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

[Răzvan Crainea]

Hello!

Yes, it is definitely possible, its just a matter of configuration. If 
you find any problems during development, the entire OpenSIPS comunity 
is here to help :).

Good luck!

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 01/05/2016 12:54 PM, suganthi karthick wrote:
> 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 <razvan at opensips.org 
> <mailto:razvan at opensips.org>> 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 Solutions
>     www.opensips-solutions.com <http://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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>>
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