Re: voip quality/bandwidth/latency techniques

Linux Advanced Routing and Traffic Control

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dale wrote:
I have voip quality issues I would like to minimize. I have a ~= 3M/384k (Comcast) cable modem and a CentOS based Linux router (SME 7, 2.6.9 kernel) with 5 NAT'd devices (3 PCs "DHCP", 2 Vonage adapters "static 10.10.2.10-11"). The quality problems are audio cutting out and popping. I tried the following (see below) based on a Cookbook example, but I still have audio popping. I have noticed popping corresponding with web browsing, etc. Any suggestions that may improve voip quality? Also, are there any network metric capture/graphing tools that are helpful to analyze these type of issues. I'm thinking of a graph that shows various network metrics. I could watch the graph while using the phone and correlate graph spikes with audio pops. Thanks in advance for your help.

Dale
Hi Dale, you have to guarantee both delay (latency) and bandwidth for clean VoIP calls. With HTB you can't do that (although it's still better than nothing). You should use HFSC scheduler. If you have working HTB script than changing it to HFSC is quite simple. Good start is here http://linux-ip.net/articles/hfsc.en/ also there are plenty of exapmles using HFSC for VoIP on the web. Note that you have to guarantee latency and bandwidth on both incoming and outgoing direction (i can see only outgoing in your example). There are other things to tweak such as MTU, VoIP chunk size and others, but start with HFSC shaping.
/ak
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