That snip of documentation for JB is pretty old and obsolete I'd say, it does not mention transcoding + FEC .
But where transcoding is used/needed is like a "point of termination".

Relevant docs (and best practices) about OPUS + JB + FEC here:
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec
If there's no JB then it can't be any FEC decoding.

If you just do OPUS->FS(bridge)->OPUS then no point of enabling the JB, it only adds delay.

Cheers,
Dragos

On Tue, May 12, 2020 at 10:23 AM Alexander Haugg <Alexander.Haugg@c4b.de> wrote:

Hi,

 

I try to find out the best practice to use the OPUS features.

The scenario is:

 

Client A -------> (JitterBuffer) FreeSwitch -----------------------> Client B 

               <---------------------- FreeSwitch (JitterBuffer) <--------

 

Is this the correct way?

 

In the documentation for OPUS the message is, JitterBuffer is needed to use FEC.

But in the documentation for the JitterBuffer is written:

              If both sides of a bridge are RTP and both sides have a jb, its fairly

              useless.  In fact if anything, it can worsen call quality.

 

              You should only run jitterbuffers at points of termination change of

              protocol.  Examples, if FS was hosting a conference or IVR, if you are

              bridging the call to a phone for instance, you want to not use a

              jitterbuffer because you want to preserve the original timestamps so

              your phone can use its own jitterbuffer.

 

 

I think, package lost is the start indicator for FEC and normally the RTP sequence number is the indicator for package lost, but the sequence number is valid for the own SSRC space only.

That means, if I have lost packages from Client A to Freeswitch is this information on the other side to client B missing, correct?

 

Now the question, what is the best practice handle this scenario?

 

Thanks a lot!

Alex

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