What Is RTMP Streaming? Real-Time Messaging Protocol Explained

Blog 1 min read | Sep 12, 2023 | JW Player


Real-Time Messaging Protocol (RTMP) has been around for decades, and while more reliable alternatives like HTTP Live Streaming (HLS) and WebRTC have emerged, RTMP streaming is still the most common format for ingest.

Due to the death of Adobe Flash, RTMP is no longer an end-to-end technology for live online video streaming—it now needs to be paired with playback-friendly alternatives like HLS and MPEG-DASH. However, thanks to its long history, low latency, encoder support, and social media requirements, RTMP streaming remains a reliable solution for distributing live streams.

Are you new to RTMP streaming and its nuances? We’ve got you covered. Below, we’ll break down everything you need to know about RTMP streaming, including what it is, how it works, variations, alternatives, and how to get started.

What Is RTMP Streaming?

RTMP streaming supports live online video streaming by transmitting audio and video data from a host streaming server to a video player. It uses Transmission Control Protocol (TCP) technology to distribute content from an encoder to a digital video host through a process known as ingestion.

Before the downfall of Flash, RTMP was an end-to-end live streaming protocol. However, it now works with modern-day players like HLS and DASH to deliver content. RTMP is known for its low-latency streaming and minimal buffering, making it an excellent solution for streaming high-quality video and providing top-notch user experiences.

Most encoders and media servers support RTMP, but that’s primarily due to its history and the long-term dominance of Flash Player. Many broadcasters and streamers compress their stream with an RTMP encoder and then repackage it for playback in an HTTP-based format.

You can use RTMP streaming to deliver content from your encoding software to multiple endpoints. For example, you might use JW Player to capture, record, and transmit your live streams to Twitch, YouTube Live, Facebook, and Instagram simultaneously instead of using (and relying upon) a single social platform.

RTMP consists of three main components:

  • RTMP server: The server represents the core of the live streaming workflow, receiving incoming streams and sharing them to the RTMP clients. Its main role is to ensure seamless client-server transfer.
  • RTMP client: The RTMP client has the crucial role of receiving the live video and audio streams from the server and displaying them to the viewer. RTMP clients range from web browsers and mobile apps to specialized streaming software.
  • RTMP protocol: The RTMP protocol establishes the guidelines and mechanisms for delivering media content across networks. It combines real-time communication and adaptive bitrate streaming, resulting in seamless exchange of control messages between server and client.

RTMP Specifications

  • Video Codecs: H.264, VP8, VP6, Sorenson Spark®, Screen Video v1 & v2
  • Audio Codecs: AAC, AAC-LC, HE-AAC+ v1 & v2, MP3, Speex
  • Playback Compatability: Flash Player (which is discontinued), Adobe AIR, and RTMP-compatible players
  • Protocol Variations: RTMPT, RTMPE, RTMPS, and RTMPF

Pros and Cons of RTMP Streaming

RTMP streaming isn’t a perfect solution, and it’s showing its age with a lack of support from modern-day browsers and devices. Here’s a list of pros and cons to consider with RTMP streaming:


  • Low latency
  • Long history
  • Broad encoder and media server support
  • Minimal buffering
  • Required for some social media streaming


  • No longer supports playback
  • No continued updates or support
  • Likely to be replaced by modern alternatives

RTMP Live Streaming Workflow

RTMP streaming works by establishing a communication connection between the host server and the player. Maintaining this pathway enables reliable transmission of audio, video, and data.

The RTMP connection is created through a 3-step process: handshake, connection, and streaming.

1. Handshake

The client (often the encoding software, platform, or hardware) starts a connection request with the server to which it will stream the live video content. During this handshake process, the host client exchanges 3 packets of data:

  • The client sends a packet explaining which version of RMTP is being used.
  • The client sends a timestamp, and the receiving server sends back a packet with a timestamp of when it received it.
  • After establishing communication, the client server sends a copy of the timestamp, and the server returns it.

Once this exchange is completed, the handshake process is over and moves on to the connection phase.

