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RTP and NAT - How They Work Together With VoIP

April 18, 2012 | Comments: 0 | Views: 206

Despite the problems of configuration of SIP clients with various SIP providers, it's actually a miracle that they're able to do it so easily. If you were to delve into the guts of a single SIP call, you will be confounded by the amount of complexity that goes on behind each. Internet applications such as VoIP have several layers to them. It is indeed possible that no single person understands the complete entirety of every layer. The SIP protocol has come to represent the industry standard in VoIP when it comes to setting up and tearing down VoIP calls. When two devices communicate, they use the SIP to invite each other to a call and to close it as well.

But the actual audio which follows utilizes a different protocol altogether. The RTP protocol is what is generally used for the streaming of media which is required for two-way voice communication. For a single call, two RTP channels have to be set up and they take over where SIP leaves off. Unfortunately, many SIP clients sit hidden behind NAT firewalls and this poses a problem for the RTP protocol to take over after the SIP protocol has performed its duties. One might say that this is one of the biggest hindrances to the general implementation of VoIP at large quite apart from the politics involved.

RTP stands for Real-Time Transport Protocol and each connection has a control channel as well - RTCP. Thus, two ports are opened for each RTP connection. RTCP stands for Real-Time Transport Control Protocol and it works in tandem with RTP helping it obtain feedback about the specifics of the single RTP connection such as audio quality, jitter etc. This quality of service information allows a client to improve its own stream if possible. The actual RTP connection utilizes an even-numbered port with RTCP taking up the next higher odd-numbered port.

We have seen that the type of codec used can significantly alter the call quality as well as the delay experienced by the two people at each end. If the feedback from RTCP is not positive, it's possible for the two clients to switch over to another codec hoping to improve both the call quality and the round-trip time.

There are actually quite a number of difficulties for SIP clients sitting behind NAT firewalls to properly initiate an RTP connection after a successful SIP one. There are several solutions to this problem which we will take a look at in further articles.

Bhagwad is an expert consultant on SIP account creation. He also specializes in hosted VoIP servers. Click here to find out more.

Source: EzineArticles
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