[Freeswitch-users] Jitterbuffer

Dragos Oancea doancea at orange-vallee.net
Fri Nov 20 13:50:55 MSK 2015


Hi

Josh, FS master is not 1.6.2 , it's the latest master branch . Get it 
from git and compile it.
1.6.2 does not have the new JB.

I assume you don't transcode at all and you use G.711 along the path.
Try using ILBC between the OBI200 and Freeswitch , so you would do 
transcoding
with Jitter Buffer and PLC enabled. It would be better if you don't 
transcode at all,
but your PSTN provider should offer ILBC or other compressed codec that 
you should try and you would also need to rely on them for PLC .You 
should probably enable the Jitter Buffer on the bridge if you do 
transcoding :
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/>

Regards,
Dragos

PS:
Obihai supports Opus in their phones: 
http://www.obihai.com/partner/obi1000 .


On 11/20/2015 07:22 AM, Russell Treleaven wrote:
> 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 
> <mailto: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
>     <mailto: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 <mailto: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>
>             [mailto: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
>             <mailto: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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