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<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 class="moz-signature" cols="72">Răzvan Crainea
OpenSIPS Solutions
<a class="moz-txt-link-abbreviated" href="http://www.opensips-solutions.com">www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 01/05/2016 09:12 AM, suganthi
karthick wrote:<br>
</div>
<blockquote
cite="mid:CAK8SOLjnGwTzCG67=sHxoPSYaDJ2v=gLQQyOhH-GaGnHyODNCQ@mail.gmail.com"
type="cite">
<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
moz-do-not-send="true"
href="mailto:suganthi.mkk@gmail.com" target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:suganthi.mkk@gmail.com">suganthi.mkk@gmail.com</a></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>
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