It would be great Daniel if you could do it. I will be more than happy to test it. A function parameter would be more flexible than a module parameter. I expect it shouldn't affect the behavior of the external application CallControl.
Thanks in advance :) Reda On Fri, Feb 24, 2012 at 08:09, Daniel-Constantin Mierla <mico...@gmail.com>wrote: > Hello, > > I am not using mediaproxy at all (but nathelper/rtpproxy), neither the > call control module, but making an option (module parameter or function > parameter) for call control to bind to another module like media proxy, > should not be big deal if it is all you are looking for -- I can look over > it and send a patch if you are going to help testing it. I cannot do it > these days, though, being out of the office. > > Cheers, > Daniel > > > On 2/23/12 8:59 PM, Reda Aouad wrote: > > First, I am posting about the wrong behavior of CallControl (or most > probably Kamailio modules) which leaves no option. I should be the only > one deciding about how to handle timeouts. If I decide to take some risk, > no module should oblige me to do otherwise. > > Mediaproxy detects ONLY RTP timeouts from BOTH parties, because linux > conntrack rules it uses are bi-directional. If a single party stops sending > RTP for whatever reason (connection lost, codec with silence detection > used, ....), mediaproxy doesn't care and doesn't act upon it. This is a > feature, and a wanted one, to mainly support voice-detecting codecs. Think > also about conferences for example, in which only a single person talks for > a long time while others are silent and don't send RTP. > > Single-side RTP timeout because of a real problem (loosing network > connection for example) should be handled with other methods, such as SIP > session timers. > > MY POINT IS : I don't see it practical to handle RTP flows for EVERY > call to handle the least probable scenario: an RTP timeout from both (or > all) parties. > > If I understood well, mediaproxy updates the CDR when it detects an RTP > timeout from both parties. CallControl can look in the CDR to debit the > correct balance, instead of attaching itself to the dialog module to detect > dialog termination. > > This is an extract from the call_control module : > > Even when mediaproxy is unable to end the dialog because it was not > started with engage_media_proxy(), the callcantrol application is still > able to detect calls that did timeout sending media, by looking in the > radius accounting records for entries recorded by mediaproxy for calls that > did timeout. These calls will also be ended gracefully by the callcontrol > application itself. > > > Unless there is something I miss.. > > I also opened a bug about the issue because call_control doesn't have > the same behavior with OpenSips. It doesn't force mediaproxy. > > Reda > > > > On Thu, Feb 23, 2012 at 20:00, Jeff Brower <jbro...@signalogic.com> wrote: > >> Reda- >> >> > It's clear but not necessary. It can look at radius records fixed by >> > mediaproxy on RTP timeout to debit the correct balance as well. And why >> > also force it on postpaid calls which it doesn't control at all ? >> >> I don't understand how you plan to tear down Kamailio calls that suffer >> RTP time-out? >> >> -Jeff >> >> > What happens is cost and performance issues for additional calls passing >> > through my mediaproxy server, which I didn't plan for at first. No audio >> > issue at all. >> > >> > Reda >> > >> > >> > >> > On Thu, Feb 23, 2012 at 11:58, Sammy Govind <govoi...@gmail.com> wrote: >> > >> >> Reading from the module docs its clear why it needs to engage media/rtp >> >> proxy to start,stop billing or timer of a call. so what happens when it >> >> engages mediaproxy on unwanted calls !? audio-issues? >> >> >> >> >> >> On Thu, Feb 23, 2012 at 1:21 PM, Reda Aouad <reda.ao...@gmail.com> >> wrote: >> >> >> >>> Thanks Sammy. I didn't get any reply yet. >> >>> >> >>> CallControl is an application used with CDRTool for prepaid calls. It >> >>> calculates the maximum call duration based on the user's balance. >> Once the >> >>> call's duration limit is reached, it sends BYE to both calling >> parties, >> >>> terminating the call. At the end of a prepaid call, terminated either >> by >> >>> the user or by CallControl, it debits the user's balance according to >> the >> >>> call's duration. >> >>> >> >>> The Call_Control module interfaces with this external application. >> >>> >> >>> call_control function is called in Kamailio's cfg to check if the user >> >>> has prepaid or postpaid account, and get the max call duration for >> prepaid >> >>> users. CallControl controls only prepaid calls, not postpaid ones. >> >>> >> >>> So call control and NAT traversal using mediaproxy are two differents >> >>> things which i can't link, since I don't want mediaproxy for every >> call. >> >>> And since the function call_control is called on every invite before >> >>> knowing if the user has a prepaid account or not, it engages >> mediaproxy for >> >>> every call. >> >>> >> >>> CallControl relies on mediaproxy to detect RTP timeouts and debit the >> >>> correct balance from a prepaid account based on the last instant the >> >>> mediaproxy saw an RTP packet. >> >>> >> >>> But why to force using mediaproxy with no choice? And why to force it >> for >> >>> every call, whether it falls under CallControl's control or not? >> >>> >> >>> I am using Kamailio 3.2. >> >>> >> >>> >> >>> Reda >> >>> >> >>> On 23 févr. 2012, at 07:21, Sammy Govind <govoi...@gmail.com> wrote: >> >>> >> >>> Hi, >> >>> I can see you posting multiple times on both proxies listings so I'm >> sure >> >>> you havent heard back from anyone.I am not at all familiar with your >> >>> functions in email but could it be possible for you to determine on >> which >> >>> calls you need to engage mediaproxy and on which not to, then on the >> base >> >>> of that flag use the call_control function ! >> >>> your problem is complicated for me atleast. I hope somebody could >> answer >> >>> you accurately and precisely. >> >>> >> >>> btw, what are you using in real? opensips or kamailio, which version? >> and >> >>> in what context you need to use the call_control function? >> >>> >> >>> Thanks, >> >>> Sammy >> >>> >> >>> On Thu, Feb 23, 2012 at 12:45 AM, Reda Aouad <reda.ao...@gmail.com >> >wrote: >> >>> >> >>>> Hi, >> >>>> >> >>>> When I use the function call_control( ) of the call_control module, >> it >> >>>> automatically engages mediaproxy if it finds the mediaproxy module >> loaded. >> >>>> If the mediaproxy module is not loaded, call_control doesn't even >> try to >> >>>> engage it. >> >>>> >> >>>> I need mediaproxy for NAT traversal in some cases, but don't want it >> to >> >>>> be engaged on every call. >> >>>> >> >>>> How can I disable this behavior? >> >>>> >> >>>> Thanks >> >>>> Reda >> >>>> >> >>>> _______________________________________________ >> >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> >>>> sr-users@lists.sip-router.org >> >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >>>> >> >>>> >> >>> _______________________________________________ >> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> >>> sr-users@lists.sip-router.org >> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >>> >> >>> >> >>> _______________________________________________ >> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> >>> sr-users@lists.sip-router.org >> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >>> >> >>> >> >> >> >> _______________________________________________ >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> >> sr-users@lists.sip-router.org >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> > sr-users@lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> >> > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla -- > http://www.asipto.comhttp://linkedin.com/in/miconda -- > http://twitter.com/miconda > >
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