<a href="http://pastebin.com/CCwmAfZK">http://pastebin.com/CCwmAfZK</a><div><br></div><div><a href="http://pastebin.com/CCwmAfZK"></a>I noticed that LPC warning only comes for outbound calls I initiate using "Originate" API.</div>
<div><br></div><div>Malay<br><br><div class="gmail_quote">On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">That's L16 not LPC, you need L16 to play files.<br>
Why don't you just put the whole log of the call on pastebin.<br>
<div><div></div><div class="h5"><br>
<br>
On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>> wrote:<br>
> Following is a section from the log:<br>
> ---------------------------<br>
> 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec<br>
> sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160 samples<br>
> 64000 bits<br>
> 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN TTS<br>
> cepstral<br>
> 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw Codec<br>
> Activated<br>
> 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996 Speaking text:<br>
> <break strength='medium'/>We must verify your identity.<br>
> 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done speaking<br>
> text<br>
> 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec<br>
> Activated L16@8000hz 1 channels 20ms<br>
> 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done playing<br>
> file<br>
> 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated<br>
> L16@8000hz 1 channels 20ms<br>
> 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec<br>
> Activated L16@8000hz 1 channels 20ms<br>
> ---------------------------<br>
> So I see when I play a file (using StreamFile / PAGD), it activates L16,<br>
> which the wiki pages says is not recommended. So should I deactivate it? If<br>
> so, how?<br>
> Now, I have not done any setting out of default / ordinary that comes with<br>
> the build. I am playing WAV file that is generated by Cepstral SWIFT command<br>
> line tool (text to WAV). The file format is "Wave PCM signed 16 bit, 8000<br>
> Hz, 128 kbps, mono".<br>
> Thank you for help so far.<br>
> Malay<br>
> On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale<br>
> <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
>><br>
>> look at your SIP traffic and console log.<br>
>><br>
>> enter "sofia global siptrace on" followed by "console loglevel debug"<br>
>> at the cli and make the call.<br>
>><br>
>><br>
>> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi <<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>><br>
>> wrote:<br>
>> > I have no idea where to look for this setting.<br>
>> > This is in modules.conf.xml<br>
>> > <!-- Codec Interfaces --><br>
>> > <load module="mod_spandsp"/><br>
>> > <!--<load module="mod_voipcodecs"/>--><br>
>> > <load module="mod_g723_1"/><br>
>> > <load module="mod_g729"/><br>
>> > <load module="mod_amr"/><br>
>> > <load module="mod_ilbc"/><br>
>> > <load module="mod_speex"/><br>
>> > <load module="mod_h26x"/><br>
>> > <load module="mod_siren"/><br>
>> > <!--<load module="mod_celt"/>--><br>
>> > <!--<load module="mod_opus"/>--><br>
>> > Apart from settings I posted in my previous post, where else to look for<br>
>> > disabling LPC?<br>
>> > Malay<br>
>> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale<br>
>> > <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
>> >><br>
>> >> is your inbound call using LPC? you don't want to be using LPC and<br>
>> >> expect anything to sound good that's for sure.<br>
>> >> It would not just magically say that unless something you are doing has<br>
>> >> LPC?<br>
>> >><br>
>> >><br>
>> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi<br>
>> >> <<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>><br>
>> >> wrote:<br>
>> >> > Hello,<br>
>> >> > I updated to the latest FS version last week.<br>
>> >> > I started getting the following warning when speech / sound is played<br>
>> >> > on<br>
>> >> > the<br>
>> >> > call.<br>
>> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC<br>
>> >> > payload<br>
>> >> > 7<br>
>> >> > added to sdp wanting ptime 90 but it's already 20 (G7221:115:20),<br>
>> >> > disabling<br>
>> >> > ptime."<br>
>> >> > I read sections on codecs and negotiations.<br>
>> >> > Following are the settings from vars.xml (I have not changed them):<br>
>> >> > <X-PRE-PROCESS cmd="set"<br>
>> >> ><br>
>> >> > data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br>
>> >> > <X-PRE-PROCESS cmd="set"<br>
>> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/><br>
>> >> > Also, there is no codec related setting in sip_profiles files<br>
>> >> > and sofia.conf.xml file.<br>
>> >> > I am playing audio files using Cepstral TTS during the call.<br>
>> >> > Can someone please help me understand these settings? And if they are<br>
>> >> > appropriate?<br>
>> >> > Thank you.<br>
>> >> > Malay<br>
>> >> > _______________________________________________<br>
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>> >> ><br>
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>> >> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> ><br>
>> >> ><br>
>> >><br>
>> >><br>
>> >><br>
>> >> --<br>
>> >> Anthony Minessale II<br>
>> >><br>
>> >> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
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>> >><br>
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>> >><br>
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>> ><br>
>> ><br>
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>> ><br>
>> ><br>
>><br>
>><br>
>><br>
>> --<br>
>> Anthony Minessale II<br>
>><br>
>> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
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>><br>
>> AIM: anthm<br>
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>><br>
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><br>
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><br>
><br>
<br>
<br>
<br>
--<br>
Anthony Minessale II<br>
<br>
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<br>
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</div></div></blockquote></div><br></div>