[OpenSIPS-Users] Can OpenSIPS can be used as a WebRTC gateway for JsSIP client and WebRTC client?

Răzvan Crainea razvan at opensips.org
Tue Jan 5 09:40:26 CET 2016


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 Solutions
www.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 
> <suganthi.mkk at gmail.com <mailto:suganthi.mkk at gmail.com>> 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.
>
>
>
>
>
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