Live streaming is a trend in this digital age. People use live streaming services to share their thoughts, events, and experiences in real time. More than just sharing their lives with others, people also use live streaming services for business purposes like products. Streaming live video content helps businesses to increase their sales and establish themselves as key players in the industry. You may hear these common acronyms like RTMP, SRT, HLS, UDP, and many others when you are a live streamer. Video streaming protocols ensure that video playback will be smooth and uninterrupted. Understanding these streaming protocols can help optimize your live stream for better performance and security. These live video streaming protocols give the viewer a good viewing experience, which is why they’re so important!
Since the internet is a jungle of information, getting your head around all the different protocols is hard. That’s why we’ve simplified things for you in this blog post! If you want to know more about these video live-streaming protocols or how they work, keep reading!
What is Video Streaming Protocol?
Video streaming protocols are rules that govern how video content is streamed across a network. This video streaming technology is designed to be efficient, flexible, and reliable without compromising video streaming quality. Video streaming protocols allow the video playback to adapt as needed, for example, if a viewer’s bandwidth is insufficient to view high-quality video. Video streaming protocols are typically used in streaming services that require high-quality video playback or low latency, such as live broadcasting. It’s possible with this protocol technology.
Major Types of Streaming Protocols
The basic concept of video streaming protocols is clear to us. Now let’s go over the types of video streaming protocols used by live streamers and comparison among these protocols.
Traditional Streaming Protocols
The traditional video streaming protocols are people’s first experience with RTMP and RTSP. Traditional protocols allow you to stream your content using the player you already have, such as Adobe Flash or Apple QuickTime Player. You can send a link to someone, and they will play it back in their media player if they’re on an operating system that supports it.
Traditional protocols are the most efficient type of streaming protocol. But they require new technology on both ends of the connection to view or distribute your content. More specifically, the user must install adobe Flash on a computer for this service to work using traditional video playback protocols. That’s why many people have stopped using this method for online video streaming. Most modern browsers do not support Adobe Flash, so getting people who use modern browsers is harder.
RTMP (Real-Time Messaging Protocol)
RTMP refers to the real-time messaging protocol, the second most popular video streaming protocol after HLS. This protocol is also known as Adobe Flash Media Server. Macromedia created RTMP in 1995, and Adobe Systems owned it until 2005.
RTMP is one of the most flexible protocols. It allows you to adjust the streaming quality for different network speeds and can be adapted to manage high-frequency spikes in traffic.
The RTMP live streaming protocol also provides encryption support through Adobe Access. However, this is not built into modern web browsers like Chrome or Firefox, so you need to install an RTMP plugin (like Flash Media Live Encoder) and add a custom tag for each video player.
The main drawback of the Adobe Flash Media Server is that it’s not compatible with iOS or HTML players. That’s why this protocol is less user-friendly than HLS.
How Does RTMP Streaming Work?
RTMP works by allowing the client to request a stream from the server. The Adobe Flash Media Server can provide different quality levels, depending on whether network bandwidth is sufficient or not. In case of a low connection speed for a given video setting, the real-time messaging protocol will automatically switch to a lower video resolution and frame rate.
The RTMP live streaming protocol can also adapt based on different network conditions. If the bandwidth drops, it will automatically reduce the quality of the stream until viewers get buffered. This flexibility allows you to adjust video streams in real-time for different devices and internet connections.
RTSP (Real-Time Streaming Protocol)
Real-time streaming protocol-RTSP is a primitive real-time transmission protocol with low latency that uses transmission control protocol (TCP) as transport protocol and allows reception from servers. The client establishes and controls one or more synchronized audio and video streams. It is specifically designed to be used in video transmissions.
It allows to control the servers for streaming, establishes a control session between the receiving client and the sender, and acts as remote control of the multimedia servers.
This protocol, for example, is used to control the Play, Pause, REW… functions of a Video On Demand system from a receiving client.
RTSP can also use UDP as its transport protocol, but it is not as reliable and secure as TCP
RTSP is similar to the HTTP Live Streaming (HLS) protocol. RTSP servers often work with Real-Time Transport Protocol ( RTP ) and Real-Time Control Protocol (RTCP) to deliver media streams.
It is an excellent choice for surveillance systems, IoT devices, and mobile SDKs. This protocol is prevalent for controlling various types of drones.
RTSP was used primarily by RealNetworks RealPlayer. RTSP requires a dedicated server for streaming and does not support content encryption or retransmission of lost packets. It relies on RTP protocol in conjunction with RTCP for streaming media delivery.
New Streaming Technologies & HTTP-Based Protocols
Several new streaming technologies and protocols use the Hypertext Transfer Protocol (HTTPS) as their foundation. These are discussed below:
HTTP (Hypertext Transfer Protocol) or HLS Protocol
HTTP is the most common protocol used by live streamers, also called HLS. In 2009, Apple introduced this protocol in its iOS devices, and since then, it has been used by many live streamers. An HLS live streaming protocol allows you to break down a video file into multiple small files to make a long story short, and then the server will then distribute these videos for playback.
Since smart TVs, Android, and iOS support HLS, it’s one of the most popular protocols.
