I am getting the error after installing latest git from freeswitch, where as it was working from old git 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder error! Regards Sam
Some more information please... G729 has patents that mean it can't be used unlicensed. FreeSWITCH either has mod_com_g729 for a full featured licensed version, or mod_g729 which operates in passthrough mode for G729-G729 calls (it needs no license since for these calls there is no encoding/decoding step, the already encoded data is just passed straight through). You will never have had mod_g729 working in the past for a transcoding call, so that won't have changed. I would say that it's either: - Your config files have changed - The endpoints are offering different codecs from before - Something changed in git about the codec negotiation Can you pastebin a debug level log of the calls, and enable siptrace? Those will show what codecs are being offered, selected and show the codec negotiation. It'd also be useful to know what your sip profile config files look like since there are several options that adjust how the negotiation is done. -Steve On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote:
I am getting the error after installing latest git from freeswitch, where as it was working from old git
2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder error!
Regards Sam
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Hi, This was working earlier and the config file have not changed , only upgraded to latest git. have attached the file codec.txt . Regards Sam On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com> wrote:
Some more information please...
G729 has patents that mean it can't be used unlicensed. FreeSWITCH either has mod_com_g729 for a full featured licensed version, or mod_g729 which operates in passthrough mode for G729-G729 calls (it needs no license since for these calls there is no encoding/decoding step, the already encoded data is just passed straight through). You will never have had mod_g729 working in the past for a transcoding call, so that won't have changed.
I would say that it's either: - Your config files have changed - The endpoints are offering different codecs from before - Something changed in git about the codec negotiation
Can you pastebin a debug level log of the calls, and enable siptrace? Those will show what codecs are being offered, selected and show the codec negotiation. It'd also be useful to know what your sip profile config files look like since there are several options that adjust how the negotiation is done.
-Steve
On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote:
I am getting the error after installing latest git from freeswitch, where as it was working from old git
2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder error!
Regards Sam
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G7222 call fails with incompatible destination because it's not enabled on the server. G729 call starts a bridge to user/7013@192.168.2.190. They only offer G711 ulaw. That means it's a G729 -> G711 call requiring transcoding. There is no way that can work using mod_g729. If you believe that that user has G729 enabled, repeat the test with sip trace enabled. 'sofia global siptrace on' That'll let you see the INVITE w/SDP sent to the user. If G729 isn't in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in the SDP and the user is only responding with G711 then it's a problem on the Cisco endpoint. -Steve On 6 January 2011 10:10, Sam <u2nsam@gmail.com> wrote:
Hi,
This was working earlier and the config file have not changed , only upgraded to latest git. have attached the file codec.txt .
Regards Sam
On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com> wrote:
Some more information please...
G729 has patents that mean it can't be used unlicensed. FreeSWITCH either has mod_com_g729 for a full featured licensed version, or mod_g729 which operates in passthrough mode for G729-G729 calls (it needs no license since for these calls there is no encoding/decoding step, the already encoded data is just passed straight through). You will never have had mod_g729 working in the past for a transcoding call, so that won't have changed.
I would say that it's either: - Your config files have changed - The endpoints are offering different codecs from before - Something changed in git about the codec negotiation
Can you pastebin a debug level log of the calls, and enable siptrace? Those will show what codecs are being offered, selected and show the codec negotiation. It'd also be useful to know what your sip profile config files look like since there are several options that adjust how the negotiation is done.
-Steve
On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote:
I am getting the error after installing latest git from freeswitch, where as it was working from old git
2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder error!
Regards Sam
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http://pastebin.freeswitch.org/14934 http://pastebin.freeswitch.org/14935 These are the paste bins where calls with g722 and G729 call fails but g729 calls rings and gets disconnected. In past it was working with both the codec on the same server as no there is no config change. Regds Sam On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre@gmail.com> wrote:
G7222 call fails with incompatible destination because it's not enabled on the server.
G729 call starts a bridge to user/7013@192.168.2.190. They only offer G711 ulaw. That means it's a G729 -> G711 call requiring transcoding. There is no way that can work using mod_g729.
If you believe that that user has G729 enabled, repeat the test with sip trace enabled. 'sofia global siptrace on'
That'll let you see the INVITE w/SDP sent to the user. If G729 isn't in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in the SDP and the user is only responding with G711 then it's a problem on the Cisco endpoint.
