VoIP Protocols

The protocols allow many types of features and determine the way in which the VoIP components interact with one another.
Two standards regulate multimedia delivery: ITU ( International Telecommunications Union) and IETF (Internet Engineering Task Force).
Some vendors are using proprietary schemes. Also, not all components of the protocols are used.
Each of the voice protocols has its own advantages and disadvantages and each has a different approach to VoIP.
The most used are:


H323


It is the recommended by ITU.
Its standards were the first used for multimedia delivery over LAN technologies.
H323 Network is based on Media Gateways and Gatekeepers.
They provide call routing an communication with end user devices.

Gateways interface with other Networks like the PSTN.
Gatekeepers function as central servers for call control, admission, bandwidth management and call signaling.
The version two of H323 presents problems for heavier specifications, shortcoming of standards scalability expecially in large-scale deployment.
The following versions solve these problems.

The best use of H323 protocol is in enterprise VoIP applications.




RTP (Real Time Transport Protocol)



The RTP protocol provides features for real-time applications.
Two IP addresses establish the session between two end points.
It is an application built on UDP.
RTCP is not a must for RTP to work, but it helps to provide feedback on the quality of data delivery done by RTP.
Although the feedback does not say where the problems of congestion are, it can be used to locate problems.
It can report where network performance is failing.
It can detect packet loss, delay in transmission, jitter and other problems that can occur.
This information is available on both ends and it is provided by the Media Gateways.

Using RTCP must be used into a small fraction of the section bandwidth.
This should be done not to limit the ability of the transport protocol.



MGCP ( Media Gateway Control Protocol)



It substitutes the traditional voice switches with the Media Gateways.
It is a master-slave control protocol.
The Media Gateway Controller is the “Call Agent” which manages the signaling controls, while the Media Gateway informs the Call Agent of service events.
The Call Agent instructs the Media Gateway to create and stop connections, very often to start an RTP session between two end user devices.



SIP (Session Initiation Protocol)


It is a powerful client-server protocol.
It initiates and stops the calls between speakers which can include multimedia conferences and termination calls.
It is designed to be simple and efficient.
It copies some architectural features of HTTP and SMTP.
It supports user mobility by redirecting requests to the user’s actual location with the aid of a Proxy Servver or a SIP Server.




MEGACO/H.248



It is similar to MGCP.
The Media Gateways provide conversion and sources of calls, while the Media Gateway Controllers control the call.
Its primary focus is the standardization of IP telephony hardware.
It is scalable in implementing a simple minimal design.
The hardware cost is related to the capabilities provided.