The main advantage of the HLS lives streaming protocol is that there are no special requirements for video players because it uses a standard server and player. The HLS live streaming video protocol is flexible because it allows the video playback to adapt as needed, for example, if a viewer’s bandwidth is insufficient to view high-quality video. That’s why it’s popular, as it’s an adaptive bitrate protocol.
Why Should You Use HTTP or HLS streaming protocol?
The great advantage of this protocol is its high compatibility. Desktop browsers, Android devices, smart TVs, and iOS mobile devices are compatible with HLS. HTML5 video players are also natively compatible with HLS, compared to HDS and RTMP.
Aside from HLS’ broader device support and HTML5 player support, this protocol is one of the most secure on the market. That makes HLS the perfect protocol for streaming live video to massive viewer bases.
The only notable downside to the HLS protocol is the high latency when broadcast lives.
Pros of HTTP or HLS Protocol:
- No special requirements for video players
- Flexible, Adaptive bitrate streaming support allows the quality of live streaming to adapt to viewers’ bandwidth.
- High adaptability with desktop browsers, smart TVs, Android devices, and iOS.
- HTML video players are natively compatible without extra plugins or codecs.
- Highly secure with built-in encryption and authentication protocols
Cons of the HLS Streaming Protocol:
- Low latency streaming is a challenge using this protocol.
- Requires streaming media servers with fast download speeds.
- High bandwidth consumption for each stream (when compared to other protocols).
RTMP (Real-Time Media Flow Protocol)
RTMFP is Adobe’s proprietary streaming protocol. It was created as a part of the Flash platform and has some significant differences compared to RTMP.
It uses UDP for connectionless transmissions instead of TCP and supports multicast streams on top of unicast connections. It means that you can send multiple streams across one network connection, reducing bandwidth usage and latency.
RTMFP also allows you to control stream quality based on viewers’ internet speeds using a unique streaming protocol called HPACK compression. However, this only works with Adobe Flash as there is no support for HTML players or iOS devices.
HDS (HTTP Dynamic Streaming)
HTTP Dynamic Streaming is the evolution of RTMP by Adobe. It is a flash-based transmission protocol that is in the process of being deprecated. It allows adaptive bitrate streaming and has a high quality. HDS protocols support low latency streaming, although it is higher than RTMP due to the fragmentation and encryption.
Video codecs: H.264, VP6
Audio Codecs : AAC, MP3 Latency: 6-30 seconds
MPEG-DASH (Moving Picture Expert Group- Dynamic Adaptive Streaming Over HTTP)
Moving Pictures Expert Group (MPEG) develops an open-source protocol Dynamic Adaptive Streaming over HTTP (DASH), an industry-standard alternative to Apple’s HLS.
MPEG-DASH is one of the most modern streaming adaptive bitrate protocols and is considered one of the best in the industry.
This low latency protocol is codec independent, supports Encrypted Media Extensions (EME) and Media Source Extension (MSE), which are standards-based APIs for browser-based Digital Rights Management (DRM)
The big downside is that being a direct competitor to Apple and its HLS, MPEG-DASH is not compatible with Safari and iOS.
Video codecs: Independent codec
Audio codecs: Independent audio codec
Variant formats: MPEG-DASH CENC
Latency: 6-30 seconds
CMAF (Common Media Application Format): A Low Latency Protocol
Low Latency CMAF for MPEG DASH is an emerging technology for accelerating HTTP-based video delivery. Delivers ultra-fast videos at scale by using shorter data segments.
Playback Compatibility: Any player not optimized for low latency CMAF for DASH can fall back to standard DASH behavior.
Benefits: Low latency joins HTTP-based streaming
Latency: 3 seconds or less
RTP (Real-Time Protocol)
RTP is a real-time streaming protocol for the actual-time transmission of multimedia data- audio and video files in unicast or multicast mode. It works on the UDP protocol. It has a series of special real-time characteristics, such as time code on the video and the number of sequences.
It has a series of limitations, like; it doesn’t include recovery of packet loss. It has mechanisms to compensate for any minor loss of data. But it does not guarantee delivery, nor does it ensure that the packages arrive in order, the packages arrive on time, or the quality of service.
RTP requires a loaded buffer before starting the reproduction to ensure the quality of the service. This buffer is the buffer against the quality limitations in the sending and loss of data packets.
It involves the periodic transmission of control packets to all participants in a session. The main function is to provide feedback mechanisms to report on the quality of data distribution.
RTP is also the transport protocol used by WebRTC.
Real-Time Transport Protocol with Forwarding Error Correction (RTP-FEC)
RTP with forwarding error correction is broadly applicability to automatically repair any corruption and packet loss. Using this protocol requires more bandwidth than RTP without FEC.
RTCP (Real-Time Control Protocol)
Real-Time Control Protocol monitors transmission statistics and provides quality of service (QoS) feedback. RTCP helps synchronize multiple RTP media streams.
RSP (Resilient Streaming Protocol)
The resilient transmission protocol is the first live transmission technology that fully protects the loss of audio and video quality during a transmission, regardless of network interruptions.
RSP ensures full error-free video and audio delivery and traverses an HTTP connection’s firewall.
