Garbled Leg Recording G729 with Asynchronous ptime
Hi, Please, I was wondering if there is anything obvious that we are missing when trying to record a G729 call with asynchronous pimes (in: 20ms, out: 60ms). The problem case call flow is: inbound leg (ptime 20 ms), start recording, playback a short message, bridge to outbound (destination can only handle a ptime of 60 ms). We are using mod_com_g729's codec. The audio while on the call in normally fine (rarely there is one way speech depending on inbound/outbound provider), but *the recording has a garbled outbound (Callee) leg* (inbound/Caller voice is fine.). We have tried all the combinations of G729@20i, G729@60i, rtp-autofix-timing, and default-ptimes to no avail. We are using git HEAD. I do see this on fs_cli: 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 *Asynchronous PTIME not supported, changing our end from 20 to 60* 2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from G729@20ms@8000hz to G729@60ms@8000hz 2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa86fe340 0xa86f1e78 Should this be working, are we missing something obvious? Thank you, Kind regards, Colin
have you tried updating lately? also try global_setvar passthru_ptime_mismatch=true On Fri, Jun 8, 2012 at 3:29 AM, Colin Sindle <colins@succinct.co.za> wrote:
Hi,
Please, I was wondering if there is anything obvious that we are missing when trying to record a G729 call with asynchronous pimes (in: 20ms, out: 60ms).
The problem case call flow is: inbound leg (ptime 20 ms), start recording, playback a short message, bridge to outbound (destination can only handle a ptime of 60 ms). We are using mod_com_g729's codec. The audio while on the call in normally fine (rarely there is one way speech depending on inbound/outbound provider), but the recording has a garbled outbound (Callee) leg (inbound/Caller voice is fine.).
We have tried all the combinations of G729@20i, G729@60i, rtp-autofix-timing, and default-ptimes to no avail.
We are using git HEAD.
I do see this on fs_cli:
2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 Asynchronous PTIME not supported, changing our end from 20 to 60 2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from G729@20ms@8000hz to G729@60ms@8000hz 2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa86fe340 0xa86f1e78
Should this be working, are we missing something obvious?
Thank you,
Kind regards,
Colin
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hi Anthony, Thank you very much for your response. Sorry, I meant that we are using the _latest_ git HEAD at time of writing. I have now tried passthru_ptime_mismatch=true, and it fixed the recording of some of the cases of asynchronous ptime, but also broke other previously "working" cases. Kind regards, Colin On 8 June 2012 21:03, Anthony Minessale <anthony.minessale@gmail.com> wrote:
have you tried updating lately? also try global_setvar passthru_ptime_mismatch=true
On Fri, Jun 8, 2012 at 3:29 AM, <colins@succinct.co.za> wrote:
Hi,
Please, I was wondering if there is anything obvious that we are missing when trying to record a G729 call with asynchronous pimes (in: 20ms, out: 60ms).
The problem case call flow is: inbound leg (ptime 20 ms), start recording, playback a short message, bridge to outbound (destination can only handle a ptime of 60 ms). We are using mod_com_g729's codec. The audio while on the call in normally fine (rarely there is one way speech depending on inbound/outbound provider), but the recording has a garbled outbound (Callee) leg (inbound/Caller voice is fine.).
We have tried all the combinations of G729@20i, G729@60i, rtp-autofix-timing, and default-ptimes to no avail.
We are using git HEAD.
I do see this on fs_cli:
2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 Asynchronous PTIME not supported, changing our end from 20 to 60 2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from G729@20ms@8000hz to G729@60ms@8000hz 2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa86fe340 0xa86f1e78
Should this be working, are we missing something obvious?
Thank you,
Kind regards,
Colin
participants (2)
-
Anthony Minessale
-
Colin Sindle