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.
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 mailto:shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com mailto:shaheryarkh@googlemail.com
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
Thanks. I appreciate your help.
Thank you.
On Wed, Dec 11, 2013 at 8:48 AM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
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:
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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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
-
Daniel-Constantin Mierla -
Muhammad Shahzad