[Freeswitch-users] Possible to disable core codecs?
Malay Thakershi
mthakershi at gmail.com
Tue Apr 10 22:00:40 MSD 2012
Are these licenses supported on windows? If not, how can I use G729 on it?
One license is 1 encoder/decoder per channel. So if I make 30 concurrent
calls, do I need 30 licenses?
Is the $10 fee per month or one-time?
If I can't use G729 at all, what could be my options to save bandwidth +
get acceptable audio quality?
Thanks for responses.
Malay
On Tue, Apr 10, 2012 at 12:34 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:
> TTS needs a real license for G729 - please check here how to purchase it:
> http://www.freeswitch.org/node/235
>
> The free G729 modules is for passthrough only, if you generate audio on
> the FS instance, it will require encoder/decoder licenses.
>
> /Peter
>
> ________________________________
> Från: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] för Malay Thakershi [
> mthakershi at gmail.com]
> Skickat: den 10 april 2012 07:22
> Till: FreeSWITCH Users Help
> Ämne: Re: [Freeswitch-users] Possible to disable core codecs?
>
> Upgraded FS to latest version before coming back for help.
>
> Now, I get error saying:
> [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
> [ERR] switch_core_io.c:1081 Codec G.729 encoder error!
>
> I think what is happening is, it agrees to use G.729 codec but as soon as
> TTS is opened, there is an error. I read in documentation that mod_g729 is
> for free codec.
>
> Please help.
>
> Here is the debug trace of the call:
> ----------------------------------
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5572 Remote SDP:
> v=0
> o=root 29993 29993 IN IP4 IP1
> s=session
> c=IN IP4 IP1
> t=0 0
> m=audio 15582 RTP/AVP 0 8 3 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G72
> 21:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G72
> 21:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[G722
> 1:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[G72
> 21:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [telephone-event:101:8000:2
> 0:0]/[G7221:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[AMR
> :96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[AMR
> :96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[AMR:
> 96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[AMR
> :96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [telephone-event:101:8000:2
> 0:0]/[AMR:96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[SPE
> EX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[SPE
> EX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[SPEE
> X:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[SPE
> EX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [telephone-event:101:8000:2
> 0:0]/[SPEEX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G72
> 9:18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G72
> 9:18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[G729
> :18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[G72
> 9:18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:3006 Set Codec
> sofia/sipinterface_1/Phone2 at x.24
> 1.99.201 G729/8000 20 ms 160 samples 8000 bits
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_codec.c:111
> sofia/sipinterface_1/Phone2 at x.241.99
> .201 Original read codec set to G729:18
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_NEW
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:380
> (sofia/sipinterface_1/Phone2@
> IP1) State NEW
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5786
> (sofia/sipinterface_1/Phone2 at IP1) Stat
> e Change CS_NEW -> CS_INIT
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_INIT
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
> (sofia/sipinterface_1/Phone2@
> IP1) State INIT
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:85
> sofia/sipinterface_1/Phone2 at IP1 SOFI
> A INIT
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125
> (sofia/sipinterface_1/Phone2 at IP1) S
> tate Change CS_INIT -> CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
> (sofia/sipinterface_1/Phone2@
> IP1) State INIT going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_channel.c:1886
> (sofia/sipinterface_1/Phone2 at x.241.99.
> 201) Callstate Change DOWN -> RINGING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard ROUTING
>
> 2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
> +1Phone2 <Phone2>->Phone1 in context inbound
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing
> [inbound->vitel-inbound] continue=fa
> lse
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [vitel-inbound]
> destination_num
> ber(8774542559) =~ // break=on-false
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action transfer(1056 XML
> default)
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
> (sofia/sipinterface_1/Phone2@
> IP1) State Change CS_ROUTING -> CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
> (sofia/sipinterface_1/Phone2@
> IP1) State EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard EXECUTE
> EXECUTE sofia/sipinterface_1/Phone2 at IP1 transfer(1056 XML default)
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711
> (sofia/sipinterface_1/Phone2 at IP1)
> State Change CS_EXECUTE -> CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer
> sofia/sipinterface_1/Phone2 at x.24
> 1.99.201 to XML[1056 at default]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
> (sofia/sipinterface_1/Phone2@
> IP1) State EXECUTE going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard ROUTING
>
> 2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
> +1Phone2 <Phone2>->1056
> in context default
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing
> [default->CHPhoneAsmtDev] continue=f
> alse
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [CHPhoneAsmtDev]
> destination_nu
> mber(1056) =~ /^105\d$/ break=on-false
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action sleep(1000)
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action managed(clsAsmtApp)
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
> (sofia/sipinterface_1/Phone2@
> IP1) State Change CS_ROUTING -> CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
> (sofia/sipinterface_1/Phone2@
> IP1) State EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard EXECUTE
> EXECUTE sofia/sipinterface_1/Phone2 at IP1 sleep(1000)
> EXECUTE sofia/sipinterface_1/Phone2 at IP1 managed(clsAsmtApp)
>
> 2012-04-10 00:13:33.980239 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed:
> attempting to run applica
> tion 'clsAsmtApp'.
