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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