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

[Anthony Minessale]

I can promise you that much of your problems will be solved with latest
SVN.  There was a small change in the timer sync that was causing your
symptoms in specific cases and it's been resolved for months.

Did you answer the question about what phones?  I'm going to guess Cisco
based on the symptoms.

non bridge calls use a timer to make sure the audio is coming in at a steady
rate to ensure bursting RTP
is played at the correct rate.  Stopping it and restarting 2 seconds later
will cause a delay by design because you have suspended the process but not
the UDP stack.  If you don't want to use rtp-timer you can disable it in the
profile settings for rtp-timer-name by setting it to "none" or the channel
variable rtp_timer_name=none on a per call basis, this is not necessarily
recommended for everyone because it's a blocking read on the rtp socket
which is usually undesirable in an asynchronous thing like a phone call.


BTW,

Conference counts as a bridge because it has 2 threads one for each
direction




On Wed, Nov 18, 2009 at 3:33 PM, Robert L Mathews <lists at tigertech.com>wrote:

> Anthony Minessale wrote:
> > Have you tried SVN trunk?
>
> No, I haven't. I'll try it. (But also, if others want to try seeing if
> it happens, it's trivially duplicated by calling the echo test app and
> sending the freeswitch server process a "-STOP" signal, sleeping for a
> second, then sending it a "-CONT" signal.)
>
> In any case, though, I have partially found the cause of the problem --
> at least in the echo test module in 1.0.4. It's the same problem
> reported here:
>
>
> http://www.mail-archive.com/freeswitch-users%40lists.freeswitch.org/msg15800.html
>
> The two suggestions there, explicitly setting
> "rtp-autoflush-during-bridge" true and "rtp_timer_name=none", didn't fix
> it for me. (The first is no surprise because that's the default anyway.)
>
> However, setting the (undocumented?) parameter "rtp-autoflush" to true
> *did* fix it in the echo test (but not the conference).
>
> I think what's happening is that FreeSWITCH contains code to detect when
> we've "fallen behind" the RTP audio. It looks like this happens in
> rtp_common_read() of switch_rtp.c: if the code detects that there are
> "extra" RTP packets waiting during several consecutive rtp_common_read()
> calls, switch_core_timer_sync() is called to catch up.
>
> This code is apparently used during bridged calls when
> "rtp-autoflush-during-bridge" is true (the default), which explains why
> I don't have the problem during SIP-to-SIP calls.
>
> However, I'm guessing that echo test calls somehow aren't considered
> "bridged" by that code. Therefore the code isn't being used unless
> "rtp-autoflush" is true.
>
> That thread suggests that this is probably a phone or network problem,
> but it seems to me that even if all the timing is perfect, this problem
> will happen if the freeswitch server thread doesn't get enough CPU time
> to retrieve a packet before the next one arrives. For example, if
> packets arrive every 20 ms but high load on the server causes the
> process to sleep for 25 ms, so that two packets are waiting the next
> time the process runs, it will never catch up that extra packet -- the
> echo test or conference will now be permanently 20 ms behind. And if
> that happens again, it will now be 40 ms behind, and so on. That
> explains why the lag slowly increases over time.
>
> Does that make sense? I don't quite understand why the "catch up" code
> isn't always used for all RTP streams: if an RTP packet poll repeatedly
> shows that there are extra audio packets waiting, it seems to make sense
> that we'd always want to catch up.
>
> Anyway, as I said, I'm still having the conference problem, even with
> "rtp-autoflush" enabled. So I need to try the svn trunk version and see
> if it still happens, then track it down further if so. I will report back.
>
> --
> Robert L Mathews, Tiger Technologies
>
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-- 
Anthony Minessale II

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