This week, we started off our blog about VoIP with An Introduction to VoIP Technology and plan to write more blogs on this topic to help our office space Toronto customers, our virtual office clients and our blog readers to understand and make informed decisions about how VoIP services can be a cost saving strategy for their businesses.
VoIP is a proving to be a self-improving technology that responds to customer demands, business dynamics and new innovations available from developers. Ever since its introduction, VoIP technology has revolutionized the way phone calls are made over the Internet. In the early days of VoIP, calls were often choppy like that of the old “walkie-talkie.” You had to wait until the other party was finished his or her sentence before you could respond. One of the other early challenges for VoIP was call quality. At that time, the exchange of voice relied heavily on the strength of the internet connection and there was a high risk of calls being dropped and disruptive lag times – even amongst calls on the same internet node.
The biggest improvements in VoIP have come about because of the availability of high-speed internet connections with more bandwidth. Another factor that improved call quality was using smaller data packets and voice compression. VoIP requires two classes of protocols – a signaling protocol that is used to set up, disconnect and control the calls and telephony features, and a protocol to carry speech packets. The RTF (Real-Time Transport Protocol) carries speech transmission which will work with any signaling protocol.
An IP phone or “softphone” generates a voice packet every 10, 20, 30 or 40 ms (milliseconds), depending on the provider’s implementation. The 10 to 40 ms of speech can be compressed, uncompressed and even encrypted. It is important to note that it takes many packets to carry just one word, so the faster your VoIP service processes the packets, the better the quality of the call. Basically, the shorter the packet creation delay, the more network delay the VoIP call can tolerate. Shorter data packets cause less of a problem if the packet is lost. Short packets require more bandwidth, however, because of increased packet overhead. Longer data packets can contain more speech bytes and reduce the bandwidth requirements – but they can also produce a longer construction delay on the receiving end. For this reason, most VoIP services are typically 20 or 30 ms sized packets to balance bandwidth and call quality.
Shorter packets have higher overhead. There are 54 bytes of overhead carrying the voice bytes. As the size of the voice field gets larger with longer packets, the percentage of overhead decreases – therefore the needed bandwidth decreases. In other words, bigger packets are more efficient than smaller packets.
Just like the early voice compression technologies that were designed for undersea cables (where bandwidth was limited and expensive), VoIP uses the same voice compression technologies that were created to reduce this bandwidth requirement of overseas calls. Therefore, voice compression is not new. Voice compression is also used for digital cellphone calls and calls made to the international space station to save on data.
Another aspect that has changed is in the way that VoIP is delivered. More and more hardware developers of traditional phones and phone systems have entered the marketplace with their IP phone technology systems. Companies like Cisco – who have long been known for developing network hardware such as routers, nodes and switches – is also making VoIP phones and VoIP phone adapters that allow consumers to use any phone for their VoIP services. An online article on the Forbes Magazine website called Cisco VoIP Phones Affected By On Hook Security Vulnerability, not only pointed out the potential for security issues with VoIP, but how companies like Cisco have to stay ahead of the security threats. So even large companies like Cisco must keep vigilant to ensure its systems are always “hacker-proof.”