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Early Pitfalls Of Voip Calls

In the early days of VoIP, pioneer adopters encountered poor quality calls with line static

, audio gaps and garbled transmissions - unflattering experiences that eventually led to some wariness and hesitation about the new technology. Fortunately, it's also a double-edged sword as these same pitfalls drove industry vanguards to undertake the intensive research and development on both the software and hardware components needed to ensure better quality VoIP calls. A lot of the initial troubles can be attributed to latency and jitter - the twin issues that arise mostly from network problems.

Latency and jitter defined

Latency is technically defined as the delay in the transmission of an audio signal from the moment it enters a system or network to the time it leaves the loop. Any processing along the way - such turning the media to packets at the beginning of the journey and media compression using a VoIP codec, to traversing gateways and firewalls that examine each packet as they go in and out of networks, to the moment the packets are re-converted to voice at the receiving line - result in some form of delays, usually measured in milliseconds. Jitter, on the other hand, is the variation in latency that happens unpredictably due to queuing delays, such as when the network is overloaded or when the queue for a particular equipment or gateway is more than the usual. Simply put, a packet becomes affected by jitter when it arrives early or after its expected arrival time.

During the early days of VoIP, latency in calls passing through the Internet ranged from 400 to 4000 milliseconds. The current industry benchmark pegs the maximum acceptable latency at 150 milliseconds, the same benchmark established for POTS lines, but VoIP users would mostly likely notice call degradation once latency exceeds 200 milliseconds.


System improvements


Both latency and jitter may result in packet delays or losses that significantly downgrade transmission quality for real-time applications like VoIP. Note that packets that arrive early or late are normally useless, as the system would not wait for delayed packets nor hasten to arrange these in correct order - thereby causing gaps in voice transmissions or call failures. Fortunately, current systems have improved hardware and software to minimize latency, while jitter buffers and packet-loss concealment devices installed in receiving VoIP systems store packets as they arrive then send them out in the correct order to the receiving telephone line.

The availability of fiber optic cabling in some parts of the world - thanks in part to the frenetic laying out done by start-up companies that went belly up during the dotcom bubble in the late 1990s - and the availability of these laid-out lines for dirt-cheap prices helped substantially in improving VoIP and enabling it to survive the initial backlash of poor quality. As more companies began to use the existing fiber-optic loops to offer broadband services with higher bandwidth and speeds, VoIP calls eventually became better and less problematic.

Other improvements introduced to lessen latency and jitter include the use of flap routes to avoid passing through congested network lines and using hardware with more effective compression and packetizing techniques that address signal attenuation particularly for long distance calls. All these improvements enabled VoIP to leapfrog into wider acceptance, as VoIP calls became at par or better than calls made over traditional telephones lines.

by: Andrew Wiggin
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