[Freeswitch-users] FS to a Sonus SIP trunk

[David Ponzone]

Sonus sucks.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

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Le 26/07/2011 à 16:52, michael knop a écrit :

> 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
> 
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