Real-time messaging protocol (RTMP) is a widely used streaming format. It’s been around for many years and has grown to become an essential tool for broadcasters, network operators, and many other industries. However, some misconceptions about RTMP have made it less popular than it could be.

But what is RTMP exactly? How does it work? And should you be using it for your next live stream?

Find out about this and more below.

What Is RTMP?

RTMP is a network protocol or a system used to stream media content over the internet based on Transmission Control Protocol (TCP) technology. 

TCP is one of the components that make up the internet protocol suite. The other important component is the internet protocol, also referred to as IP.

RTMP is a network protocol or a system used to stream media content over the internet.

Together, TCP and IP act as communication bridges between the application and network layers. Think of it this way; the application layer encompasses what you would normally interact with, like Mozilla Firefox browser or any other user application. 

For your Firefox browser to load up a webpage, it must send a request to the website’s server. Upon receiving the request, the server sends the requested resource (i.e., a video stream, pre-recorded video on YouTube, or Html code for the webpage).

To maintain effective communication (i.e., avoid the correspondence from being dropped or delayed), the message must be dismantled into smaller pieces known as packets. This is done on the sender side, and once the message is received, it’s reassembled for the user.

TCP is the component that deals with breaking the message into packets or smaller pieces that can be transmitted effectively and efficiently.

The IP layer acts as the forwarding agent that determines the best routes for transmitting the packets over the internet. 

The RTMP protocol is used by many popular media players, including Adobe Flash Player, VLC Media Player, QuickTime Player, and Windows Media Player. RTMP is also supported by some web browsers, including Google Chrome and Mozilla Firefox.

The primary concern for most users with a streaming solution is how it delivers the content. If the streaming resolution quality is low, it’ll be a dealbreaker for most consumers. Similarly, a streaming solution with high latency and buffers or loads for too long before playing the content won’t fare well.

This is where RTMP shines. Since its development, RTMP has guaranteed low latency, minimal buffering, and one of the best streaming resolution qualities provided the network connection is strong and fast enough.

Another plus for RTMP is its ability to support mass streaming concurrently and without major issues.

However, despite being around for many years, RTMP has recently come under heightened scrutiny because the system is insecure for its users. 

Here is how the security vulnerability(s) comes about:

First, the RTMP protocol does not have any encryption built into it. Therefore, any communication or packet transfer while the RTMP is used is open to being listened to by unauthorized groups or man-in-the-middle type attacks.

Another factor that contributed to RTMP’s security vulnerability is that its source code was proprietary for a long time. Proprietary software (i.e., software whose ownership and control rights are limited to the entity that developed or bought it) typically receives security patches that come up regularly, but it isn’t enough.

New vulnerabilities crop up frequently, and the community built around open-source software guarantees relatively more frequent and better security patches. This is what RTMP missed out on to enhance its security posture.

RTMP Variations

RTMP variations include the following:

• Real-Time Messaging Protocol Server (RTMPS) – is quite like RTMP only that it has encryption, i.e., secure sockets layer (SSL) and Transport Layer Security (TLS) enabled, and supports all players with Flash player enabled. It’s used in scenarios where it’s vital to prevent the tampering or unauthorized access of data in transit.
• Encrypted Real-Time Messaging Protocol (RTMPE) – it’s a very versatile streaming protocol that utilizes both Transport Control Protocol (TCP) and User Datagram Protocol (UDP) to transmit data. RTMPE also encrypts all data transmissions using Adobe’s proprietary encryption to avoid unauthorized access and tampering.
• Real-Time Messaging Protocol Tunnel (RTMPT) – RTMPT employs a tunneling mechanism to bypass firewalls that typically block all RTMP traffic. In practice, RTMPT requires the client to send a modified HTTP request to the server, which responds with an almost similar HTTP transmission. The client and server use a session ID; once a connection is established, data transmission can begin between the two.
• Real-Time Media Flow Protocol (RTMFP) – RTMFP is a step-up version of the RTMP in that it employs a different encoding format UDP to achieve high-performance media streaming.

History of RTMP Streaming

Real-Time Messaging Protocol (RTMP) was initially a proprietary protocol developed by Macromedia for streaming audio, video, and data over the internet between a Flash player and a server.

RTMP is now used by numerous popular online services such as Facebook, Twitch, and Twitter for live video streaming.

