As we mentioned above, VoIP is the technology that allows you to make and receive calls over the internet. In fact, this is the telephony component of most top cloud voice solutions (like RingCentral):
The goal: to replicate traditional telephones as much as possibleóand, in many ways, improve on those traditional phones. It does this through packet switching technology and multiple internet protocols working together to duplicate certain telephone functions.
Sounds complicated right?
Letís try to uncomplicate things by breaking down how VoIP works and how its many components work together.
Packet switching technology
So, we mentioned packet switching. This is actually how VoIP replicates traditional telephone networks or public-switched telephone networks (PSTN). Instead of copper-wired networks, VoIP uses computer networks to transmit calls.
These computer networks include local area networks, personal networks, and of course, the internet, which is the biggest computer network in the world:
Packet switching technology
What goes on is that voice data (yours in this case) is broken down into small packets, then it travels across various networks, including the internet, until it reaches its destination (the person youíre calling) where itíll be assembled again to your voice data.
Then it goes in reverse: the voice of the person youíre calling travels back to you in the same order until you hear their voice on your end.
Types of VoIP protocols
To make all this happen, VoIP needs a little help. This is where internet protocols come in. Internet protocols are rules that you apply for routing data. This way, your voice and the voice of the person youíre calling will follow certain rules so it wonít get lost in the network.
The most important VoIP-related protocols are:
Transport Control Protocol (TCP) or User Datagram Control (UDP): These two are the most common delivery protocols used in VoIP. They dictate the behavior of how the data will be deliveredódo you prioritize speed or quality?
If you want speed, UDP is the preference. What happens is that even if the connection isnít established, itíll deliver the data regardless. So, if you or the person you called has a bad connection, the voice data will still be delivered. Whether you receive it or not is the problem. That is why most IT people call it the ìsend and prayî method.
TCP, on the other hand, is all about quality delivery. It establishes a connection first before delivering data. Itís all about making sure that your voice data is delivered and received. By making sure the connection is established, however, speed is sacrificed.
Real-Time Transport Protocol (RTP): If TCP or UDP is the delivery guy, RTP is the lifter. This protocol determines how your voice data is carried over to the person youíre calling. It tells the receiving end the media type format of the audio (your voice), the security that protects your voice data, sender identification (the IP address where data is coming from), and more.
Session Initiation Protocol (SIP): We touched on this before, but SIP is the one responsible for emulating most of the functions of a regular telephone. If TCP/UDP and RTP are all about delivery and transport of your voice, SIP takes care of everything else like dialing, establishing a connection, and terminating the call.