[Freeswitch-users] Call latency in conferences and echo test increases over time

[Brian West]

Can you tell us what kind of phone you're using?  And have you tried  
this on SVN trunk?

/b

On Nov 17, 2009, at 4:18 PM, Robert L Mathews wrote:

> I'm using FreeSWITCH 1.0.4.
>
> When I make a call from a SIP phone to either a conference or an echo
> test on the FreeSWITCH server, the latency ("lag") starts off very low
> -- a fraction of a second. But as several minutes of time goes by,  
> the 
> lag increases. After, say, 15 minutes, the lag will reach a couple of
> seconds, making conference calls unusable.
>
> This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is
> handling the RTP media.
>
> If I hang up and immediately call back in (even to the same  
> conference),
> the lag is reset to almost zero. If I put the call on "hold" and  
> take it
> off hold, the lag is also gone.
>
> During testing, I've found that this may be related to the freeswitch
> app on the server not getting all the CPU time it wants.
>
> If I suspend the freeswitch process for two seconds and then resume  
> it,
> the sound stops for two seconds, as I'd expect. But the echo/ 
> conference
> calls that were active are then lagged by two seconds until they  
> hang up
> (or get put on hold), even after freeswitch is resumed and getting all
> the CPU time it needs.
>
> This is easily reproduced by making a SIP call to the echo test  
> module,
> then:
>
>  pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch
>
> Any echo test or conference call that was in progress will then be
> permanently lagged by two seconds. However, any SIP-to-SIP calls that
> were in progress will not become lagged.
>
> Of course, killing it with -STOP is an artificially nasty thing to do.
> But it effectively just prevents it from being scheduled on the CPU  
> for
> a short period of time, and I can duplicate the same behavior (more
> gradually) by just increasing the load on the machine to the point  
> that
> the freeswitch app isn't getting much CPU time.
>
> Just for the record, I get the same results from several different
> phones and several different Internet connections, all of which have a
> ping latency of under 40 ms to the server. This problem does not  
> happen
> using the same phones and network connections to an asterisk server.
>
> Throwing out an even more complicated example that I've encountered:  
> If
> I have a SIP-to-SIP call going from party A to party B and I stop the
> process for two seconds, it doesn't permanently introduce lag to that
> call, as I mentioned. But if a third person (party C) starts
> eavesdropping on the call and presses "3" to make it a three way call,
> and then I suspend it for two seconds, the call between A and B isn't
> lagged, but what party C hears and sends *is* lagged.
>
> Any ideas on how to fix this? Do other people see the same thing
> happening? As I said, the gradual increase in lag over a long period  
> of
> time makes long conferences unusable, unfortunately.
>
> -- 
> Rob
>
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