Is there a way to get “direct media” between 2 WebRTC endpoints ? WebRTC should normally provide end-to-end RTP to both endpoints, but this is not the case with Asterisk as it stays in the middle for the RTP streams.
The reason for my question is that end-to-end RTP is really a must for hosted environments. For an on-premise service this is less important, but for a hosted service it is mandatory to keep RTP local for local extension to extension calls. So like for SIP clients also WebRTC clients and mix of both types need direct media for a hosted PBX service.
I’m aware that some Asterisk built-in features would be lost with direct media, as is the same for direct media using chan-sip or PJSIP, but that’s not an issue compared to bandwidth saving and less latency offered via direct media. Also codec limitations are not a problem : G.711 is fine for local extension to extension traffic.
If there is a way to solve this, then I have a second question : how to get direct media between a PJSIP and a WebRTC client ? SRTP is normally a prerequisite for WebRTC and I noticed that SRTP is also restricting direct media with SIP at this moment. As a workaround SRTP could maybe avoided until a solution (e.g. ZRTP ?) is available ?