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

[suganthi karthick]

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> 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 <
> <suganthi.mkk at gmail.com>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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