VoIP and Internet Telephony Basics: Important terms and concepts

VoIP and Internet Telephony Basics: Important terms and concepts
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Voice over Internet Protocol is a set of protocols working together to provide telephony service thatís similar to regular landlines.
Before we go over the end-to-end process of how Voice over IP works, it is important that you familiarize yourself with some basic concepts that will help you understand how the technology works as a whole.
IP Packets for VoIP Transmission
Voice and data are encapsulated here. During a VoIP call, the voice or audio is captured and segmented into tiny samples which are then carried as payloads. These samples are attached to IP packets that will be transmitted between different networks, both public and private.
Codecs for conversion of analog voice data
Also known as codec-decoder, it converts analog voice signals (like your voice) into digital voice formats that are ready to be transmitted over the Internet. It also converts the digital form back to analog for the human ear.
Transmission Control Protocol (TCP) vs. User Datagram Protocol (UDP) for VoIP delivery
These are the two most common methods for delivering VoIP protocols over the Internet.
TCP-IP: packets sent using this method guarantees that data is delivered; and if itís lost, it will be transmitted. Think quality over speed.
UDP-IP: packets are sent over the networks with no guarantee it will be received. Think speed over quality.
Real-time Transport Protocol (RTP) for carrying VoIP audio stream
This protocol is responsible for carrying audio and video streams during VoIP transmissions. It carries the information about the file, including its format and media type.
Session Initiated Protocol (SIP) and SIP Servers for VoIP signaling
SIP is the protocol that is responsible for mimicking regular telephony. It is a standard signaling protocol that establishes, manages, and terminates real-time communications over IP networks.
SIP servers, on the other hand, accept SIP requests and respond to them. There are many types of SIP servers, with the Proxy server being the most known (since it facilitates connections to establish Voice over IP). Other SIP servers include Registration Server, Presence Server, and Redirect Server.
End-to-end process of VoIP
Using what we learned, here is an example of how a VoIP call works, with each step showing what is going on in the background for every action taken.
Person A calls Person B (the calling party uses the SIP signal to try to establish a connection via the Proxy server–RingCentral uses TCP to deliver SIP)
Person B answers the call (the called party accepts the request from Proxy Server using SIP)
Person A and Person B talk (IP packet containing voice data is delivered via UDP or TCP; RTP transmits the voice data, while Codecs convert analog voice to digital format and vice-versa. All these are happening in real-time.)
Person A or Person B ends the call (the SIP signal is used to terminate the connection)
Making VoIP calls to regular phone lines or PSTN through VoIP gateways
VoIP gateways are used to convert traditional telephony connections into VoIP connections and vice-versa.
What happens is that when calls are initiated from a traditional landline phone, the gateway will translate the multiplexed voice into packetized voice sample. It will then follow the same call process discussed above.
While thereís more to hosted VoIP transmissions than what is discussed here, this should give you a general idea of how the technology works.

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