As online video consumption continues to grow, so does the need for efficient streaming technologies. WebRTC and HLS have emerged as two popular solutions, each serving different streaming needs. While WebRTC is optimized for real-time communication, allowing direct peer-to-peer connections, HLS excels in delivering adaptive video streams across a wide range of devices. Understanding the strengths, limitations, and ideal use cases of each is essential for developers and businesses alike. In this article, we'll explore the core differences between WebRTC and HLS, helping you make an informed choice based on your streaming requirements.
WebRTC (Web Real-Time Communication) is a technology that enables real-time communication directly between web browsers, without the need for a third-party plugin. It provides a platform for developers to build applications that support features like video conferencing, voice calling, and screen sharing.
WebRTC (Web Real-Time Communication) is a technology that enables real-time communication directly between web browsers, without the need for a third-party plugin. It provides a platform for developers to build applications that support features like video conferencing, voice calling, and screen sharing.
Here are some of the key features of WebRTC:
Direct connection: WebRTC establishes a direct connection between the sender and receiver, bypassing intermediaries. This eliminates the latency and potential bottlenecks associated with server-based communication.
Reduced latency: By cutting out the middleman, WebRTC ensures a more responsive and real-time experience for users.
Audio and video: WebRTC enables the transmission of high-quality audio and video streams in real-time.
Data sharing: In addition to media, WebRTC allows for the exchange of data between applications, making it suitable for collaborative tools and interactive experiences.
Beyond media: Data channels provide a mechanism for exchanging data other than audio and video, such as text messages, application-specific information, or control signals.
Collaborative applications: This feature is particularly useful for applications that require real-time collaboration, such as online gaming, collaborative editing, and remote assistance.
JavaScript integration: WebRTC offers JavaScript APIs that developers can use to easily incorporate real-time communication features into their web applications.
Developer-friendly: The APIs are designed to be intuitive and accessible, making it easier for developers to build WebRTC-enabled applications.
Universal reach: WebRTC is supported by major web browsers, ensuring that your applications can reach a wide audience.
Consistent experience: Users can enjoy a consistent experience across different devices and browsers.
Encryption: WebRTC employs encryption to protect user data and prevent unauthorized access.
Authentication: It also includes mechanisms for authenticating users to ensure the security of communication.
Handling large numbers of users: WebRTC is designed to handle a large number of concurrent connections, making it suitable for large-scale applications like video conferencing platforms and online gaming servers.
Efficient resource utilization: WebRTC optimizes resource usage to ensure efficient performance even with many users.
The below diagram visually represents the flow of communication and data between web browsers, highlighting the key steps and components involved in establishing and maintaining real-time connections.
In essence, the WebRTC workflow is a process of establishing a direct connection between two web browsers and exchanging media streams and data in real-time. It involves various steps, including signaling, negotiation, SDP (Session Description Protocol), and ICE (Interactive Connectivity Establishment) to ensure a smooth and reliable communication experience.
HLS (HTTP Live Streaming) is a technology developed by Apple that enables the delivery of live video content over the internet. It works by breaking down a live stream into small, pre-segmented video files called ‘chunks’, typically in the MPEG-TS format. These segments are then delivered to the viewer over HTTP, allowing for flexible and efficient streaming.
Server-based: HLS requires a server to process the live stream and create the segmented files.
Adaptive bitrate: HLS supports adaptive bitrate streaming, which allows the server to dynamically adjust the video quality based on the viewer's network conditions. This ensures optimal playback quality, even in environments with fluctuating bandwidth.
HTTP delivery: HLS uses HTTP, a widely supported protocol, to deliver the segmented video files. This makes it compatible with a wide range of devices and platforms.
Fragmented MP4 (fMP4): HLS can also use fMP4 segments, which are smaller and more efficient than MPEG-TS segments.
Playlist files: HLS uses playlist files (typically in the .m3u8 format) to provide information about the available segments and their playback order.
Here's a breakdown of how HLS works:
WebRTC's combination of real-time capabilities, user experience benefits, business advantages, and technical benefits make it a compelling choice for developers and businesses seeking to build interactive and engaging web applications.
While HLS (HTTP Live Streaming) is primarily known for its server-based architecture and adaptive bitrate capabilities, customers often choose it for the following additional reasons:
HLS's combination of wide compatibility, adaptive streaming, scalability, and other benefits make it a popular choice for businesses and content creators seeking to deliver high-quality live streaming experiences.
Virtual assistant devices
The virtual digital assistant market is expected to grow by a factor of three by 2021, according to Tractica. One-way conversational assistants such as Amazon Alexa, Google Assistant, and Apple's Siri serve the majority of this market, and the devices they run on have become highly sought-after pieces of technology.
