Hi Razvan, Thank you.
We gone through this tutorial and tried to setup openSIPS to setup for handling JsSIP clients and that is working for us, Here we used the RTPEngine and openSIPS as mentioned. So, the current flow is "JsSIP client A --> openSIPS >--> RTPEngine <--> SIP client" and this is working. But we need to do the following flow. JsSIP client A --> "openSIPS + RTPEngine --> conference bridge Platform <--> SIP client" Here, we need RTPEngine only for DTLS and ICE, but the media needs to go through the conference bridge. Is this possible? Whether RTPEngine will work in such a way? Thanks. On Tue, Jan 5, 2016 at 4:13 PM, Răzvan Crainea <[email protected]> wrote: > Hi, Suganthi! > > You can find here[1] a tutorial about how you can configure OpenSIPS 2.1 > to stay between your WebRTC customers and your SIP gateways. > > [1] http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 01/05/2016 11:13 AM, suganthi karthick wrote: > > Thank you so much. > > We have a conference bridge platform, and we need to integrate openSIPS > with the platform. > We have certain init functions, config functions and some media related > functions that needs to be handled in openSIPS. > Also the conference platform will handle the media, so media needs to be > send to the Motion Platform. > > How this can be handled with openSIPS? It will be helpful if you give some > overview on how to start work on top of openSIPS for this purpose. Since we > are new to the development, your suggestions would be great for us. > > Thank you. > > On Tue, Jan 5, 2016 at 2:10 PM, Răzvan Crainea <[email protected]> > wrote: > >> Hello, Suganthi! >> >> You can use OpenSIPS 2.1 (for WebSockets signalling) and RTPengine (for >> media, DTLS, ICE, etc. handling). OpenSIPS 2.2 also comes with an alpha >> version of Secure WebSockets. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 01/05/2016 09:12 AM, suganthi karthick wrote: >> >> Thanks for the reply. >> >> Whether OverSIPS has support for ICE,STUN,DTLS-SRTP? >> >> Since the existing conference bridge platform is in C implementation, we >> thought of using openSIPS >> >> Thanks. >> >> On Tue, Jan 5, 2016 at 12:12 PM, suganthi karthick < >> <[email protected]>[email protected]> wrote: >> >>> Hi all, >>> >>> I need to implement a WebRTC gateway for an existing conference bridge. >>> The WebRTC gateway has to support Signaling, ICE, DTLS-SRTP. The webrtc >>> clients can be JsSIP or any JSON based webrtc client. >>> >>> The conference bridge is an existing working one for SIP clients, and I >>> am trying to add webrtc support for that. >>> >>> The webrtc gateway needs to be implemented in a way like a library >>> because it needs to be integrated into the existing platform. >>> >>> There are some init functions and config function from the existing >>> conference platform, based on which the webrtc gateway has to be >>> configured. >>> >>> Also, when a webrtc call come from a webrtc client, it needs to handle >>> the signaling and the media(RTP) has to go to the conference bridge >>> platform. >>> >>> Do you have some suggestion on whether openSIPS can be used for this >>> purpose? >>> >>> Your suggestions will be helpful. >>> >>> Thanks. >>> >>> >>> >> >> >> _______________________________________________ >> Users mailing >> [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing > [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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