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<tt>Hello!<br>
<br>
Yes, it is definitely possible, its just a matter of
configuration. If you find any problems during development, the
entire OpenSIPS comunity is here to help :).<br>
<br>
Good luck!<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 12:54 PM, suganthi
karthick wrote:<br>
</div>
<blockquote
cite="mid:CAK8SOLj406CE0_C-TnofcvxbYDXN3v5=w=amdBsh79Sg_daFBw@mail.gmail.com"
type="cite">
<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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true" 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
moz-do-not-send="true"
href="mailto:razvan@opensips.org"
target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:razvan@opensips.org">razvan@opensips.org</a></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 moz-do-not-send="true" 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
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>
</blockquote>
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