Update: Call starts with good sound quality. After the following log message sound is choppy: 2011-07-28 12:07:22.639151 [DEBUG] sofia.c:5094 Duplicate SDP v=0 o=Sonus_UAC 8739 8900 IN IP4 XXX.XXX.XXX.XXX s=SIP Media Capabilities c=IN IP4 YYY.YYY.YYY.YYY t=0 0 m=audio 20320 RTP/AVP 8 0 18 100 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=maxptime:10 /micha Am 26.07.2011 16:52, schrieb michael knop:
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