18 Dec
2014
18 Dec
'14
7:30 p.m.
On 12/18/14 12:11, Andrey Utkin wrote:
Hi!
I need to establish calls between WebRTC and usual SIP clients
(exactly, sipml/jssip and linphone-android).
I used configs from https://github.com/caruizdiaz/kamailio-ws and
latest git master HEAD of both kamailio and
rtpengine. I got calls from webrtc to android working correctly (but only with
Firefox browser), even with video. But in other directions i have some
issues because of lack of RTP delivery or RTP timeouts.
I have some logs to show you regarding this:
https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works),
https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't
work).
I would really love to get some quick help from anyone. For direct
manual fixing, I can give a couple of hundreds of bucks.
Looking forward impatiently for reply from anyone having something to say.
Write error on RTP socket usually indicates an incorrect network setup,
for example trying to use a source address for IP packets which isn't
bound to any local network interface (especially if you're sitting
behind NAT), or local iptables rules rejecting outgoing IP packets, or
missing IP routes to the destination. Perhaps post your network setup
and the CLI arguments to rtpengine you're using.
cheers