[Freeswitch-users] FS to a Sonus SIP trunk

[michael knop]

Yehavi,

unfortunately changing the provider is not an option. If I can’t get 
FreeSWITCH to work with Sonus I can’t switch to FreeSWITCH. Then if have 
to stay using Asterisk.

To me it looks like that its nondeterministic how much time it takes 
until the sound is choppy. Sometimes sound quality is brilliant for 
several minutes. Sometimes the call starts with choppy sound.

/micha


Am 27.07.2011 20:54, schrieb Yehavi Bourvine:
> Hello Micha,
>    How much time does it take until the sound is choppy?
> We have been connected to Sonus provider up to a week ago. Incoming
> calls started being choppy after about 15 minutes (outgoing calls were
> ok). We also had inconsistent problems with DTMF. We ended it by
> changing a supplier...
>                       __Yehavi:
>
> 2011/7/26 michael knop <michael.knop at hcu-hamburg.de
> <mailto:michael.knop at hcu-hamburg.de>>
>
>     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
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org