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

[Răzvan Crainea]

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 Solutions
www.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 <razvan at opensips.org 
> <mailto:razvan at opensips.org>> 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 Solutions
>     www.opensips-solutions.com <http://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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