[Freeswitch-users] Poor audio quality due to rtp timing issues

[Anthony Minessale]

Whatever you are talking to on the other end is negotiating 30 ms and
actually sending 20.  Sounds like the sipura/cisco bug.

Go in the UI for the device and find the packet time param which is set to
30 and change it to 20.

>From our wiki: http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo

SPA3102 DEVICE CONFIGURATION:

NOTE: There is a bug in the default configuration of the SPA-3102 and other
Linksys devices that sets the RTP Packet Size to .030, this should be set
to .020 to avoid problems.

*admin -> advanced -> Voice -> PSTN*

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On Wed, Mar 20, 2013 at 8:02 PM, Garey Arrington
<gareyarrington at yahoo.com>wrote:

> When making inbound and outbound calls I am getting slightly less than
> acceptable audio quality after a few seconds.  The machine is a 64 bit
> Centos 5.5 vm running on Xen with kernel version: 2.6.18-194.el5
>
> I am getting warnings like with each call:
>
> 2013-03-21 04:22:54.655968 [WARNING] mod_sofia.c:1274 Asynchronous PTIME
> not supported, changing our end from 30 to 20
> and
> 2013-03-21 04:22:54.056083 [WARNING] switch_time.c:578 Increasing global
> timer resolution to 10ms to handle interval 30
>
>
> I have tried the fix that I have consistently found for this issue which
> is to add this to external.xml:
>
> <param name="rtp-autofix-timing" value="false"/>
>
>
> But this makes the audio quality so bad you can hardly make out anything.
>
>
> Can someone point me in a better direction for a solution to this problem?
>
>
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-- 
Anthony Minessale II

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