What is a video streaming server?

A streaming server is a specialized server that handles the delivery of video and audio content to viewers over the internet. To offer quality streaming, streaming servers process multimedia data under time constraints. This prevents any jerkiness or hindrance in the video delivery and playback.

A streaming server typically has 3 subsystems:

Communicator – It involves application layer and transport protocols which are implemented on the server. In other words, it manages the network communication between streaming servers and clients. It handles how the data flows over the network, ensuring smooth and uninterrupted streaming. It manages streaming protocols like HTTP, RTSP and aspects like buffering and bandwidth optimization.

Operating system – The OS provides the necessary environment for all the processes and applications to run within the server. It manages the server’s hardware resources such as memory, CPU. It includes optimizations to handle high volume of data throughput and connections which are common while streaming.

Storage System – Here the actual streamable content is stored. It needs to ensure fast and reliable access to the media files. It has an effective caching mechanism and redundancy to ensure high performance and availability.

Video streaming architecture

Video Compression

Video compression is crucial in streaming technology. It reduces the size of the video files, making them easier to transmit over the internet. Compression is achieved using codecs (encoder/decoder). Common codecs include H.264, H.265, and VP9. The process involves two key steps:

• Encoding – It compresses the video file, reducing its size without significantly compromising quality. This is done through techniques like inter-frame compression, which only stores changes from one frame to the next rather than entire frames.
• Decoding – On the user’s end, the compressed video is decompressed by the decoder to be viewed. Efficient compression balances file size reduction and video quality.

Application-Layer QoS Control

Quality of Service (QoS) at the application layer ensures a smooth streaming experience. It involves:

• Adaptive Bitrate Streaming – Dynamically adjusting video quality based on the user’s internet speed to prevent buffering.
• Network Monitoring – Continuously monitoring network conditions to adapt the stream accordingly.
• Resource Allocation – Prioritizing network resources for critical components of the stream to maintain quality.

Continuous Media Distribution Services

This section refers to the services that enable continuous, uninterrupted delivery of streaming content. Key elements include:

• Content Delivery Networks (CDNs) – Distributed network of servers that deliver content efficiently by caching it closer to the end-user.
• Load Balancing – Distributing network traffic across multiple servers to ensure stability and reduce latency.
  Redundancy – Implementing backup systems to avoid service interruptions.

Streaming Servers

Streaming servers are specialized servers designed to efficiently deliver streaming content. They handle:

• Client Requests – Managing simultaneous requests from a vast number of users.
• Data Transmission – Utilizing protocols like RTMP (Real-Time Messaging Protocol) or HLS (HTTP Live Streaming) for efficient data transmission.
• Storage and Retrieval – Storing large volumes of media and retrieving them quickly upon request.

Media Synchronization Mechanisms

Synchronization ensures the audio and video tracks play in harmony. This includes:

• Lip Sync – Matching audio with the corresponding video (especially speech).
• Frame Synchronization – Ensuring frames are displayed at the correct rate.
• Buffer Management – Balancing buffers to avoid delays between audio and video.

Types of streaming servers

Streaming servers are categorized based on the content they deliver, functionality, and technology they use.

Live streaming servers – These servers broadcast live events in real-time. The process includes live audio and video feeds, encoding and then transmitting over the internet.

On-demand streaming servers – These servers store pre-recorded videos and media files and deliver upon request from the user. The services include movie streaming, playing recorded sessions, podcasts and more.

Audio streaming servers – These servers are designed for audio content (streaming music, podcasts, radio broadcasts)

Peer-to-Peer (P2P) Streaming Servers – Unlike traditional client-server approach, in P2P streaming, a peer simultaneously acts as a client and a server. A video can be downloaded from a set of peers and simultaneously uploaded to another peer set. This significantly reduces the bandwidth requirement.

Private Streaming Servers – Private servers have restricted access and more security. They are primarily used within organizations for training, webinars, and internal communications.

Functionality of a live streaming server

Capture and Encode – The server receives live video and audio feeds, typically from a camera or a computer. These feeds are then encoded, meaning they are compressed into a digital format suitable for transmission over the internet. Common encoding formats include H.264 for video and AAC for audio.

