[asterisk-users] force_avp ignored?

[Daniel Pocock]

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?