-Steve
On 6 January 2011 10:10, Sam <u2nsam@gmail.com> wrote:
Hi,
This was working earlier and the config file have not changed , only upgraded to latest git. have attached the file codec.txt .
Regards Sam
On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com> wrote:
Some more information please...
G729 has patents that mean it can't be used unlicensed. FreeSWITCH either has mod_com_g729 for a full featured licensed version, or mod_g729 which operates in passthrough mode for G729-G729 calls (it needs no license since for these calls there is no encoding/decoding step, the already encoded data is just passed straight through). You will never have had mod_g729 working in the past for a transcoding call, so that won't have changed.
I would say that it's either: - Your config files have changed - The endpoints are offering different codecs from before - Something changed in git about the codec negotiation
Can you pastebin a debug level log of the calls, and enable siptrace? Those will show what codecs are being offered, selected and show the codec negotiation. It'd also be useful to know what your sip profile config files look like since there are several options that adjust how the negotiation is done.
-Steve
On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote:
I am getting the error after installing latest git from freeswitch, where as it was working from old git
2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder error!
Regards Sam
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http://lists.freeswitch.org/mailman/options/freeswitch-users
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G722.2 ====== This codec is not enabled on your server. 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000] You are calling with G722.2. G722.1 is enabled, but G722.2 is not. G729 ==== It rings because Cisco returns 180 Ringing with no SDP. At that time no codec has been selected. Anything you hear will be generated by your SIP client. The codec is selected in the 200 OK w/SDP. That is the time the codec is picked. FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs: m=audio 24636 RTP/AVP 18 98 99 0 8 101 13 The Cisco only responds with PCMU: m=audio 31600 RTP/AVP 0 101 ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 ) Either your FS config has changed so that is offering more codecs than before, or the Cisco's config has changed to prefer G711 over G729. -Steve On 6 January 2011 10:45, Sam <u2nsam@gmail.com> wrote:
http://pastebin.freeswitch.org/14934 http://pastebin.freeswitch.org/14935
These are the paste bins where calls with g722 and G729 call fails but g729 calls rings and gets disconnected. In past it was working with both the codec on the same server as no there is no config change.
Regds Sam
On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre@gmail.com> wrote:
G7222 call fails with incompatible destination because it's not enabled on the server.
G729 call starts a bridge to user/7013@192.168.2.190. They only offer G711 ulaw. That means it's a G729 -> G711 call requiring transcoding. There is no way that can work using mod_g729.
If you believe that that user has G729 enabled, repeat the test with sip trace enabled. 'sofia global siptrace on'
That'll let you see the INVITE w/SDP sent to the user. If G729 isn't in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in the SDP and the user is only responding with G711 then it's a problem on the Cisco endpoint.
-Steve
On 6 January 2011 10:10, Sam <u2nsam@gmail.com> wrote:
Hi,
This was working earlier and the config file have not changed , only upgraded to latest git. have attached the file codec.txt .
Regards Sam
On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com> wrote:
Some more information please...
G729 has patents that mean it can't be used unlicensed. FreeSWITCH either has mod_com_g729 for a full featured licensed version, or mod_g729 which operates in passthrough mode for G729-G729 calls (it needs no license since for these calls there is no encoding/decoding step, the already encoded data is just passed straight through). You will never have had mod_g729 working in the past for a transcoding call, so that won't have changed.
I would say that it's either: - Your config files have changed - The endpoints are offering different codecs from before - Something changed in git about the codec negotiation
Can you pastebin a debug level log of the calls, and enable siptrace? Those will show what codecs are being offered, selected and show the codec negotiation. It'd also be useful to know what your sip profile config files look like since there are several options that adjust how the negotiation is done.
-Steve
On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote:
I am getting the error after installing latest git from freeswitch, where as it was working from old git
2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder error!
Regards Sam
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This sip profile parameter would allow the G729 call to work, providing G729 is enabled on the Cisco: <param name="disable-transcoding" value="true"/> This will mean the INVITE sent to the Cisco on the bleg only includes the codec from the aleg. The aleg is already using G729, so the Cisco will only be offered G729. It is then forced to either accept G729 or fail the call with Incompatible Destination. -Steve On 6 January 2011 10:59, Steven Ayre <steveayre@gmail.com> wrote:
G722.2 ======
This codec is not enabled on your server.
