On Fri, 2008-01-11 at 02:39 +0100, Andreas Granig
wrote:
Jerome,
In my opinion it depends on the policy of the VoIP provider rather than
on technical issues.
Agreed, the definitive solution does not exists, and is policy and
environment-dependent.
Proper implementation of RFC 4028 of all involved
UACs might render RTP
analysis useless, if it's in line with the policy of the the VoIP
provider to have some minutes of tolerance in their CDRs in case of
missing BYEs (the tolerance can be controlled by the provider via the
defined headers). If that is still unacceptable by a provider, there
maybe should be some SIP/RTP-aware billing engines in place though.
What I don't get here is the "minutes of tolerance". Typically, RTP
timeout is in the same order of time, for what I've seen. Do you use an
RTP timeout of a few seconds only ? If so, clearly the issues I've
mentionned earlier are even worse with such a short timeout.
Talking about policy, I would say it is in the best interest of every
provider to limit the amount of "potentially post-BYE, hard-to-bill"
minutes. But even with RTP detection, this is hard to acheive.
Or maybe you're thinking more of a hybrid solution ? Like RTP timeout
triggering a SIP ping, which in return, if failling, triggers call
termination. But this is really tough to handle, particulary the case
when you don't have RTP but the UAC is still responding to SIP
signaling.
I'm really curious, could you give me a real-world example of an
RTP-detection based soution providing sub-minute dead UA detection ?
My provider (about 50.000 customers, few milion minutes per month)
detects RTP streams and terminate calls after 10 seconds without RTP.
VAD is disabled, of course :-) Its solution is based on propritary telco
SIP to SS7 gateway...
BTW: The same thing I do with Asterisks behind my OpenSER (10 secs RTP
timeout), but I'm very interested in SIP pinging. Could you point me to
some hint/working example how to do it with SIP proxy (like OpenSER)?
Thanks in advance,
kokoska.rokoska