Also you can use is WebRTC2SIP gateway which can connect to any traditional SIP server, like freeswitch or asterisk.<div><br></div><div><a href="http://code.google.com/p/webrtc2sip/">http://code.google.com/p/webrtc2sip/</a><br>
<div><br></div><div>Thank you.</div><div><br><br><div class="gmail_quote">On Tue, Dec 4, 2012 at 11:43 AM, <a href="mailto:qasimakhan@gmail.com">qasimakhan@gmail.com</a> <span dir="ltr">&lt;<a href="mailto:qasimakhan@gmail.com" target="_blank">qasimakhan@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I have successfuly made SIP-SIP (Both Audio &amp; Video) calls using OverSIPs as WebRTC-SIP gateway and using Opensips 1.7 with VIA patch for WebRTC. <br>
<br>Regards,<br>Qasim<br><br><div class="gmail_quote">On Sat, Dec 1, 2012 at 2:34 AM, Ros P <span dir="ltr">&lt;<a href="mailto:rosphotos82@gmail.com" target="_blank">rosphotos82@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<div><br></div><div>Has anyone had any luck in implementing a webrtc call? I have been able to register using the webrtc2sip gw and sipml5 client but no audio goes through. As this seems to be a client problem (<a href="http://code.google.com/p/sipml5/issues/detail?id=39" target="_blank">issue 39</a>), does anyone have any other suggestions? Anything successfully tested?</div>


<div><br></div><div>Kind Regards,</div><div>ros</div>
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