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
You are calling with G722.2.
G722.1 is enabled, but G722.2 is not.
G729 ====
It rings because Cisco returns 180 Ringing with no SDP. At that time no codec has been selected. Anything you hear will be generated by your SIP client.
The codec is selected in the 200 OK w/SDP. That is the time the codec is picked.
FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs: m=audio 24636 RTP/AVP 18 98 99 0 8 101 13 The Cisco only responds with PCMU: m=audio 31600 RTP/AVP 0 101 ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
Either your FS config has changed so that is offering more codecs than before, or the Cisco's config has changed to prefer G711 over G729.
-Steve
On 6 January 2011 10:45, Sam <u2nsam@gmail.com> wrote:
http://pastebin.freeswitch.org/14934 http://pastebin.freeswitch.org/14935
These are the paste bins where calls with g722 and G729 call fails but g729 calls rings and gets disconnected. In past it was working with both the codec on the same server as no there is no config change.
Regds Sam
On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre@gmail.com> wrote:
G7222 call fails with incompatible destination because it's not enabled on the server.
G729 call starts a bridge to user/7013@192.168.2.190. They only offer G711 ulaw. That means it's a G729 -> G711 call requiring transcoding. There is no way that can work using mod_g729.
If you believe that that user has G729 enabled, repeat the test with sip trace enabled. 'sofia global siptrace on'
That'll let you see the INVITE w/SDP sent to the user. If G729 isn't in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in the SDP and the user is only responding with G711 then it's a problem on the Cisco endpoint.
-Steve
On 6 January 2011 10:10, Sam <u2nsam@gmail.com> wrote:
Hi,
This was working earlier and the config file have not changed , only upgraded to latest git. have attached the file codec.txt .
Regards Sam
On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com> wrote:
Some more information please...
G729 has patents that mean it can't be used unlicensed. FreeSWITCH either has mod_com_g729 for a full featured licensed version, or mod_g729 which operates in passthrough mode for G729-G729 calls (it needs no license since for these calls there is no encoding/decoding step, the already encoded data is just passed straight through). You will never have had mod_g729 working in the past for a transcoding call, so that won't have changed.
I would say that it's either: - Your config files have changed - The endpoints are offering different codecs from before - Something changed in git about the codec negotiation
Can you pastebin a debug level log of the calls, and enable siptrace? Those will show what codecs are being offered, selected and show the codec negotiation. It'd also be useful to know what your sip profile config files look like since there are several options that adjust how the negotiation is done.
-Steve
On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote:
I am getting the error after installing latest git from freeswitch, where as it was working from old git
2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder error!
Regards Sam
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Where does it compare these codecs, 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G7221:115:32000:20:48000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G7221:107:16000:20:32000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[PCMU:0:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[PCMA:8:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G729:18:8000:20:8000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G7221:115:32000:20:48000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G7221:107:16000:20:32000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[PCMU:0:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[PCMA:8:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G729:18:8000:20:8000] I need to add g722 Regds Sam On Thu, Jan 6, 2011 at 4:34 PM, Steven Ayre <steveayre@gmail.com> wrote:
This sip profile parameter would allow the G729 call to work, providing G729 is enabled on the Cisco: <param name="disable-transcoding" value="true"/>
This will mean the INVITE sent to the Cisco on the bleg only includes the codec from the aleg. The aleg is already using G729, so the Cisco will only be offered G729. It is then forced to either accept G729 or fail the call with Incompatible Destination.
-Steve
On 6 January 2011 10:59, Steven Ayre <steveayre@gmail.com> wrote:
G722.2 ======
This codec is not enabled on your server.
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
You are calling with G722.2.
G722.1 is enabled, but G722.2 is not.
G729 ====
It rings because Cisco returns 180 Ringing with no SDP. At that time no codec has been selected. Anything you hear will be generated by your SIP client.
The codec is selected in the 200 OK w/SDP. That is the time the codec is picked.
FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs: m=audio 24636 RTP/AVP 18 98 99 0 8 101 13 The Cisco only responds with PCMU: m=audio 31600 RTP/AVP 0 101 ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
Either your FS config has changed so that is offering more codecs than before, or the Cisco's config has changed to prefer G711 over G729.
