Jitter buffer in IP phone

[p-beauregard]

Is the jitter buffer adjustable in Cisco IP phone? Also, what are the specs? I'm interested in knowing what is the maximum delay before a phone considered a packet not useful and drop it.

[thomas.chen]

You can use cisco quality of service techniques to improve voice quality.

[p-beauregard]

Yes, I know. But we are testing it over the internet trough a VPN. At certain time in the day the quality is very good, almost no lost packet but at night the quality drops and the number of lost packets raise. The access method is cable modem.

[dgoodwin]

Then it is most certainly related to the lack of QoS as the packets traverse one or more Internet hops. Even if you were able to mark the VPN packets that were important with ToS or DSCP markings, Internet core and border routers don't usually do any fancy queuing at this point, except for the special circumstances where some ISPs might offer VPN services with guarantees SLAs.

[saridder]

I've heard that the IP phone drops packets after 400ms, but ITU specifications specifiy that voice quality degrades at the 150ms mark. Try dial-up via ISDN and implement fragmenting with MLPPP.

Perhaps you have data comming out that's creating serialization for the voice packets, plus all the traffic increases that occour in the afternoon/evening.