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 &lt;s.safarov(a)gmail.com&gt;om>: > 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(a)lists.kamailio.org&gt; wrote: > >> Hi guys, >> Can anybody explain to me this flow. >> >> https://raw.githubusercontent.com/havfo/WEBRTC-to-SIP/master/images/webrtc-… >> >> 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(a)lists.kamailio.org >> >