-Steve
On 6 January 2011 10:45, Sam <u2nsam@gmail.com> wrote:
http://pastebin.freeswitch.org/14934 http://pastebin.freeswitch.org/14935
These are the paste bins where calls with g722 and G729 call fails but g729 calls rings and gets disconnected. In past it was working with both the codec on the same server as no there is no config change.
Regds Sam
On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre@gmail.com> wrote:
G7222 call fails with incompatible destination because it's not enabled on the server.
G729 call starts a bridge to user/7013@192.168.2.190. They only offer G711 ulaw. That means it's a G729 -> G711 call requiring transcoding. There is no way that can work using mod_g729.
If you believe that that user has G729 enabled, repeat the test with sip trace enabled. 'sofia global siptrace on'
That'll let you see the INVITE w/SDP sent to the user. If G729 isn't in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in the SDP and the user is only responding with G711 then it's a problem on the Cisco endpoint.
-Steve
On 6 January 2011 10:10, Sam <u2nsam@gmail.com> wrote:
Hi,
This was working earlier and the config file have not changed , only upgraded to latest git. have attached the file codec.txt .
Regards Sam
On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com>
wrote:
Some more information please...
G729 has patents that mean it can't be used unlicensed. FreeSWITCH either has mod_com_g729 for a full featured licensed version, or mod_g729 which operates in passthrough mode for G729-G729 calls (it needs no license since for these calls there is no encoding/decoding step, the already encoded data is just passed straight through). You will never have had mod_g729 working in the past for a transcoding call, so that won't have changed.
I would say that it's either: - Your config files have changed - The endpoints are offering different codecs from before - Something changed in git about the codec negotiation
Can you pastebin a debug level log of the calls, and enable
siptrace?
Those will show what codecs are being offered, selected and show the codec negotiation. It'd also be useful to know what your sip profile config files look like since there are several options that adjust how the negotiation is done.
-Steve
On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote: > I am getting the error after installing latest git from freeswitch, > where as > it was working from old git > > > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only > usable in > passthrough mode! > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 > encoder > error! > > > Regards > Sam > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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in show codec i see ; show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc i have this in vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h ,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h ,G7221@16000h,PCMU,PCMA,G729"/> Regds Sam On Fri, Jan 7, 2011 at 12:07 PM, Sam <u2nsam@gmail.com> wrote:
Where does it compare these codecs,
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G7221:115:32000:20:48000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G7221:107:16000:20:32000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[PCMU:0:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[PCMA:8:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G729:18:8000:20:8000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G7221:115:32000:20:48000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G7221:107:16000:20:32000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[PCMU:0:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[PCMA:8:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G729:18:8000:20:8000]
I need to add g722
Regds Sam
On Thu, Jan 6, 2011 at 4:34 PM, Steven Ayre <steveayre@gmail.com> wrote:
This sip profile parameter would allow the G729 call to work, providing G729 is enabled on the Cisco: <param name="disable-transcoding" value="true"/>
This will mean the INVITE sent to the Cisco on the bleg only includes the codec from the aleg. The aleg is already using G729, so the Cisco will only be offered G729. It is then forced to either accept G729 or fail the call with Incompatible Destination.
-Steve
On 6 January 2011 10:59, Steven Ayre <steveayre@gmail.com> wrote:
G722.2 ======
This codec is not enabled on your server.
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
You are calling with G722.2.
G722.1 is enabled, but G722.2 is not.
G729 ====
It rings because Cisco returns 180 Ringing with no SDP. At that time no codec has been selected. Anything you hear will be generated by your SIP client.
The codec is selected in the 200 OK w/SDP. That is the time the codec is picked.
FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs: m=audio 24636 RTP/AVP 18 98 99 0 8 101 13 The Cisco only responds with PCMU: m=audio 31600 RTP/AVP 0 101 ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
Either your FS config has changed so that is offering more codecs than before, or the Cisco's config has changed to prefer G711 over G729.
-Steve
On 6 January 2011 10:45, Sam <u2nsam@gmail.com> wrote:
http://pastebin.freeswitch.org/14934 http://pastebin.freeswitch.org/14935
These are the paste bins where calls with g722 and G729 call fails but g729 calls rings and gets disconnected. In past it was working with both the codec on the same server as no there is no config change.
Regds Sam
On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre@gmail.com> wrote:
G7222 call fails with incompatible destination because it's not enabled on the server.