Amazon uses WebRTC for Alexa as well as several other products, including its online meetings and conferencing software, Amazon Chime, and a browser-based version of its Alexa devices.
Google Duplex uses real-time communications and artificial intelligence (AI) that allows users to have natural conversations and carry out real-world tasks over the phone. Similarly, Google Dialogflow enables developers to build voice and text-based conversational interfaces, including voice apps and chatbots.
Surveillance
Privacy and protection concern from governments, businesses, and consumers continue to rise as our society becomes more connected. According to IHS Markit, 130 million surveillance cameras are expected to ship globally in 2018, compared to fewer than 10 million shipped in 2006. Though surveillance has classically required at least one human being, that has begun to change. Now, facial recognition, pattern recognition, infrared, and other AI algorithms are aiding surveillance systems in identifying malicious behavior. With WebRTC, these systems can send automated alerts, which video surveillance technologies such as Amaryllo and ring.com already are doing.
Internet of things (IoT)
Globally, about 127 new devices are connected to the Internet every single second, according to a McKinsey Global Institute report. The IoT is experiencing significant growth, and the most important thing coming out of the IoT is the data passed between the machine-to-machine connections. This is what makes WebRTC prime for use in the IoT. One of many examples of how the IoT benefits from WebRTC is DroneSense, a software platform that powers drone and uses real-time communications for video conferencing.
Connected and self-driving automobiles
The number of autonomous vehicles expected by 2040 is estimated at more than 33 million globally, IHS Markits has reported. Cadillac, Mercedes, and Tesla are each in the process of implementing self-driving cars, among others.
Self-driving technology companies are implementing real-time communications, especially WebRTC, in high-profile self- and assisted-driving automobiles, like those from Waymo. It has the potential to be an easy-to-implement way to communicate, monitor, and control cars.
HLS (HTTP Live Streaming) isn't inherently designed for real-time interaction like WebRTC. However, it can be used in situations that have a real-time element, especially when combined with other technologies. Here are some potentials "real-time adjacent" use cases for HLS:
Important note: While HLS can be used in situations with real-time elements, it's not ideal for applications requiring ultra-low latency and true two-way communication. For those needs, WebRTC is a better choice.
In some scenarios, adopting a hybrid approach that combines WebRTC and HLS can be highly effective, especially when balancing the needs for real-time interactivity and broad video delivery. Here's a deeper dive into how developers might implement such a solution:
Real-time interaction with WebRTC:
Use case: WebRTC excels in scenarios where low-latency communication is crucial, such as live chat, video conferencing, or interactive components during a live event.
Implementation: Developers can integrate WebRTC for real-time interactions by setting up signaling servers to handle peer-to-peer connections. Libraries and frameworks like simple-peer or peerjs can simplify the WebRTC integration. WebRTC's peer-to-peer data channels allow for efficient, low-latency messaging between users.
Optimization: Ensure that the WebRTC configuration is optimized for minimal latency and high quality. This involves fine-tuning codec settings (e.g., VP8 or H.264), handling network conditions, and implementing appropriate error recovery mechanisms.
Broad video delivery with HLS:
Use case: HLS is well-suited for delivering video content to a large audience with varying network conditions. It's ideal for streaming pre-recorded or live video content that doesn't require real-time interaction.
Implementation: Implement HLS by encoding video into multiple bitrates and segmenting it into chunks. This can be achieved using tools like FFmpeg for encoding and segmenting. Deploy an HLS-compatible media server (e.g., AWS Elemental MediaPackage) to serve the content and handle adaptive bitrate streaming.
Optimization: Optimize HLS streams for performance by configuring segment lengths, tuning bitrate ladders, and ensuring efficient CDN distribution. Implement manifest prefetching and caching strategies to reduce startup times and improve playback performance.
The choice between WebRTC and HLS depends largely on your application's specific requirements. WebRTC is ideal for real-time, interactive experiences, while HLS is better suited for broader compatibility and adaptive streaming. To choose the right technology, consider factors such as latency, interactivity, device support, and scalability.
At FastPix, we use HLS to deliver adaptive bitrate streaming, ensuring smooth, high-quality playback on any device, regardless of network conditions. HLS excels in scalability and reliability, making it perfect for widespread content delivery.
Choose WebRTC when you need low-latency, real-time communication and interactivity, such as video conferencing, live collaboration, or gaming applications.
HLS is ideal for delivering high-quality video content to a large audience where real-time interactivity is not critical. It is suitable for live events, streaming services, and content delivery that requires adaptive bitrate streaming.
Yes, a hybrid approach can combine WebRTC and HLS. Use WebRTC for real-time interactions and HLS for broad video delivery, optimizing each for their respective strengths.
WebRTC is used in virtual assistants, surveillance systems, Internet of Things (IoT) devices, and connected/self-driving automobiles for real-time communication and data exchange.