Hi,
I have an Asterisk 17.0 WebRTC-SIP gateway using PJSIP and I want to negotiate rtcp-fb packets on both ends. The SIPml WebRTC client negotiates rtcp-fb for the H264 codec, but Asterisk does not include rtcp-fb in the SDP Offer that it generates to the SIP endpoint. Does PJSIP/Asterisk support relaying rtcp-fb offers/answers? I would like to make this work so proper RTCP video packets can be received by both endpoints in case key-frames are needed.
Thank You
There is no relay. Each SDP offer/answer is independent. The rtcp-fb attributes are only present for WebRTC enabled endpoints.
Thank you for the information. Do you have any recommendations on how Asterisk/PJSIP can deal with video key-frame needs? If the SIP endpoint can send an RTCP PLI packet on the rtcp port, is there any way for asterisk to let the WebRTC endpoint know that a key-frame has been requested via RTCP? WebRTC will generate a key-frame when it receives a PLI/FIR packet. Thanks.
Theoretically it would pass through the core and get sent out as an RTCP FIR.