How Does a VoIP phone System Work?

VoIP’s presence has changed the way businesses and consumers use voice communication.

It has never been possible to make cheap calls whether national or international but VoIP has introduced features that bring cheap voice calling into the 21st century.

VoIP phone system works very differently as compared to the analog phone systems which have been used for decades.

First, VoIP does not need a separate infrastructure or physical phone lines. However, VoIP functions over an existing framework which is utilized for delivering internet all over the world.

Therefore, it is known as Voice over Internet Protocol- VoIP phone calls are transmitted via the internet instead of copper lines. Although VoIP provides a lot of benefits, the biggest advantage of VoIP is that it is based on the Internet protocols instead of telecommunication standards.

While VoIP has gradually been rising in popularity and usage, not many people recognize how it functions. Even though the regular customer can get by without really understanding how VoIP functions, it is necessary that business executives understand the VoIP before implementing it.

Packet Switching Vs. Circuit Switching

Usually, phone calls made through a landline travel by TDM circuits to connect two ends of the call, over long and short distances. For landline calls, the voice signals have to be transmitted and the circuit has to be held up throughout the duration of the phone call. On the contrary, VoIP used packet switch technology which is similar to the way data is sent through the internet like email.

As the Internet was not actually intended to facilitate real-time voice communication like VoIP, a number of protocols were developed and implemented to make voice calls over the last few years.

Fundamental Functioning of VoIP Phone Systems

Throughout a VoIP call, the voice signals are transformed into data packets which then travel separately from each other to the call destination. There the data packets are reassembled and transformed back into audio signals that can be heard by the receiver. In general with Internet data, it doesn’t matter in what sequence the data packets are received and if a number of data packets are dropped, the missing packets will just be resent. Nonetheless the same doesn’t hold true for real-time communication. The data packets have to be reassembled in a particular sequence so that it makes sense to the other person. In addition, missing data packets can also cause silence or choppy conversations. Each tool that is registered on a VoIP network has an exclusive IP address that is dynamically assigned which means it is not lasting as a phone number. When a VoIP call is dialed, a signal is sent to a soft switch to know the current IP address of several VoIP endpoints such as computers, desk phones etc. if a specific soft switch doesn’t get the information then the request gets passed on till it gets to a soft switch which does have the data. As the other endpoint is located, VoIP establishes a connection so that the two-way communication can start.

Elements of the VoIP System

Codecs

These are software algorithms that compress audio signals and transform them into data packets which are then transmitted over the Internet. In addition, the similar algorithms work at the destination to transform data packets into audio. A number of codecs do not compress the data which can enhance audio quality although end up using a huge sum of bandwidth for one phone call. To allow several simultaneous calls, the most frequently of used codecs depend on compression. Suppliers have to get the accurate stability between compression and quality in order to provide audible conversations that do not damage Internet bandwidth.

Protocols

Several pieces are involved in the VoIP network such as endpoints, soft switches, software, codecs etc. Protocols are required to make sure that these different hardware and software pieces can work mutually to complete a call. Protocols are used to define standards that state how devices can connect to each other, how endpoints are recognized along with which audio codecs to utilize. H.323 and SIP are two of the extensively utilized protocols however they differ drastically. Whilst H.323 was initially designed for video conference and was afterward modified for VoIP calls, SIP was particularly developed for enabling real-time voice communication over the Internet. In addition, the H.323 is a telecommunication standard and was produced by the International Telecommunication Union but SIP was standardized by the Internet Engineering Task Force.

QoS or Quality of Service

As VoIP uses packet switch technology, errors that influence the data packets can badly impact voice calls. Such as jitter, latency, packet loss etc. are frequent on the Internet however typical processes are usually unaffected.

On the other hand, these issues become important for phone calls. High latency means that callers will experience wait, the caller will begin talking thinking that the other person has stopped when they have not.

Jitter happens when data packets are received in the incorrect sequence or get postponed which can disrupt conversation flows. In the same way, dropped packets can cause missing words or occasionally complete sentences.

VoIP providers use call monitoring to make sure quality of service or QoS. A variety of algorithms are utilized to get the average quality of a call which is called MOS or Mean Opinion Score.

Companies that use hosted VoIP service are at the mercy of the providers when it comes to right quality of service settings however a number of modifications within their own network can also add to voice quality.

A large number of customers and companies switch to VoIP as it is reasonably priced as compared to conventional phone service. Although low costs are VoIP’s main strength, call quality is used to be its flaw. It is one of the major reasons why many people are switching VoIP providers.

On the other hand, the fast innovation within the VoIP business will make sure that the technology continues to get better and for the greater part of the world’s users, there is no going back to the simple old telephone system.