[OpenSER-Users] RTP-Proxy

kokoska rokoska kokoska.rokoska at post.cz
Fri Jan 11 09:08:08 CET 2008




Jerome Martin napsal(a):
> 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





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