Transcoding – Some servers also transcode the stream, which means converting the original encoded content into multiple formats or qualities. This is crucial for adaptive streaming, where the quality of the stream adjusts based on the viewer’s bandwidth.

Delivery – After encoding, the server distributes the stream to viewers. This can be done directly (in smaller setups) or via a Content Delivery Network (CDN) for larger audiences to reduce latency and buffering.

Live streaming server protocols and technologies

Streaming protocol is a set of rules governing how data travels from one device to another. For online video delivery, both streaming protocols and HTTP-based protocols are used. For example, RTMP offers fast video delivery while HTTP protocols optimize viewing experience.

RTMP

Adobe defines RTMP as “Adobe’s Real Time Messaging Protocol (RTMP) provides a bidirectional message multiplex service over a reliable stream transport, such as TCP [RFC0793], intended to carry parallel streams of video, audio, and data messages, with associated timing information, between a pair of communicating peers.”

Flash player and RTMP were the main players for live streaming for many years. Even with the decline of Flash, still, RTMP encoders are the most used streaming protocol for ingest. Mostly, live video streams are initially encoded and later transcoded using a streaming server for content delivery to various devices and players. The most common live streaming workflows include RTMP and HLS.

HLS

Developed by Apple, HLS streaming is a widely used streaming protocol. It was originally used both for live streaming and on-demand video content over HTTP to iOS devices. Now, it is platform agnostic.

HLS is also an adaptive bitrate streaming protocol, meaning the video quality is adjusted in real-time as per the viewer’s internet speed. Initially, compared to RTMP, HLS had higher latency but later Apple introduced low-latency HLS that reduced latency to 2 seconds or less.

Supported Codecs and Formats:

• Video Codecs – H.264, H.265/HEVC (High Efficiency Video Coding).
• Audio Codecs – AAC (Advanced Audio Coding), MP3.
• Transport/Package Format – MPEG-2 TS (Transport Stream)

MPEG-DASH

Developed by Moving Pictures Expert Group (MPEG), MPEG-DASH is an alternative to HLS. Being open-source, it is highly customizable for audio and video codecs. It supports adaptive bitrate (ABR) protocol, delivering best quality video as per the viewer’s network connectivity.

MPEG-DASH allows use of any encoding standard unlike HLS that requires H.264 or H.265. Also, iOS devices cannot play video that is delivered over MPEG-DASH.

Similar to other streaming protocols, MPEG-DASH breaks video into small chunks that are transmitted over HTTP.

• Chunked Transfer Encoding – In this method, as soon as the segment of a video chunk is ready, it is sent to the client. In other protocols, there is a wait time for the entire segment to be encoded and packaged before sending. This approach significantly reduces latency and makes MPEG-DASH an ideal streaming protocol for live streaming.
• No minimum queue required – DASH doesn’t require a minimum number of segments to be queued before playback can start. This lowers latency because the video can begin playing as soon as enough data has been received, rather than waiting for a larger buffer to fill up.
• Codec Agnostics – MPEG-DASH is not tied to any specific video codec. A codec is a method for encoding and decoding video and audio data.

Setting up a live streaming server

Hardware requirements

Processing power – live streaming is a highly resource intensive process. To As the number of concurrent streams and resolution increases, the CPU consumption for video encoding in real-time increases. We then need powerful CPU.
Storage – To ensure smooth operation, adequate storage and RAM is required. SSDs are preferred for faster data access.
Network interface – The server should be highly reliable and have robust internet connectivity for faster data transmission.

Software and Platform selection

Operating System – Choose between Windows Server, Linux, or other server operating systems based on compatibility with your streaming software and your own expertise.
Streaming Software – Open-source software like OBS Studio (for encoding) and NGINX with RTMP module (for server-side streaming). If you want know more about how to use obs studio, you can check the blog linked.
Codec Selection – Choose the right codec (e.g., H.264, H.265) based on a balance between quality and required bandwidth.

Server configuration and optimization

Network Configuration – Ensure that your network settings, including firewalls and port forwarding, are configured to allow uninterrupted streaming.
Software Configuration – Set up your streaming software. This involves configuring input sources (e.g., cameras, microphones). Setting up encoding options, including resolution, bitrate, and frame rate.
Configuring the streaming protocol (e.g., RTMP, HLS) and stream destinations (e.g., YouTube, Twitch, or a CDN).
Security – Implement security measures like encryption, access controls, and, if necessary, a VPN.
Testing: Before going live, conduct thorough testing to ensure everything works seamlessly. Check for stream quality, latency, and stability.
Optimization – Based on the testing, optimize settings like bitrate and encoding options for the best performance. Monitor CPU and memory usage to ensure the server is not overburdened.
Scaling – Plan for scalability, especially if you expect a large number of viewers. This might involve setting up load balancing or using a CDN.

Network considerations for live streaming

Bandwidth consumption

Live video streaming consumes significantly higher bandwidth. The requirement depends on the stream’s resolution, compression and frame rate. For example, streaming at 1080p might require 4-5Mbps, while a 4k stream might need upwards of 20Mbps. Make sure the network handles maximum bandwidth requirement without hindering simultaneous internet usage.

Upload speed

The upload speed determines how much data you can send to the internet per second. Higher upload speeds allow for higher quality video and audio streams.
With a faster upload speed, you can stream at higher resolutions (like 1080p or 4K) and higher bitrates, which translates to clearer, more detailed images and better sound quality.

Latency

Low latency is essential for real-time interaction between the streamer and the audience. In interactive streams, like Q&A sessions, gaming, auctions, or live events, high latency can disrupt the flow of conversation and engagement. The closer the stream is to real-time, the more engaged the audience can be, as they can react to and interact with the content as it happens.

Content Delivery Network (CDN)

For reaching a wider audience, especially geographically dispersed, a CDN can distribute the load, reduce latency, and improve the overall viewing experience. CDNs work by caching the stream on multiple servers located closer to different viewers. During high-traffic events, a CDN can distribute the load among multiple servers, preventing any single server from becoming overwhelmed. This ensures consistent streaming quality even when viewer numbers are high.

FAQs

Which server is best for live streaming?

The best server for your live streaming website depends on your specific needs, such as the number of viewers, stream quality, and budget. Cloud-based solutions like Amazon Web Services (AWS) and Google Cloud Platform (GCP) are also highly regarded for their scalability and reliability.

What is a streaming server?

A streaming server is a specialized server that distributes audio and video content to viewers over the internet. Unlike traditional download servers, streaming servers allow users to view content in real-time without downloading the entire file first. They are optimized to manage large, continuous data flows essential for streaming media.

What is streaming server price?

The cost of a streaming server can vary widely. A basic setup using open-source software like OBS Studio can be relatively inexpensive, mainly requiring a capable computer and internet connection. Professional-grade servers can cost from a few hundred to several thousand dollars, depending on the features and scale. Cloud-based solutions often operate on a pay-as-you-go model, where costs depend on usage and bandwidth.

Jyoti

Jyoti began her career as a software engineer in HCL with UNHCR as a client. She started evolving her technical and documentation writing skills to become a full-time Content Marketer at VdoCipher.

The post Live Streaming Server Technology in 2024 appeared first on VdoCipher Blog.

We’ve compared RTMP and RTSP operational methods, benefits, limitations, and unique features like encryption and packet structures. This analysis highlights each protocol’s suitability for different streaming scenarios. Accompanying this is an infographic visually contrasting RTMP’s strengths in live streaming against RTSP’s capabilities in media control, providing a clear, at-a-glance comparison of the two protocols. This exploration offers essential insights into the roles and relevance of RTMP and RTSP in various streaming contexts.

What is RTSP?

Real-Time Streaming Protocol (RTSP) is an application-level network communication system used for transferring real-time data from multimedia sources to an endpoint device by directly communicating with the server streaming the data.

It is an application-level network protocol designed for multiplexing and packetizing multimedia transport streams, such as interactive media, video, and audio, over a suitable transport protocol. RTSP is employed in entertainment and communication systems to control streaming media servers, establishing and controlling media sessions between endpoints. Notably, RTSP does not stream multimedia itself but communicates with the server that streams the multimedia data.

How Does RTSP Work?