2. Connection

The client server and the player negotiate a connection with Action Message Format (AMF). The RTMP encoder uses AMF-encoded messages to initiate the connection request and share the connection URL, video codec, and audio codec details.

Once the player server responds, the connection is established, and the stream can launch.

3. Streaming

Once the handshake and connection steps have been completed, you can begin streaming data over the pathway. Video, audio, and data will transfer over the connection, and users can initiate commands like play and pause to engage with the content according to their own preferences.

What is Action Message Format (AMF)?

Action Message Format (AMF) is a data format defined by Adobe that’s geared towards object-oriented programming. It’s commonly used to transmit commands between the server and client in RTMP streams. With AMF, the server and client can engage in two-way communication, allowing broadcasters to send out control messages that update or alter the parameters of the stream. Thanks to AMF, viewers get a smoother and more efficient streaming experience.

Protocol Variations for RTMP

Variation protocols of RTMP have emerged to serve different purposes and use cases. Each has pros and cons that make it a better fit depending on your needs:

  • RTMP: Originally developed by Macromedia, this is the foundational input for other variations that have emerged.
  • RTMPS: The RTMPS protocol is an enhanced version of the Real-Time Messaging Protocol (RTMP), designed to secure streaming media over public networks. It works by adding a layer of security by encapsulating RTMP with TLS or SSL, protecting streams from attacks and verifying the streamer’s identity. Without RTMPS, streamers would be vulnerable to data theft, as well as quality loss due to unauthorized packet sniffing or injection.
  • RTMPE: RTMPE uses a secure 128-bit encrypted channel for data transmission between the client and server. Unlike SSL, it does not involve certificate management, making it ideal for applications prioritizing performance and speed over endpoint authentication. Additionally, RTMPE requires 15% more processing power than RTMP.
  • RTMPT: Uses tunneling (a method for sending private data over public networks) to stream live video. This allows for streaming media to be sent through firewalls which can block the direct traffic of RTMP, typically used in corporate environments or public networks. However, it often increases the latency.
  • RTMFP: RTMFP is built on User Datagram Protocol (UDP), offering advantages for live streaming media, including lower latency, reduced overhead, and improved resilience to dropped packets. Unlike RTMP, RTMFP allows direct data transfer between Adobe Flash Players without the need for a server. Nevertheless, you’ll still need a server connection for the initial setup and server-side functionality. Moreover, Flash Media Server users must also authorize network address lookup and NAT traversal services to ensure the managed usage of Flash Player.

What is an RTMP video encoder?

A RTMP encoder is the link between a live video source and the streaming platform. It captures audio and video from various sources such as cameras, computers or gaming consoles, transcodes it into the desired format, and sends it via an RTMP connection to be viewed in real time. The encoder converts the recording into data packets for transmission over the internet, allowing for a high quality and smooth streaming experience.

When selecting an encoder, it is important to consider features such as compatibility with various video formats, resolution support, and audio functions. With the right RTMP encoder, viewers can be sure to get an enjoyable experience with a high quality stream. A proper encoder also ensures that streaming is hassle-free and efficient for broadcasters, allowing them to focus on creating top-notch content.

By leveraging RTMP streaming powered by a reliable encoder, broadcasts can reach a wider audience in real-time with no interruptions. This opens up many possibilities for businesses, organizations and individuals to produce engaging content that can be quickly shared with viewers at home or anywhere in the world. Overall, RTMP streaming is a powerful tool that can make live video broadcasting more accessible and enjoyable than ever before.

Video Encoders That Support RTMP Streaming

While RTMP might not be the technology many player servers use, it’s still a popular transmission tool for connecting encoding software to live streaming platforms. Many purpose-built RTMP encoders exist on the market, but today’s top video cloud platforms tend to support RTMP streaming.

Here are a few of the top RTMP encoders:

1. JW Player

JW Player is a complete video platform for hosting, recording, streaming, and distributing live and video-on-demand (VOD) content. Its encoding software supports publishing to multiple social sites via RTMP simulcast.

With customizable options to meet your specific needs, JWP supports a wide range of broadcasting requirements. From 24/7 linear channels to event-based live streaming and on-demand video streaming up to 4K resolution, the platform ensures a seamless viewing experience across various devices including web, OTT devices, and CTV.

