I didn't see any SIP session timers in the wiki. Since I'm already using the event socket for control, my current plan is to use sched_api to play a file with a short (20ms?) clip of silence, capture the play_file event and use it to reschule another one for a couple of seconds later.
I'll let you know what happens. BB On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <m...@jerris.com> wrote: > My suggestion is to use sip session timers not rtp timeouts as rtp is > supposed to be discontinuous. That being said, we have several settings to > continuously send media, but then you are doing exactly what you said you > didn't want to do. > Mike > > On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote: > > OK, I finally got a moment to do a packet capture and take a look at the > streams. It became very clear very quickly that what happens is that during > silence the gateway still sends RTP packets to Freeswitch, but Freeswitch > doesn't send any back to the gateway. After 10s of this, the gateway says > "Oh, the RPT must be broken" and it hangs up. > > We found a way to turn off this behavior in the gateway, and the good news > is that it did indeed fix the problem. But we'd rather not rely on that as a > long-term solution because then we can't detect and drop RTP streams that > really are broken. > > So now I'm back to looking at Freeswitch to figure out how to send just a > single packet every second or so during silence. If anyone knows of a way to > do this, let me know, otherwise I'll get back to you if and when I find one. > > BB > > On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrash...@gmail.com>wrote: > >> I took a closer look at the SIP messages on the console. From it, I >> understand that it's not Freeswitch timing out, but rather FS is getting the >> "BYE" msg from somewhere else. I've tested phones and tested calling without >> going through the FS conference, though, and everything works fine. Then I >> saw something else odd inside the BYE msg: >> >> Reason: Q.850 ;cause=31 ;text="RTP Broken Connection" >> So I Googled "RTP Broken Connection" and saw several sites talking about >> AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From >> these sites it sounds like AudioCodes is rather aggressive in detecting RTP >> breaks, and is interpreting the silence from FS as a break. >> >> So now I'm looking into ways to maybe send "I'm still here" RTP packets >> from FS or to tune the gateway to be less aggressive. I can't stop and get a >> clean packet capture right now because I've got a bunch of testers working >> on it today. I'll do that sometime when the system is less busy. >> >> BB >> >> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier >> <bjbrash...@gmail.com>wrote: >> >>> I had just thought of the exact same thing. I'm trying to test that now. >>> Thanks for your input. >>> >>> BB >>> >>> On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <m...@jerris.com>wrote: >>> >>>> My guess is that its the other end killing the call due to rtp >>>> timeouts, not us killing the call. If you can confirm this the best method >>>> would be to get them not to do rtp timeouts. >>>> On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote: >>>> >>>> I'm sure that would work, but I'm worried about it sucking up >>>> bandwidth, especially since you'd need it on every caller (since otherwise >>>> the one person who had it could hang up and you'd be back to square 1). >>>> >>>> Any other ideas, or should I hunt through the code to try to override >>>> the behavior manually? >>>> >>>> BB >>>> >>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins >>>> <m...@freeswitch.org>wrote: >>>> >>>>> Check out the 'waste' member flag. I think if at least one member has >>>>> that set then RTP will get sent out even during silence. Let us know if >>>>> that >>>>> helps... >>>>> >>>>> -MC >>>>> >>>>> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier < >>>>> bjbrash...@gmail.com> wrote: >>>>> >>>>>> Hi all. >>>>>> >>>>>> The solution to this one should be short. >>>>>> >>>>>> My conference hangs up when there's 2+ users and silence for 5 sec or >>>>>> so. I'm trying to find a parameter that changes that (I'd rather it be, >>>>>> say, 60 seconds). >>>>>> >>>>>> I didn't see a parameter like this specific to conferences, so I >>>>>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's >>>>>> set >>>>>> to 300 (the default), so I'm pretty sure that's not the problem. I also >>>>>> searched through the mod_conference.c code and didn't see it, though I >>>>>> was >>>>>> only skimming. >>>>>> >>>>>> I'm not 100% convinced that this is limited to conferences, but I >>>>>> don't currently have a way to test in a non-conference environment. >>>>>> >>>>>> Anybody know how to increase the conference silence-hangup timeout? >>>>>> >>>>>> BB >>>>>> >>>>>> _____ >>>>> >>>>> >>>> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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