Kamailio + RTPEngine + TURN server to enable calling between WebRTC client and legacy SIP clients

[Monxarat]

Hi guys, 

 Can anybody explain to me this flow.

And

What will be done in 1,2,3,4. When a new connection the request flow is 1,2,3,4 order is created.

https://raw.githubusercontent.com/havfo/WEBRTC-to-SIP/master/images/webrtc-sip.png

This page is: How to setup Kamailio + RTPEngine + TURN server to enable calling between WebRTC client and legacy SIP clients. This config is IPv6 enabled by default. This setup will bridge SRTP --> RTP and ICE --> nonICE to make a WebRTC client (sip.js) be able to call legacy SIP clients.

Thanks a lot!

[Donat Zenichev]

I'm not really sure I understand your question. What will be done with what?

The dia you've sent is quite clearly showing:

- webrtc client connected via ws/wss socket to the Kamailio

- pure SIP client connected to the Kamailio via any of the transports: UDP / TCP / TLS

- webrtc client uses encrypted audio stream, rtcp-mux and has ICE candidates

- meanwhile pure SIP client uses plain RTP, has no ICE support, and uses two ports for RTP+RTCP (so no rtcp-mux support)

- kamailio keeps control on rtpengine via socket (indeed via module which has NG protocol to send commands to rtpengine's socket)

What is the IP family, is totally of no matter, it can be v4 or v6 depending on your network requirements.

So you have kind of a gate to interconnect webrtc with pure SIP.

What's the question?

Regards,

Donat Zenichev

понедельник, 4 декабря 2023 г. в 22:02:45 UTC+1, nguyenquo...@gmail.com: