<div dir="ltr"><div>Hello Micha,</div><div> </div><div> How much time does it take until the sound is choppy?</div><div>We have been connected to Sonus provider up to a week ago. Incoming calls started being choppy after about 15 minutes (outgoing calls were ok). We also had inconsistent problems with DTMF. We ended it by changing a supplier...</div>
<div> </div><div> __Yehavi:<br><br></div><div class="gmail_quote">2011/7/26 michael knop <span dir="ltr"><<a href="mailto:michael.knop@hcu-hamburg.de">michael.knop@hcu-hamburg.de</a>></span><br><blockquote style="margin: 0px 0px 0px 0.8ex; padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="gmail_quote">
Hi all!<br>
<br>
I’m trying to connect my FS to a Sonus SIP trunk. I followed the<br>
instruction at<br>
<br>
<a href="http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus" target="_blank">http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus</a><br>
<br>
but it did not work. At the beginning of a call voice quality is good.<br>
After a while it changes to choppy.<br>
<br>
I don’t know if it’s the same problem: When I call the Tetris extension<br>
via Sonus SIP trunk the sound is too fast and I’m getting log entries<br>
like the following one:<br>
<br>
[...]<br>
2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME<br>
not supported, changing our end from 20 to 10<br>
2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from<br>
PCMA@20ms@8000hz to PCMA@10ms@8000hz<br>
2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global<br>
timer resolution to 10ms to handle interval 10<br>
2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer<br>
[soft] 80 bytes per 10ms<br>
2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec<br>
sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits<br>
2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write<br>
Buffer at 160 bytes to accommodate 320->160<br>
[...]<br>
<br>
This problem is fixed by adding the following line to<br>
conf/sip_profiles/external.xml:<br>
<br>
<param name="rtp-autofix-timing" value="false"/><br>
<br>
Any hints?<br>
<br>
/micha<br>
<br>
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</blockquote></div></div>