asterisk users - Feb 2010 - SIP Disconnects from Network - Asterisk Does not hangup

Good day all!

I have an issue which has plagued me for quite sometime now...and as I close
in on its cause, I have reached a point where additional info would be
greatly helpful!

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...?!

On Mon, 2010-02-22 at 16:13 -0500, JT wrote:
> Is this something that is fixed in an update?  (Currently running 1.2)
Yes... modern versions of Asterisk support SIP session timers.  (If I
remember correctly, Asterisk 1.2 could tear down a call based on lack of
RTP data, but I never found it worked well enough in my tests to warrant
its use.)

--
Jared Smith
Digium, Inc.

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. 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.

I do not know if these settings are available in 1.2 though.

  -kkm
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