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

[Alexander Haugg]

Hi,

in the dialplan I am using "<action application="bridge" data="{media_webrtc=true}user/123 at 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.
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