[Freeswitch-users] mod_portaudio send 3 rtp packet/60ms instead of1 packet/20ms
Anthony Minessale
anthony.minessale at gmail.com
Mon May 5 10:13:21 PDT 2008
did you try setting the ptime on the rtp to 30 or 60ms
when you choose a codec in your sip settings on FS in vars.xml, instead of
PCMU try PCMU at 30i or PCMU at 60i
it may be that the other side is doing 30 or 60 ms and not telling us.
also in pablio.c in the portaudio_mod directory in ReadAudioStream func,
there is a sleep 1 ms too
if the windows is really sleeping a lot longer than that, try omitting line
158.
This probably will consume the whole cpu but if it fixes your problem it
will support the theory that the sleep on windows in inaccurate.
On Mon, May 5, 2008 at 11:38 AM, Csaba Zelei <csaba.zelei at 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*<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 at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *Im Auftrag von *Zelei Csaba
> *Gesendet:* Donnerstag, 24. April 2008 19:05
> *An:* freeswitch-users at 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
>
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> 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;
> }
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080505/04aa0055/attachment-0002.html
More information about the FreeSWITCH-users
mailing list