[SR-Users] [BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)

[Andrey Utkin]

I hope the subject line says it all.
I need to set up a backend to establish calls between browser clients
and usual SIP clients, and the abovementioned applications are of the
highest interest. (Of course, calls between same UAs must work, too.)
If Linphone is not going to work due to its internal issues, then
other established open source mobile apps for iOS and Android will be
fine.

Currently I have configured latest kamailio+rtpengine with configs
from here https://github.com/caruizdiaz/kamailio-ws

The calls pass this way currently:
jssip -> android: no sound from phone to the browser, i see that jssip
sends "sendonly" attribute for audio in INVITE's SDP. Audio from
browser to phone, and both video streams appear immediately,
everything is fine with them.

android -> jssip: video from browser to the phone appears in 1-2
_minutes_ after the call is answered. All other media streams are
fine.

The above results are the same in such browsers
[IP-] [  ] www-client/firefox-bin-31.3.0:0
[IP-] [ ~] www-client/google-chrome-unstable-41.0.2243.0_p1:0

With sipml currently i have no stable results, so it's hard to
describe what happens.


Please contact me if you have configs to make the needed things work,
or if you have experience of such things working stable, and can
configure it quickly.

-- 
Andrey Utkin