Poor audio quality due to rtp timing issues
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?
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* * * * * On Wed, Mar 20, 2013 at 8:02 PM, Garey Arrington <gareyarrington@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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participants (2)
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Anthony Minessale
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Garey Arrington