RTSP allows a client device to send an RTSP request to the server to determine options like pause, play, and record. The server returns a list of request types it can accept through RTSP. The client then sends a media description request, and the server responds with a description of the media. A setup request follows, and the server responds with transport mechanism information. Once the setup is complete, the client initiates the streaming process.

Benefits of RTSP

RTSP is beneficial in scenarios where real-time control and manipulation of streaming media are required. It allows for commands like play, record, and pause, facilitating real-time control of media streaming.

Cons of RTSP

The transmission of streaming data itself is not a task of RTSP, which could be considered a limitation. Most RTSP servers use the Real-time Transport Protocol (RTP) in conjunction with Real-time Control Protocol (RTCP) for media stream delivery, but some vendors implement proprietary transport protocols.

Key Components of RTSP

RTSP has several key components, such as:

• Options: Determines the request types the media server supports.
• Describe: Includes a URL and describes the replay data.
• Setup: Describes how to transport the bitstream.
• Teardown: Ends the streaming media session.
• Redirect: Directs the client to connect to a different media server.
• Play: Starts the specified media.
• Pause: Freeze the specified media until resumed.
• Record: Starts a recording of the media.
• Set_Parameter: Tests client or server activity and reveals values of presentation or stream guidelines.

Sample code for some of these components includes:

• Options:
  yaml
• C->S: OPTIONS rtsp://example.com/media.mp4 RTSP/1.0
CSeq: 1
Require: implicit-play
Proxy-Require: gzipped-messages

• Describe:
  rust
• C->S: DESCRIBE rtsp://example.com/media.mp4 RTSP/1.0
CSeq: 2

• Setup:
  shell
• C->S: SETUP rtsp://example.com/media.mp4/streamid=0 RTSP/1.0
CSeq: 3
Transport: RTP/AVP;unicast;client_port=8000-8001

• Pause:
  rust
• C->S: PAUSE rtsp://example.com/media.mp4 RTSP/1.0
CSeq: 5
Session: 12345678

• Record:
  rust
• C->S: RECORD rtsp://example.com/media.mp4 RTSP/1.0
CSeq: 6
Session: 12345678

• Teardown:
  rust
• C->S: TEARDOWN rtsp://example.com/media.mp4 RTSP/1.0
CSeq: 8
Session: 12345678

• Set_Parameter:
  bash
• C->S: SET_PARAMETER rtsp://example.com/media.mp4 RTSP/1.0
CSeq: 10
Content-length: 20
Content-type: text/parameters

• Redirect:
  rust
  S->C: REDIRECT rtsp://example.com/media.mp4 RTSP/1.0
CSeq: 11
Location: rtsp://bigserver.com:8001


History and Development of RTSP

RTSP was based on a 1997 proposal by Anup Rao of Netscape Communications and Rob Lanphier of Progressive Networks (now RealNetworks). The Internet Engineering Task Force (IETF) published it as RFC 2326. Rao, Lanphier, and others proposed RTSP Version 2.0 as RFC 7826 in December 2016, introducing new header values and addressing issues

What is RTMP?

Real-Time Messaging Protocol (RTMP) is a communication protocol for streaming audio, video, and data over the Internet. Initially developed by Macromedia for streaming between Flash Player and the Flash Communication Server, Adobe, which acquired Macromedia, later released an incomplete version of the protocol for public use. RTMP is essential in live video streaming technology, enabling the transfer of data over the internet. Its design facilitates the chopping of high-definition video files into small packets for faster and easier internet transmission, which are then reassembled on the viewer’s screen.

How Does RTMP Work?

RTMP operates as a TCP-based protocol, maintaining persistent connections and allowing for low-latency communication. It enhances smooth stream delivery by splitting streams into fragments, whose size is dynamically negotiated between the client and server. These fragments from different streams can be interleaved and multiplexed over a single connection. RTMP defines several virtual channels for packet transmission, including channels for RPC requests and responses, video stream data, audio stream data, and out-of-band control messages. Each packet consists of a header, specifying the channel ID, timestamp, and payload size, followed by the actual payload content.

