[SR-Users] Kamailio + mediaproxy-ng for webrtc to sip calls
Daniel-Constantin Mierla
miconda at gmail.com
Wed Dec 11 08:48:25 CET 2013
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 at hotmail.com <mailto:shari_786pk at hotmail.com>
> Email: shaheryarkh at googlemail.com <mailto:shaheryarkh at googlemail.com>
>
>
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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