All right, let's dive deeper into Voice over IP, its components, and
their functioning. In this module, we are going to explore some of the
most popular protocols and different standards, so you could gain a
deeper understanding of different protocols,
commonly used Codecs, and how they deliver the actual traffic.
Additionally,
we will cover the important aspect of Quality of Service,
QoS, that guarantees amazing media quality.
Let's start with the first protocol, which is the SIP protocol. SIP is
widely used in VoIP systems to initiate, establish, and manage
communication sessions for voice and video calls. It is a signaling
protocol responsible for setting up, modifying,
maintaining, or terminating the calls.
For instance,
if you want to hold an ongoing call to check your emails or notes for
information, if you want to initiate a conference call by adding a
third person to the ongoing call, or if you want to end the call, the
SIP protocol does it all for you.
Imagine you're trying to call your best friend on the telephone.
Instead of using the traditional phone lines, you are
using the internet to make the call.
That's what SIP is all about.
SIP stands for Session Initiation Protocol, and it's like
the traffic cop on the internet to make sure your voice
call gets to your friend's phone.
Just think about it, you are at a big party and you're looking for your
friend, so you ask the host where your friend is. The host here acts
like the SIP and tells you which room your friend is in. In the same
way, SIP is the traffic cop on the internet that helps you find your
friend's phone so you can talk to them. SIP set up the call between you
and your friend's phone, and makes sure that the call quality is good.
So the next time you make a VoIP call, remember, it may be the
SIP protocol that's connecting you to your friend's phone.
Another VoIP protocol is H.323 protocol.
It's a communication standard used for audio and video calls
over the internet. Any protocol is a set of rules needed by
computers to talk to each other.
When you make a voice or video call using H.323, your computer and computer of
the person you're calling at the destination use this protocol to communicate
and coordinate with each other for a call signaling. It helps the computers
understand each other and work together to make that call happen. Just like
the way you and your friends agreed on certain rules for playing any game,
computers that use H.323 have agreed on a set of rules for making audio and
video calls.
This makes it easier and smoother for them to communicate and allow
you to make a call without any confusion or issues.
Okay, let's talk about the RTP protocol, which stands
for real‑time transport protocol.
This protocol serves a different purpose and is used to send and receive
audio and video in real time over the internet. RTP works by taking audio
and video data and dividing it into small packets. Each packet is then
sent over the internet to the recipient. The recipient then takes all of
the packets and put them back together, just like putting a puzzle
together, to form the original audio or video. RTP also keeps the quality
of the audio and video high. It does this by adding extra information to
each packet, like the time it was sent and the order of the packets.
This way the recipient knows the right order to put the packets back
together, and the audio and video quality stays high. In short, RTP is
like a delivery truck that takes your audio and video and delivers it in
real time to your friends, making sure that it arrives in the right
order and with high quality.
However,
keep in mind that the RTP streams are typically delivered over
UDP, which is an unreliable transport mechanism.
Hence, there is no guarantee of packet delivery.
RTP focuses on speed rather than reliability, and it is
optimized for low latency and jitter,
making it well‑suited for real‑time communication. RTP is often used
in conjunction with other protocols such as SIP and H.323 to transmit the actual media streams in VoIP systems.