G729 call starts a bridge to user/7013@192.168.2.190. They only offer G711 ulaw. That means it's a G729 -> G711 call requiring transcoding. There is no way that can work using mod_g729.
If you believe that that user has G729 enabled, repeat the test with sip trace enabled. 'sofia global siptrace on'
That'll let you see the INVITE w/SDP sent to the user. If G729 isn't in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in the SDP and the user is only responding with G711 then it's a problem on the Cisco endpoint.
-Steve
On 6 January 2011 10:10, Sam <u2nsam@gmail.com> wrote:
Hi,
This was working earlier and the config file have not changed , only upgraded to latest git. have attached the file codec.txt .
Regards Sam
On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com>
wrote:
> > Some more information please... > > G729 has patents that mean it can't be used unlicensed. FreeSWITCH > either has mod_com_g729 for a full featured licensed version, or > mod_g729 which operates in passthrough mode for G729-G729 calls (it > needs no license since for these calls there is no encoding/decoding > step, the already encoded data is just passed straight through). You > will never have had mod_g729 working in the past for a transcoding > call, so that won't have changed. > > I would say that it's either: > - Your config files have changed > - The endpoints are offering different codecs from before > - Something changed in git about the codec negotiation > > Can you pastebin a debug level log of the calls, and enable siptrace? > Those will show what codecs are being offered, selected and show the > codec negotiation. It'd also be useful to know what your sip profile > config files look like since there are several options that adjust how > the negotiation is done. > > -Steve > > > > On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote: > > I am getting the error after installing latest git from freeswitch, > > where as > > it was working from old git > > > > > > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only > > usable in > > passthrough mode! > > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 > > encoder > > error! > > > > > > Regards > > Sam > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
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show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc You have no module loaded that gives G722. Add mod_spandsp to modules.conf.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/> That sets a variable. The actual definition of which codecs to use is on the sip profile, which could use the variable that's set above, or specify its own list. That should be fine as long as the profile uses the variable, but they'll only be used if the codec is loaded - see above. -Steve On 7 January 2011 06:58, Sam <u2nsam@gmail.com> wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/>
Regds Sam
On Fri, Jan 7, 2011 at 12:07 PM, Sam <u2nsam@gmail.com> wrote:
Where does it compare these codecs,
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G7221:115:32000:20:48000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G7221:107:16000:20:32000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[PCMU:0:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[PCMA:8:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [G722:9:8000:0:64000]/[G729:18:8000:20:8000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G7221:115:32000:20:48000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G7221:107:16000:20:32000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[PCMU:0:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[PCMA:8:8000:20:64000] 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare [telephone-event:127:8000:0:0]/[G729:18:8000:20:8000]
I need to add g722
Regds Sam
On Thu, Jan 6, 2011 at 4:34 PM, Steven Ayre <steveayre@gmail.com> wrote:
This sip profile parameter would allow the G729 call to work, providing G729 is enabled on the Cisco: <param name="disable-transcoding" value="true"/>
This will mean the INVITE sent to the Cisco on the bleg only includes the codec from the aleg. The aleg is already using G729, so the Cisco will only be offered G729. It is then forced to either accept G729 or fail the call with Incompatible Destination.
-Steve
On 6 January 2011 10:59, Steven Ayre <steveayre@gmail.com> wrote:
G722.2 ======
This codec is not enabled on your server.
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000] 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
You are calling with G722.2.
G722.1 is enabled, but G722.2 is not.
G729 ====
It rings because Cisco returns 180 Ringing with no SDP. At that time no codec has been selected. Anything you hear will be generated by your SIP client.
The codec is selected in the 200 OK w/SDP. That is the time the codec is picked.
FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs: m=audio 24636 RTP/AVP 18 98 99 0 8 101 13 The Cisco only responds with PCMU: m=audio 31600 RTP/AVP 0 101 ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
Either your FS config has changed so that is offering more codecs than before, or the Cisco's config has changed to prefer G711 over G729.
-Steve
On 6 January 2011 10:45, Sam <u2nsam@gmail.com> wrote:
http://pastebin.freeswitch.org/14934 http://pastebin.freeswitch.org/14935
These are the paste bins where calls with g722 and G729 call fails but g729 calls rings and gets disconnected. In past it was working with both the codec on the same server as no there is no config change.
Regds Sam
On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre@gmail.com> wrote:
G7222 call fails with incompatible destination because it's not enabled on the server.
