[Freeswitch-users] disabling ptime warning message

Mon Feb 28 22:53:57 MSK 2011

Following is a section from the log:
---------------------------
2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160 samples
64000 bits

2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN TTS
cepstral
2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw Codec
Activated
2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996 Speaking text:
<break strength='medium'/>We must verify your identity.
2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done speaking
text

2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done playing
file
2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated
L16 at 8000hz 1 channels 20ms
2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec
Activated L16 at 8000hz 1 channels 20ms
---------------------------

So I see when I play a file (using StreamFile / PAGD), it activates L16,
which the wiki pages says is not recommended. So should I deactivate it? If
so, how?

Now, I have not done any setting out of default / ordinary that comes with
the build. I am playing WAV file that is generated by Cepstral SWIFT command
line tool (text to WAV). The file format is "Wave PCM signed 16 bit, 8000
Hz, 128 kbps, mono".

Thank you for help so far.

Malay

On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> look at your SIP traffic and console log.
>
> enter "sofia global siptrace on" followed by "console loglevel debug"
> at the cli and make the call.
>
>
> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > I have no idea where to look for this setting.
> > This is in modules.conf.xml
> >     <!-- Codec Interfaces -->
> >     <load module="mod_spandsp"/>
> >     <!--<load module="mod_voipcodecs"/>-->
> >     <load module="mod_g723_1"/>
> >     <load module="mod_g729"/>
> >     <load module="mod_amr"/>
> >     <load module="mod_ilbc"/>
> >     <load module="mod_speex"/>
> >     <load module="mod_h26x"/>
> >     <load module="mod_siren"/>
> >     <!--<load module="mod_celt"/>-->
> >     <!--<load module="mod_opus"/>-->
> > Apart from settings I posted in my previous post, where else to look for
> > disabling LPC?
> > Malay
> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> is your inbound call using LPC? you don't want to be using LPC and
> >> expect anything to sound good that's for sure.
> >> It would not just magically say that unless something you are doing has
> >> LPC?
> >>
> >>
> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi <mthakershi at gmail.com
> >
> >> wrote:
> >> > Hello,
> >> > I updated to the latest FS version last week.
> >> > I started getting the following warning when speech / sound is played
> on
> >> > the
> >> > call.
> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC
> payload
> >> > 7
> >> > added to sdp wanting ptime 90 but it's already 20 (G7221:115:20),
> >> > disabling
> >> > ptime."
> >> > I read sections on codecs and negotiations.
> >> > Following are the settings from vars.xml (I have not changed them):
> >> >   <X-PRE-PROCESS cmd="set"
> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
> >> >   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >> > Also, there is no codec related setting in sip_profiles files
> >> > and sofia.conf.xml file.
> >> > I am playing audio files using Cepstral TTS during the call.
> >> > Can someone please help me understand these settings? And if they are
> >> > appropriate?
> >> > Thank you.
> >> > Malay
> >> > _______________________________________________
> >> > 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/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
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> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
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