Benefits of RTMP

• RTMP provides low-latency streaming, which is crucial for live video streams, and maintains constant contact between the server and video player, enhancing the viewing experience.
• It supports secure, cloud-based streaming through RTMPS, a more secure version of RTMP.
• RTMP is compatible with most live-streaming video services and is relatively easy to set up.

Cons of RTMP

• RTMP requires high internet bandwidth.
• It is not suitable for streaming to large audiences on multiple servers.
• Many browsers and embedded video players don’t accept RTMP streams.

Encryption

RTMP sessions can be encrypted using TLS/SSL mechanisms, wrapping the underlying RTMP session inside a normal TLS/SSL session, or using RTMPE, which wraps the RTMP session in a lighter-weight encryption layer.

HTTP Tunneling

RTMP Tunneled (RTMPT) encapsulates RTMP data and exchanges it via HTTP, particularly beneficial in scenarios where the client is behind firewalls that block non-HTTP and non-HTTPS outbound traffic. Commands are sent through the POST URL, and AMF messages through the POST body.

Packet Structure

RTMP packets sent over a TCP connection include a header and a body. The header consists of a Basic Header and a Chunk Message Header. The Basic Header is composed of a single byte indicating the Chunk Type and Stream ID. The Chunk Message Header contains metadata information such as message size, Timestamp Delta, and Message Type. The packet header specifies the channel ID, timestamp, and payload size, followed by the actual payload content.

Software Implementations

RTMP is implemented in live video encoders, live and on-demand media streaming servers, and live and on-demand clients. An example is the open-source RTMP client command-line tool ‘rtmpdump’, designed to play back or save the full RTMP stream to disk. It supports various operating systems and is a part of the major open-source repositories.

Protocol

The RTMP protocol involves a handshake process to establish a connection, involving the exchange of three packets from each side. These packets, referred to as Chunks, have a specific structure with fields setting the “epoch” timestamp and other parameters. After the handshake is complete, RTMP packets can be exchanged.

History and Development

Originally developed by Macromedia for streaming between Flash Player and Flash Communication Server, RTMP was first used to support Flash Player. Adobe, which acquired Macromedia, later released a version of the protocol specification for public use. RTMP has evolved to include multiple variations, such as RTMPS, RTMPE, and RTMPT, to adapt to different security and network requirements.

RTMP vs RTSP Comparison Summary

References

• RTMP Wikipedia – link
• RTSP Wikipedia – link

Vishal Sharma

My expertise focuses on DRM encryption, CDN technologies, and streamlining marketing campaigns to drive engagement and growth. At VdoCipher, I’ve significantly enhanced digital experiences and contributed to in-depth technical discussions in the eLearning, Media, and Security sectors, showcasing a commitment to innovation and excellence in the digital landscape.

The post RTSP Vs RTMP Protocol : A Concise Comparison appeared first on VdoCipher Blog.

With the help of low latency connections, you can broadcast music and video using the Real-Time Messaging Protocol (RTMP) from Adobe. Large amounts of audio, video, and data can be efficiently transmitted from a server to the Internet using this TCP-based network protocol. RTMP’s primary function in current live streaming setups is to transfer material from an encoder to an online video server.

In the context of its new position in live streaming, RTMP is significant, but its reach is considerably narrower than it formerly was. It is capable of low-latency streaming, a considerable advantage for broadcasters transmitting actual events in real time. Additionally, it gives minor buffering, which significantly improves the user experience. RTMP streaming is one of the finest techniques to offer streaming material with a small buffer.

RTMP technology is also used by adaptive-bitrate streaming and some web conferencing applications. Let’s explore RTMP in detail:

What is RTMP Stream?

Real Time Messaging Protocol or RTMP is used to stream multimedia data – audio and video  – between Flash Media Server and Flash Player. The chief utility of RTMP stream is in the optimization of the audio and video data transfer between the server and player. Its major use comes in Dynamic live streaming, and because of the encryption service RTMPE (which, despite of its security flaws, has a wide use). In this blog post, I expand upon –

• Use as a Streaming protocol – How RTMP evolved as a widely used real time video streaming protocol
• Differences from other streaming protocols, strengths – In what aspects it differs from newer streaming protocols such as  HLS & HDS.
• Use Cases – We also discuss certain RTMP use cases.
• RTMPE – Utility of Encrypted RTMP and the flaws in its security services.
• Flaws in Security & Streaming reliability

