[Freeswitch-users] FS to a Sonus SIP trunk
michael knop
michael.knop at hcu-hamburg.de
Wed Jul 27 14:48:03 MSD 2011
full ack
Am 27.07.2011 11:51, schrieb David Ponzone:
> Sonus sucks.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service ClientIPeva
> tel: 0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>- www.ipeva-studio.com
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>
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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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