[cisco-voip] Jitter buffering on NEC phones

[Dennis Heim]

I guess the first thing I would be curious as to what is the reason for adjusting the jitter buffer. What issues are you encountering?

Dennis Heim
Senior Engineer (Unified Communications)
CDW  Advanced Technology Services
10610 9th Place
Bellevue, WA 98004

425.310.5299 Single Number Reach (WA)
317.569.4255 Single Number Reach (IN)
317.569.4201 Fax
dennis.heim at cdw.com<mailto:dennis.heim at cdw.com>
cdw.com/content/solutions/unified-communications/<http://www.cdw.com/content/solutions/unified-communications/>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joseph Mays
Sent: Monday, February 20, 2012 10:44 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Jitter buffering on NEC phones

My apologies in advance for sending this to the Cisco VOIP list, since it's not, strictly speaking, a cisco issue, but I'm hoping to find someone here who has worked with NEC DT700 voip phones, because we are trying to get an answer to what should be a simple question, but no amount of calling to NEC has yielded an answer, and multiple searches online have proven useless.

We are trying to figure out how to set the jitter buffer in NEC DT700 phones to 160ms. Or set the buffer for 16 10ms packets. We have a template config we found on NEC's site that contains all the parameters that can be set for the phone, but no amount of begging NEC support through several layers of escalation has yielded a simple answer to what exactly are the config parameters we want. The section they are in is easy enough to identify; it should be one or two of the following parameters --

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.

Again, my apologies for sending it here, I have been very hesitant to do so for days, but we are out of other options, and at this point are just hoping that there is either someone who knows DT700 phones or knows VoIP in general enough that they can identify what we need to set for 160 ms (16 10ms packets) jitter buffer.

Joe Mays

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20120220/c688641f/attachment.html>