[asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup

[Leonja Cerebro]

Hello,
worst aspect is that - if SIP clients do not have such a timeout, and in
that case if killing an asterisk and to start it up again -
so it is nothing to do with this asterisk timeout.

Regards,

On 23 February 2010 08:44, Olle E. Johansson <oej at edvina.net> wrote:

>
> 23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson:
>
> > On 100222 1313, JT wrote:
> >> When a SIP device dials another SIP device...Asterisk connects the calls
> and
> >> displays the channel information.
> >> If one of those SIP devices hangs up, Asterisk receives the hangup
> notice
> >> and disconnects the call/channel.
> >> However - what does Asterisk do when the network cable is unplugged from
> one
> >> of the SIP devices...?!
> >
> > Jared already mentioned SIP session timers, which are supported starting
> with 1.6. Here's my experience. While I am running 1.6, the software stack
> that is used for agent softphone (PJSIP) does not support the session
> timers. If the softphone crashes in a call, the call would get stuck exactly
> as you describe.
> >
> > I am working around this problem by setting rtp timeouts in sip.conf:
> >
> > [general]
> > rtptimeout=10
> > rtpholdtimeout=300
> >
> > This means that if RTP flow stops while the agent is in the call, the
> call will be disconnected in 10 seconds. If the call was put on hold by the
> agent, it will be disconnected in 300 seconds. Your timeouts may vary.
> >
> > The caveat here is that it is perfectly normal NOT to transmit any RTP
> data in case of long silence.
> Not in Asterisk - we do not really support silence suppression. The
> recommendation is to turn it off on the phones.
>
> > This is why the SIP timers were introduced in the first place: there is
> no correct way to detect when the client is going away, as no activity is a
> good session state.
> >
> > I am able to get away with the small timeout because I set the PJSIP
> client to always transmit RTP, by turning off voice activity detection
> feature (VAD). If you want to support that feature, set rtptimeout as high
> as for how long you allow absolute silence on the line without disconnecting
> it.
>
> Just to complete this discussion - we also have the absolute timeout that
> is a lifesaver in many cases. If you set this to a time that's larger than
> the normal calls, Asterisk will hang up the call. I very often set it to two
> hours, just to make sure that if anything strange happens, all calls will be
> cancelled out at some point.
>
> /O
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