Blog Summary:

WebRTC enables real-time, low-latency communication for video calls, gaming, and instant file sharing, while HLS delivers high-quality, scalable streaming for IPTV, OTT platforms, and live events. The right implementation depends on latency, scalability, and audience size. This blog explores WebRTC vs HLS, their differences, use cases, and how businesses can leverage them for better user experiences.

Why does choosing the right streaming protocol matter? For SpaceX’s Crew Dragon Demo-2 launch, the choice was more about which technology has the shortest delay of seconds to ensure viewers stay hooked until the launch countdown hits zero.

This live product launch was streamed on NASA’s official YouTube channel and SpaceX’s website on May 30, 2020. The event used HTTP Live Streaming (HLS) for high-definition streaming to millions of viewers.

However, Web Real-time Communication (WebRTC)-powered news broadcasts and site launch viewers posted updates seconds before YouTube viewers could reach the end of the countdown. Hence, understanding when to use HLS and when to leverage WebRTC can make all the difference. 

The right streaming protocol ensures you’re in the moment, not playing catch-up. A few seconds can make or break a viewing experience when your viewers are ready to:

  • Stream a high-stakes Super Bowl match, but the goal starts buffering just as the ball is about to hit the net.
  • Air a weekly cliffhanger episode, but social media floods with spoilers before the stream catches up.

So, what goes behind this selection? Let’s fight the WebRTC vs HLS battle.

What is WebRTC?

WebRTC is much more than a streaming protocol. Released by Google in 2011 as an open-source project, WebRTC has a number of protocols, codecs, and JavaScript APIs that help implement data-sharing features in web applications.

To establish a connection, a user needs to know the exact IP address. This IP address is usually hidden behind the Network Address Translation (NAT) traversal, which makes it easier for devices to connect directly.

To facilitate data transfer, it consists of the following core components:

  • getUserMedia provides access to a camera and microphone to capture audio and video.
  • RTCDataChannel enables real-time bidirectional data exchange across browsers.
  • RTCPeerConnection allows direct communication between peers and browsers.

Normally, when a user makes an audio or video call to another user, it goes through a server. This means the data from one user, such as audio, text, or video, passes through a middleman (the server) before it reaches another user.

With WebRTC, users can create a direct bridge between themselves and make the call without any mediator. Because of its lower latency, WebRTC architecture speeds up real-time communication. It’s also more secure because the call has fewer points to intercept data. Moreover, it’s highly efficient due to less bandwidth and the cost of setting up a server.

However, WebRTC is not a scalable technology, and evolving its workflow to meet growing business demands often affects its latency.

Use Cases of WebRTC

WebRTC is used in many interactive streaming applications for business, including video conferencing, audio and video calls, sharing files and screens, streaming live events, and even emergency responses. Online gaming and smart devices with IoT also utilize WebRTC for remote monitoring and security.

Let’s understand these in detail:

Real-time communication

Suppose a rural area has a hospital but lacks specialty doctors. When a patient needs a heart or cancer specialist, the nearest doctor is about 200 miles away.

To make it convenient for the patients, the hospital sets up a WebRTC-enabled telemedicine system for real-time calling. This saves time and money while also providing instant expert care.

File sharing

WebRTC is a great option for peer-to-peer (P2P) file transfers between two devices. It doesn’t need a server to store or deliver the data, making file sharing faster, more secure, and more efficient.

It has no intermediary, is encrypted, and has no storage limits. Whether it’s an email attachment or a 10GB video, a WebRTC file-sharing app like ShareDrop instantly sends files over a direct P2P connection.

Streaming

WebRTC helps develop video streaming apps with the lowest delay of up to 0.5 seconds, enabling low-latency transmission between devices. It is almost real-time and can run in browsers without any expensive media servers.

In live tournament commentary, the audience receives reactions in real time without any buffering or long delays.

Other Use Cases

Apart from the above, some other use cases of WebRTC include live customer support chats to connect with representatives instantly. Online education with virtual classrooms also utilizes WebRTC for interactive streaming applications, digital whiteboards, and sharing screens.

Online multiplayer games and IoT are used in smart devices for near-zero delay and crystal-clear audio and video communication.

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What is HLS?

Apple introduced HTTP Live Streaming (HLS) in 2009. It revolutionized the streaming landscape by building a protocol that adapts its capability based on Internet speeds.

While watching a video or a movie, it ensures buffer-free media streaming to the user over the Internet. HLS streaming protocol enables a powerful technology called adaptive bitrate streaming (ABR), even if the Internet speed changes, instead of downloading entire content at once.

The ABR concept works like this: If the Internet speed is good, the user gets high-quality video. On the other hand, if the Internet is slow, the user gets low-quality video. Hence, HLS adjusts the quality of video according to changes in the Internet speed. It splits videos into small chunks and switches the quality based on network speed.

