[Freeswitch-users] disabling ptime warning message

Anthony Minessale anthony.minessale at gmail.com
Tue Mar 1 00:47:09 MSK 2011


You also need "sofia global siptrace on" so we can see the SIP traffic
in the log.
We have our own pastebin at http://pastebin.freeswitch.org


On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> 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:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> 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
>> >
>> >
>> > _______________________________________________
>> > 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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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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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