[Users] Detecting runaway calls

Fri Oct 28 10:05:19 CEST 2005

Hi all!

It depends on what you expect from SIP. If you want to use it as POTS 
replacement, or if you want to have full control over every terminal, 
than I suggest using proprietary protocols.

If you are dreaming the Internet dream, then the service should be 
end-2-end without a service provider inspecting all signaling, payload 
and media.

And if there are broken clients, the clients should be fixed. Of course 
you can save $$ by giving cheap (broken) phones to your costumers and 
fix it by spending $$ for a SBC. Depending on the numbers of clients you 
have, it may be cheaper using cheap phones and buy the SBC.

One big problem with SBC are the service contract - you have to buy them 
for each year to get updates. If you detect a bug and ask for the 
bugfix, they ask you for $$.

Of course if your business depends on selling minutes to the PSTN and 
being cheaper than traditionel POTS, you have to be sure that the 
billing is accurate.

And do not forget - there is a great B2BUA for free - it's called 
"Asterisk". And it will detect dead calls as well.

regards
klaus

Joachim Fabini wrote:
> Hi,
> 
> I'd also like to add my 2 cents to this topic, which
> might be slightly off-topic on this mailing list.
> Imho the problem discussed here is a major defficiency 
> of the SIP protocol, which forces most vendors to 
> implement proprietary extensions in their equipment. 
> Call supervision is a feature that might not be 
> important for best-effort services, but is of utmost
> importance when deploying SIP within a commercial 
> environment.
> 
> Have a look at, e.g. H.323. They do have several
> redundant mechanisms to detect stale calls (e.g. 
> at H.225 RAS level (IRQ/IRR/IACK/INAK), and at 
> call signaling level (Status request/Status)). These 
> messages offer detailed information on all ongoing
> calls within a terminal/gateway - either on sending
> an explicit request or by asking the endpoint to 
> provide this information on a periodical base.
> Beside this, RRQ/RCF also provides re-registration 
> features similar to REGISTER timers in SIP.
> 
> Again imho the clean solution for SIP was to have 
> an equivalent of these features defined in the 3261 core. 
> It's too late for this now. What still can (and should) be 
> done is an RFC that defines an extension to SIP which 
> implements call supervision.
> 
> Concluding: It's obvious that _real_ call supervision 
> that is capable to detect malicious, non-standard 
> compliant terminals (which, e.g. signal voice but send 
> video over the voice channel) requires some border 
> gateways featuring RTP content sniffing. But this is 
> imho no excuse for SIP not providing standardized 
> means capable to supervise calls on standard-compliant
> SIP devices.
> 
> regards
> --Joachim
> 
> 
> 
>>-----Original Message-----
>>From: users-bounces at openser.org 
>>[mailto:users-bounces at openser.org] On Behalf Of Bogdan-Andrei Iancu
>>Sent: Donnerstag, 27. Oktober 2005 21:02
>>To: Daryl Sanders
>>Cc: users at openser.org
>>Subject: Re: [Serusers] Re: [Users] Detecting runaway calls
>>
>>Daryl,
>>
>>normally, a PSTN provider who offer quality service must 
>>offer a way to detect hanged calls.
>>There is also a detection mechanism based on media (on GW) - 
>>at least CISCO GW have.
>>
>>regards,
>>bogdan
>>
>>Daryl Sanders wrote:
>>
>>
>>>This is exactly what I am trying to overcome. Do third party PSTN 
>>>providers typically employ any other means of monitoring a call? I 
>>>would think the would automatically disconnect the call 
>>
>>after a certain 
>>
>>>amount of time if there was no media detected.
>>>
>>>On 10/27/05, Iqbal <iqbal at gigo.co.uk> wrote:
>>> 
>>>
>>>
>>>>problem with session timers, is if you have several 
>>
>>different routes 
>>
>>>>and you dont control all the end gateways....then its a pain, so I 
>>>>would think you would need a plan B, just in case to ensure 
>>
>>you double 
>>
>>>>checked, but agree with bogdan, b2bua i great but I think 
>>
>>the overhead 
>>
>>>>is not needed.
>>>>
>>>>Iqbal
>>>>
>>>>Bogdan-Andrei Iancu wrote:
>>>>
>>>>   
>>>>
>>>>
>>>>>Hi Greg,
>>>>>
>>>>>that's the fortunate case of having already a B2BUA on the path :).
>>>>>But only for this particular case, I find the usage of a B2BUA a 
>>>>>little bit to  "heavy"....
>>>>>
>>>>>Session Timer is actually doing a great job ;)
>>>>>
>>>>>regards,
>>>>>bogdan
>>>>>
>>>>>Greg Fausak wrote:
>>>>>
>>>>>     
>>>>>
>>>>>
>>>>>>The re-INVITEs definitely work.  I run some setups with Jasomi 
>>>>>>inline (a commercial inline b2bua).  I configure them to 
>>
>>reINVITE, 
>>
>>>>>>works like a charm.  If the reINVITE isn't answered (in either
>>>>>>direction) the Jasomi sends BYEs in both directions.  I believe 
>>>>>>other commercial gear works that way.
>>>>>>
>>>>>>Also, I believe Session-Timers are supported by Cisco 
>>
>>gateways, but 
>>
>>>>>>I don't have experience with those.
>>>>>>
>>>>>>-g
>>>>>>
>>>>>>
>>>>>>On 10/27/05, Daryl Sanders <daryl.sanders at gmail.com> wrote:
>>>>>>
>>>>>>
>>>>>>       
>>>>>>
>>>>>>
>>>>>>>Thanks for the info guys! It sounds like I need to do a little 
>>>>>>>reading up on cseq to determine if this will even work, 
>>
>>or find a 
>>
>>>>>>>PSTN gateway provider that supports Session-Timers.
>>>>>>>
>>>>>>>- Daryl
>>>>>>>
>>>>>>>On 10/27/05, Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote:
>>>>>>>
>>>>>>>
>>>>>>>         
>>>>>>>
>>>>>>>
>>>>>>>>by sending re-INVITEs from the middle of the path you will 
>>>>>>>>increase the cseq number differently on each side...so you will 
>>>>>>>>need to synchronize the cseq value when some in-the-dialog 
>>>>>>>>requests are passing through your proxy....and that's quite 
>>>>>>>>complicated....
>>>>>>>>
>>>>>>>>regards,
>>>>>>>>bogdan
>>>>>>>>
>>>>>>>>Daryl Sanders wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>           
>>>>>>>>
>>>>>>>>
>>>>>>>>>Would it be possible to fake a REINVITE then check the 
>>
>>response 
>>
>>>>>>>>>to determine if a call is still in session? Just 
>>
>>brainstorming...
>>
>>>>>>>>>- Daryl
>>>>>>>>>             
>>>>>>>>>
>>
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