HLS encodes the media content into numerous variants with different bitrates and resolutions. Then, it divides these variants into segments of short durations (2 to 10 seconds long). Lastly, HTTP delivers these segments to the devices through a Content Delivery Network (CDN).

HLS streaming protocol has another variant called low-latency HLS (LL-HLS), which Apple introduced to minimize latency by reducing the delay between capturing video and playing it.

So, if a user is watching a live Super Bowl football match, the regular HLS will stream a goal 10 to 30 seconds after it happens. Conversely, the LL-HLS will stream it within 3 to 5 seconds—almost in real-time.

Use Cases of HLS

HLS allows multiple video and audio stream resolutions while also handling HD live streaming with little buffering or latency. Its technology of adjusting bandwidth based on network speeds delivers uninterrupted playback streaming.

Let’s understand these use cases:

Live Event Streaming

Streaming live events like music concerts or the Super Bowl is smoother with HLS because it adapts to each viewer’s network speed. This means that it will stream in HD for fast networks and switch to SD for slower speeds. Even if a connection drops from a higher resolution, the streaming continues irrespective of the device.

Video-on-demand (VOD) Services (OTT Platforms)

HLS streaming protocol is one of the most widely used applications for VOD. Popular platforms include Hulu, Netflix, Prime, etc., which allow users to stream videos anytime without downloading. Based on the network speed, users can watch on any device. More than 85% of the global streaming platforms use HLS for VOD due to its adaptive streaming.

Mobile Streaming

HLS for mobile streaming enables audio and video delivery to a wide audience through different media players. Since many browsers, such as Chrome, Safari, and Edge, and various OSs, like Android and iOS, support HLS, CDN costs are also reduced.

This saves bandwidth by dynamically adjusting the resolution, such as 1080p to 480p when the network fluctuates across 3G, 4G, or Wi-Fi.

Corporate Communications

For corporate communications, the HLS streaming protocol primarily delivers pre-recorded and live video content. This includes training videos and event streams for global teams across different networks.

Check out our guide on building a live streaming app development for encrypting sensitive content such as internal training, meetings, presentations, product demos, webinars, etc.

E-learning Platforms

In e-learning platforms, HLS is used to deliver video lectures and other educational content, such as recorded lectures, courses, and live classes. It’s ideal for streaming without buffering issues where the network has inconsistent speed. It makes it easy to deliver webinars, live classes, video libraries with recorded lectures, and even transcripts to smartphones, tablets, or desktops.

IPTV Services

HLS in Internet Protocol Television (IPTV) delivers TV channels and on-demand videos over the Internet instead of satellite or cable TV. HLS encodes videos in different formats by using the server to split them into small chunks and deliver them over HTTP. The IPTV player then fetches the next video segment through an M3U8 playlist file.

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In-depth Difference Between WebRTC VS HLS

WebRTC vs HLS is a battle between two highly accepted streaming protocols. Streaming protocols are delivery mechanisms that break up a video into smaller chunks, transport it, and rearrange it.

Since videos are heavyweights, they require a proper setup to create a journey with multiple stops from their source to the user. The two most popular streaming protocols are WebRTC and HLS. Let’s take a closer look at how they differ.

Here’s a quick tabular comparison between WebRTC and HLS:

Particulars HLS WebRTC
Device Compatibility All except Linux All
Latency High, 6 to 30 seconds for HLS; with LL-HLS, it’s up to 3 seconds. Low, sub-500 milliseconds
Security Medium, HTTP Secure (HTTPS) protocol, Common Encryption (CENC, fmp4, DRM High, Mandatory Secure Real-time Transport Protocol (SRTP), available ones with platform
Supported Codecs H.264, H.265 H.264, H.265, VP8, VP9
Delivery Methods and Speed TCP, Server-based buffering is delayed by a few seconds P2P, UDP, Instant
Video Quality High focus on quality High focus on speed
Use Cases VOD, Live streaming, mobile streaming, IPTV, OTT Voice and video calls, online conferences, chat
Scalability Easily scalable with LL-HLS; however, not widely accepted Low, only for up to 50 viewer-base, requires a CDN for larger user base

Let’s understand the HLS vs WebRTC differences in depth:

Streaming Latency

HLS has a higher latency (5-30 seconds) due to transmission control protocol (TCP)- based segment fetching. However, Apple introduced LL-HLS at the Apple Worldwide Developer Conference (WWDC) 2019. This variant reduced HLS latency from 10-30 seconds to under 3 seconds.

Conversely, WebRTC minimizes interference and pushes media instantly, achieving ultra-low latency of around 300ms.

