What Is VoIP, Really?

VoIP From the Ground Up: Part 5 of 10

We have spent four posts building up to this. You understand how phone calls used to work over dedicated copper circuits. You know what a network is and how data moves across it in packets. You understand why voice is uniquely difficult for a packet-based network.

Now we can talk about what Voice over Internet Protocol actually is, not as a marketing term, but as a mechanical process.

The basic idea

VoIP is the process of taking a live human voice, converting it into digital data, splitting that data into packets, sending those packets over an internet protocol network, and converting them back into audible sound at the other end. All in real time.

That is the whole idea. The complexity is in the execution.

Step by step

Here is what happens from the moment you speak into a VoIP phone to the moment the other person hears you.

Step 1: Capture

Your phone's microphone picks up the sound waves from your voice and converts them into an analog electrical signal. This is identical to what happens on a traditional phone. The microphone does not know or care whether it is connected to a copper phone line or a digital network. It just turns sound into electricity.

Step 2: Digitize

The analog signal from the microphone is converted into digital data. A component called an analog-to-digital converter samples the electrical signal thousands of times per second and records each sample as a number. The standard rate for voice is 8,000 samples per second, though some systems use higher rates for better quality.

This gives you a stream of numbers that represent the shape of your voice's sound wave at each instant. This is raw digital audio.

Step 3: Compress

Raw digital audio uses a lot of bandwidth. Sending it uncompressed across a network would consume far more resources than necessary, especially since human speech has a lot of predictable patterns that can be represented more efficiently.

A codec (short for compressor-decompressor) takes the raw digital audio and compresses it. Different codecs make different trade-offs between audio quality and bandwidth usage. Some prioritize crystal-clear audio at the cost of higher bandwidth. Others squeeze the audio down to a fraction of the size while still sounding acceptable.

Common VoIP codecs include G.711, which uses 64 kilobits per second and sounds very close to a traditional phone call, and G.729, which compresses down to about 8 kilobits per second at the cost of some audio fidelity. Newer codecs like Opus can adapt dynamically, adjusting compression on the fly based on network conditions. For more on how codecs affect quality scoring, see our post on MOS scores.

Step 4: Packetize

The compressed audio gets broken into small chunks, typically 10 to 30 milliseconds of audio per chunk. Each chunk is placed inside a packet along with header information that tells the network where it came from, where it is going, and what order it belongs in. An additional protocol layer called the Real-time Transport Protocol adds timing and sequencing information specifically designed for media streams.

At this point, your voice has been transformed from sound waves into numbered, addressed packets of data, ready to travel across a network alongside web traffic, email, and everything else.

Step 5: Transmit

The packets leave your device, pass through your local network equipment (switches, router, firewall), cross your internet connection, enter your internet service provider's network, traverse the broader internet through multiple hops, and eventually reach your VoIP provider's infrastructure. From there, the packets are routed to the other participant's connection and make the reverse journey into their local network and onto their phone.

Each packet travels independently. As we covered in Part 3, they may take different routes. They may experience different amounts of delay. Some may not arrive at all.

Step 6: Buffer

The receiving device collects the arriving packets and places them in a jitter buffer. This is a small holding area where packets wait briefly so the device can sort them into the correct order and smooth out variations in arrival timing. The buffer introduces a small amount of deliberate delay in exchange for more consistent audio playback.

Step 7: Decompress

The codec on the receiving end takes the compressed audio data from each packet and reverses the compression process, reconstructing the digital audio stream. If any packets were lost, the codec may attempt to fill in the gap by estimating what the missing audio sounded like based on the surrounding packets. This is called packet loss concealment.

Step 8: Play

The digital audio is converted back to an analog electrical signal, which drives the speaker in the receiving phone's handset or headset. The listener hears your voice.

This entire sequence, all eight steps, is happening continuously in both directions for the entire duration of the call. And the total time from your mouth to the other person's ear needs to stay below about 150 milliseconds for the call to feel natural.

What VoIP replaced

Looking at this process, you can see both the appeal and the risk of VoIP compared to the traditional phone system.

The traditional system had a dedicated circuit from end to end. The local loop between your phone and the central office was analog, but the trunk lines between central offices were digital, using PCM encoding since the 1960s. Still, there was no compression, no packetization, no buffering, and no competition with other types of traffic. A dedicated circuit carried your call the whole way. Simple, reliable, and limited to one purpose.

VoIP replaced that with a complex chain of conversions, compressions, and routing decisions that all have to execute flawlessly in real time. The upside is that voice can now travel over the same infrastructure as everything else, saving enormous amounts of money and adding flexibility that was never possible with circuit switching. The downside is that every step in the chain is a potential failure point.

The finger-pointing problem

Here is the practical reality that most VoIP users eventually face.

When a call sounds bad, the problem could be at any point in the chain. Your local network could be congested. Your internet connection could be dropping packets. Your internet service provider could have a routing issue. The path between your provider and your VoIP provider could be congested at a peering point. Your VoIP provider's infrastructure could be overloaded.

You will call your VoIP provider and they will tell you the problem is your internet connection. You will call your internet service provider and they will tell you the connection looks fine on their end. Neither one is necessarily lying. They just cannot see the full picture.

This is the core problem that VoIP Test was built to address: our free VoIP quality test gives you independent data that shows exactly where in the chain the breakdown is happening.

What comes next

The next two posts zoom in on the two halves of the path. First, we will look at your local network, the part you own and control, and what you can do to optimize it for voice traffic. Then we will look at the internet side, the part you do not control, and what that means for diagnosing problems.

This is Part 5 of the VoIP From the Ground Up series. Part 6: Your Local Network, The Part You Control covers the equipment and configuration that directly impacts call quality inside your building.