In this article, we will look at what RTP (Real-Time Transport Protocol) is, where it came from, what makes it unique, and how it sends data little by little. We will also explain how it works step by step.
What is the RTP Protocol for Online Media Communication?
RTP (Real-Time Transport Protocol) sends multimedia content over a network in real-time. This standard adds segments, sequence numbers, and time labels to the info before sending it to the next step.
They initially developed the RTP protocol to broadcast real-time traffic to multiple points. Also, it performs for single-point broadcasting.
Moreover, people use this standard for extra services like videos, IP phones, and VoIP (Voice over IP) for online talking.
RTP gives us info about everyone in the online meeting and keeps the quality of data transfer good. Thus, it uses a helper control protocol called RTCP.
It also recreates the content on the receiver without waiting for the process to download the entire range of a large data segment.
Because it’s a standard protocol, it supports coding, timestamps, sequence numbers, and other parts. This way, it delivers data to the destination most suitably and securely.
The RTP protocol, which they standardized in RFC 1889, supports various audio and video extensions. Some of these are standard formats like WAV, GSM, or MPEG. It can also be performed in private ways.
RTP History and Development
RTP, which allows us to communicate live with our close circle in our daily online work, dates back to the 1990s.
They felt the need for online creations as the Internet became more common. Precisely, they needed something for transmitting sound and video to users. So, the Internet Engineering Task Force (IETF) started working on RTP.
As a result of their efforts, in 1996, the IETF introduced RTP in its first official document, RFC 1889. The initial version primarily aimed to transmit audio and data on the communication channel. However, with subsequent developments, they made significant progress with RFC 3550 in 2003.
Among these advances, error management stands out as the most remarkable. In addition, they increased codec support with RTP/AVP (Audio-Visual Profile).
Initially, RTP focused on providing only audio and video transmission. But as time passed, it began supporting things like meetings and live videos. At this point, it boosted online gaming in the computer game industry. Additionally, corporate environments frequently started using IP phones.
In the finale, RTP is vital for today’s web-based apps and services. That’s why it works with the Internet of Things or IoT. This means all the cool devices we have nowadays.
RTP Features
It uses and manages some fundamental structures while transmitting real-time data on the web. Here are some of these features:
- Timestamping
It sends each packet marked with timestamps to the receiving device so that it can play them at the right time. In this way, it corrects packet delays during the timestamping process.
- Sequence Numbering
The RTP protocol enables data to be sent to the receiving device sequentially, ensuring it receives the same sequential order on the other side. This prevents loss and delays, providing the sent packets are received reliably.
- Payload Type
To determine the type of data, use this header. In other words, it defines the kind of multimedia between two receivers. In short, the Payload field helps the user.
- Synchronization Source Identifier (SSRC)
One of its most important features is the SSRC ID. RTP recognizes the remote device using it in real-time streams. For instance, each device identifies with this method in a multi-user conference session.
- Marker Bit
RTP focuses on specific moments using the marker bit value. In this way, it determines the frame structure in video traffic.
- Error Detection and Correction
As we know, the UDP protocol does not provide security in data transfer. Therefore, RTP uses CRC (Cyclic Redundancy Check) while UDP exchanges real-time info.
- Standalone Codec Support
RTP helps different apps work together smoothly by letting them use other types of sound and video.
Understanding the Working Logic of RTP
RTP is mainly used to send data in real-time. It does this by including specific info for a particular app. It also changes the necessary parts and organizes them for sending.
Moreover, it works based on your app instead of being restricted to a specific layer.
This protocol uses UDP instead of TCP because TCP is unsuitable for apps needing real-time contact. It simply assigns user ports for each type of media sent.
Afterward, it sets up and sends the data created using any coding method correctly for transmitting data.
It can send data to one specific place or many places at once. It also sets up resources that can repack data if needed. In simpler terms, it uses numbers to figure out if any data is missing in the stream.
How Does RTP Work?
The main goal of the Real-Time Transport Protocol (RTP) is to send and receive audio and data online. Breaking data into smaller pieces called packets can send both small and large amounts of data.
It also includes a timestamp in these prepared packets. So, the sending device informs the receiving device when it sends the package. This way, the receiving person or device organizes the stream data accordingly.
After this process, the device that receives the packets simultaneously plays the video or audio smoothly.
However, the receiving device checks the sequence numbers of the data it receives before collecting and playing them. This ensures that we place the multimedia content in the correct order.
Afterward, it looks at the payload info inside the packet and recognizes the content. This data could be a video, audio, or something else. In this case, the receiver understands what is what by looking at it.
After all these processes, it uses a method to check if there is an error in the data stream. If there is a packet loss due to the network, it corrects them.
As a result, RTP will work in the background when you have a live chat with a friend from your PC while connected to the Internet.
Usage Areas of RTP Protocol
The Real-Time Transport Protocol can be used differently depending on your needs. But the most important ways to use it are:
- Audio and Video Conferences
The best thing about RTP is that it lets you do video calls. Using this protocol, you can chat with people globally in real moments.
- Live Broadcasts
Live broadcasts on TV, which are a part of our lives, are made possible through RTP. This lets you watch a real-time event, like a concert prepared in one location.
- Video Games
RTP ensures low latency for PC gamers worldwide when playing games with friends.
- Medical Applications
In the health sector, doctors can control the status of their patients through a remote connection.
- Training Area
In the always-changing world of education, this super-advanced protocol helps teachers quickly teach their students remotely.
- Collaboration Tools
Whether big or small, employees who work for companies can have project meetings online to make them more accessible.
- Telecommunications Industry
In telecom, they provide voice transmission using RTP in call centers.
- Security Area
They use the real-time transmission protocol to make security cameras in buildings work correctly.
Real-Time Transport Protocol Packet Structure
The Real-Time protocol always uses a 12-digit structure at the beginning of the message. But, during a video conference, it uses particular nodes that provide info at specific times.
After the header, it contains optional additions. However, the source determines the data carried by RTP.
The RTP header design allows carrying only the fields required for various types of apps.
- Version (V) – 2 Bits: The first two bits identify the version of the protocol.
- Padding (P) – 1 Bit: It declares that RTP data has padding to fill a specific size block. The last byte in the UDP message refers to the padding size.
- Extension (X) – 1 Bit: It indicates an extension header.
- CSRC Count (CC) – 4 Bits: It tells how many markers there are after the fixed header.
- Marker (M) – 1 Bit: It points out particular things.
- Payload Type (PT) – 7 Bits: It explains how the application should understand the info.
- Sequence Number – 16 Bits: The person sending stuff adds one to it every time they send something, and the person receiving it uses it to find out if anything got lost.
- Timestamp – 32 Bits: It shows when the first eight bits of the sent data were captured.
- SSRC – 32 Bits: It tells us where the package comes from.
- CSRC – 32 Bits: The value mixers show if the info has been changed.
Frequently Asked Questions (FAQ) About RTP
- What is RTP, and What Does It Mean?
- What Uses RTP?
- Why Do They Use RTP?
- What is an Example of RTP?
- Is RTP the Same as TCP?
TolgaBagci
Hi, I'm Tolga, a computer expert with 20 years of experience. I help fix computer issues with things like hardware, systems, networks, virtualization, servers, and operating systems. Check out my website for helpful info, and feel free to ask me anything. Keep yourself in the loop about the newest technologies!
