[Freeswitch-users] Conference silence timeouts

Bradley Brashier bjbrashier at gmail.com
Thu Aug 13 14:48:55 PDT 2009

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.


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.
> BB
>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at 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 <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...
>>> -MC
>>>   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?
>>>> BB
>>>> _____
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