Kamailio + RTPEngine + TURN server to enable calling between WebRTC client and legacy SIP clients
Hi guys, Can anybody explain to me this flow. https://raw.githubusercontent.com/havfo/WEBRTC-to-SIP/master/images/webrtc-s...
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
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-s...
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
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:
- INVITE message with ICE and encryption in the SDP;
- 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-s...
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
It is a complex question. For WebRTC client requires TLS encrypted connection and something like openssl library a required. For TCP/UDP sip client I think an external library is not required.
Sergey
On Tue, Dec 5, 2023 at 9:02 AM Monxarat nguyenquocchinhdev@gmail.com wrote:
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:
- INVITE message with ICE and encryption in the SDP;
- 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-s...
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
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Monxarat -
nguyenquocchinhdev＠gmail.com
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Sergey Safarov