RTMP at a glance

Even though RTMP is officially deprecated, you can use it with certain transcoders in various broadcasting processes and circumstances. Up until the early 2010s, Flash Player and RTMP dominated the live streaming distribution landscape. RTMP is resource-intensive since it necessitates a separate streaming server. When Adobe announced that Flash would be discontinued in December 2020, it was clear that this was the end of the road. Anyone may create products and technologies to provide video, audio, and data thanks to Adobe’s release of RTMP as an open specification. With the protocol available, various novel solutions might be created without the need for Flash by using RTMP for all or a portion of the transmission process.

Key benefits of RTMP:

• You can securely transmit or display audio and video in various formats, including endless others.
• Real-time message posting or very low transmission delay
• Gives you an advantage over HTTP-enabled video streams in terms of clever search functionality. A flash player is required to enjoy this feature.
• For the delivery of packets, RTMP provides several virtual channels that operate independently of one another.

Real-Time Messaging Protocol (RTMP) may appear a relic of a bygone age, yet many platforms continue to support it for RTMP intake even if it hasn’t been used for delivery since Flash’s demise. RTMP is still essential now despite how live streaming has evolved.

Video formats supported include- H.264 (the preferred option) and VP8, as well as alternative codecs, including Screen Video v2 and AMF, SWF, FLV, and F4V.

Supported audio formats-  mp3, AAC-LC, HE-AAC, and AAC (recommended).

Is RTMP Stream Optimal for Low Latency Transmission?

RTMP stream does not use either of the 80/443 ports(HTTP/HTTPS) ports but instead uses the 1935 port. An exclusive port for video packet transmission signifies that the RTMP stream protocol is suitable for transmission of low latency content – essentially live streaming of video content, and streaming of media without buffering. However, by using this port RTMP stream becomes vulnerable to being blocked by certain corporate firewalls.

RTMP stream is theoretically optimal for low latency streaming as the RTMP pointer is always in sync with the media server at the exact point of the stream. This means that if there is a signal interruption for say 5 seconds, the flash player will not display the signal for 5 seconds, but would resume in real-time when the signal over the network resumes. The stream should therefore never lag in spite of poor network, although the quality of the video may suffer.

Adaptive Streaming through RTMP

RTMPE is suitable for live streaming as it does not download any video packets except at the time of streaming – there is no buffering of content as would occur with a progressive media player. The bitrate can be dynamically decided to optimize user experience over the available network. Dynamic streams consist of multiple single streams of the same content, all of the different quality. RTMP stream relies on dynamically adaptive content transfer to maintain video playback at periods of uneven network quality

RTMP stream is distinct from Progressive Streaming Players

Progressive media players, a category that includes popular video players Youtube, Vimeo & VdoCipher, allow video portion of the future video part to be downloaded as well. Essentially it allows for a future buffer in the player. Progressive Streaming is now a common feature across all popular video hosting services across the internet. The shifting of video metadata from the end of the media file to the front allows this possibility. Thus, before the download of the complete media file, the media player has all the information that it requires to start playback.

In progressive streaming, the media is stored in the temporary directory of the associated web browser. This is the concept on which most downloaders or download software/plugins work. They are able to fetch this data that is coming into the browser.

With RTMP there is no storage in a temporary directory, and instead of continued direct streaming between the two servers. It is conceptually live and does not have any future buffer even for recorded videos.

Encrypted RTMPE & Security

Encrypted RTMP (RTMPE) wraps the RTMP stream session in a lightweight encryption layer. Through Encrypted RTMPE, the streaming protocol provides low-level stream encryptions for high-traffic sites. RTMPE uses Anonymous Diffie-Hellman key exchange method. In this algorithm two parties – the media server and the flash player – establish a shared secret key over an insecure channel. However, the initial point of secret key sharing is unsecure. The media server cannot verify the identity of the media player. This leaves the player vulnerable to man-in-the-middle attacks at session initializations.

