VoIP Quality : Jitter, Dropped and Choppy Calls

Posts: 1 · New Forum Member Joined: May 23, 2004 Posts: 1

This is a very sensitive issue! I will explain it in detail. Most problems with Voip are Internet (ISP) related. Basically stems from line quality problems. I would say speech quality has a few hurdles to make it perfect.

Initially, IP wasn't designed to carry voice or real-time media. So IP was designed for data. Well voice however is very sensitive to delay and data is sensitive to packet-loss. And then add jitter, which are variations in delay. Oh, don't forget real-time voice over the wire uses the UDP protocol. Okay, UDP wasn’t designed with error checking. UDP doesn't care if user A got packets from user B. The challenge lies specifically in these areas: BANDWIDTH, delay, packet-loss and QOS (Quality Of Service). To begin with delay which voice seems to be extremely intolerant to relate with one another. Maybe while on a Voip call, packets suddenly change course or simply decide to take a different route. So, some packets are passing through different routers to get to its destination. Maybe a router on the Internet put some of your packets in a queue longer than others. Or a Core router within the Internet or ISP might be just taking a dump. This can cause problems for a few days, weeks or even longer than months. Until, a tech fixes the issue.

All of these issues can cause calls to be dropped, voice to have jitter and choppy conversations. The jitter problem can also be known as variations in delay. Most Voip devices have jitter compensation built into the hardware like data buffering. Simply, this process involves caching and syncing data before it is played to the ear. The device can keep adjusting to delay as it keeps varying. Just at least this delay stays consistent of course.

Another factor with Voip would be little or none of the packets with your voice samples are lost from user A to user B. It is possible that network nodes across the Internet could be overflowing to the result of lost packets. Specially, the route between the caller and receiver. This stems back to UDP in the sense that UDP does not check to see if packets were lost in route containing your voice. I'm sure most people have witnessed this choppiness on a call.

Hopefully, ISP's will implement later versions of DOCSIS on every network. The later versions of DOCSIS have all kinds of goodies like for one “QOS”. So UDP traffic can be prioritized first hand. It would be critical to have voice traffic processed quickly. The voice traffic won’t be waiting in a routers queue while a huge file transfer is occupying the available bandwidth. I would say “QOS” would be essential to reduce congestion as well as minimizing packet-loss.

I was wondering if anyone else can give more information to understand these hurdles. Post more questions for answers and answers for questions. Idea

Posts: 183 · Vonage Forum Master Joined: May 05, 2004 Posts: 183

Redisgning the Internet?