VoIP is one of the latest technologies creating a buzz…
How Does A VoIP System Work?
The introduction of VoIP has changed the way consumers and businesses utilize voice communication.
Making phone calls has never been cheaper and VoIP enables a whole suite of features that bring voice calling into the 21st century.
How VoIP phone systems work differs considerably from the analog systems that we have gotten used to over the last few decades.
For one thing, VoIP does not require separate infrastructure or phone lines. Instead it functions on top of the existing framework that is used for delivering Internet throughout most of the world.
This is precisely why it is called Voice over Internet Protocol – VoIP calls are transmitted over the Internet rather than copper lines. While VoIP offers many advantages, it’s biggest strength is that it is based on Internet protocols rather than telecommunication standards.
Even as VoIP has steadily been growing in popularity and usage, not a lot of people understand how it functions. Although the average consumer can get by without truly understanding how VoIP works, it is essential that business executives understand the VoIP phone system before implementing it.
Packet Switching Vs. Circuit Switching
Normally phone calls made on a landline travel through TDM circuits to connect two people, over long and short distances. The voice signal is transmitted as is and the circuit is held open for as long as the phone call continues.
VoIP on the other hand utilizes packet switching technology i.e. the same way other data is sent over the Internet such as email.
Since the Internet was not really designed to enable real-time voice communication such as VoIP, several protocols were developed and implemented to facilitate voice calls over the last few years.
Basic Functioning of VoIP Phone Systems
During a VoIP call, the voice signals are converted into data packets which then travel independently of each other to the destination. There they are reassembled and converted back into audio signals that can be heard by human beings.
Generally with Internet data, it does not matter much in what order the packets are received and if some packets are dropped – the missing ones will simply be resent.
However the same does not hold true for real-time communication – the packets have to be reassembled in a specific order in order to make sense to the person. Additionally missing packets can lead to silence or choppy conversations.
Every device that is registered on a VoIP network has a unique IP address that is dynamically assigned i.e. it is not permanent like a phone number. When a VoIP call is initiated, a signal is sent to a soft switch which knows the current IP address of various VoIP endpoints (desk phone, cell phone, computer etc.).
If the particular soft switch does not have that information, the request is passed on further until it reaches a soft switch that does have the data. Once the other endpoint is located, a connection can be established and two-way voice communication can begin.
Parts of the VoIP System
Codecs are software algorithms that compress audio signals and convert them into data packets that can be transmitted over the Internet. The same algorithms also work at the destination to reconvert data into audio.
Some codecs do not compress the data which can improve audio quality but end up utilizing vast amounts of bandwidth for one phone call. In order to enable multiple concurrent calls, most of the commonly used codecs rely on compression.
Providers have to strike the right balance between compression and quality in order to deliver audible conversations that do not strain Internet bandwidth.
A number of pieces are involved in the VoIP network – endpoints, soft switches, software, codecs etc. Protocols are necessary in order to ensure that these disparate hardware and software pieces can work together to complete a call.
Protocols are used to define standards that dictate how devices can connect to each other, how endpoints are identified as well as which audio codecs to use.
H.323 and SIP are two of the most widely used protocols but they differ significantly. While H.323 was originally designed for videoconferencing and later adapted for VoIP calls, SIP was specifically developed for enabling real-time voice communication over the Internet.
Additionally, the H.323 is a telecommunication standard and was created by the International Telecommunication Union whereas SIP was standardized by the Internet Engineering Task Force.
QoS or Quality of Service
Since VoIP utilizes packet switching technology, issues that affect the data packets can severely impact voice calls. For instance jitter, latency, packet loss etc. are common enough on the Internet but normal processes are generally unaffected.
However these issues become significant for phone calls. High latency means that callers will experience delays – the user at one end will start talking thinking that the other person has finished when they have not.
Jitter happens when packets are received in the wrong order or get delayed which can interrupt conversation flows. Similarly dropped packets can result in missing words or sometimes whole sentences.
VoIP vendors employ call monitoring in order to ensure quality of service or QoS. Various algorithms are used to determine the average quality of a call which is called MOS or Mean Opinion Score.
Businesses that utilize hosted VoIP service are at the mercy of the vendors when it comes to appropriate QoS settings but some changes within their own network can also contribute to voice quality.
Most consumers and businesses switch to VoIP because it is inexpensive compared to traditional phone service. While low prices are VoIP’s biggest strength, call quality is used to be its weakness.
It is one of the biggest reasons cited by people when they are switching VoIP vendors, especially by businesses for home voice communication is the backbone for various workflows.
Nevertheless the rapid innovation within the VoIP industry will ensure that the technology will continue to improve and for the majority of the world’s users, there is no going back to the plain old telephone system.
Recent blog posts
How Do I Know When It’s Time to Ditch Skype?
Skype was – and still remains – one of the most well-known services for making voice calls over the Internet. Long before VoIP became a household word and a necessary upgrade for businesses, Skype was the default choice for millions of users. Originally launched in 2003, the service quickly became popular before being purchased by…Read more
New VoIPstudio Salesforce Lightning Integration Announced!
When you think of innovative technology, enterprise communication is not the word that springs to mind. However this space has seen a lot of innovation and is completely transformed from what it used to look like in the previous century. In a world where customers expect companies to serve their needs at any time and…Read more
Meet the VoIPstudio team at the Chester Business Show! Sign up during the event and get a 90-day free trial of our VoIP service.
Are you looking for a hosted VoIP solution for your business but don’t know where to start? Meet us at the Chester Business Show UK on Wednesday 29th March, stand #112 and you can to talk to us in person about making that transition. Drop by our booth #112 and talk to one of our…Read more
30-day free trial
You can take a 30-day FREE trial of VoIPstudio with absolutely no obligations.
A simple web-based portal gives you total control over all your telephony requirements.
Ready to take your business to new heights?