RTMPS is an RTMPE over a secure TLS/SSL connection. The underlying algorithm of RTMPE remains the same, but a secure SSL connection precludes a man-in-the-middle attack from unknown third parties. However there is still the case in which the client can grants access to a third party.  Through this user can download the stream content.  In that case the video download is easy, leaving the content provider vulnerable to the potential piracy from end-users for illegal distribution.

Security

As explained, RTMP stream never has a block of data, only a pointer to the current live data. Owing to this no local storage of data RTMPE occurs. RTMPE can therefore, with an additional layer of security, be a means for video transmission locally. As discussed, RTMPE was quite different from other streaming players in concept of its streaming protocol. Thus, not many downloaders or plugins are built to grab it. Thus, it has a certain level of protection sometimes pitched with it.

RTMPDump to Download RTMP Videos

RTMP dump is an RTMP toolkit to connect to Flash Media servers just like normal flash players. It stores the data from the RTMP connection, sometimes used to capture the stream from the server. This allows local saving of the video file. This opens a means by which TV shows and videos on demand being streamed via this can be downloaded to cause piracy.

RTMP Stream and Content Delivery Networks(CDN)

RTMP also requires direct constant connection with a Flash Media Server throughout the period of playback, as there is no local storage. Every flash client has to maintain state with the media server all the time.  This results in higher CDN cost.

Currently, CDNs have started to phase out RTMP. Although CDNs Akamai and Amazon Cloudfront have committed to supporting its streaming, they accept that usage traffic has declined considerably over the previous years. Limelight supports RTMP for video ingestion from content provider to CDN server, which routs to the video user through HTTP.

How does RTMP work?

RTMP relies on three essential components to function:

1. Handshake

1. Connection

1. Stream

Adobe no longer supports the RTMP protocol complicates a relatively straightforward task.

RTMP is mainly used in professional broadcast contexts nowadays.

Here is a concise explanation of the RTMP process.

Step 1: The Handshake

The client and server engage in brief interactions during the handshake phase. Initially, the client transmits what is typically known as a header. Essentially, this is a cryptographic signature. Immediately after the transmission of the title, the client transmits 1536 bytes of random data.

Step 2: The Connection

As soon as the handshake is complete, the connection phase begins. The connection step includes an AMF (Action Message Format)-encoded data exchange. This defines a standard for communication between the client and server, including broad criteria for video playback, frame size, and bandwidth.

Step 3: The Stream

After establishing the connection and communication protocols, the stream is begun. This phase permits the execution of critical user instructions such as play and pause.

RTMP protocol variations

There are several variants of RTMP, the most common of which is RTMP itself, as well as RTMPS, RTMPE, RTMPT, and RTMFP. Within live streaming, each video protocol is used for somewhat distinct goals, which contrast with one another.

Let’s look at the streaming protocol’s specific benefits of each of these different variants.

1. RTMP

RTMP proper is the version of the RTMP protocol that has been around the longest. This is the streaming format created by Macromedia, which Adobe eventually acquired. It served as the basis for the other standards included on this list.

2. RTMPS

SSL certification is used to produce a more secure stream when RTMPS streaming is being used. 

3. RTMPE

Macromedia was the company that first invented RTMPE, which was a different, more secure way of streaming video. In 2022, it will have just a marginal user base. Still, it does not take advantage of SSL security certification (which remains one of the predominant standards for securing data and content over the internet).

4. RTMPT

Video is being tunneled using RTMPT’s streaming service. Tunneling is a method for transmitting confidential information via public networks. Even though this may be helpful in some circumstances, it is well recognized to add additional delay to the process or workflow.

5. RTMFP

Real-Time Media Flow Protocol, often known as RTMFP, is an RTMP variant constructed using UDP rather than TCP. This technology is the foundation for a wide variety of video conferencing solutions and social networking applications that include video live stream chat capabilities.

RTMP encoding

As previously mentioned, RTMP technology is often utilized to send video material from a live stream encoder to a streaming platform. To stream in this protocol using an RTMP-based configuration, an RTMP encoder is necessary.

Which Encoders Support RTMP?

Broadcasters have alternatives to software and hardware encoders for RTMP encoding. 

Let’s explore some of the most popular RTMP encoder options:

• OBS Studio:

OBS Studio is a free, open-source encoding software application for novice broadcasters. If you want know more about how to use obs studio, you can check the blog linked.

