VOIP over Broadband quality issues

[mark.blanchfield]

We have a few remote broadband sites using VOIP. Some are terminated by a Pix 501 and some are using an 1811. We have been having voice quality issues. About 1/3 of the calls experience clipping and/or distortion. I have QOS configured using policy maps on the 1811 and still am having issues. Is there a good tool to trobleshoot the VOIP? All ping tests seems to be fine but I am looking for a better tool to trobleshoot these issues. We have a combination of Tadiran and Cisco IP phones. I was under the impression the Cisco phones can self correct the jitter using the jitter buffer. Thanks.

[sundar.palaniappan]

Hi,

You can use SAA for network monitoring. It will help you measure jitter/latency across the WAN. I don't come from the voice side but my understanding is the IP phones can only tolerate certain level of jitter. You might want to look at the SPEC for the type of IP phones you have to check how much jitter is acceptable.

Here's the link for configuring SAA to monitor network. Look under the heading 'jitter operation' for configuration setup. To solicit more useful responses you might want to post this in the IPT forum.

http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps1831/products_configuration_guide_chapter09186a00800ca63e.html#1001551

HTH

Sundar

[chuck.brule]

Mark:

Do you have a Linux machine anywhere around? If so, I would strongly recommend Ethereal as a packet trace analysis tool.

For reference purposes, VoIP under most standard implementations is worried about the following three quality issues:

1) The connection must suffer no more than 200 ms round trip delay. Times of 150ms or less are strongly recommended.

2) The packet loss percentage can be no more than 1%. Closer to zero is recommended.

3) Jitter (which is defined as the difference between the delay from one packet to the next) can be no more than roughly 20 ms. This means if you have three packets sent, one takes 80 ms, one takes 85 ms and one takes 75 ms, your Jitter is 10 ms.

Three things I might suggest:

1) Look at your policy map on the 1811... how are you classifying VoIP traffic and are you giving it enough bandwidth in the priority queue?

2) Are you using G.711? If so, I would DEFINATELY switch to G.729; your voice quality will noticably improve at a great savings of bandwidth.

3) If you are already using G.729, have you considered using Compressed RTP? cRTP will shave about 40 - 50% of the bandwidth from each packet.

If you have any questions, please don't hesitate to contact me.

Chuck Brule

CCNA

VoIP & Data Networks Customer Engineer

Verizon