I'm using v11.11 I tried setting: force_avp=yes in a SIP peer in sip.conf and it seems to be ignored. The WebRTC client sends an INVITE with "RTP/SAVPF" and Asterisk is still sending back 183 and 200 responses with the UDP/TLS/RTP/SAVPF string Are there some limitations with this option or does it depend on any other settings? Is there any debugging I can enable to understand what is going wrong?
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