• Wirecast:

Telestream’s Wirecast is a popular software encoder. This software’s starting price of $599 makes it more suitable for advanced broadcasters.

• VidBlaster:

VidBlaster is a live streaming software application with editing, mixing, and encoding options. It is an excellent alternative for broadcasters seeking to add a professional touch to their programs. The software costs $9 per year, to begin with.

• vMix:

VMix is another live streaming software package with encoding features and pricing levels to accommodate any budget. Currently, plans vary from zero dollars to $1,200 for a lifetime license.

• TriCaster:

NewTek’s TriCaster is a set of encoding devices. This series contains both portable and fixed choices.

• Teradek:

Teradek is a hardware manufacturer that provides a variety of encoding devices.

RTMP Apps for iPhone

In addition to the tools mentioned above, iOS applications enable RTMP streaming from an iPhone. These applications are developed for mobile live streaming.

Here are some of the top iOS and RTMP-compatible live streaming applications:

• Larix Broadcaster
• Twitch. tv
• nano stream
• Wirecast Go
• Airmix Solo
• Broadcast Me
• LinkedIn Live

Although it is feasible to stream from an iPhone, professional-grade equipment is preferred for streaming. Nonetheless, an iPhone or iPad would do if you are streaming on-site and do not have mobile live streaming equipment.

HLS and DASH as an alternative for RTMP

We have sufficient knowledge of the RTMP streaming protocol. Examine protocols that may serve as alternatives to RTMP.

HLS

HLS is one of the most used protocols for streaming media. Like HTTP, it can send on-demand video data, but it also excels at live streaming. HLS was created by Apple and was first exclusive to iOS, but now it is compatible with the time of end devices and browsers. The primary benefit of HLS is the clear separation of video data. HLS employs a protocol already supported by most end devices and is thus compatible with many devices. 

Due to the transfer of individual data packets, users may modify the stream quality without interrupting or refreshing the stream. HLS makes it feasible to load the following data segment at a lower bit rate if a user has a momentary Internet connection time. Hence, you can avoid buffering of a stream by momentarily delivering a stream with lesser quality. This method is known as Adaptive Bitrate Streaming.

DASH

DASH is a streaming protocol that segments data similarly to HLS. It has the benefit of dividing video files into significantly smaller chunks than HLS does. This enables the broadcast stream to respond to changes on the client side and transition between various speeds more rapidly. While HLS segments are around 10 seconds long, DASH segments are just 2 to 4 seconds long, providing significantly more flexibility. 

DASH, unlike HLS, is a standard protocol; however, not all end devices support it. iOS-based devices are incompatible with DASH streaming. DASH may be incompatible with other devices since it is not inherently compatible with HTML5. Despite this, DASH is a popular and commonly used streaming protocol, particularly for live broadcasts owing to its efficient and adaptable functioning.

The Real-Time Messaging Protocol (RTMP) is essential in various live streaming activities. It is necessary to remember that RTMP is not extinct; instead, it only serves in more supporting capacities than it did in the past. This is something to keep in mind.

Alternative to RTMP – High Secure & Smoother Streaming

As seen above, it has primary two problems –
1. Poor streaming due to no future buffer and constant server to player connections.
2. Downloadable through RTMPdump compromises security from piracy.

VdoCipher offers Google Widevine DRM encryption + Apple Fairplay DRM encryption, viewer specific watermarking, and backend authentication. This ensures the highest security from any downloader or plugin looking to hack content. The OTP given to the specific viewer is authenticated at the backend, precluding access from anyone except the intended video user.

VdoCipher has a progressive buffer-conserving player. On seeking the video to past or future timeline, the buffer remains conserved, thereby demanding lower bandwidth. Our html5 video player is optimized for serving at both low and high bandwidth – VdoCipher serves viewers in tier II and tier III cities in India, we also have major clients across Africa. Our streaming has been highly effective even at lower connection speeds.

For a full version 5 GB free trial , please visit VdoCipher.

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Siddhant Jain

CEO, VdoCipher. Writes about video tech, ed tech and media tech.

www.vdocipher.com/

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