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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
