RTP Real Time Transport Protocol SIP

Rohit Bajaj
Rohit Bajaj used Ask the Experts™
on
Hi,
I have communication between client 1 and client 2 over SIP phones established. The signaling part is done and media is flowing between the SIP client 1 and 2. Now suppose the client 1's ip changes or port changes in between media flow during RTP . How will this be handled so as the communication continues properly.

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If the IP address of either endpoint changes during the call then the call is lost because the traffic can no longer find its destination. IP addresses shouldn't change during a call anyway; if they do this indicates a fault somewhere.

The range of ports used by RTP is determined when the session is established and, again, won't change unless there is an error.

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Commented:
I am bit confused here. As to why cant client1 inform client2 about the change in ip address or port ??
Client1 is having the direct ip of client 2
Why would there be a change in IP address of either client during the call?
Beyond the reason why a client needs to change its IP, the SIP protocol is not meant to handle such situations. That does not mean it can't be use to manipulate the media sessions, however a renegotiation needs to occur for the other endpoint to know what the new IP, codec, DTMF method, port and such will be used, that part would signify a brief period of no media.

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