[Freeswitch-users] WebRTC sipml5 client hangs up after 120 seconds on bridged calls to sip phone

[Karsten Horsmann]

Hello,

i play around with freeswitch master on centos 6.4 and webrtc to sip-phones.
if i call from the browser (chrome) to a snom, the call ends after 120
seconds.
I see that the sipml5 client is sending then a bye message.

The other direction works fine. From snom to the webrtc phone, no hangup
after 120 seconds.
And from webrtc to the moh extension works fine too.

Any special settings to resolve that?

I made an sofia trace, both phones are on the same sip-profile.
http://pastebin.freeswitch.org/21291


my git version is d1268e81036ce8ce00de8ee22f387cdbf43a7203

--
Kind Regards
Mit freundlichen Grüßen
*Karsten Horsmann*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130810/631f646f/attachment.html