>
> 2012-04-10 00:13:34.180422 [DEBUG] switch_cpp.cpp:1172
> CoreSession::seHangupHook, hangup_func: 00000
> 000
>
> 2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:1227 Not an outbound
> call.
>
> 2012-04-10 00:13:34.780969 [INFO] switch_cpp.cpp:1227 caller_id_number:
> Phone2
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3258 AUDIO RTP
> [sofia/sipinterface_1/Phone2 at x.2
> 41.99.201] 10.25.20.202 port 16542 -> IP1 port 15582 codec: 18 ms: 20
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_rtp.c:1669 Not using a timer
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send
> payload to 101
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive
> payload to 101
>
> 2012-04-10 00:13:34.780969 [DEBUG] mod_sofia.c:754 Local SDP
> sofia/sipinterface_1/Phone2 at x.241.
> 99.201:
> v=0
> o=FreeSWITCH 1334018272 1334018273 IN IP4 64.22.232.56
> s=FreeSWITCH
> c=IN IP4 64.22.232.56
> t=0 0
> m=audio 16542 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:731 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244
> (sofia/sipinterface_1/Phone2 at x.241.99.
> 201) Callstate Change RINGING -> ACTIVE
>
> 2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:599 Channel
> [sofia/sipinterface_1/Phone2 at x.2
> 41.99.201] has been answered
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel
> sofia/sipinterface_1/Phone2 at x.241.99.20
> 1 entering state [completed][200]
>
> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:35.420577 [INFO] switch_cpp.cpp:1227 OK. Connected to
> customer DB.
>
> 2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS
> flite
>
> 2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
> Activated
>
> 2012-04-10 00:13:35.460613 [DEBUG] switch_ivr_play_say.c:2160 Speaking
> text: <break strength='medium
> '/>Hello.<break strength='medium'/>Welcome
>
> 2012-04-10 00:13:35.481120 [DEBUG] sofia.c:5561 Channel
> sofia/sipinterface_1/Phone2 at x.241.99.20
> 1 entering state [ready][200]
>
> 2012-04-10 00:13:35.481120 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
>
> 2012-04-10 00:13:35.481120 [ERR] switch_core_io.c:1081 Codec G.729 encoder
> error!
>
> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2354 done
> speaking text
>
> 2012-04-10 00:13:35.500650 [NOTICE] switch_cpp.cpp:1227
> ab127cc9-a729-4ea8-b4e0-6863f3ab243f-Inside
> clsAsmtApp.Run (args ''); HookState is CS_EXECUTE.
>
> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS
> flite
>
> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
> Activated
>
> 2012-04-10 00:13:35.520180 [DEBUG] switch_ivr_play_say.c:2160 Speaking
> text: <break strength='medium
> '/>We must verify your identity.
>
> 2012-04-10 00:13:35.540686 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
>
> 2012-04-10 00:13:35.540686 [ERR] switch_core_io.c:1081 Codec G.729 encoder
> error!
>
> 2012-04-10 00:13:35.540686 [DEBUG] switch_ivr_play_say.c:2354 done
> speaking text
>
> 2012-04-10 00:13:35.560216 [DEBUG] switch_ivr_play_say.c:1306 Codec
> Activated L16 at 8000hz 1 channels
> 20ms
>
> 2012-04-10 00:13:35.560216 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
>
> 2012-04-10 00:13:35.560216 [ERR] switch_core_io.c:1081 Codec G.729 encoder
> error!
> ----------------------
>
> On Thu, Apr 5, 2012 at 1:58 PM, Avi Marcus <avi at avimarcus.net<mailto:
> avi at avimarcus.net>> wrote:
> Try absolute_codec_string<
> http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
>
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>
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