[cisco-voip] Jitter buffering on NEC phones

[Andreas Sikkema]

> phone.audio.rfc2833.dtmf.relay="0" phone.audio.onhook.dtmf.mode="1"
> phone.audio.pcmu.pref="4" phone.audio.pcma.pref="5"
> phone.audio.g722.pref="3" phone.audio.g729.pref="6"
> phone.audio.g7221.24kbps.pref="1" phone.audio.g7221.32kbps.pref="2"
> phone.audio.g729.vad="1" phone.audio.pcmu.ptime="80"
> phone.audio.pcma.ptime="80" phone.audio.g722.ptime="80"
> phone.audio.g729.ptime="80" phone.audio.g7221.24kbps.ptime="80"
> phone.audio.g7221.32kbps.ptime="80"
>
> The codec we are using is G711, so it's probably among the default values,
> rather than the values aimed specifically at g722 or others.

And

> Almost all of the issues identified by wireshark at 50ms jitterbuffer are
> packets dropped for exceeding the jitter buffer.

That's because you have the packetization time set to 80ms, even for
G.711 (pcma or pcmu depending on which G.711 you're using). Those
values are not settings I see very often, so it's probably no wonder
that endusers are complaining. Since 80 msec between each packet is
larger than the wireshark jitterbuffer so every other packet is being
dropped especially since the wireshark jitterbuffer is not adaptive as
far as I'm aware.

Is 80ms a value you've chosen yourself? Usually these are 20 or 30ms...

-- 
Andreas Sikkema