G729 call starts a bridge to user/7013@192.168.2.190. They only offer G711 ulaw. That means it's a G729 -> G711 call requiring transcoding. There is no way that can work using mod_g729.
If you believe that that user has G729 enabled, repeat the test with sip trace enabled. 'sofia global siptrace on'
That'll let you see the INVITE w/SDP sent to the user. If G729 isn't in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in the SDP and the user is only responding with G711 then it's a problem on the Cisco endpoint.
-Steve
On 6 January 2011 10:10, Sam <u2nsam@gmail.com> wrote: > Hi, > > This was working earlier and the config file have not changed , > only > upgraded to latest git. > have attached the file codec.txt . > > > > Regards > Sam > > > > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre@gmail.com> > wrote: >> >> Some more information please... >> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH >> either has mod_com_g729 for a full featured licensed version, or >> mod_g729 which operates in passthrough mode for G729-G729 calls >> (it >> needs no license since for these calls there is no >> encoding/decoding >> step, the already encoded data is just passed straight through). >> You >> will never have had mod_g729 working in the past for a transcoding >> call, so that won't have changed. >> >> I would say that it's either: >> - Your config files have changed >> - The endpoints are offering different codecs from before >> - Something changed in git about the codec negotiation >> >> Can you pastebin a debug level log of the calls, and enable >> siptrace? >> Those will show what codecs are being offered, selected and show >> the >> codec negotiation. It'd also be useful to know what your sip >> profile >> config files look like since there are several options that adjust >> how >> the negotiation is done. >> >> -Steve >> >> >> >> On 6 January 2011 09:43, Sam <u2nsam@gmail.com> wrote: >> > I am getting the error after installing latest git from >> > freeswitch, >> > where as >> > it was working from old git >> > >> > >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is >> > only >> > usable in >> > passthrough mode! >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec >> > G.729 >> > encoder >> > error! >> > >> > >> > Regards >> > Sam >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users@lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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For one you have G7222 ... secondly you do not have mod_spandsp loaded. /b On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/>
Regds Sam
Which module should i load for g7222 ? Regds Sam On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote:
For one you have G7222 ... secondly you do not have mod_spandsp loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h ,PCMU,PCMA,G729"/>
Regds Sam
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Sam, Brian pointed out 2 mistakes:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/>
Here, you have G7222. It is G722. And for G722, you need mod_spandsp, as he said in his last reply. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/01/2011 à 06:33, Sam a écrit :
Which module should i load for g7222 ?
Regds Sam
On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote: For one you have G7222 ... secondly you do not have mod_spandsp loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/>
Regds Sam
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I have now , <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,G729"/> The G722 started working, but i wanted to know for G722.2 which module works? Regds Sam On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone@ipeva.fr>wrote:
Sam,
Brian pointed out 2 mistakes:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G7222,PCMU,PCMA,G729"/>
Here, you have G7222. It is G722.
And for G722, you need mod_spandsp, as he said in his last reply.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
*Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. **IPeva** décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.* * *
Le 10/01/2011 à 06:33, Sam a écrit :
Which module should i load for g7222 ?
Regds Sam
On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote:
For one you have G7222 ... secondly you do not have mod_spandsp loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h ,PCMU,PCMA,G729"/>
Regds Sam
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G722.1 is in mod_siren David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/01/2011 à 10:28, Sam a écrit :
I have now ,
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,G729"/>
The G722 started working, but i wanted to know for G722.2 which module works?
Regds Sam
On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone@ipeva.fr> wrote: Sam,
Brian pointed out 2 mistakes:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/>
Here, you have G7222. It is G722.
And for G722, you need mod_spandsp, as he said in his last reply.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 06:33, Sam a écrit :
Which module should i load for g7222 ?
Regds Sam
On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote: For one you have G7222 ... secondly you do not have mod_spandsp loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/>
Regds Sam
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David, G.722 G.722.1 and G.722.2 codecs are 3 different codecs. Sam, G.722 - mod_spandsp G.722.1 - mod_siren G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it requires patent licenses Regards, -Steve On 10 January 2011 09:39, David Ponzone <david.ponzone@ipeva.fr> wrote:
G722.1 is in mod_siren David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 10:28, Sam a écrit :
I have now ,
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,G729"/>
The G722 started working, but i wanted to know for G722.2 which module works?
