5 Dec
2023
5 Dec
'23
9:02 a.m.
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(a)gmail.com>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> 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
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