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@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Dragos Oancea
Sent: den 12 november 2015 10:14
To: freeswitch-users@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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