[Freeswitch-users] codec error

Fri Jan 7 09:37:48 MSK 2011

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
> >>> >> >
> >>> >> > _______________________________________________
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> >>> >> >
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> >>> >> >
> >>> >>
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