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: _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users .

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(a)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(a)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(a)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 >>>>>> >>>>>>

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 _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

Daryl Sanders wrote: Yes they should - but there a thousands of GW providers which have no glue about their equipment :-( klaus _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

Hi Greg, keeping the call control on your hands is an vital need...With the tomorrow release I prepared the first steps for a light dialog support based on Routing mechanism - complete FROM changing and restoring for the whole dialog was the prove of concept :). By being able to control the call from somewhere in the middle, I target especially issue no 2 and why not 4 and 5. ....and all this by avoiding the B2BUA....... regards, bogdan Greg Fausak wrote: _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

Hi Greg, keeping the call control on your hands is an vital need...With the tomorrow release I prepared the first steps for a light dialog support based on Routing mechanism - complete FROM changing and restoring for the whole dialog was the prove of concept :). By being able to control the call from somewhere in the middle, I target especially issue no 2 and why not 4 and 5. ....and all this by avoiding the B2BUA....... regards, bogdan Greg Fausak wrote: _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users .

Bogdan, can you give us more details of your plans? We have to decide how to deal with the problem of killing a call in progress and detect an hunged-up call. I'd like to avoid the use of a B2BUA, so I'd like to know if, when and how you plan to obtain this. I really cannot understand how you will achieve this simply with a SER module. You need some way to store all the data of dialogs in progress... Don't you? Thanks. Bogdan-Andrei Iancu wrote: