[Freeswitch-users] WebRTC using RTP/AVP instead of RTP/SAVPF

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.