Who Scored: HLS is sufficient for most applications. However, interactive streaming applications such as auctions and telehealth may require WebRTC’s ultra-low latency, which makes it ideal for building real-time video streaming app interactions.

Adaptive Bitrate for Videos

HLS and LL-HLS excel in video quality through ABR streaming, which allows smooth playback across multiple devices and network conditions. In contrast, though WebRTC lacks ABR, it enhances real-time latency and video quality with simulcasting and selective video coding (SVC).

WebRTC’s selective forwarding units (SFU) architecture dynamically adjusts bitrate, optimizing playback. HLS with ABR generates multiple bitrate versions of a video indexed in a multivariant playlist.

Who Scored: The HLS player selects segments based on network conditions. However, these segments are six or more seconds long. This can make adaptation slow, sometimes causing buffering if a high-bitrate version is fetched on a weaker connection.

Scalability and Performance

LL-HLS provides better latency but remains in development. It has CDN integration and the ability to scale to thousands of viewers with up to 3 seconds of latency. WebRTC excels in ultra-low latency but was originally designed for smaller, chat-based interactions.

However, developers and vendors are actively working on scalable WebRTC solutions, which would enable larger deployments with real-time interactivity.

Who Scored: While HLS scales through CDNs, WebRTC achieves scalability through SFU-based architectures, making both viable for high-demand streaming applications.

Security and Encryption

With secure real-time transport protocol (SRTP), WebRTC employs a highly secure encryption method ideal for sensitive applications. Though WebRTC lacks digital rights management (DRM) support, it compensates with TLS encryption for traffic handling and securing data in transit.

HLS and LL-HLS both provide strong protection against video content with authentication, DRM, common encryption (CENC), and token authentication. They rely on HTTPS encryption, which is secure but less robust than SRTP.

Who Scored: Identity management must be implemented separately for WebRTC. While the HLS streaming protocol excels in content protection, WebRTC remains a strong choice for real-time encrypted communication.

Network Bandwidth and Stability

HLS uses ABR to pre-encode multiple bitrate versions and select the best version based on network conditions. However, it has a slower response time with 6+ second segment lengths, creating lags and buffering issues when fetching a high bitrate on a weak connection.

WebRTC works on packet-based delivery, SFUs, modern codecs like AV1 and VP9, SVC architecture, and user datagram protocol (UDP) to provide better control over congestion, resolution, and frame rates, reducing bandwidth requirements.

Who Scored: Unlike HLS, WebRTC can choose to retransmit lost packets or move on with congestion control algorithms to estimate bandwidth dynamically. HLS supports HEVC (H.265) but lacks native SVC support, making WebRTC more adaptable for real-time, low-latency applications.

Similarities Between WebRTC and HLS

HLS vs WebRTC is a powerful battle between two streaming technologies that excel in different use cases. HLS is great for large-scale, high-quality interactive streaming applications, and WebRTC is perfect for instant, peer-to-peer communication.

However, their shared focus on device compatibility, adaptive streaming, security, and low latency makes them essential for modern video-driven applications.

Let’s understand how:

Adaptive Resolution for Streaming

Both WebRTC and HLS are streaming protocols that can optimize the streaming resolution in real time based on the network speed and ensure the best possible video quality.

On the one hand, HLS uses ABR to dynamically adjust the quality of video by automatically lowering or increasing the resolution to prevent buffering. Similarly, WebRTC also adjusts the bitrate in a real-time session, such as a video call. It reduces the resolution or frame rate in unstable network conditions.

Security

Both technologies prioritize data security and encryption to ensure media is delivered privately and safely. HLS streaming protocol gate keeps the content to prevent unauthorized access to content with encryption mechanisms like AES-128 and DRM. WebRTC ensures End-to-End Encryption (E2EE) across video calls, meetings, conferences, and other types of sensitive communication.

Cross-platform Compatibility

Both streaming protocols work seamlessly across multiple devices and browsers, offering compatibility for cross-platform accessibility. They both support major browsers like Chrome, Safari, Firefox, and Edge.

Moreover, they also ensure broad accessibility across smartphones, smart TVs, media players, and desktops to engage users in real-time interactions without any external software or plugins.

Future Trends for WebRTC

WebRTC provides frictionless audio and video conferencing without the need for additional plugins or software, earning itself a high adoption rate across various sectors.

Moreover, it also addresses the demand for cost-effective tools that function smoothly on mobile platforms owing to the immense growth of smartphones and tablets.

Let’s understand what the future holds:

Integration with IoT and AR/VR

WebRTC is set to integrate with revolutionary technologies like the Internet of Things (IoT), 5G, and Augmented Reality/Virtual Reality (AR/VR) for real-time video feeds, alerts, and streams. 360-degree video streaming for meetings and simulations, guided video calls for remote diagnostics, and interactive social VR will also take over.