Regds Sam
On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
Sam, Brian pointed out 2 mistakes:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/>
Here, you have G7222. It is G722. And for G722, you need mod_spandsp, as he said in his last reply. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 06:33, Sam a écrit :
Which module should i load for g7222 ?
Regds Sam
On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote:
For one you have G7222 ... secondly you do not have mod_spandsp loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/>
Regds Sam
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G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it requires patent licenses
Correction: there is mod_amrwb However, because of the patent licences it is passthrough-only so it may not suit your needs. Regards -Steve On 10 January 2011 11:38, Steven Ayre <steveayre@gmail.com> wrote:
David, G.722 G.722.1 and G.722.2 codecs are 3 different codecs.
Sam, G.722 - mod_spandsp G.722.1 - mod_siren G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it requires patent licenses
Regards, -Steve
On 10 January 2011 09:39, David Ponzone <david.ponzone@ipeva.fr> wrote:
G722.1 is in mod_siren David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 10:28, Sam a écrit :
I have now ,
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,G729"/>
The G722 started working, but i wanted to know for G722.2 which module works?
Regds Sam
On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
Sam, Brian pointed out 2 mistakes:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/>
Here, you have G7222. It is G722. And for G722, you need mod_spandsp, as he said in his last reply. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 06:33, Sam a écrit :
Which module should i load for g7222 ?
Regds Sam
On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote:
For one you have G7222 ... secondly you do not have mod_spandsp loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/>
Regds Sam
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thanks Steve ! Regds Sam On Mon, Jan 10, 2011 at 5:11 PM, Steven Ayre <steveayre@gmail.com> wrote:
G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it requires patent licenses
Correction: there is mod_amrwb
However, because of the patent licences it is passthrough-only so it may not suit your needs.
Regards -Steve
On 10 January 2011 11:38, Steven Ayre <steveayre@gmail.com> wrote:
David, G.722 G.722.1 and G.722.2 codecs are 3 different codecs.
Sam, G.722 - mod_spandsp G.722.1 - mod_siren G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it requires patent licenses
Regards, -Steve
On 10 January 2011 09:39, David Ponzone <david.ponzone@ipeva.fr> wrote:
G722.1 is in mod_siren David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 10:28, Sam a écrit :
I have now ,
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221@32000h,G7221@16000h ,G722,PCMU,PCMA,G729"/>
The G722 started working, but i wanted to know for G722.2 which module works?
Regds Sam
On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
Sam, Brian pointed out 2 mistakes:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h
,G7222,PCMU,PCMA,G729"/>
Here, you have G7222. It is G722. And for G722, you need mod_spandsp, as he said in his last reply. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 06:33, Sam a écrit :
Which module should i load for g7222 ?
Regds Sam
On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote:
For one you have G7222 ... secondly you do not have mod_spandsp
loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h
,G7222,PCMU,PCMA,G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h ,PCMU,PCMA,G729"/>
Regds Sam
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Sure :) David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/01/2011 à 12:38, Steven Ayre a écrit :
David, G.722 G.722.1 and G.722.2 codecs are 3 different codecs.
Sam, G.722 - mod_spandsp G.722.1 - mod_siren G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it requires patent licenses
Regards, -Steve
On 10 January 2011 09:39, David Ponzone <david.ponzone@ipeva.fr> wrote:
G722.1 is in mod_siren David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 10:28, Sam a écrit :
I have now ,
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,G729"/>
The G722 started working, but i wanted to know for G722.2 which module works?
Regds Sam
On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
Sam, Brian pointed out 2 mistakes:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/>
Here, you have G7222. It is G722. And for G722, you need mod_spandsp, as he said in his last reply. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 10/01/2011 à 06:33, Sam a écrit :
Which module should i load for g7222 ?
Regds Sam
On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian@freeswitch.org> wrote:
For one you have G7222 ... secondly you do not have mod_spandsp loaded.
/b
On Jan 7, 2011, at 12:58 AM, Sam wrote:
in show codec i see ;
show codecs type,name,ikey codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.723.1 6.3k,mod_g723_1 codec,G.729,mod_g729 codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc
i have this in vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/>
Regds Sam
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participants (4)
-
Brian West
-
David Ponzone
-
Sam
-
Steven Ayre