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Voice over Internet Protocol (VoIP) is one the fastest growing telecommunication technologies of 2019. The VoIP system converts analog audio signals into digital data which is transmitted over the internet to allow free phone calls. Additionally, VoIP services now include video conferencing, group chats, and integration with Customer Relationship Management (CRM) systems.
Almost all businesses have switched from traditional landlines to VoIP for cost-saving. Research says companies save somewhere between 50% – 70% on average when they switch to VoIP. The industry is booming and is expected to reach 204.8 billion corporate users by 2020, who will account for $86.2 billion in revenues.
Your organization very likely also uses VoIP for communication. The systems have become increasingly complex, so it pays to understand how they work and make the best use out of them.
Three Types of VoIP
IP phones look just like traditional phones but don’t require an external adapter to convert the signals into digital data. In place of the RJ-11 socket, they have an RJ-45 connection and are connected directly to a router.
The majority of the VoIP systems use analog Telephone Adaptor (ATA).. An ATA is a small device that converts analog signals into digital signals. You connect one end of the ATA to a landline phone and the other to an internet router or computer. Some ATAs come with software that needs to be installed on your computer for added security.
This method doesn’t require any installation of hardware at all. By downloading and installing VoIP software on your computer, you can make computer-to-computer calls.
Circuit switching has been used for decades to place calls between two parties. With a circuit switch, the connection between two callers remains open for the entire duration of the call, and no other data can be transmitted. Sounds are converted into digital data and transmitted through a complicated network of fiber optic cables along with other voice calls. The connections between the two ends of the line remain open with a consistent transmission rate of 64KB/second. A 5-minute phone call transmits approximately 5MBs of data across both ends.
VoIP uses the more advanced technology of packet switching. Packet switching only transmits data when there is noise on either end of the connection. Packet switching cuts data transmission in half since only one party speaks at a time. Just like internet connections, VoIP breaks down the data and packages ‘payloads’ inside numerous ‘packets.’ These packets are sent over the internet through the least congested paths. Each packet takes a different route to reach its end destination. The built-in receiver uses the header file to arrange the packets in order and decrypts them. Packet switching is very efficient since it cuts the data into half and doesn’t require dedicated circuits to transmit data. Additionally, packet switching leaves computers and phones free to receive data from multiple sources at a time.
VoIP in Action
Here’s an example of how a typical office call works when using VoIP:
- The ATA receives a signal as you pick up the receiver.
- The ATA transmits a signal (dial tone) that lets you know you’re connected to the internet.
- As you dial your colleague’s number, the ATA stores the data temporarily.
- The number you dial is sent to the phone provider of your VoIP service company. The number is verified.
- The phone provider uses mapping to convert the phone number into the relevant IP address. The two devices are connected using a soft switch and your colleague’s phone rings.
- As your colleague picks up their phone, a session is initiated, so both devices expect to receive data packets from the other source and assemble them.
- When either of you speaks the analog signal is converted into data packets and transmitted to the other end where they’re converted into an analog signal through ATA or IP phone.
- At the end of the conversation, you hang up the phone.
- The soft switch receives a signal from the ATA or IP phone and terminates the session.
The quality of your phone call depends on the Internet Service Provider and the bandwidth you’ve chosen.
A codec, short for coder-decoder converts analog signals into digital data that can be transmitted over the internet in the form of packets.
Different codec sample the analog signal at different rates. Sampling is used to break down the audio into distinct sample pieces. The rate of sampling determines the quality of sound. Lower rates miss out on data but even at the lowest rates, humans are unable to recognize a loss of audio. Some examples of rates of codec sampling are:
- 8,000 times/second
- 32,000 times/second
- 64,000 times/second
Most VoIP services use the G.729A codec, which samples audio signals at the rate of 8,000 times in one second. Even though there’s only a slight difference between the audio qualities, you can opt for a higher sampling rate when selecting your VoIP service package.
The rule that determines whether to transmit data or not is based on the absence of noise is called CS-ACELP. This algorithm drastically improves the efficiency of the VoIP system.
A numbering system is required to filter phone calls and route them to their designated locations. In the U.S., the North American Numbering Plan (NANP) is responsible for the numbering system.
However, VoIP technology uses IP addresses instead of the traditional NANP system. IP addresses are assigned at random by the DHCP server on the network, each time there’s a new connection the IP address changes. The VoIP system first translates the NANP number to an IP address and then finds the correct IP address of that number. The translation of NANP number to IP addresses and the identification of the IP address is handled by a soft switch running a mapping process.
The soft switch is installed on the central call processor of the VoIP service provider. The soft switch database contains information on VoIP users and their numbers. If the database doesn’t contain a user’s information, the soft switch places a request to other soft switches. Upon response from a soft switch request, the user’s IP address is identified, and the information is relayed to the IP phone.
Protocols are the set of transmission standards used by devices to communicate over the internet. Different protocols have different rules regarding how codec, devices, and soft switches work in combination. To avoid compatibility issues, the International Telecommunication Union (ITU) has created a standard for all VoIP services to use: H.323.
The H.323 is a set of complex protocols that specify configurations for phone calls, video conferencing, data sharing and other applications.
With the increase in standardization of protocols and technological advancements, VoIP services today offer a lot more than just phone calls. New VoIP providers offer features like auto-attending, video conferencing, call parking, call forwarding and integration with CRM software. You can use this web page to compare the most popular VoIP providers of 2019.