[Freeswitch-users] Audio delays from conferences under heavy CPU load on Windows

[Peter Olsson]

Try to start with the param "--monotonic-clock", it might enable a more
accurate clock on these kind of systems.


2014-02-04 Markus von Arx <mkvonarx at gmail.com>:

> Hi
>
> I've got a nasty problem with FreeSwitch conferences: after some time, I
> observe that audio that goes through FreeSwitch conferences gets more and
> more delayed (by the FreeSwitch conferences). Sometimes it's only 2
> seconds, but I have also observerd audio delays up to 5:30 minutes. Audio
> is never missing, only delayed, and even transmitted correctly after many
> minutes of delays.
>
> More details:
> - FreeSwitch 1.2.17, 64bit
> - OS: Windows Server 2012 64bit and Windows 7 64bit (observed on both
> platforms)
> - audio codec: G.711 alaw only
> - external channels: only SIP channels (mod_sofia), RTP with 20ms packets,
> G.711 alaw, all SIP endpoints in the local LAN on the same Ethernet switch.
> - dialplan: very very simple; only two actions: answer and connect every
> incoming/outgoing SIP channel directly to its private conference
> (mod_conference); and I set and export the following variables on all
> channels: rtp_disable_vad_in=true, rtp_disable_vad_out=true,
> send_silence_when_idle=-1, bridge_generate_comfort_noise=-1,
> suppress_cng=true
> - my own application: does some logic over mod_event_socket (on
> localhost); basically connects/disconnects these conferences using loopback
> channels (mod_loopback)
>  -> simple example: two SIP channels from SIP users 1001 and 1002,
> dialplan creates two conferences 1001c and 1002c, my application connectes
> 1001c and 1002c with a loopback channel, audio from SIP user 1001 travels
> through SIP channel to FreeSwitch conference 1001c, then through loopback
> channel to conference 1002c, then through SIP channel to SIP user 1002. The
> audio delay I'm talking about is that what user 1001 speaks into his phone
> arrives at user 1002 N seconds later instead of almost immediately.
> - mod_conference config: rate=8000, interval=20, max-members=99,
> energy-level=0, *-sound="", comfort-noise=0, conference-flags=audio-always
> - mod_sofia: rtp-timer-name=soft, suppress-cng=true, vad=none
> - timer: soft timer (no other available on Windows)
>
> Usage scenarios:
> - failing scenario 1 (low load): 34 SIP channels, 34 conferences, 97
> loopback channels, 228 FreeSwitch channel objects (34 + 2*97)
>  -> ~24% CPU load
> -> audio delays only sometimes, a bit unpredictable, mostly none, but
> sometimes in the region of 2-3 seconds
> - failing scenario 2 (medium load): ...
> -> ~51% CPU load
>  -> audio delays start slowly, in the region of 2-10 seconds, grow slowly
> with time while keeping channels and conferences alive
> - failing scenario 3 (high load): 66 SIP channels, 66 conferences, 385
> loopback channels, 836 FreeSwitch channel objects (66 + 2*385)
>  -> ~74% CPU load
> -> audio delays start almost immediately: 18 seconds delay after 15
> minutes, 41 seconds delay after 3 hours, 5:30 minutes delay after 19 hours.
>
> => My guess: it could be that under heavy CPU load, the (soft) timers that
> are responsible for the RTP audio streams don't get fired reliably anymore
> but sometimes (often) too late.Theoretically this should not cause an ever
> increasing delay, as the next timer event after the delayed one should be
> queued a bit earlier. But still somehow the delay happens and it looks like the
> mixed RTP packets from the conferences are sent later and later and the
> delay is never recovered.
>
> => Any ideas? Is this a known issue? Maybe some combination of Windows,
> soft timers, high CPU load, loopback channels, disabled CNG/VAD, long
> running conferences? Maybe a FreeSwitch (timer) bug? Any help would be
> appreciated!
>
> Thanks and best regards,
> Markus
>
>
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