The first public release of RTMP was in 2002. In 2009, Adobe released an open-specification version of RTMP known as OpenRTMP. The main difference between RTMP and OpenRTMP is that with OpenRTMP, any media server can be used instead of just Flash Media Server (FMS).

There’s also more flexibility with the open RTMP spec regarding how developers can secure or configure peer-to-peer functionality. This is meant to encourage innovation and collaboration via competition and open access among developers to develop the ideal RTMP solution.

The Main Principle

RTMP uses a technique called “streaming” to deliver content. That means the data is transferred in small pieces called “chunks.” The chunks are reassembled at the other end, so the user can watch or listen to the content without waiting to download it completely.

There are two parts to how RTMP operates: First and last-mile delivery.

First-mile delivery generally involves transmitting media from the encoder to the server using RTMP. Last-mile delivery refers to transmitting media from the server to the user’s device. A Flash player or an equally capable tool is used in this second part. There are reports that Adobe is dropping all support for Flash; consequently, this means an end to last-mile delivery.

In response, the industry has embraced Hypertext Transfer Protocol (HTTP), a more efficient streaming solution.

RTMP variations like RTMPT currently employ HTTP to encapsulate and transmit media.

How RTMP Ingest Works

This is probably one of RTMP’s saving graces that has kept it going this long. As the world moved away from viewing media on computers and towards mobile viewing, RTMP faced a challenge.

For one, RTMP relied on the Adobe Flash player to deliver its seamless streaming experience, but there was a slight problem. Mobile devices did not support Adobe Flash player; essentially, RTMP became useless to users who wanted the same streaming services on their mobile devices.

In answer, Apple developed the HLS protocol to support the streaming functionality on mobile devices.

It was only reasonable to expect that RTMP would become obsolete. Fortunately, it lived on with RTMP ingest, creating its niche as the ideal protocol to transport media from the encoder to the server.

RTMP ingest prioritizes low-cost encoders to function and generally offers low-latency streaming to users.

It involves three main components:

1. The Handshake

When a client wants to connect to an RTMP server, it first needs to establish a handshake. This process begins with the client sending a “connect” request to the server, which includes information about the client and the type of connection it is attempting to make.

The server then responds with a “connected” message, which includes information about the server and the type of connection that has been established.

Finally, the client and server exchange messages to confirm that they are both still connected and to negotiate any parameters needed for the connection.

2. The Connection

The primary purpose of the RTMP ingest connection is to provide a means for streaming media content from a source to a destination.

The media source could be a live camera feed, pre-recorded video, audio, or other media. The destination is usually a streaming media server, which distributes the content to viewers.

There are three components to an RTMP ingest connection:

• The encoder converts the video and audio signal into a digital format that can be transported over the internet.
• The transport: This is the medium over which the encoded signal is sent from the encoder to the server; typically, this is done via UDP or TCP.
• The server receives the encoded signal and makes it available to viewers (usually by packaging it into a format like Flash).

3. The Streaming

When a user streams content to a media server, the server must first encode the incoming video and audio feed before sending it out to all connected clients.

The process of encoding and reformatting video and audio to a standard file format is called transcoding. It involves converting the input signal into a form that can be played on various devices.

More on streaming there are two types of streaming: live and on-demand. Live streaming refers to broadcasting in real-time, while on-demand streaming is when users can watch content conveniently.

Live streaming requires a constant connection between the client and server, while on-demand streaming does not.

RTMP uses TCP to maintain a persistent connection between the client and server, allowing low-latency streaming. However, RTMP is not well suited for on-demand streaming.

RTMP Alternatives For Ingest

SRT and WebRTC are the major contenders that can match up with or exceed RTMP capabilities. Here’s a brief preview of the two alternatives:

Secure Reliable Transport (SRT)

SRT fills the gaps RTMP couldn’t manage, such as maintaining low latency streaming even when the user is connected to a relatively unreliable network. This presents it as an excellent choice for both live and on-demand streaming.

Since it’s open source, the limits to its capabilities are infinite, and there’s no worry about development support being withdrawn.

Web Real-Time Communications (WebRTC)

WebRTC wins with its browser-based publishing. WebRTC HTTP Ingest Protocol (WHIP) is also in the works, and what this means for users is that they’ll be able to stream with only a web browser instead of bothering with encoders as with RTMP.

RTMP Alternatives For Egress

Top on the list of RTMP egress alternatives is HTTP Live Streaming (HLS), MPEG-DASH, and WebRTC.

