On Mon, May 5, 2008 at 11:38 AM, Csaba Zelei <csaba.zelei@gmail.com> wrote:

With a little hack I can make mod_portaudio to send rtp packets with ~24ms, ~16ms delay alternately on linux. This result in a constant 4ms jitter but its better than the original. (diff attached)
However on windows the delta between rtp packets is 15-32 ms randomly, with occasionally high 70-100ms delta.
I also tried to tweak the windows timer without success.
Does anybody has any idea how to make windows xp more accurate?

Sluschny, Thomas wrote:
as you can see here:

http://jira.freeswitch.org/browse/MODENDP-40

i have this problem all the time (the error mentioned in this issue was only related with this).

It has to do with windows handle sleep() method, you has say sleep(1) for 1ms but on my

machine it waits 15ms (it depends on your hardware, other PCs behave different!). So i tested around with high performance counters.

For now i ignore that problem an set jitterbuffers on other device big enough.

Thomas

Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Zelei Csaba
Gesendet: Donnerstag, 24. April 2008 19:05
An: freeswitch-users@lists.freeswitch.org
Betreff: [Freeswitch-users] mod_portaudio send 3 rtp packet/60ms instead of1 packet/20ms

Dear all,

I tried to use FS in client mode, starting calls with mod_portaudio to our providers gateway ( a Cirpack softswitch )
I experienced that there is 2-3 sec delay in the call, its choppy and robot like.
I tested it with a softphone, and an ip phone and everything was fine. I traced back the problem to mod_portaudio sending 3 rtp packet in 60ms instead of 1 packet/20ms.

Here is an rtp statistic from a call: (see http://pastebin.freeswitch.org/4307 for the complete list and sip trace)

Packet    Sequence    Delta (ms)
42 26138    0.00
43    26139    0.02
46    26140    45.69
47    26141    0.02
48    26142    2.96
52    26143    56.31
53    26144    5.75
54    26145    0.02
58    26146    51.99
59    26147    0.03
60    26148    2.96
63    26149    42.95
65    26150    17.06
66    26151    0.02
67    26152    2.90
71    26153    56.99
72    26154    0.03
73    26155    0.02

Did anyone else experience similar problems?
Is this the desired behaviour, because portaudio get data in 60ms interval or can I set it to 20ms somehow?

Thanks,

Csaba Zelei
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Index: mod_portaudio.c
===================================================================
--- mod_portaudio.c (revision 8260)
+++ mod_portaudio.c (working copy)
@@ -121,6 +121,7 @@
int ring_interval;
GFLAGS flags;
switch_timer_t timer;
+ switch_timer_t sync_timer;
} globals;

@@ -282,7 +283,15 @@
}

switch_set_flag_locked(tech_pvt, TFLAG_IO);
+
+ /* Start Synchronization Timer */
+ //Is it ok to always use 20ms? What about the 160 sample????
+ if ( switch_core_timer_init(&globals.sync_timer,"soft",20,160,switch_core_session_get_pool(session)) != SWITCH_STATUS_SUCCESS)
+ {
+ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Sync Timer failed!!\n");
+ }

+
/* Move Channel's State Machine to RING */
switch_channel_set_state(channel, CS_RING);

@@ -412,6 +421,8 @@
}

switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s CHANNEL HANGUP\n", switch_channel_get_name(switch_core_session_get_channel(session)));
+ /* Destroy timer */
+ switch_core_timer_destroy(&globals.sync_timer);

return SWITCH_STATUS_SUCCESS;
}
@@ -542,12 +553,17 @@
switch_mutex_lock(globals.device_lock);

get_samples:
-
+
if ((samples = ReadAudioStream(globals.audio_stream, globals.read_frame.data,
globals.read_codec.implementation->samples_per_frame,
&globals.timer)) == 0) {
+
+ //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "No data reset timer\n");
+ switch_core_timer_sync(&globals.sync_timer);
switch_yield(1000);
+
goto get_samples;
+
} else {
globals.read_frame.datalen = samples * 2;
globals.read_frame.samples = samples;
@@ -562,7 +578,9 @@
status = SWITCH_STATUS_SUCCESS;
}
switch_mutex_unlock(globals.device_lock);
-
+
+ switch_core_timer_next(&globals.sync_timer);
+
return status;
}

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