice versa. It is also capable of managing transfer capacity. That can be used for example when the maximum number of simultaneous connections in LAN are specified within the network management – in such case the gatekeeper can refuse to deploy multiple connections if the value is exceeded. The H.323 gateway is used for interconnecting the H.323 domain with another type of network. It secures conversion of signalization and multimedia information. The Multi Control Unit (MCU) must be incorporated into the system whenever a conferencing discussion among three or more participants must be performed. The biggest advantage is the conversion of the multicast broadcast to the unicast broadcast because mixing of audio and video data is not done at the end-user devices but actually at the MCU. The only truly necessary component in telecommunication networks utilizing H.323 is the terminal, which is usually a telephone set. Each terminal must be capable of providing voice services, the support for video transmission is not obligatory.
H.323 is used for providing services under the "classic" approach, which counts on "dumb" telecommunication terminals and "smart" telephone exchanges. This means that most of the system logic as well as support for services are concentrated in the telephone exchange. This type of approach makes the end-user devices cheaper but it requires higher investment into the telephone exchange. In addition, deployment of new services is somewhat troublesome because in addition to the replacement of terminals, it is often necessary to perform at least an upgrade of the telephone exchange.
With H.323, the Real-Time Transport Protocol (RTP) is used for transmission of multimedia information, which, as opposed to signalization, utilizes the less-reliable transmission via UPD.
The Session Initiation Protocol (SIP) was created in 1998 under patronage of IETF (Internet Engineering Task Force). As opposed to H.323, the implementation of which is more similar to classic telecommunication services, the text-oriented SIP is very similar to most-widely used Internet protocols like HTTP or SMTP. Thanks to that, it is quite easily readable by humans and usually no protocol analyzers are necessary. SIP, as opposed to H.323, is not a self-contained platform; it is fundamentally the signalization protocol only. Same as with H.323, the RTP protocol is used for transmission of multimedia information. Despite the option of utilizing TCP, the UDP is being used in most of the cases, which has a positive effect on network load.
SIP creates a client-server type connection, whereat, same as for example with HTTP, it is a stateless protocol. Therefore, each of the nodes must be equipped with both, the client and the server part. The most important and simultaneously the only obligatory equipment of the SIP network are the client terminals called the User Agents. In this regard, the SIP and H.323 do not differ. The other, this time the optional components, are the SIP Proxy Server, SIP Registrar Server, and SIP Redirect Server. Same as with H.323, these components are optional only in a virtual sense. If you want to have a functional telephone system, you will need them. The SIP Registrar Server is the key element of the functional telephone network; it is used for registration of the User Agents and their subsequent addressing. The SIP Proxy Server is used for routing the conversations based on information for SIP Registrar Server. Thus, all signalization at SIP Proxy Server is translated and SIP is not transparent. The SIP Redirect Server is used for similar purposes as SIP Proxy Server; however, it does not hand over the signalization but only refers to the other node. Its usage is not frequent in common circumstances.
In real-time deployment, differentiation between the SIP Registrar Server and the SIP Proxy Server is more or less rubbed out because most implementations cover these functions with single hardware. This is the reason why literature sources in configuration of various end-user devices commonly interchange or simplify these terms to a sole "SIP Server".
Differences between H.323 and SIP
Already from the aforementioned information, we can get partial idea about differences between the technologies built on protocol sets of H.323 and SIP. The SIP is being currently preferred over H.323. Mainly the fact in favor of SIP is its significantly higher simplicity, extendibility and analyzability as opposed to H.323, provided all possibilities remain intact. These favorable characteristics are accompanied by lesser effort on centralization of services as well. As mentioned earlier, the H.323, same as with all classic telephone services, presupposes all logic to be integrated in the telephone exchange and the end-user devices utilizing the services of telephone exchange only. SIP prefers the opposite approach, which in the end admittedly makes the end-user devices more expensive, but on the other hand, it significantly lowers the cost of acquiring the telephone exchange. In addition, the deployment of new functions can be done through a sole exchange of end-user devices. Moreover, large product portfolio of end-user devices is available by default for SIP in all price categories. The market also offers relatively large number of various telephone exchanges and the open-source solutions can be used for complex installations. Nevertheless, for the home-office use or for the use within small-size organizations, the best choice would be to utilize a ready-to-use solution, which could be for example Signamax 065-9101 SIP Private Branch eXchange, which integrates the function of SIP Private Branch eXchange and the function of SIP Proxy Server into a single compact device.
For the deployment of VoIP technology within your home-office or your organization, you will be most likely interested in differences affecting practical design or run of the telephone system. These systems are shown in the table below:
 | SIP | H.323 |
| Throughput via NAT | + via SIP Proxy Server or via STUN technology | - H.323 Proxy |
| Conference Calls | - limited options; depends on end-user devices only | + Audio & Video Conferencing; option of their management |
| Call Billing | - SIP Proxy Server must be in the call route for the entire duration of the conversation | + H.323 terminals report the start and end time of the call to the gatekeeper |
| Availability and Price | + excellent availability of all components; very low prices | - smaller product portfolio; higher prices |
| Usage | + used by most VoIP operators | - mostly intradepartmental networks within large organizations |
Investment into SIP devices is much more beneficial for smaller and simple installations suitable for home-offices and small and mid-size companies due to lower prices and wider portfolio of the end-user devices. Nevertheless, H.323 is still being used in professional product portfolios of the big players offering complete systems of all telephone network components. H.323 is also used for videoconferencing more often because of current technological insufficiency of SIP. The choice is dependent on the technology your telecommunication operator uses. The conditions in the Czech Republic call for usage of SIP, it is therefore more beneficial for the customers to build their systems on SIP platform.
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