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[e.schmidbauer at gmail...]

hi, great job with the celt module. im eager to use it but im not sure how. are there any sip clients that use the celt codec? or is there some other way to use the celt codec to play audio in a conference? thanks.

[brian at freeswitch.org]

Well I used mod_portaudio on my Mac with mod_celt to my FreeSWITCH box
on linux when we developed the module. Works great!

/b

On Dec 30, 2008, at 11:56 AM, e schmidbauer wrote:

[e.schmidbauer at gmail...]

Could you explain in a more detail how you set that up?

On Tue, Dec 30, 2008 at 1:06 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

http://wiki.freeswitch.org/wiki/Freeswitch_softphone

/b

On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:

[e.schmidbauer at gmail...]

i have port audio setup but when i do a 'pa call <extension>' it enters the conference using the L16 codec. is there a way to use celt codec instead of the L16?

On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

You need to use CELT between FS and another FS box, L16 is from the PA
to the Conference no need to encode it to celt and then decode it
again.. it never hits the wire.

/b

On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:

[brian at freeswitch.org]

OK here try this..

in portaudio.conf.xml


<param name="sample-rate" value="48000"/>
<param name="codec-ms" value="10"/>




in dialplan/default.xml


<extension name="sip_uri">
<condition field="destination_number" expression="^[url=sip:(.*)$]sip:(.*)$[/url]">

<action application="bridge" data="{absolute_codec_string=CELT@48000h@10i}sofia/${use_profile}/[url=sip:$1]sip:$1[/url]"/>
</condition>
</extension>


save that
then


pa call [url=sip:886@taz.bkw.org:5080]sip:886@taz.bkw.org:5080[/url]


/b





On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:

[e.schmidbauer at gmail...]

i did as your said and got some errors in the console....
after i run pa call sip:886@ww2.bwrl.org:5080 i get.....
2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing relaxxplayer->sip:886@ww2.bwrl.org:5080 in context default
2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid Profile
2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW]
2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
2008-12-30 14:45:25 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER

any suggestions?


On Tue, Dec 30, 2008 at 2:06 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

Yes actually call sip:886@taz.bkw.org:5080 I set it up specifically
for you to test

/b

On Dec 30, 2008, at 1:50 PM, e schmidbauer wrote:

[e.schmidbauer at gmail...]

tried 'pa call sip:886@taz.bkw.org:5080'

2008-12-30 15:15:42 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel portaudio/sip:886@taz.bkw.org:5080 [a220653c-d6ae-11dd-af32-1ff260d7b236]
2008-12-30 15:15:42 [NOTICE] mod_portaudio.c:1586 place_call() Channel [portaudio/sip:886@taz.bkw.org:5080] has been answered
2008-12-30 15:15:42 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->sip:886@taz.bkw.org:5080 in context default
2008-12-30 15:15:42 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid Profile
2008-12-30 15:15:42 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW]
2008-12-30 15:15:42 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
2008-12-30 15:15:42 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER
2008-12-30 15:15:42 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup portaudio/sip:886@taz.bkw.org:5080 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]

still got errors....


On Tue, Dec 30, 2008 at 3:05 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

You don't have the default config ... did you modify the dialplan to
have the propler sip_uri extension.. you only had to change one line.


On Dec 30, 2008, at 2:16 PM, e schmidbauer wrote:

[e.schmidbauer at gmail...]

success! i was able to connect to your box using the CELT codec. thanks for your help.
could you tell me how your end is configured so i can try this on my box?

On Tue, Dec 30, 2008 at 3:32 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

Its setup to answer and playback... with CELT@48000h allowed on the
sofia profile.. playing a stream from my DirecTV (XM Radio 20on20 @ 48k)

btw how did it sound?

/b

On Dec 30, 2008, at 2:43 PM, e schmidbauer wrote:

[e.schmidbauer at gmail...]

sound quality is amazing like im listening to music on my own computer. can you show me the dialplan config for the extension?

On Tue, Dec 30, 2008 at 3:47 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

<extension name="xmradio">
<condition field="destination_number" expression="^886$">
<action application="answer"/>
<action application="playback" data="local_stream://xmradio"/>
</condition>
</extension>


and local stream is just pointed at a shout cast server on my Mac that
is plugged into my DirecTV receiver

/b


On Dec 30, 2008, at 2:54 PM, e schmidbauer wrote: