Hi Serfey Safarov, Thanks for answering.

I have one question. Regarding the right block (1: WebRTC client -> SIP Server) and the left block (SIP client), will a library be used here? Or is this functionality handled in Kamailio?

2023年12月5日(火) 15:44 Sergey Safarov <s.safarov@gmail.com>:

1) INVITE message with ICE and encryption in the SDP;

2) INVITE message and then response without ICE and encryption in the SDP (if client does not support ICE);

3) internal message with SDP content (Kamailio request with received SIP message and RTPengine response with SDP for delivery to next hop);

4) RTP stream without encryption and without ICE/STUN (if client does not support ICE);

5) RTP stream with ICE/STUN and encryption.

On Tue, Dec 5, 2023 at 12:48 AM nguyenquocchinhdev--- via sr-dev <sr-dev@lists.kamailio.org> wrote:

Hi guys,
 Can anybody explain to me this flow.
https://raw.githubusercontent.com/havfo/WEBRTC-to-SIP/master/images/webrtc-sip.png

And
What will be done in 1,2,3,4,5. When a new connection the request flow is 1,2,3,4,5 order is created.

1: webRTC client -> SIp Server
2: SIP server -> SIP client
3: Kmailio -> RTPengine
4: SIP client -> RTPengine
5: RTPengine -> webRTC client
_______________________________________________
Kamailio (SER) - Development Mailing List
To unsubscribe send an email to sr-dev-leave@lists.kamailio.org