Hello,

there were some config snippets on mailing lists (maybe sr-dev) about this topic.

Peter just published his slides at last event he presented on this topic:
- http://www.slideshare.net/crocodilertc/webrtc-websockets

Last part has also snippets of Kamailio config.

Cheers,
Daniel

On 09/12/13 14:04, Muhammad Shahzad wrote:

Hi,

According to documentation, using kamailio's rtpproxy-ng module with mediaproxy-ng service, it is possible to make webrtc to sip calls and vice versa,

However i am stuck since morning to make JSSIP (in chrome) to phonerlite (in Windows 8) calls. There is not working example or sample code anywhere either. So i was wondering if anyone has actually tries that successfully and would care to share some samples for us.

So, far i tried "+SP" flags for phonerlite to JSSIP calls and "-sp" for JSSIP to phonerlite calls in "rtpproxy_manage" method. Apparently both calls connects but then drop after a few seconds of ACK. Which indicate the problem is likely to be on mediaproxy-ng end rather then kamailio..

Thank you.

--

Mit freundlichen Grüßen

Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk@hotmail.com
Email: shaheryarkh@googlemail.com

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