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RTP Real Time Transport Protocol SIP

Posted on 2014-11-28
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Last Modified: 2014-12-03
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.

Thanks
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Question by:Rohit Bajaj
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Assisted Solution

by:Perarduaadastra
Perarduaadastra earned 200 total points
ID: 40470144
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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Author Comment

by:Rohit Bajaj
ID: 40470248
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
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Expert Comment

by:Perarduaadastra
ID: 40470314
Why would there be a change in IP address of either client during the call?
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Accepted Solution

by:
José Méndez earned 300 total points
ID: 40470340
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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