Introduction
As we already know, Internet services can be used for several functions or operations. One of the services provided by the Internet, can be used to create a voice communication from one device to another device located in either the same building or in different cities across the world, as does by the normal landline calls.
You might have heard about VoIP, which is an acronym for Voice over Internet Protocol or in more common terms, phone service over the Internet. This VoIP is used to create or make a common phone call as with a normal mobile phone service carrier. If you have a reasonable quality Internet connection, you can get phone service delivered through your Internet connection instead of from your local phone company service carrier. The VoIP was originally launched on the Internet in 1995.
VoIP Technologies
In this article, we will explain what VoIP is, how the service works, meaning of some technical terms and protocols associated with VoIP, and end with some examples of VoIP services/software.
Enjoy the reading!🙂
Definition and Overview of VoIP
Voice over Internet Protocol (VoIP), also called IP telephony, internet telephony, or internet calling, is a method and collection of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the Internet, rather than through the public switched telephone network (PSTN), also known as plain old telephone service (POTS).
At the simplest level, a VoIP phone system is a way of transmitting voice calls over IP networks. It is a means of making phone calls using an internet connection, rather than making a call using a landline.
VoIP telephony
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A voice over IP converts voice into a digital signal, compresses the converted digital voice, and sends it over the internet. A VoIP service provider sets up the call between all the participants. On the receiving end, the digital voice data is then uncompressed into the sound that can be listen through the handset or speakerphone.
How Does VoIP Work?
A VoIP phone system is a technology to make phone calls through your internet connection instead of a regular landline or a mobile network. A VoIP system converts analog voice signals into digital signals over your Internet broadband connection. A VoIP server is used to connect calls to other telephone networks.
With a regular phone call, a specific physical path is provided by a service provider; usually a dedicated phone company (Public Switched Telephone Network - PSTN). That path goes between yourself and the number that you call. The regular phone call system also utilises the traditional telephony infrastructure, which means the network of phone lines crossing the country.
With a VoIP service, calls are transmitted differently. The audio at your end of a call (your voice) is converted into digital packets. It's easiest to think of this digital packet as being like a single unit of data package in an envelope, in the same way as traditional envelopes contain what you've written.
The steps and principles involved in originating VoIP telephone calls are similar to traditional mobile network and it involve signalling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, the digital information is packetized and transmission occurs as Internet Protocol packets over a packet-switched network. A typical VoIP configuration involves a desk phone and a Session Initiation Protocol (SIP) server, which is typically a VoIP service provider.
It transports media streams using special media delivery protocols that encode audio and video with audio codecs and video codecs. Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on narrowband and compressed speech, while others support high-fidelity stereo codecs.
The most widely used speech coding standards in VoIP are based on the linear predictive coding (LPC) and modified discrete cosine transform (MDCT) compression methods.
The conversion of audio voice signals into digital packets is handled by 'codecs'. Codecs can be either hardware devices or software-based processes. They compress the voice signals and then encode them as digital signals. The codec takes what you say and pop it into the digital envelopes.
The data packets are then transmitted via IP. This can be either across a Local Area Network (LAN) or online. They're often transmitted via the Real-Time Transport Protocol (RTP). Or, if not, via the Secure Real-Time Transport Protocol (SRTP). The latter is simply an encrypted version of the former. This stage of the process is like a postman picking up your envelopes and taking them to the destination.
The data packets reach their destination almost instantly. The data packets then need to be decoded and decompressed at the destination point. This is handled by codecs. The codecs take the digital data and convert it back to audio signals. The recipient of your call heard your voice as they would down a normal phone line. The codecs at their end of the transmission open the envelopes for them to read.
Apart from the technicalities, the process of making a VoIP call doesn't have to be much different to a standard phone call. To make the calls, you can either use hardware or a software-based VoIP phone.
The hardware-based VoIP phone is very much like a traditional desk phone or handset. It will look almost identical and can be used in all the same ways. That doesn't only mean making calls. Most VoIP phones also let you use voicemail, make internal calls, employ call routing systems or auto-attendants, and call transfer tasks. Some even have screens, allowing for video conferencing.
Software-based VoIP phones are often called 'softphones'. They're apps or programs that can be install on a mobile phone or computer. The interfaces of those apps or programs replace a traditional phone handset. They're often designed to look similar and can be used either via a touchscreen or keyboard. Calls through these phones typically used a headset and microphone. They can also use a computer's built-in microphone and speakers. Calls and SMS text messages may be sent via Wi-Fi or the carrier's mobile data network.
Messenger calls
Examples of VoIP
Today, there are many ways you or your business organization and employees can use VoIP to communicate better with three different types of VoIP services, which will enhance the communication experiences. These three types of VoIP communication are:
- Computer-to-Computer
- Computer-to-Phone
- Phone-to-Phone
Every time you use your Mac or Windows computer to call someone using the internet, you are using VoIP. For instance, when you use Skype or Facebook Messenger, you are using VoIP software.
Here are the most common examples of VoIP applications:
- Nextiva
- Aircall
- Zoiper
- Skype
- Google Meet
- Viber
- Facebook Messenger
- Zoom
Whatsapp calls
Protocols Used in VoIP Service (VoIP Protocols)
Voice over IP has been implemented with proprietary protocols and protocols based on open standards in applications such as VoIP phones, mobile applications, and web-based communications.
