[Freeswitch-users] disabling ptime warning message

Malay Thakershi mthakershi at gmail.com
Tue Mar 1 00:22:04 MSK 2011


http://pastebin.com/CCwmAfZK

<http://pastebin.com/CCwmAfZK>I noticed that LPC warning only comes for
outbound calls I initiate using "Originate" API.

Malay

On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> That's L16 not LPC, you need L16 to play files.
> Why don't you just put the whole log of the call on pastebin.
>
>
> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > 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
> >> > 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:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
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> > 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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