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I'm trying to setup a call between webRTC based client (olympus) and a standard one (x-lite i.e.). The call is failing (480). I believe it is because of SDP negotiation failed. Currently I use standard telestax mediaserver setup.

Can restcomm be configured in a way, that it transcodes the stream (and modifies codec negotiation), so webRTC based clients can call the standard sip ones ?

Thank you very much in advance.

Hubert

Hubert Zegota
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    Can you provide more information, like RestComm logs messages or configuration details? – hrosa Aug 20 '15 at 12:03
  • I made a workarund for that. I created a simple application in RVD, that simple dial the sip URI of another SIP proxy. In such way restcomm seems to transcode the connection, and it works properly. – Hubert Zegota Sep 08 '15 at 09:28
  • Hi yes flow work from webRTC client -> RVD -> Sip Client. but how can we do it from a Sip Client -> RVD App -> webRTC – user987760 Jul 02 '18 at 09:09

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