<div dir="ltr"><div>Hi Razvan,<br><br></div><div>Thank you.<br><br></div><div>We gone through this tutorial and tried to setup openSIPS to setup for handling JsSIP clients and that is working for us,<br><br></div><div>Here we used the RTPEngine and openSIPS as mentioned.<br><br></div><div>So, the current flow is "JsSIP client A --> openSIPS >--> RTPEngine <--> SIP client" and this is working.<br><br></div><div>But we need to do the following flow.<br><br>JsSIP client A --> "openSIPS + RTPEngine --> conference bridge Platform <--> SIP client"<br><br></div><div>Here, we need RTPEngine only for DTLS and ICE, but the media needs to go through the conference bridge.<br><br></div><div>Is this possible? Whether RTPEngine will work in such a way?<br><br></div><div><br></div>Thanks.<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Jan 5, 2016 at 4:13 PM, Răzvan Crainea <span dir="ltr"><<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<tt>Hi, Suganthi!<br>
<br>
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.<br>
<br>
[1] <a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1" target="_blank">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1</a><br>
<br>
Best regards,<br>
</tt><span class="">
<pre cols="72">Răzvan Crainea
OpenSIPS Solutions
<a href="http://www.opensips-solutions.com" target="_blank">www.opensips-solutions.com</a></pre>
</span><div><div class="h5"><div>On 01/05/2016 11:13 AM, suganthi
karthick wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>
<div>Thank you so much.<br>
<br>
</div>
We have a conference bridge platform, and we need to
integrate openSIPS with the platform.<br>
</div>
We have certain init functions, config functions and some
media related functions that needs to be handled in openSIPS.<br>
</div>
<div>Also the conference platform will handle the media, so
media needs to be send to the Motion Platform. <br>
<br>
</div>
<div>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.<br>
<br>
</div>
Thank you.<br>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, Jan 5, 2016 at 2:10 PM, Răzvan
Crainea <span dir="ltr"><<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> <tt>Hello, Suganthi!<br>
<br>
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.<br>
<br>
Best regards,<br>
</tt>
<pre cols="72">Răzvan Crainea
OpenSIPS Solutions
<a href="http://www.opensips-solutions.com" target="_blank">www.opensips-solutions.com</a></pre>
<div>
<div>
<div>On 01/05/2016 09:12 AM, suganthi karthick wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div>
<div dir="ltr">
<div>
<div>Thanks for the reply.<br>
<br>
</div>
Whether OverSIPS has support for
ICE,STUN,DTLS-SRTP?<br>
<br>
</div>
<div>Since the existing conference bridge platform
is in C implementation, we thought of using
openSIPS<br>
<br>
</div>
Thanks.<br>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, Jan 5, 2016 at
12:12 PM, suganthi karthick <span dir="ltr"><<a href="mailto:suganthi.mkk@gmail.com" target="_blank"></a><a href="mailto:suganthi.mkk@gmail.com" target="_blank">suganthi.mkk@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div>Hi all,<br>
<br>
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.<br>
<br>
The conference bridge is an existing
working one for SIP clients, and I am
trying to add webrtc support for that.<br>
<br>
The webrtc gateway needs to be implemented
in a way like a library because it needs
to be integrated into the existing
platform.<br>
<br>
There are some init functions and config
function from the existing conference
platform, based on which the webrtc
gateway has to be configured. <br>
<br>
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.<br>
<br>
Do you have some suggestion on whether
openSIPS can be used for this purpose?<br>
<br>
</div>
<div>Your suggestions will be helpful.<br>
<br>
Thanks.<br>
<br>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
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