[BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)
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 writes:
jssip -> android: no sound from phone to the browser, i see that jssip sends "sendonly" attribute for audio in INVITE's SDP.
same here. perhaps it is better to wait a year or two before the moving targets (both specs and implementations) stabilize.
while waiting, it may also turn out that webrtc itself becomes irrelevant, since a new sdpless rtc api is in the works.
-- juha
2014-12-22 8:43 GMT+02:00 Juha Heinanen jh@tutpro.com:
Andrey Utkin writes:
jssip -> android: no sound from phone to the browser, i see that jssip sends "sendonly" attribute for audio in INVITE's SDP.
same here. perhaps it is better to wait a year or two before the moving targets (both specs and implementations) stabilize.
while waiting, it may also turn out that webrtc itself becomes irrelevant, since a new sdpless rtc api is in the works.
Juha, I've reported this issue and it is fixed now: https://github.com/versatica/JsSIP/issues/283
Update: my installation works fine with latest JsSIP & Linphone Android, except for this still happening in ~30% of calls:
android -> jssip: video from browser to the phone appears in 1-2 _minutes_ after the call is answered. All other media streams are fine.
I'm using www-client/google-chrome-unstable-41.0.2251.0_p1 and http://tryit.jssip.net .
Andrey Utkin writes:
Juha, I've reported this issue and it is fixed now: https://github.com/versatica/JsSIP/issues/283
yes, indeed. i gave it a try and sendonly was not there anymore.
-- juha
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Andrey Utkin -
Juha Heinanen