<div>Hi,</div><div><br></div><div>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).</div><div><br></div><div>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 <b>the recording has a garbled outbound (Callee) leg</b> (inbound/Caller voice is fine.).</div>
<div><br></div>We have tried all the combinations of G729@20i, G729@60i, rtp-autofix-timing, and default-ptimes to no avail.<br clear="all">
<div><br></div><div>We are using git HEAD.</div><div><br></div><div>I do see this on fs_cli:</div><div><br></div><div><div><font face="'courier new', monospace" size="1">2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa8894898 (nil)</font></div>
<div><font face="'courier new', monospace" size="1">2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xa8894898 (nil)</font></div><div><font face="'courier new', monospace" size="1">2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 <b>Asynchronous PTIME not supported, changing our end from 20 to 60</b></font></div>
<div><font face="'courier new', monospace" size="1">2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from G729@20ms@8000hz to G729@60ms@8000hz</font></div><div><font face="'courier new', monospace" size="1">2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa86fe340 0xa86f1e78</font></div>
</div><div><br></div><div>Should this be working, are we missing something obvious?</div><div><br></div><div><br></div><div>Thank you,</div><div><br></div><div><br></div><div>Kind regards,</div><div><br></div><div><br></div>
<div>Colin</div><div><br></div>