Enhanced Interactivity

WebRTC will offer a standard framework. For instant communication, this framework can embed real-time communication features directly into mobile browsers. Small and medium enterprises (SMEs) will also be able to leverage WebRTC for efficient communication across geographically remote location teams.

AI-powered enhancements

Artificial Intelligence (AI) and WebRTC will enhance real-time interactive streaming applications with voice recognition, natural language processing, and sentiment analysis. Optimized video compression, object recognition, and real-time translation will also enhance AI-generated videos.

Future Trends for HLS

HLS will evolve into more personalized and interactive technology, combining the power of AI and 5G. On one hand, users will be able to discover more enjoyable content with AI.

On the other hand, the integration of 5G will facilitate faster streaming speeds. 4K and 8K content will also become more common with immersive streaming.

Let’s understand what the future looks like for HLS:

Low-latency HLS (LL-HLS)

LL-HLS will power real-time communication by reducing latency from 30 seconds to under 5 seconds, making it ideal for live events like sports streaming. 5G network and EDGE computing will further aid higher resolutions of 4K and 8K with less buffering and process HLS streams with reduced delays.

Adaptive Bitrate Optimization

AI and ML will work together to personalize video streaming and change video quality based on user behavior. AI-driven analytics will also optimize ABR to deliver real-time content recommendations for personalized viewing.

Content Delivery Network (CDN) Integration

The low-latency delivery of video content will be further optimized by edge caching and improved CDN services. Cloud gaming and interactive streaming will also evolve, with LL-HLS becoming more dominant in cloud gaming with minimal lag for real-time interactions.

Security Enhancements

HLS will integrate blockchain to enhance the security and protection of video content over the Internet and prevent unauthorized distribution and access. It will also include advanced DRM protection for IPTV and OTT services.

WebRTC or HLS? Find Out Which is Right for You

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WebRTC Vs HLS: Which is Better?

The choice between WebRTC and HLS depends on your business needs and the type of user experience you want to offer. Choosing the right streaming protocol depends on factors like latency requirements, scalability, and use case complexity.

Are video conferencing, online gaming, telehealth, and instant file sharing your priorities? Then, WebRTC is the perfect peer-to-peer communication protocol designed for low-latency, real-time interactive applications.

Or do you want to optimize bandwidth usage, deliver adaptive streaming, and support millions of concurrent users? In that case, HLS is the deal-breaker for on-demand and live video streaming services like IPTV, OTT platforms, mobile streaming apps, and large-scale live events.

Conclusion

Are you looking for a real-time communication solution or a scalable media streaming service?

WebRTC and HLS streaming protocols serve different purposes—WebRTC powers real-time interactions, while HLS ensures smooth, scalable streaming. The former is ideal for video calls and live collaboration, while the latter supports IPTV, mobile streaming, and live event broadcasting.

At Moon Technolabs, we specialize in custom WebRTC and HLS development. We are a leading WebRTC development company that ensures businesses get the right streaming solution for seamless user experiences.

Whether you need a real-time collaboration platform or a buffer-free streaming service, our expert team delivers tailored solutions to match your goals. Contact us today to build the next-generation video, streaming, or communication app!

FAQs

01

Is WebRTC good for streaming?

Yes, WebRTC is a great choice for live streaming in interactive applications such as video conferencing and online gaming because it adjusts video quality based on the network speed and bandwidth. Since it has low latency, it can easily work across various browsers without any communication delay or additional software.

02

Does Twitch use WebRTC or HLS?

Twitch uses HLS (HTTP Live Streaming) by transcoding the streams from RTMP (Real-Time Messaging Protocol) into HLS. HLS is better suited for larger audiences than WebRTC, which is designed for low-latency video calls. It is highly adaptable to the bandwidths of different browsers and offers multiple video quality options for different viewer bases.

03

Is WebRTC a full duplex?

Yes, WebRTC is a full duplex protocol since it allows simultaneous data transfer between two peers in real-time communication. It enables direct communication of both media and data channels between devices where users can speak and listen at the same time without relying on a central server.

04

Can WebRTC and HLS be combined in a single streaming solution?

Yes, WebRTC and HLS can be combined into hybrid streaming solutions, using the former for source streaming and the latter for delivering videos to viewers. In live events, WebRTC is an excellent option for real-time interactions, and HLS is best for handling different network bandwidths for large audiences.
About Author

Jayanti Katariya is the CEO of Moon Technolabs, a fast-growing IT solutions provider, with 18+ years of experience in the industry. Passionate about developing creative apps from a young age, he pursued an engineering degree to further this interest. Under his leadership, Moon Technolabs has helped numerous brands establish their online presence and he has also launched an invoicing software that assists businesses to streamline their financial operations.