[SR-Users] SIP - WebRTC gateway

Zappasodi Daniele D.Zappasodi at selta.it
Mon Mar 31 15:27:34 CEST 2014


Hello, 
I'm figuring out the best approach to deploy a bridge between Websocket\Webrtc and SIP\rtp.
Can Kamailio (+mediaproxy-ng or something else) operate as a full Webrtc\SIP gateway (signaling, audio or video transcoding, ICE and so on)? 

Some months ago I found the architecture described here http://www.kamailio.org/wiki/devel/rtcweb_breaker that proposes to introduce a new RTCWeb 
Breaker.
Is it just a proposal or is Kamailio moving following this approach? 
If Kamailio really requires a RTCWeb Breaker, what are the main issues against using Doubango webrtc2sip with Kamailio? Performance? Interoperability? License? ...

Thanks 
Daniele

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