[asterisk-dev] SIP session-timers: concept, discussion

[Raj Jain]

Session timers is a pretty useful feature in my opinion. The value of it is
clear when the media is not flowing through the Asterisk. We should
definitely implement it in Asterisk. 

As a data point, majority of the end-points support session-timers (60%
according to the last SIPit).

http://www1.ietf.org/mail-archive/web/sip/current/msg18959.html
 
Raj


> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com 
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Grey Man
> Sent: Wednesday, July 18, 2007 3:43 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] SIP session-timers: concept, discussion
> 
> > ---- Original Message ----
> 
> > From: John Todd <jtodd at loligo.com>
> 
> > To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> 
> > Sent: Wednesday, 18 July, 2007 6:52:36 AM
> 
> > Subject: Re: [asterisk-dev] SIP session-timers: concept, discussion
> 
> 
> >
> > Any implementation of Session Timers into Asterisk, alas, will not
> 
> > cure your problems according to the limited data you've 
> provided.  It
> 
> > sounds like the calls are being "answered" at the far end, and then
> 
> > audio continues to flow between the endpoints (since you're already
> 
> > using rtptimeout, I assume you're handling media.)
> 
> 
> >
> > I  expect your international carriers are not providing adequate
> 
> > answer supervision or disconnect supervision.  This can't be solved
> 
> > by any code in Asterisk - it requires clubbing your carriers with a
> 
> > threat of moving to a better network.
> 
> 
> >
> > The Session Timers will resolve other, more subtle problems, like a
> 
> > carrier network or a SIP UA that has lost it's mind or has lost it's
> 
> > network.  We already keep state in Asterisk - this just forces the
> 
> > state to be refreshed, and if that fails, hang up.  It won't cure
> 
> > problems with calls that "look" normal, at least as far as Asterisk
> 
> > can see.
> 
> 
> 
> 
> Hi John,
> 
> I understand what you are syaing and the truth is I don't 
> know why we get calls that don't hangup on our Asterisk 
> servers. I only ever find up afterwards when a customer 
> complains about being over charged so it's not something that 
> can be easily diagnosed. The porblem has been occurring for 
> the whole time I've been operating Asterisk over the last 3 
> years and across various versions. It's not a huge deal as at 
> one or two calls a month it's an annoyance rather than a big issue.
> 
> I do have the media going through my servers but I don't 
> believe that mechanism is foolproof hence the enthusiasm with 
> the SIP session timers approach, conceptually it sounds a lot 
> more robust. I actually put a time limit of 3 hours on all 
> calls in order to catch instances where the rtptimeout does 
> not hangup the call. I suspect in some cases the user is 
> accidentally putting the calls on hold by pressing the 
> on-hook button as a lot of queries come from simultaneous 
> calls to the same destination. It could be that the UA is 
> putting a call on hold re-dialing and then for whatever 
> reason forgetting about the original call. The user then 
> hangs up and one or both calls don't get BYEs. In theory 
> there should be no RTP from the UA and the rtponholdtimeout 
> should kick in but in some cases it's not. The SIP session 
> timers would be a good belts and braces approach. If the UA 
> has been hung up then it should generate an error response to 
> the UPDATE or reINVITE and then Asterisk can  hangup the call 
> irrespective of any RTP timers.
> 
> As for international carriers since a lot of the calls we get 
> disputed are to out of the way places where the telecoms 
> infrastructure may not be the best I suspect that either the 
> signalling is getting screwed up so calls do not get hungup 
> from the callee end or the carrier may choose not to hangup 
> incoming calls from their own end in order to maximise 
> revenue. The latter opinion being more of a a conspiracy than 
> scientific theory.
> 
> Irrespective I do have a problem with calls not getting 
> hungup as reliably as I believe they should be and at the 
> very least the SIP session timers can only improve that 
> situation not make it worse.
> 
> Regards,
> 
> Greyman.
> 
> 
> 
> 
> 
> 
> 
> 
>       
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