How Voice over IP Has Met the Challenges of Quality of Service (QoS)
Voice over IP (VoIP) has established itself as the most economical and effective way of providing voice communication in an organization and helping it scale to its needs. What is pretty amazing about this fact is that Internet Protocol (IP), was never primarily designed to be able to handle real-time audio communication, let alone be able to do so with a comparable quality of service (QoS). The evolution of VoIP has been one in which this technical hurdles has gradually been overcome but thankfully the efforts of researchers and engineers have made it a reality.
When VoIP was nascent in the early part of the last decade, many people wrote it off as a novelty and pointed to the jitter, delay, and garbling of early attempts as a sign that it was not going anywhere. However, the dedication of skilled professionals and the application of some communication science has paid off. It is now a major force in communication. Let u s take a look at how this was accomplished.
A traditional public switched telephone network (PSTN) offers 64kbps of bandwidth in both directions. That is not a level of bandwidth that can be reliably delivered over most internet connections, especially when you consider that a packet switched phone connection must share a network with a flurry of other traffic.
The sampling rate needed for VoIP is typically around 8kbps, as telephone communication generally occurs at frequencies between 0.5 and 3.5 KHz. Any considerations in reducing bandwidth by lowering sampling rate not only mean signal degradation, they also run up against the Nyquist Theorem. This theorem is fundamental to almost every form of digital media and states that you cannot reliably digitize a signal with a sample rate lower than twice the highest spectrum frequency. Lowering the bit depth, which provides the range of signal values of each sample, can lower bandwidth but it will turn the signal into gibberish pretty quickly.
The real answer to this is an approach that has also been taken to address the needs of streaming video – compression. Low-loss compression and CODEC standards such as G.711, G.726 and G.729 that allow compression and decompression to happen on the fly, allow us to put more fidelity into the voice signal given the restraints of low-bandwidth.
VoIP is a technology that continues to expand, improve, and evolve. Contact a trusted and reliable VoIP expert to learn more.
