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 <[email protected]
<mailto:[email protected]>> 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
<[email protected] <mailto:[email protected]>> 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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