[Freeswitch-users] Conference silence timeouts

Bradley Brashier bjbrashier at gmail.com
Thu Aug 13 15:24:19 PDT 2009

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 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 <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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