One of JWP’s key strenghts lies in its effective monetization and engagement strategies. To be specific, JWP enables you to monetize your content across different devices and platforms by offering robust content protection, including studio-approved DRM protection like PlayReady, Widevine, and FairPlay. And if that’s not enough, additional safeguards like token signing and geo-blocking will enhance your content’s security.

To maximize your revenue, you can take advantage flexible monetization strategies, including SSAI, SVOD, and TVOD models.

Engage your audience with features like Live+VOD stitching, live clipping, social media sharing, and smooth transitions from live to on-demand content.

Lastly, JWP’s support and monitoring services are well-regarded for their excellence and scalability. Thanks to the continuous proactive monitoring by an in-house NOC team and skilled support engineers, your systems will always be taken care of. Automated alerts and adaptable Support SLA packages complete the comprehensive support ecosystem.

If you choose JWP, you’ll reap the benefits of a cloud-based, highly scalable platform with a cloud-hosted solution that ensures high availability and readiness to handle peak demand.

2. OBS Studio

Open Broadcaster Studio (OBS) is an open-source video platform with encoding functionality. It integrates with today’s popular video platforms to live stream to websites, applications, and social media platforms.

OBS Studio is relatively easy to learn, making it a beginner-friendly tool for those just getting started with RTMP streaming. You’ll find recording and encoding tools, mixing features, overlays, and broad hardware support.

3. vMix

vMix is a live streaming software tool designed for Windows operating systems. It provides encoding capabilities and allows you to push and pull video content from social media platforms. vMix’s player offers features like instant replay, scoreboards, and slow motion.

4. Wirecast

Wirecast is a live streaming solution with comprehensive controls for switching between multiple video and audio sources. Its encoding tools support RTMP streaming, making it a great desktop and mobile live streaming option.

While Wirecast has plenty of advanced features to suit your use case, it can be dense and hard to learn. Plus, the high price point often makes it a non-negotiable for startups and small businesses.

5. VidBlasterX

VidBlasterX is a subscription-based live video software encoder with robust customization options. VidBlasterX Studio focuses on HD full-screen support and high-quality production, while VidBlasterX Broadcasts adds more module capacity and direct support from a developer.

RTMP alternatives

While RTMP is the most popular protocol for ingest, other alternatives may suit your needs and use case better. Below, we’ll explore a few other options that have picked up steam in recent years.

HLS Streaming Protocol

HTTP Live Streaming (HLS) is a widely adopted streaming protocol developed by Apple Inc, used for transferring video and audio data from media servers to the audience’s screen. The protocol enables adaptive bitrate streaming for both live and on-demand content, while dynamically adjusting video quality based on the viewer’s internet connection.

What’s more, HLS provides easy setup, reliable and scalable streaming, low latency, and content protection support. Nevertheless, it’s important to note that due to its proprietary nature, HLS may require transmuxing and has limited compatibility. By seamlessly streaming your content with HLS, you can deliver a superior viewing experience to your audience while ensuring optimal performance and flexibility.

RTSP Streaming Protocol

RTSP stands for Real-Time Streaming Protocol. It’s an older low-latency streaming protocol that’s typically used with closed-circuit televisions (CCTV) and surveillance systems. It’s typically used for ingesting data, but most platforms will repackage for delivery with a more playback-friendly alternative.

SRT Streaming Protocol

Secure Reliable Transport (SRT) protocol is an open-source solution built for low-latency streaming over public networks. It’s still an emerging technology and has yet to gain as broad support with encoders, decoders, and players as RTMP, but it’s becoming a more popular option. SRT streaming tends to provide high-quality, low-latency live video even over unpredictable networks.

The main difference between RTMP and SRT is that RTMP lacks timestamps in packet headers, requiring large buffers and fixed time intervals for packet transmission. In contrast, SRT includes timestamps for each packet, reducing the need for buffering. Plus, SRT also excels in handling problematic video and requires less bandwidth, resulting in a smoother stream for viewers. With RTMP being outdated since 2012, SRT is expected to replace it as a more efficient and reliable streaming protocol.

