Speech coding/rx jitter/buffer audio packet size

[PA0TCA]

Is there any rule of the tumb settings that can be used for setting
the value of the RX jitter buffer/delay size and audio packet size,
on Internet connections with roundtrip delays around 20ms-50 ms?
And have they be the same on both ends?

[sm2o]

Hi
The default settings quality (Q) = 2, Rx-jitter buffer size (J) =4, Rx Jitter delay (D) = 3 and audio packet size (S) = 20 ms works very good for SSB, FM etc. in most installations using ex ADSL. For CW operations with the sidetone from the radio, lower latencys is preferable. If you have a good Internet connection with enough bandwidth ( up to 1 Mbit/s) .Try to lower S to 10 then 5 or even 2. If there is no dropouts J and D can also be lower down to 2 resp 1.
When using for example mobile Internet (3G) the other way can be necessary. Eg reduce bandwidth Q = 0 and maybe increase J and D if there is dropouts.
The setting do not have to be the same in both end. But the Q must be set to codecs using the same sampling rate in both ends. Ex Q=0 in one end and Q=2 in the other works fine. When I use the Control Panel via 3G, I used to set Q=0 in control end and Q=2 in radio end.

The document http://www.remoterig.com/firmwares/datarates090529.pdf describes the theory and how the different settings affect the delays and bandwidth needs, buts its very theoretical.

Good luck, with the tests.

73 de mike/sm2o

[PA0TCA]

Hello Mike,

thanks for the information. I was using an Internet ADSL connection with a 17 ms aroundtrip delay but intermitent going up to 80 ms, this produced a hickup in de received audio. Changed the settings for RX jitter buffer size to 5, and RX jitter delay to 10 (audio packet size 20 and quality 2). Works very good now for voice and all data modes.
Best regards,
Otto