Low, predefined latency ensures nothing is buffered or packet loss due to network instability at the broadcast site.
RIST (Reliable Internet Stream Protocol)
RIST, which stands for Reliable Internet Streaming Protocol, is a low-latency, high-availability protocol suitable for long-distance applications. It is an open-source, open-specification transport protocol for reliable video transmission over lossy networks with low latency and high quality. It is currently under development within the “RIST Activity Group” of the Video Services Forum.
MSS (Microsoft Smooth Streaming)
MSS is Microsoft’s Adaptive Bitrate Video Streaming Protocol, designed with DRM security measures to protect content from piracy.
It is a hybrid media delivery method that works like streaming but relies on HTTP progressive download.
MSS protocol allows responsive delivery to all Microsoft devices. The protocol cannot compete with other HTTP-based formats.
Video codecs: H.264, VC-1
Audio Codecs: AAC, MP3, WMA
Latency: 6-30 seconds (lower latency is only possible when tuning)
SRT (Secure Reliable Transport Protocol)
It is an open-source protocol powered by Haivision. SRT is oriented to transporting video and audio signals with ultra-low latency under UDP.
SRT is a pure transport protocol independent of the codec. It guarantees that what is sent and received in the decoder is identical and without loss. It offers native AES encryption, making transmission security at the link level under encryption. It allows the firewall to pass through the entire workflow by supporting both send and receive modes.
Possesses the ability to detect network performance concerning latency, packet loss, jitter, and bandwidth available. SRT’s advanced integrations can use this information to guide the start of streaming or even to adapt endpoints to changing network conditions.
In a transmission stream under SRT, these three steps would be an example of how it works:
1. Encode the captured video in h264 (or h265, mpeg2)
2. Wrap this encoded data in MPEG-TS data
3. Wrap the MPEG data- TS in an SRT packet and send them to their destination.
More and more Broadcast manufacturers are adopting this protocol for video over IP networks, remote work, video in the cloud, the transmission of content under uncontrolled networks, etc.
ZIXI
Although Zixi is more of a platform that allows multiple protocols, it is also a proprietary protocol under license, which has the characteristic of being a very secure transmission system since it uses AES encryption. Zixi encapsulates the contents with proprietary technology based on FEC and ARQ.
The Zixi TS (Zixi protected transport) protocol is UDP- based, and dynamically adjusts to different network conditions. It employs proprietary error correction techniques to deliver error-free live video with ultra-low latency (less than 300 ms ).
It has network linkage, dynamic link quality evaluation, and optimizes transmission based on bandwidth.
WEBRTC (Web Real-Time Communications)
WEBRTC is an open-source protocol with JavaScript API for sharing videos between browsers. It offers almost very low latency, almost instantaneous voice, and video transmission to and from any browser, best for video conferencing. Its design is based on chat applications, more than video and audio, used in other areas.
Video codecs: H.264, VP8, VP9
Audio codecs: Opus, iSAC, iLBC
Playback compatibility: Chrome, Firefox, and Safari support WebRTC without any plugins
Benefits: Speedy and browser-based
Video Latency: less than 500 milliseconds delivery
MPEG-TS (Moving Pictures Expert Group(MPEG) Transport Stream)
The MPEG transport stream (MPEG-TS, MTS) or simply Transport Stream (TS) is a video container format, MPEG standard used for the transmission and storage of audio, video, and system and program information data (PSIP ). It is used in transmission systems such as DVB, ATSC, and IPTV.
TS is fast because it immediately starts streaming the video stream, typically MPEG-2. Other protocols, such as HLS, have to negotiate first which flow is best for your connection. HLS is slower but better than TS (due to a higher possible bit rate) and more reliable than TS.
CMAF (Common Media Application Format)
Apple and Microsoft came to the MPEG forum with a proposal. A new standard is called the Common Media Application Format (CMAF).
It is a new protocol, which has optimized workflows and reduced latency. This protocol is a container for files.
CMAF itself is a multimedia format. But by incorporating it into a broader system aimed at reducing latency, leading organizations are driving the industry forward.
An optimized stream divides the video into small chunks of a specified duration, published immediately after encoding. That way, delivery is near real-time, while subsequent snippets are still being processed.
Castr Supports these Streaming Protocols:
Castr is one of the most popular multistreaming software worldwide. It is an excellent solution for anyone who wants to stream simultaneously on YouTube, Facebook, other social platforms, and his websites.
Our video streaming technology supports the following protocols:
- RTMP
- RTSP
- MPEG-TS
- WebRTC
- SRT
- HLS
Conclusion
Video streaming protocols are vital for the distribution of video content. As we can see, each one has its features and characteristics to serve a different purpose with benefits such as reduced latency or excellent compression rates. It would help if you chose an appropriate protocol based on your specific needs and use cases. While some are open-source, others are proprietary and require a license. Due to its speed and reliability, the market is moving towards MPEG-TS, but you should not disregard other protocols like HLS or WebRTC. Ultimately, it all comes down to what works best for you.
We hope you have found this guide helpful in understanding video streaming protocols and their functions. Stay tuned for more posts on related topics!
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