Here’s a brief preview of the alternatives:

HLS and MPEG-DASH

These two are virtually the same, only that HLS is proprietary while MPEG-DASH is open-source.

The best thing about these two is they’re designed to deliver low-latency, optimum media quality and even work with unreliable network connections.

WebRTC is also a noteworthy alternative to RTMP egress solutions.

Are RTMP and Flash Dying?

The short answer is: probably not. The long answer is a bit more complicated.

A steady rise in the popularity of HTML5 and the proliferation of capable alternatives of Flash may make it seem like RTMP and Flash are dying. But that’s not the case.

Flash has been on the decline for a while now, losing a significant market share to HTML5 in recent years, and its once-dominant position in the video world is now consistently under threat.

Nevertheless, it still has a significant presence on the web and is used by many popular sites, including YouTube and Facebook.

As for RTMP, it’s still widely used for streaming audio and video content. However, its future is less certain than Flash’s.

Adobe has announced that it will stop supporting RTMP in 2020, which could mark this protocol’s end. Nevertheless, there are plenty of alternatives based on RTMP, so it will likely continue to be used in some form or another for years to come.

So, Should You Stream Using RTMP?

It depends. Have a look at some of the pros and cons of using RTMP.

Pros

• It’s very stable. Compared to other alternatives in the market, it’s highly unlikely to experience any disruption or downtime while using an RTMP-enabled service.
• Low latency and minimal buffering. RTMP is unique in this regard, which means users can watch videos in better resolution, and it’ll take significantly less time for the media to load.
• Compatibility. RTMPS’ robust and reliable nature encouraged more manufacturers to design their products to integrate with RTMP easily

Cons

• RTMP requires a persistent connection between the client and server, which can be problematic if there are network disruptions
• As a proprietary software, there’s little flexibility for power users.

FAQS

How Do I Use Wave.video for Streaming via RTMP?

If you’re looking to stream video via RTMP, Wave.video is a great option. Here’s how to use it:

1. Create an account with Wave.video and login if you haven’t already.
2. Choose the video you want to stream.
3. Navigate to the “Destinations” page on Wave.video and click on “Custom RTMP”
4. Next, you’ll have to find the server URL and the stream key for the content you’re trying to stream. To do this, navigate to the website with the stream you want to view.
5. You can use a third-party app or extension to extract the URL and key if you don’t know how to go about it.
6. Copy the server URL and stream key.
7. Paste the server URL and stream key into Wave.video.
8. Create or schedule your stream.
9. Open the live studio and start streaming.

That’s all there is to it, quick and easy!

Which Encoders Support RTMP?

There are many encoder hardware and software that support RTMP. Some of these include:

• Adobe Media Encoder
• OBS Studio
• Elemental Server
• TriCaster
• Wirecast
• vMix
• TeraDek
• Wowza Streaming Engine
• Niagara Video

RTMP vs. RTSP – Which Is Better?

RTMP and RTSP are protocols for streaming audio, video, and data over the internet. They are similar in many ways, but some key differences make them ideal for different situations or preferences.

Here is a quick rundown on the key differences between the two:

• RTMP is better for live streaming, while RTSP is better for on-demand streaming.
• RTMP has lower latency, while RTSP can provide higher-quality video.
• RTMP requires a Flash Media Server, while RTSP can work with any media server.

So, which protocol is better? It’s all down to your specific needs.

RTMP is a good choice if you need low latency and don’t mind using Flash. RTSP might be ideal if you need a high-quality video or want to use a non-Flash media server.

What Is Action Message Format (AMF)?

AMF is a binary format for encoding and transmitting data over the internet, and it’s often used in conjunction with RTMP.

AMF allows for transmitting data incompatible with RTMP, such as ActionScript objects. It also enables the efficient exchange of data between Flash applications and servers.

What Is an RTMP URL and How to Get It from Facebook or YouTube?

An RTMP URL is a unique identifier used to stream live video content to various platforms.

It typically contains an IP address, domain name, and port number.

You must create a live-streaming event on either platform to get it from YouTube or Facebook. Once you’ve done this, you’ll be able to find the RTMP URL in the event’s settings.

Final Thoughts

RTMP has undoubtedly made its mark in the world. Is it on its way out? As an egress solution, possibly, ingest not at all!

Even as other equally or more capable alternatives come up, RTMP will remain relevant in media transmission and streaming.