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.
webrtc spec requires savpf, so no you can’t do webrtc without it. You can do ice without savpf but most of the browsers have removed this capability.
On Jan 16, 2018, at 5:15 AM, Alexander Haugg <Alexander.Haugg@c4b.de> wrote:
Hi,
in the dialplan I am using „<action application="bridge" data="{media_webrtc=true}user/123@my.sip.domain <mailto: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?
participants (2)
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Alexander Haugg
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Michael Jerris