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

[Karsten Horsmann]

Hello List,

answering myself to that problem.

With deactivated session-timers *) it seems to work. Iam not sure if i
broke then something other.
Maybe the snom firmware is buggy. I will try an upgrade to latest and try
another hardphone too.

*) in the sip-profile:
<param name="enable-timer" value="false"/>



2013/8/10 Karsten Horsmann <khorsmann at gmail.com>

> 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*
>



-- 
Mit freundlichen Grüßen
*Karsten Horsmann*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130811/3a72cff8/attachment.html