[Freeswitch-users] Conference silence timeouts
mike at jerris.com
Thu Aug 13 22:47:29 PDT 2009
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.
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.
> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at 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
> 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.
> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com
> > wrote:
> I had just thought of the exact same thing. I'm trying to test that
> now. Thanks for your input.
> On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>
> 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?
>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins
>> <msc at 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...
>> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at 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?
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