how to use celt codec
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.
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:
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.
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> wrote:
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:
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.
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http://wiki.freeswitch.org/wiki/Freeswitch_softphone /b On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
Could you explain in a more detail how you set that up?
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> wrote:
http://wiki.freeswitch.org/wiki/Freeswitch_softphone
/b
On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
Could you explain in a more detail how you set that up?
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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:
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?
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="^sip:(.*)$"> <action application="bridge" data="{absolute_codec_string=CELT@48000h@10i}sofia/${use_profile}/sip: $1"/> </condition> </extension> save that then pa call sip:886@taz.bkw.org:5080 /b On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:
i have port audio setup but when i do a 'pa call <extension>' it enters the conference using the L16 codec. is there a way to use celt codec instead of the L16?
On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian@freeswitch.org> wrote: http://wiki.freeswitch.org/wiki/Freeswitch_softphone
/b
On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
Could you explain in a more detail how you set that up?
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i did as your said and got some errors in the console.... after i run pa call sip:886@ww2.bwrl.org:5080 i get..... 2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing relaxxplayer->sip:886@ww2.bwrl.org:5080 in context default 2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid Profile 2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER] 2008-12-30 14:45:25 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER any suggestions? On Tue, Dec 30, 2008 at 2:06 PM, Brian West <brian@freeswitch.org> wrote:
OK here try this.. in portaudio.conf.xml
<param name="sample-rate" value="48000"/>
<param name="codec-ms" value="10"/>
in dialplan/default.xml
<extension name="sip_uri">
<condition field="destination_number" expression="^sip:(.*)$">
<action application="bridge" data="{absolute_codec_string=CELT@48000h@10i}sofia/${use_profile}/sip:$1"/>
</condition>
</extension>
save that then
pa call sip:886@taz.bkw.org:5080
/b
On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:
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> wrote:
http://wiki.freeswitch.org/wiki/Freeswitch_softphone
/b
On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
Could you explain in a more detail how you set that up?
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Yes actually call sip:886@taz.bkw.org:5080 I set it up specifically for you to test :P /b On Dec 30, 2008, at 1:50 PM, e schmidbauer wrote:
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?
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> wrote:
Yes actually call sip:886@taz.bkw.org:5080 I set it up specifically for you to test :P
/b
On Dec 30, 2008, at 1:50 PM, e schmidbauer wrote:
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?
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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:
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]
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> wrote:
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:
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]
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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:
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?
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> wrote:
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:
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?
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<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:
sound quality is amazing like im listening to music on my own computer. can you show me the dialplan config for the extension?
participants (2)
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Brian West
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e schmidbauer