[Freeswitch-users] OPUS: Jitter Buffer for bridged calls to use FEC?

Alexander Haugg Alexander.Haugg at c4b.de
Tue May 12 09:22:48 UTC 2020


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!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200512/97f72fb6/attachment.html>

More information about the FreeSWITCH-users mailing list