[Freeswitch-users] Jitterbuffer

[Russell Treleaven]

Are you sure the jitter/loss in on the fixed wireless side?

Is the freeswitch<->pstn sip?

Is freeswitch transcoding between the legs?

There is a tool called mtr to measure the loss/jitter along the path.

If freeswitch is not transcoding then it doesn't need a jitter buffer.
You could actually use bypass_media to take freeswitch out of the media
path and see if that helps.
https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview


On Thu, Nov 19, 2015 at 11:07 PM, Josh Richesin <jlrichesin at gmail.com>
wrote:

> Hello all!
>
> Thanks for the recommendations on upgrading to FS master, (Assuming that
> is 1.6.2?)   and using the new Jitter Buffer and OPUS codec.
> Unfortunately, after researching and working with our ATA vendor OBI, they
> do not support OPUS.  So that has me form a few more questions on this
> issue.
>
> 1) Is there any documentation on the new Jitter Buffer being used?
>
> 2) Is there any reason to believe that upgrading from our current Free
> Switch Version 1.4.18 to 1.6.2 would fix the issue with the current codec
> of g711, even tho I can not use opus?
>
> 3) Below is the real issue that I am trying to fix:
>
> obi200------------> fixed wireless --------> FreeSwitch ------PSTN
>
> Call Quality issues are mainly reported by the end user on the PSTN
> network.  Users report that the calls are cutting in and out but our users
> on the obi200 side hear them perfectly.  Any ideas on settings to try to
> optimize in this situation?
>
> I know that it is a network related issue, with high jitter, however I
> know there must be a fix as other VOIP providers can ride the same network
> path and have no issues at all.
>
> 4) Main reason for not just upgrading is that I am using fushionPBX, and
> they do not yet support 1.6.2, and also trying to figure out an actual
> upgrade plan so that I do not loose all the customers voicemail's, VM
> greetings passwords and such.
>
> Any help is greatly appreciated with what direction I should go.
>
> Thanks!
>
> Josh
>
> On Fri, Nov 13, 2015 at 9:45 AM, Brian West <brian at freeswitch.org> wrote:
>
>> We probably need to wipe confluence clean of the incorrect jitter buffer
>> docs now that its been completely rewritten in the past few months.
>>
>> On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström <Ola.Backstrom at loxysoft.se
>> > wrote:
>>
>>> Dragos,
>>>
>>>
>>>
>>> Your example to enable jitter buffering uses the letter ”p”. Is that a
>>> typo or does it indeed mean something?
>>>
>>> >  You can enable the Jitter Buffer in the dial plan:
>>> >  <action application="set" data="jitterbuffer_msec=2p:25p:"/>
>>>
>>> I see no such usage in the documentation at
>>> https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but
>>> just “60:200:20”.
>>>
>>>
>>>
>>> Best regards
>>>
>>> /Ola
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Dragos
>>> Oancea
>>> *Sent:* den 12 november 2015 10:14
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* Re: [Freeswitch-users] Jitterbuffer
>>>
>>>
>>>
>>> Hi Josh,
>>>
>>> Use FS master, it has a new audio Jitter Buffer.
>>>
>>> Then you can use Opus with this setting in the codec config:
>>> <param name="use-jb-lookahead" value="1"/> .
>>> The other side must be sending FEC so you could profit from it.
>>>
>>>
>>> You can enable the Jitter Buffer in the dial plan:
>>> <action application="set" data="jitterbuffer_msec=2p:25p:"/>
>>> And then enable PLC :
>>> <action application="set" data="rtp_jitter_buffer_plc=true"/>
>>>
>>> Regards,
>>> Dragos
>>>
>>>
>>> On 11/12/2015 07:51 AM, Josh Richesin wrote:
>>>
>>> Hello,
>>>
>>> I am running FreeSwitch for our small Wireless ISP.   Obviously, as a
>>> wireless ISP and ITSP, we have a lot of jitter and some packet loss.  Have
>>> a few questions for this setup.
>>>
>>> 1) What is the best codec to use when there is going to be jitter and
>>> some packetloss?
>>>
>>> 2) The ATA's that we are using have adaptive jitter buffers, should I
>>> enable jitterbuffer on Freeswitch as well?  If so, should it be via sofia
>>> profiles?
>>>
>>> The problem that I am trying to solve is this...
>>>
>>> Our voice customers on our freeswitch have a lot of issues with "calls
>>> breaking up"  When they use another voice provider, like say vonage or
>>> something running off of a metaswitch, they do not have the same issues.
>>>
>>> Any help with this would be appreciated!  Thanks in advance.
>>>
>>> Josh
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>>
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>>>
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>>>
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>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
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>>
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>>
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>
>
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>
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