Following is a section from the log:<div>---------------------------</div><div><div>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</div>
<div><br></div><div>2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN TTS cepstral</div><div>2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw Codec Activated</div><div>2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996 Speaking text: <break strength='medium'/>We must verify your identity.</div>
<div>2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done speaking text</div><div><br></div><div>2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@8000hz 1 channels 20ms</div>
<div>2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done playing file</div><div>2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated L16@8000hz 1 channels 20ms</div><div>2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@8000hz 1 channels 20ms</div>
<div>---------------------------</div><div><br></div><div>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?</div>
<div><br></div><div>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".</div>
<div><br></div><div>Thank you for help so far.</div><div><br></div><div>Malay</div><br><div class="gmail_quote">On Mon, Feb 28, 2011 at 12:50 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;">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>
<div><div></div><div class="h5"><br>
<br>
On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi <<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>> 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 <<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 on<br>
>> > the<br>
>> > call.<br>
>> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC 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>
>> > data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br>
>> > <X-PRE-PROCESS cmd="set" 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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>> --<br>
>> Anthony Minessale II<br>
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Anthony Minessale II<br>
<br>
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</div></div></blockquote></div><br></div>