Need help on getting rtp stream on sip server

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Hi Benny, et al,
With your help (thanks a lot!) and from the sample stateful_proxy.c, I was able to create sip server which does registrar and proxy functions. This setup allows two soft-phones to registrar themselves and make calls between them. However the rtp stream for the call is established directly between the User Agents and not via the sip server. Hence  I would  like to know what changes should I have to perform to make the rtp go through the sip server. Appreciate your help.

Thanks,
Senthil.





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