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Problems with VoIP

Problems with VoIP
Are you considering making the move to cloud VoIP? You have, no doubt, heard your share of shining success and equally gloomy horror stories about colleagues that have moved to VoIP. And they are all true!

Let’s look at some of the things that can go wrong with a VoIP implementation and discuss what steps can be taken to fix them or avoid them completely.

Latency is the time between the moment a voice packet is transmitted and the moment it reaches its destination. When this time is long (roundtrip voice delays of 250ms), callers will notice echoes or other degradation in voice quality.

Although the voice packets that make up a VoIP call travel within your network, they also are subject to delays outside of your network. You can reduce or eliminate latency by ensuring that voice packets are given high priority over other data packets in your network. Generally accepted guidelines tell us that if we can ensure that voice packets in your own network have transit latencies of considerably less than 150 ms, you can go a long way towards improving overall VoIP call voice quality.

There are a number of proven techniques for prioritizing VoIP traffic. And you don’t have to be a trained network engineer to make this happen. A quality VoIP router, like the CompuVoIP cloud-based managed router, can solve many of these issues and help you achieve high-quality VoIP calls in your organization.

Jitter is the measure of variability of latency across a network over time. The sending side transmits packets in a continuous stream and spaces them evenly apart. Because of network congestion, improper queuing, or configuration errors, the delay between packets can vary instead of remaining constant. Jitter is another cause of poor VoIP call quality.

We again look to the VoIP router to come to the rescue. When a router receives an audio stream for Voice over IP (VoIP), it must compensate for the jitter that is encountered. The mechanism that handles this function is the playout delay buffer. The playout delay buffer (aka de-jitter buffer) buffers these packets and then plays them out in a steady stream to the digital signal processors (DSPs) to be converted back to an analog audio stream.

Most residential internet service offerings are optimized for web surfing, not for VoIP calling. Voice packet transmission requires an additional set of internet protocols

These days, ‘business class’ internet service is just about ubiquitous. Make sure you are subscribed to your internet provider’s ‘business class’ internet service if you want to ensure high-quality VoIP calls. Chances are that you are already subscribed to this level of internet service as it usually includes symmetric bandwidth levels for both upload and download.

At CompuVoIP, we assess your network and make recommendations and adjustments to ensure the highest level of VoIP phone call quality. Learn more about us at