Hi,

in the dialplan I am using „<action application="bridge" data="{media_webrtc=true}user/123@my.sip.domain" />“

The media description header have ever the RTP/SAVPF as set.

The most Clients wich support webrtc and ICE have the possibiliti to work without SRTP (that’s nice vor debugging problems).

Is it possible to set RTP/AVP for webrtc calls?

Thanks a lot.