There are a number of protocols employed to facilitate VOIP communication services. Real Time Protocol (RTP) is the most popular standard adopted for transmitting audio and video packets between communicating computers all over the world. However for people communicating over a Voip, it being susceptible to an eavesdropping is always a cause of concern.



Hence to combat such issues and find the remote device which can negotiate means by which media could flow between two devices efficiently, separate Call Signaling Protocols were developed not very long ago. The most popular being H.323 and (SIP) Session Initiation Protocol.

Fundamentally, H.323 and SIP allow users to do the same thing but differ significantly in design and facilities. Where H.323 is senior in years, SIP is relatively newer and evolving. Infact both can do the job, though H.323 is superior in a number of ways like: better interoperability with the PSTN, fully supports video and data conferencing, procedures in lines to provide controls for conference audio and video streaming and lip synchronization. Message encoding is compact and suitable for narrow and as well broadband connections, Flexible addressing mechanisms, including URLs and E.164 numbers. While SIP has limited support for video no protocol or support for controlling the conference and no mechanism for lip synchronization as of now.

However SIP was designed exclusively to support VOIP program and hence it is still in the stage of infancy, evolving and creating its mark with some developed features like caller and callee authentication via HTTP mechanism. SIP also addresses security issues well by defining end-to-end authentication and encryption using PGP or S/MIME.

It shall be sensible to conclude that Call Signal Protocols are still at the stage of facilitating VOIP, to the best of their abilities and hence there is a long way to go before we could decide which of the two is better. As of now H.323 and SIP, they both have their share of space in the market and a long way to go!