VoIP Protocols & Standards

- a summary or tutorial about the protocols used with Voice over IP, VoIP networks and systems.

Voice over IP, VoIP Includes:
VoIP basics VoIP phones How to choose best VoIP service provider

Voice over Internet Protocol, VoIP, has seen an enormous level of growth in recent years and it is likely that this with continue, or even increase in the foreseeable future. The reason for this level of growth in the use of VoIP results from the cost savings it provides, the increase in flexibility, and the fact that the same network can be used for voice and data.

Although the first VoIP networks started to appear as early as 1995, and these VoIP networks used proprietary protocols to enable the IP data to be exchanged. However with the rapid growth, and the need for VoIP traffic to be routed globally, the need for established standards to control the exchange of data arose. In view of the fact that there were a number of different requirements, several standards were developed and are in use today.

Although it may be surprising at first sight that there are several VoIP protocols, these different protocols are required for different reasons and they have been deployed in many systems. Even though the concept of one unified VoIP protocol or set of protocols would seem ideal, this is unlikely to happen in view of the extent to which the different protocols have already been deployed. Additionally different companies and organizations will use the VoIP protocol which best meets their requirements and accordingly, they will want to maintain what they are currently using. This will mean that there will be resistance to any change, especially as interfaces have been devised to enable interoperability.

VoIP protocols overview

Although working together, there are a number of different organizations and bodies that are mentioned when referring to VoIP protocols:

IETF This is the Internet Engineering Task Force. It is a community of engineers that defines some of the prominent standards used on the Internet (including VoIP protocols) and seeks to spread understanding of how they work.
ITU the International Telecommunication Union. This is an international organization within the United Nations System used by where governments and private sector companies to coordinate and standardize telecommunications networks, services and standards on a global basis.

In addition to the organizations involved, there is also a variety of different VoIP protocols and standards.

G711: G711 is possibly the most widely used VoIP vocoder and it is the standard for transmitting uncompressed packets. G.729 is the standard for compressed packets. Many equipment vendors also use their own proprietary codecs. Voice quality may suffer when compression is used, but compression reduces bandwidth requirements. There are many other vocoders/ codecs that are used with varying data rates and providing different levels of voice quality.
H.248: H.248 is an ITU Recommendation that defines "Gateway Control Protocol" and it is also referred to as IETF RFC 2885 (Megaco). It defines a centralized architecture for creating multimedia applications and it extends MGCP. H.248 is the result of a joint collaboration between the ITU and the IETF and it is another VoIP protocol.
H323: The signalling protocol is used to control and manage the call. It includes elements such as call set up, clear down, call forwarding and the like. The first protocol to be widely used for VoIP was H323. However this is not a particularly rigorous definition and as a result other variants have been developed.
Megaco This is also known as IETF RFC 2885 and ITU Recommendation H.248. H.248 defines a centralized architecture for creating multimedia applications.
Media Gateway Control Protocol (MGCP) This is also known as IETF RFC 2705. It defines a centralized architecture for creating multimedia applications, and it is therefore a VoIP protocol.
Real-Time Transport Protocol (RTP) This VoIP protocol is defined under IETF RFC 1889 and it details a transport protocol for real-time applications. RTP provides the transport mechanism to carry the audio/media portion of VoIP communication and is used for all VoIP communications.
RTP : RTP, Real Time Protocol, is a data exchange protocol and this can handle both audio and video. RTP handles the data exchange, but in addition to this a codec is required. Where voice is used a vocoder is used (a codec can be used for any form of data including audio, video, etc).
Session Initiation Protocol (SIP) This is also known as IETF RFC 2543 and it defines a distributed architecture for creating multimedia applications. SIP, Session Initiation Protocol, is now being widely adopted as the main standard is a far more rigorous protocol for signalling and is the one that is most widely used now.
Skinny: One other signalling protocol that was used was known as "Skinny" and is a Cisco Proprietary protocol and is from Nortel and anotheris called Unistem. In view of this there are often interfacing problems.

Centralised and distributed architectures

One of the advantages of VoIP is that it does not legislate for the architecture of the network that carries the data. Early telecommunications networks used a centralised structure where all the intelligence was contained at the switching station or exchange. With the advent of packet technology, the routing and intelligence can be distributed to where it is most convenient to locate it. This may be by having a distributed architecture, or a centralised one.While both architectures can be employed with VoIP, the type of architecture does have an impact on the optimum VoIP protocols to use. This is one of the reasons why a number of VoIP protocols are used, and will remain to be used.