[Freeswitch-users] FS to a Sonus SIP trunk

[michael knop]

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 at 20ms@8000hz to PCMA at 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... at 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