[Freeswitch-users] Conference silence timeouts

Bradley Brashier bjbrashier at gmail.com
Fri Aug 14 11:41:41 PDT 2009

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.


On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <mike at 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 <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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