[asterisk-dev] SIP ghost calls

[Klaus Darilion]

On Thu, November 30, 2006 18:49, Hans Petter Selasky said:
> On Thursday 30 November 2006 18:03, Zoa wrote:
>> Google for rtptimeout
>>
>> Zoa
>
> Yes, I found "rtptimeout" and "rtpholdtimeout". But I think "qualify" is
> more
> interesting.


qualify does not help you. What if the phone crashes and reboots. Then
qualify would succeed, as the phone has rebooted.

rtptimeout is IMO the best solution. SIP phones should send RTP packets at
regular intervals, e.g. every 30 seconds.

Of course this does not work if the RTP stream is directly between two SIP
clients. In this case you can use session timer.

Another solution would be to not implement the whole session timer thing,
but only a configurable reINVITE timer. E.g. in sip.conf:
keepalivereinvite=600

which cause Asterisk  to send reINVITE (with the same SDP as before). If
the reINVITE fails, the call will be terminated.

regards
klaus

>
> What about "outside bridges", when Asterisk does not forward the RTP
> stream?
>
> On Thursday 30 November 2006 18:09, Volkov Alexei wrote:
>> Hi Hans Petter!
>>
>> First of all you do not will pay for unconnected call.
>> If someone answer your ghost call, i think it disconnect quickly for two
>> reasons
>>     INVITE not confirmed
>>     empty RTP stream for some predefined time (3 min by default) leads
>> to call disconnect.
>>
>> And last, but not least, try to use qualify=3000 in sip friend. It makes
>> asterisk to peroidicaly ping sip endpoint to track the reachability.
>
> Whould it be an idea of Asterisk would qualify a SIP phone also when it is
> connected, to see if it is still present. That would work in any telephone
> state, and not just when one is receiving RTP packets or the call is on
> hold?
>
> Is "qualify" the equivalent to STATUS_ENQUIRY in the EuroISDN world ?
>
>>
>> > Hi,
>> >
>> > I'm playing around with a new SIP phone connected to Asterisk 1.2.13.
>> I
>> > am performing some tests, and one of them is the "cable unplug" test.
>> >
>> > During a call I unplugged the ethernet cable, and to my surprise, the
>> > call was not disconnected. Even after 20 minutes. This is very bad!
>> >
>> > Can anyone explain why Asterisk does not ping the SIP phone regularly,
>> > and if there is no reply, disconnect all associated calls?
>> >
>> > Just imagine what happens if two persons call eachother, using SIP
>> phones
>> > over the PSTN network, and both disconnect their phones. Who is going
>> to
>> > pay for the ghost call, which might last forever?
>> >
>
> Thanks,
> --HPS
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