in show codec i see ;<br><br>show codecs<br>type,name,ikey<br>codec,AMR,mod_amr<br>codec,G.711 alaw,CORE_PCM_MODULE<br>codec,G.711 ulaw,CORE_PCM_MODULE<br>codec,G.723.1 6.3k,mod_g723_1<br>codec,G.729,mod_g729<br>codec,H.261 Video (passthru),mod_h26x<br>
codec,H.263 Video (passthru),mod_h26x<br>codec,H.263+ Video (passthru),mod_h26x<br>codec,H.263++ Video (passthru),mod_h26x<br>codec,H.264 Video (passthru),mod_h26x<br>codec,PROXY PASS-THROUGH,CORE_PCM_MODULE<br>codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE<br>
codec,Polycom(R) G722.1/G722.1C,mod_siren<br>codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE<br>codec,Speex,mod_speex<br>codec,iLBC,mod_ilbc<br><br>i have this in vars.xml<br><br><X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G7222,PCMU,PCMA,G729"/><br>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221@32000h,G7221@16000h,PCMU,PCMA,G729"/><br><br><br><br>Regds<br>Sam<br><br><br><br><div class="gmail_quote">On Fri, Jan 7, 2011 at 12:07 PM, Sam <span dir="ltr"><<a href="mailto:u2nsam@gmail.com">u2nsam@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Where does it compare these codecs,<br><br>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]<br>
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]<br>
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]<br>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]<br>
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]<br>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]<br>
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]<br>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]<br>
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]<br>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]<br>
<br><br>I need to add g722<br><br>Regds<br><font color="#888888">Sam</font><div><div></div><div class="h5"><br><br><div class="gmail_quote">On Thu, Jan 6, 2011 at 4:34 PM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">This sip profile parameter would allow the G729 call to work,<br>
providing G729 is enabled on the Cisco:<br>
<param name="disable-transcoding" value="true"/><br>
<br>
This will mean the INVITE sent to the Cisco on the bleg only includes<br>
the codec from the aleg. The aleg is already using G729, so the Cisco<br>
will only be offered G729. It is then forced to either accept G729 or<br>
fail the call with Incompatible Destination.<br>
<br>
-Steve<br>
<div><div></div><div><br>
<br>
On 6 January 2011 10:59, Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>> wrote:<br>
> G722.2<br>
> ======<br>
><br>
> This codec is not enabled on your server.<br>
><br>
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec<br>
> Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000]<br>
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec<br>
> Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000]<br>
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec<br>
> Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000]<br>
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec<br>
> Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000]<br>
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec<br>
> Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]<br>
><br>
> You are calling with G722.2.<br>
><br>
> G722.1 is enabled, but G722.2 is not.<br>
><br>
> G729<br>
> ====<br>
><br>
> It rings because Cisco returns 180 Ringing with no SDP. At that time<br>
> no codec has been selected. Anything you hear will be generated by<br>
> your SIP client.<br>
><br>
> The codec is selected in the 200 OK w/SDP. That is the time the codec is picked.<br>
><br>
> FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs:<br>
> m=audio 24636 RTP/AVP 18 98 99 0 8 101 13<br>
> The Cisco only responds with PCMU:<br>
> m=audio 31600 RTP/AVP 0 101<br>
> ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )<br>
><br>
> Either your FS config has changed so that is offering more codecs than<br>
> before, or the Cisco's config has changed to prefer G711 over G729.<br>
><br>
> -Steve<br>
><br>
><br>
> On 6 January 2011 10:45, Sam <<a href="mailto:u2nsam@gmail.com" target="_blank">u2nsam@gmail.com</a>> wrote:<br>
>> <a href="http://pastebin.freeswitch.org/14934" target="_blank">http://pastebin.freeswitch.org/14934</a><br>
>> <a href="http://pastebin.freeswitch.org/14935" target="_blank">http://pastebin.freeswitch.org/14935</a><br>
>><br>
>> These are the paste bins where calls with g722 and G729 call fails but g729<br>
>> calls rings and gets disconnected.<br>
>> In past it was working with both the codec on the same server as no there is<br>
>> no config change.<br>
>><br>
>><br>
>> Regds<br>
