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

[Tito Cumpen]

Suganthi,

Rtpengine will translate the SAVPF media channels into standard avpf or
ago. By SAVPF mean rtp with dtls and it will work as a media relay if ice
is supported or not on either side. As far as webrtc SAVPF(DTLS) ICE are
needed. I have a deployment that translates SAVPF into avpf and it works
with no issues.
On Jan 5, 2016 5:54 AM, "suganthi karthick" <suganthi.mkk at gmail.com> 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>
> 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 <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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