Hello all, can anyone please point me to the right direction on making the audio from one side of the SIP plugin forward the RTP? I can successfully use rtp_forward on the videoroom and audiobridge plugins, but I don’t find in the documentation nothing doing something similar for the SIP plugin, other than making a SIP call.
I will appreciate your assistance.
Okay, found it. Found my question already asked on this site and a blog article describing a solution is here:
(Of course, I had searched exhaustibly for the answer before writing the question, the search bar could have done a better job, but as it happens many times things start showing up only after you ask the question)
Thanks
The SIP plugin doesn’t support the RTP forward request yet. The RTP support you see in the SIP plugin is the output of the negotiation that happens with SIP peers.
Good to know. Thanks Lorenzo
CS
Hi Lorenzo, the first time I read the blog I understood it better 
…but now that I am re-reading it I will appreciate you clarification. For a Proof of Concept I see it is you are making a SIP call from point A to point B with Drachtio in the middle. But I don’t see how I to trigger a second INVITE… I mean when going from point A to B the first INVITE has to be received by point B and then point B replies, and then a bunch of other messages are exchanged between A and B, don’t know with Drachtio in the middle if the messages are just passed thru.
I am trying to make a phone call, I want Sofia-SIP to call a cell phone (point B) with Asterisk making the outside call possible.
After the call connects I want to start capturing the RTP. Once the connection is established and both parties are talking, how am I going to make point A trigger a new INVITE this time so that Drachtio reads it and creates the SIP?
Thanks very much for your attention and congrats on the awesome product
Then you don’t need my Drachtio experiment or the AudioBridge. The SIP plugin only supports sending RTP to a single peer (the calleer/callee), and we don’t have any RTP forwarding to external components in that plugin. Until that’s implemented (no plan for that yet), you may have better luck instructing Asterisk to duplicate the call so that you get the RTP for monitoring purposes.