Video Streaming Protocols : All You Need To Know

Role of Video Streaming Protocols in Present Times

The demand for OTT services has been witnessing a meteoric rise in recent times for which tools that can transmit video data in a seamless, uninterrupted manner are in need of the hour. This is where video streaming protocols come into play, acting as the invisible backbone that ensures smooth and seamless playback for movies, shows, and live streams. They act as a governing medium for defining rules and procedures about how the video information is encoded, packaged and delivered from server to device.   

Over the years, video streaming protocols have undergone significant evolution. But with the advent of more technologically advanced protocols like HTTP Live Streaming (HLS) and Real-time Transport Protocol (RTMP), it has led a progressive revolution in video streaming that has introduced dynamic functionalities such as the adaptive bitrate for automatic adjustment to varying bandwidth situations and low latency streaming for realtime applications. This blog delves into the video streaming protocols that empower the OTT evolution and its emerging technologies that will give a path to more secure and efficient processes.

Video Streaming Protocols That Are Currently Leading The Video Streaming Scene

1) HTTP Live Streaming (HLS): 

HLS is one of the most prominent video streaming protocols for delivering content on OTT platforms. It uses HTTP protocol, popularly used for web surfing, to send large components of web content into small manageable chunks over the internet for efficient delivery. Serving as a dynamic and powerful tool for optimizing content, it adjusts the video according to the user’s device on which it is being played to ensure a seamless watching experience. It smoothly adapts to the available bandwidth of the network and conditions of the playback options with the help of Adaptive Bitrate (ABR), making it a reliable streaming technology for content delivery. This allows users to play lower-quality videos over slower connections and enjoy higher-quality videos in case of higher network bandwidth availability. 

Developed by Apple, it is continuously being improved with enhancements such as Low Latency HLS for real-time experiences and CMAF (Common Media Application Format) for enhanced interoperability across platforms.   

Features:

- Efficient Buffering: For streaming video content, smooth buffering is one of the most essential aspects of a seamless watching experience. HLS guarantees efficient buffering by pre-loading and downloading the content ahead of its playback position, ensuring interruption-free streaming, even in case of net connectivity problems.

- Code Agnostic: HLS allows freedom to content providers in choosing the codecs themselves (like H.264, HEVC) based on their specific content requirements and focuses on delivery mechanism instead, which involves packaging, segmentation and delivery. This aids HLS in adapting to emerging encoding technologies without any problems by not limiting itself to specific codecs. 

- Security: HLS facilitates content protection with the implementation of security messages including encryption and Digital Rights Management (DRM). By encrypting the content, HLS makes it unreadable and impossible to access without a decryption key to prevent unauthorized access. DRM systems such as Widevine and FairPlay ensure effective management of distribution and decryption keys for only authorized users and applications.

2) MPEG-Dash:

Also known as Dynamic Adaptive Streaming over HTTP (DASH), MPEG-Dash comes under the adaptive bit-rate protocol category that supports high-quality streaming with adjustable quality depending on the network bandwidth of the user’s device. It uses segment-based delivery and relies on Media Presentation Description (MPD) files that provide detailed information about bitrates, resolutions and codecs. 

Features:

- Standardization and Interoperability: Backed by the international standard ISO/IEC 23009, MPEG-Dash ensures seamless interoperability and standardization across platforms and various implementations of the protocol. This benefits content providers by ensuring wider reach through reduced fragmentation and and negates the requirement for catering to specific platform-based needs. 

- Code Agnostic: MPEG-Dash provides similar functionality as HLS by giving content providers the freedom to choose the most appropriate codec for their content. 

- Signaling Mechanism: The signaling mechanism plays an essential role in communicating vital information about available representations (bitrates, resolutions etc) to the user’s device and guides the playback process. It uses two types of signaling mechanisms:

In-bound signaling: In this form of signaling, video segments contain necessary information such as available representations and switching cues which removes the requirement to add separate signaling files and comes with the cost of heavy segment sizes.

Out-of-band signaling: It makes use of separate file sizes to contain information about available representations and switching cues and is delivered through HTTP or MPD files. 

3) Real-Time Messaging Protocol (RTMP):

RTMP is considered highly ideal for live streaming content such as broadcasts, webinars, and online gaming streaming as it provides exceptional performance in terms of low latency and real-time streaming. It focuses on a consistent and persisting connection between the software and the server, facilitating immediate data transfer, and reducing delays between the streaming server and video being played on the user's device. With the help of reliable data delivery, RTMP ensures there are no errors in the streaming process and the content is delivered accurately. 

Features:

- Stream Recording: RTMP provides the functionality of recording the video on the server side, which allows for playback later or during post-production editing. With this functionality, live event videos can be archived or can be used for (Video-On-Demand) content. 

