[Freeswitch-users] Audio rate changes and clipping on playback

[Michael Jerris]

Doing packet tagging in a cross platform way is a bit difficult, and  
we have just never tried to do it when there are other simple ways to  
handle.  It would probably involve adding the ability to do qos  
tagging to APR.  If someone is willing to get patches into apr for  
this, it would make sense to look at adding that support to freeswitch  
at some point in the future.

Mike


On Jan 26, 2008, at 4:10 PM, Peder @ NetworkOblivion wrote:

> Why is it that FS doesn't offer QoS packet tagging as a feature?  It
> seems kind of kludgy to run another program on the same box to tag the
> packets before they are sent out.  I know everything in FS is pretty
> well thought out, so I am sure there is a reason, I am just curious  
> what
> it is.
>
> Peder
>
>
> Brian West wrote:
>> Dale,
>> 	What SVN rev are you running?  Is this a xen or other virtual
>> instance?  I have done extensive testing in this area and haven't  
>> seen
>> this... It could be related to QoS bits.  Have you tried to use
>> iptables to tag your traffic?  Check this link http://wiki.freeswitch.org/wiki/QoS/
>>
>> FreeSWITCH doesn't tag any traffic with any QoS bits and Asterisk
>> does... So you might have to use the above URL to cause the traffic  
>> to
>> get tagged and prioritized on the ADSL router.
>>
>> /b
>>
>>
>> On Jan 26, 2008, at 7:29 AM, freeswitch at dalethatcher.com wrote:
>>
>>> Hello all!
>>>
>>> I'm experiencing some problems with clipping and the rate of
>>> playback of
>>> wav files from Freeswitch.  I've not raised this before as I thought
>>> the
>>> problems were due to hosting over an ADSL line.  However I've now  
>>> seen
>>> the same issues running from a dedicated host.
>>>
>>> My test setup is:
>>>
>>> 	Client <-> NAT ADSL <-> Freeswitch
>>>
>>> I've replicated using both the Gizmo and FWD clients from Windows.
>>> If I
>>> substitute Asterisk with Freeswitch then the audio comes over fine.
>>> The
>>> two test samples are one of myself and the fpm-sunshine.wav that is
>>> bundled with asterisk.
>>>
>>> I've tried playing around with the codecs but see the same problems
>>> regardless.  Also I've performed a test with a conference hosted on
>>> the
>>> Freeswitch instance with myself and friend dialling in using Gizmo  
>>> and
>>> the audio is fine.  Which implied to me that the link between  
>>> Gizmo to
>>> Freeswitch is ok.
>>>
>>> So I'm guessing that there is something wrong with the samples  
>>> that is
>>> causing the problem?  They're both in this format:
>>>
>>> $ file fpm-sunshine.wav
>>> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono
>>> 8000 Hz
>>>
>>> My deployment platform is Debian stable and the Freeswitch SVN
>>> revision
>>> is 7343.  I've tried looking in Jira but keep getting a system error
>>> page (for search terms 'audio', 'wav' and others).
>>>
>>> Is anyone else seeing the same issues or has a solution?
>>>
>>> thanks,
>>>
>>> - Dale
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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