A variety of functions are needed to implement VoIP communication. Some protocols perform multiple functions, while others perform only a few and must be used in concert. These functions include:
- Network and transport: Creating reliable transmission over unreliable protocols, which may involve acknowledging receipt of data and retransmitting data that wasn't received.
- Session management: Creating and managing a session (sometimes glossed as simply a "call"), which is a connection between two or more peers that provides a context for further communication.
- Signaling: Performing registration (advertising one's presence and contact information) and discovery (locating someone and obtaining their contact information), dialing (including reporting call progress), negotiating capabilities, and call control (such as hold, mute, transfer/forwarding, dialing DTMF keys during a call [e.g. to interact with an automated attendant].
- Media description: Determining what type of media to send such as audio or video, how to encode/decode it, and how to send/receive it (IP addresses, ports, etc.).
- Media: Transferring the actual media in the call, such as audio, video, text messages, files, etc.
- Quality of service: Providing out-of-band content or feedback about the media such as synchronization, statistics, etc.
- Security: Implementing access control, verifying the identity of other participants - computers or people, and encrypting data to protect the privacy and integrity of the media contents and/or the control messages.
- Session Initiation Protocol (SIP): Connection management protocol developed by the IETF
- H.323: One of the first VoIP call signaling and control protocols that found widespread implementation.
- Media Gateway Control Protocol (MGCP) - Connection management for media gateways.
- H.248: Control protocol for media gateways across a converged internetwork consisting of the traditional PSTN and modern packet networks.
- Real-time Transport Protocol (RTP): Transport protocol for real-time audio and video data.
- Real-time Transport Control Protocol (RTCP): Sister protocol for RTP providing stream statistics and status information.
- Secure Real-time Transport Protocol (SRTP): Encrypted version of RTP.
- Session Description Protocol (SDP): A syntax for session initiation and announcement for multi-media communications and WebSocket transports.
- Inter-Asterisk eXchange (IAX): Protocol used between Asterisk PBX instances
- Extensible Messaging and Presence Protocol (XMPP): Instant messaging, presence information, and contact list maintenance.
- Jingle: For peer-to-peer session control in XMPP.
- Skype protocol: Proprietary Internet telephony protocol suite based on peer-to-peer architecture.
Definition of VoIP Technical Terms
Considering the number of collections for so many technical jargons associated with the Internet telephony services, it will be appropriate to briefly explain some of these terms to a newbie in VoIP journey.
- Bandwidth:
The capacity of a network to transmit data from one point to another in a given time period. It is often measured in 1000 bits per second (kbps). The higher the amount of available bandwidth, the more VoIP calls a network can support. If your internet connection has very limited bandwidth, the quality of a call will be lower. - Codecs:
Codecs can be either hardware devices or software-based processes. They’re used to compress, encode and decompress data. In the case of VoIP, codecs convert audio voice signals into digital data packets. They then compress these digital signals for transmission and re-convert it at the ‘other end’ of a call. - DSL:
DSL stands for ‘Digital Subscriber Line’. It refers to the traditional phone technology that lets a broadband connection be carried over existing phone lines. All the while still allowing analogue phone signals to travel along the same copper lines. - IP:
The 'IP' in 'voice over IP' is an acronym which is short for Internet Protocol. The IP provides a standard set of rules for transmitting and receiving data online. These standardised rules let devices running on different platforms communicate with one another. The IP also provides basic rules for transmitting packets of data. It does not establish the connection for doing so or order the packets being transmitted. That’s handled by transmission or transport protocols. - IVR:
IVR or Interactive Voice Response is a feature of traditional telephony. It’s the interactive service that lets callers use menus and handles call forwarding and call transfers - basically, an upgraded version of an auto-attendant. Think ‘press one for the accounts department’ etc. - Latency:
The time taken for the transmission of data. The higher the latency, the greater the delay between the start of a transmission and data being received at the other end. High latency can be an issue for VoIP - especially if you're using it for video conferencing calls. Voice delay is noticeable with a latency above around 150 milliseconds. Considerably more than that and a conversation will be difficult. - PBX:
PBX stands for Private Branch Exchange. It’s the name given to a private telephone network used within a business or organisation. It’s your PBX which lets you press a button on a desk phone to reach someone else in your office. - RTP:
The Real Time Transport Protocol. An internet protocol that often transmits the data packets related to VoIP calls. It also carries audio or video streams for other forms of multimedia communication. Secure Real Time Transport Protocol – The encrypted (and so, secure) variant of the RTP. - SIP Trunking:
A way of delivering voice communication over the internet. It’s in many ways an alternative to VoIP. It typically involves connecting a PBX to the internet. This gives added control to a user but does require a fair bit more equipment to get up and running than a VoIP phone system. - Softphone:
The name given to software or apps used to make VoIP calls. In the case that you use your computer, mobile device, or tablet and not a VoIP phone. Most ‘softphones’ have an interface that looks like a phone handset when it appears on your screen. It will have a keypad, display and be workable via touchscreen or computer keyboard. - VoIP:
Voice over Internet Protocol is the technology that lets you make voice calls via an internet connection. Audio is repackaged as digital data and transmitted to its destination almost instantly. It’s becoming a new business communications standard for firms of all shapes and sizes.
VoIP phone services works even better than the traditional landline phone because it offers many more features than what analog phone service could ever provide.
There is no doubt, the buzz and impacts VoIP is creating today is the future in the making. The internet is more than a place where we make search queries, it has become a modern system where we can connect with others on multiple levels.
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