On Thu, Nov 29, 2012 at 2:42 AM, Jeremy Jongepier jeremy@autostatic.com wrote:
On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
Dear Peter, Thansk for your fast response. I highly appreciate it. Is there any way that I can convert the RTP/SAVPF into general media profile that PSTN GW support ? So that I can get that call working/
Best Regards, Roy.
I think at the moment only a patched Asterisk might be able to do this: http://code.google.com/p/sipml5/wiki/Asterisk But I haven't got this to work yet because I get the feeling the patch Doubango provides is incomplete. Or maybe webrtc2sip by the same makers: http://code.google.com/p/webrtc2sip/ I've tried setting up a webrtc2sip server today but it crashes after a few minutes, is horrible to set up (the web GUI is lightyears behind Siremis) and for the moment I can't get it to work because I can't find any proper documentation on how to set it up.
I think it will be better for you to report the problem to sipml5/doubango mailing list. Many users reported that it works (including me). Mamadou is a kind and helpfull guy.
If anyone could build webrtc support into rtpproxy or any other media proxy that can work together with Kamailio I'd be more than happy to test it.
Regards,
Jeremy
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users