- Metadata feature: It facilitates the transmission of metadata along with the video content such as captions or content information like song names, venue or member data etc that can be displayed during its streaming.

- Simplified Setup: With a simplified setup, people with basic technical knowledge can deploy live streaming with RTMP protocol which makes it accessible for wider usage.

4) Secure Reliable Transport (SRT): 

SRT is an open-source protocol that is designed for reliable and secure delivery of content, especially in terms of unstable networks. With its ability to solve low-latency challenges such as packet loss, bandwidth limitations, and video transport, it is considered idea for live streaming, video conferencing, and remote broadcasting. By involving aspects from both User Datagram Protocol (UDP) and Transmission Control Protocol (TDP), SRT delivers seamless content delivery maintaining the qualities of low latency and reliability.  

Features:

- Reliable Delivery: SRT makes sure that the content is delivered even in cases of data loss and jitter by utilizing error correction mechanisms such as Forward Error Correction and retransmission that maintain integrity and reliability in unstable scenarios. Additionally, it sends redundant information to the sender side so that in case of data loss, the redundant data can be used to keep the streaming quality smooth and minimize delays.  

 

- Firewall Traversal: It facilitates wide usage in real-life applications with its ability to dynamically adjust itself towards traverse firewalls and network address translations and transmit data seamlessly.

 

- Low Latency: By utilizing mechanisms such as selective acknowledgments, packet ordering techniques and forward error correction (FEC) to correctly identify corrupted or missing pockets, and ensure only correct information is transmitted along with redundant data for additional backup, SRT provides effective content delivery with low latency.

5) WebRTC (Web Real-Time Communication):

WebRTC is an open-source project used for real-time and peer-to-peer communication that can be easily conducted over web browsers. It is a fairly new technology that provides software developers with application programming interfaces created in JavaScript. That comes with extensive compatibility support for video, audio and text-based content and doesn’t require any additional plugins or software downloads to run.

- Peer-to-Peer Communication: It removes the requirement of a central server, enabling the establishment of direct connections between devices and facilitating data exchange between peers without intermediaries. Without a central server, it substantially reduces latency and dependence on external servers by distributing the workload and responsibility across involved devices. Through direct communication between peers, WebRTC is able to handle a large number of connections taking place simultaneously without putting excessive load on servers.

- Media Capturing & Streaming: WebRTC utilizes the Media Capture and Streams API to access user devices like webcams and microphones. This API provides methods for requesting access to media devices, handling permission requests, and capturing the media stream. Developers can specify constraints like resolution, frame rate, and audio quality to control the captured media.

Emerging Technologies in Video Streaming Protocols:

1. ABR (Adaptive Bitrate Streaming) Evolution: The future of video streaming protocols will witness personalized adaptation based on user preferences, device capabilities, and content type, optimizing user experience and resource usage. Different devices have varying processing power and display capabilities. Adaptive bitrate streaming would enable providing dynamic video streaming experiences to a larger audience with diverse needs. It would offer automatic adjustment of video content delivery from low quality to high-quality videos to the user’s device ranging from low bandwidth and limited capabilities to great net connectivity and advanced models. 

2. AI-powered Content Delivery Networks (CDNs): The future video streaming protocols will witness the confluence of artificial intelligence and Content Delivery Networks (CDNs) that will revolutionize the OTT scene with content delivery based on personalized, optimized and secure video transmission. CDNs powered with AI will provide precise prediction for user demand, optimize routing and pre-cached content as per real-time network status and change user behavior for a seamless and personalised viewing experience for the user. 

3. Next-generation Video Codecs: Along with video streaming protocols, video codecs offer high-efficiency compression for video data that makes it easier to deliver content at faster loading times without compromising its quality. Advanced video codecs like AV1 and VVC( Versatile Video Coding) are gaining traction in terms of popularity for their significant improvements in video compression, and high-quality video delivery at lower bit rates as compared to their older counterparts like H.264 and HEVC.

Final Thoughts

As the ever-expanding world of online video demands reliable and efficient delivery, video streaming protocols continue to ensure smooth playback of our favorite content. From humble beginnings to the adoption of features like adaptive bitrate streaming and low-latency options, these protocols have constantly evolved to meet user needs. The future of video streaming protocols is set to witness the rise of next-generation video codecs like AV1 and VVC. These codecs offer significant compression improvements, paving the way for higher-quality video at lower bitrates. With collaborative efforts towards standardization and optimization, video streaming protocols will continue to shape the way we consume and enjoy online video content for years to come. If your business is looking to develop its own set of dynamic video solutions or tech-based solutions in general, we’d love to be your partner in the journey. You can contact us here, and our experts will get back to you in 24 hours.