WebRTC Streaming Protocol

Web Real-Time Communication (WebRTC) is a browser-based protocol that delivers lightning-fast video and audio streaming to and from major web browsers. Initially, it was designed for chat-based applications, but it’s expanded into video and audio—and it’s also evolving to expand beyond just browser-based publishing. In some situations, WebRTC works well as an end-to-end technology, but it requires encoding know-how to ensure top-notch performance and playback quality.

Choosing between RTMP and WebRTC might not be a straightforward decision, as both technologies have their own strengths and limitations. For instance, RTMP offers reliable data transmission with slightly higher latency, making it suitable for large-scale live streaming. On the other hand, WebRTC provides near real-time latency and better browser support, making it ideal for two-way conferencing or real-time device control. Lastly, The scalability and cloud support of RTMP are superior, while WebRTC has the advantage of native API integration and easier development. The choice depends on the specific requirements of the use case, so keep reading to learn the factors that will help you make an easier decision.

How to Choose the Right Protocol for Your Live Streaming

As you can see above, you have options when it comes to choosing a protocol for your live streaming. However, narrowing things down and deciding on one can be tricky.

Most of the time, you aren’t limited to a single streaming protocol. You might use one protocol for capture and another for playback. Encoders, players, and video platforms are able to repackage live streams for accessible playback across more devices, applications, browsers, and regions.

For example, you might use a video platform to ingest an RTMP stream and transcode it to an HTML5 player with an HTTP-based protocol like HLS—this is the most common type of live streaming workflow. It allows you to reduce your latency while still maximizing playback quality and performance.

The best cloud video platforms can consume RTMP streams from mobile apps, encoders, and IP cameras and prepare them for playback-friendly players. And just about every modern player supports HLS, making it the perfect protocol combination.

Below, we’ve outlined a few different criteria to help you find the right solution.

Playback Destination

Where do you plan on your audience consuming your content? If it’s going to be over social media platforms (like Facebook, YouTube, or Twitch), you’ll want to use RTMP. Most social players accept RTMP, and it has broad support across encoders.

If you want to stream to and from a web browser, we’d recommend going with WebRTC. WebRTC is simple and easy to launch, empowering you to go live in seconds and transport high-quality, low-latency video and audio data to your viewers.

Speed and Simplicity

WebRTC lets you skip using an encoder altogether, making it a go-to option when speed and simplicity are your priorities. You’ll need a little bit more technical know-how when working with WebRTC, but it’s a great option if you can invest the developer resources.

Quality and Playback Performance

SRT protocol provides high-quality, low-latency streams, making it an excellent choice if your focus is high resolution and top-notch performance. However, it isn’t supported by common encoders and players like RTMP, meaning you’ll need to do more research on your video stack before making a decision on your protocol.

Interactivity and Engagement

WebRTC is one of the best live streaming protocols if sub-second delivery and near real-time response are important. This is critical with emergency response and remote monitoring systems, but it’s also growing in importance for streamers and gamers to interact in real time with their audiences through voice and chat.

Start a Free Trial and See for Yourself

JW Player provides a complete end-to-end video platform for publishers, broadcasters, sports, gaming, and everyone in between. Build our players into your own mobile applications, OTT apps, or web pages with our robust APIs and SDKs. We’ll encode and transcode your uploads for optimal playback across devices, locations, and bandwidths—and you can even RTMP simulcast to multiple social media networks.

JW Player isn’t just an encoding tool, though. It’s a comprehensive video platform that lets you connect and engage with your audiences on a global scale:

  • Play: Use our HTML5 video player to deliver high-quality, buffer-free experiences with customized branding and user experience options.
  • Stream: Deliver VOD or live stream content across devices and locations to your audience.
  • Monetize: Maximize your revenue by monetizing with subscriptions, pay-per-view models, or advertisements.
  • Engage: Keep your audience engaged and consuming your content with Article Matching and Recommendation features.

Want to take your video content to the next level? Expand your video’s reach and streaming capabilities like never before, and sign up today.