Hi all!

I’m trying to connect my FS to a Sonus SIP trunk. I followed the
instruction at

  http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus

but it did not work. At the beginning of a call voice quality is good.
After a while it changes to choppy.

I don’t know if it’s the same problem: When I call the Tetris extension
via Sonus SIP trunk the sound is too fast and I’m getting log entries
like the following one:

[...]
2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME
not supported, changing our end from 20 to 10
2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from
PCMA@20ms@8000hz to PCMA@10ms@8000hz
2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global
timer resolution to 10ms to handle interval 10
2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer
[soft] 80 bytes per 10ms
2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec
sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits
2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write
Buffer at 160 bytes to accommodate 320->160
[...]

This problem is fixed by adding the following line to
conf/sip_profiles/external.xml:

<param name="rtp-autofix-timing" value="false"/>

Any hints?

/micha

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