Also you can use is WebRTC2SIP gateway which can connect to any traditional SIP server, like freeswitch or asterisk. http://code.google.com/p/webrtc2sip/ Thank you. On Tue, Dec 4, 2012 at 11:43 AM, qasimakhan@gmail.com <qasimakhan@gmail.com>wrote:
I have successfuly made SIP-SIP (Both Audio & Video) calls using OverSIPs as WebRTC-SIP gateway and using Opensips 1.7 with VIA patch for WebRTC.
Regards, Qasim
On Sat, Dec 1, 2012 at 2:34 AM, Ros P <rosphotos82@gmail.com> wrote:
Hello,
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 (issue 39<http://code.google.com/p/sipml5/issues/detail?id=39>), does anyone have any other suggestions? Anything successfully tested?
Kind Regards, ros
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