[Freeswitch-users] codec error

Sam u2nsam at gmail.com
Fri Jan 7 09:58:18 MSK 2011 

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 at 32000h,G7221 at 16000h
,G7222,PCMU,PCMA,G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221 at 32000h
,G7221 at 16000h,PCMU,PCMA,G729"/>



Regds
Sam



On Fri, Jan 7, 2011 at 12:07 PM, Sam <u2nsam at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org
>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >
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>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> > http://www.freeswitch.org
>> >>> >> >
>> >>> >> >
>> >>> >>
>> >>> >> _______________________________________________
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>> >>> >
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>> >>> >
>> >>>
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>
>
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