[Freeswitch-users] disabling ptime warning message

[Anthony Minessale]

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
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>> >> >
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>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
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>> >>
>> >> 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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>> >
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>> >
>>
>>
>>
>> --
>> 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
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>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
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>
>
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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FreeSWITCH Developer Conference
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