>> Sam<br>
>><br>
>><br>
>><br>
>> On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>> wrote:<br>
>>><br>
>>> G7222 call fails with incompatible destination because it's not<br>
>>> enabled on the server.<br>
>>><br>
>>> G729 call starts a bridge to user/<a href="mailto:7013@192.168.2.190" target="_blank">7013@192.168.2.190</a>. They only offer<br>
>>> G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.<br>
>>> There is no way that can work using mod_g729.<br>
>>><br>
>>> If you believe that that user has G729 enabled, repeat the test with<br>
>>> sip trace enabled.<br>
>>> 'sofia global siptrace on'<br>
>>><br>
>>> That'll let you see the INVITE w/SDP sent to the user. If G729 isn't<br>
>>> in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in<br>
>>> the SDP and the user is only responding with G711 then it's a problem<br>
>>> on the Cisco endpoint.<br>
>>><br>
>>> -Steve<br>
>>><br>
>>><br>
>>><br>
>>> On 6 January 2011 10:10, Sam <<a href="mailto:u2nsam@gmail.com" target="_blank">u2nsam@gmail.com</a>> wrote:<br>
>>> > Hi,<br>
>>> ><br>
>>> > This was working earlier and the config file have not changed , only<br>
>>> > upgraded to latest git.<br>
>>> > have attached the file codec.txt .<br>
>>> ><br>
>>> ><br>
>>> ><br>
>>> > Regards<br>
>>> > Sam<br>
>>> ><br>
>>> ><br>
>>> ><br>
>>> > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>> wrote:<br>
>>> >><br>
>>> >> Some more information please...<br>
>>> >><br>
>>> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH<br>
>>> >> either has mod_com_g729 for a full featured licensed version, or<br>
>>> >> mod_g729 which operates in passthrough mode for G729-G729 calls (it<br>
>>> >> needs no license since for these calls there is no encoding/decoding<br>
>>> >> step, the already encoded data is just passed straight through). You<br>
>>> >> will never have had mod_g729 working in the past for a transcoding<br>
>>> >> call, so that won't have changed.<br>
>>> >><br>
>>> >> I would say that it's either:<br>
>>> >> - Your config files have changed<br>
>>> >> - The endpoints are offering different codecs from before<br>
>>> >> - Something changed in git about the codec negotiation<br>
>>> >><br>
>>> >> Can you pastebin a debug level log of the calls, and enable siptrace?<br>
>>> >> Those will show what codecs are being offered, selected and show the<br>
>>> >> codec negotiation. It'd also be useful to know what your sip profile<br>
>>> >> config files look like since there are several options that adjust how<br>
>>> >> the negotiation is done.<br>
>>> >><br>
>>> >> -Steve<br>
>>> >><br>
>>> >><br>
>>> >><br>
>>> >> On 6 January 2011 09:43, Sam <<a href="mailto:u2nsam@gmail.com" target="_blank">u2nsam@gmail.com</a>> wrote:<br>
>>> >> > I am getting the error after installing latest git from freeswitch,<br>
>>> >> > where as<br>
>>> >> > it was working from old git<br>
>>> >> ><br>
>>> >> ><br>
>>> >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only<br>
>>> >> > usable in<br>
>>> >> > passthrough mode!<br>
>>> >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729<br>
>>> >> > encoder<br>
>>> >> > error!<br>
>>> >> ><br>
>>> >> ><br>
>>> >> > Regards<br>
>>> >> > Sam<br>
>>> >> ><br>
>>> >> > _______________________________________________<br>
>>> >> > FreeSWITCH-users mailing list<br>
>>> >> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> >> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>>> >> ><br>
>>> >> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>>> >> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>> >> ><br>
>>> >> ><br>
>>> >><br>
>>> >> _______________________________________________<br>
>>> >> FreeSWITCH-users mailing list<br>
>>> >> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> >> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>>> >><br>
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>>> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>> ><br>
>>> ><br>
>>> > _______________________________________________<br>
>>> > FreeSWITCH-users mailing list<br>
>>> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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>>> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>> ><br>
>>> ><br>
>>><br>
>>> _______________________________________________<br>
>>> FreeSWITCH-users mailing list<br>
>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>><br>
>><br>
>> _______________________________________________<br>
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>><br>
>><br>
><br>
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</div></div></blockquote></div><br>
</div></div></blockquote></div><br>