WebRTC and HLS (HTTP Live Streaming) are two popular technologies used for real-time communication and low-latency video streaming respectively. With the rising importance of applications like video conferencing, live streaming, remote collaboration, etc, there is often a debate around WebRTC vs HLS for delivery of such services. The objective of this blog is to provide a comparative analysis between WebRTC vs HLS across various parameters like latency, infrastructure needs, complexity, use cases, and mortae.
WebRTC (Web Real-Time Communication) is an open-source project that enables real-time peer-to-peer communication capabilities like voice calling, video chat, and file sharing directly within web browsers. The key components of WebRTC include media streaming protocols like RTP and APIs for accessing cameras, microphones, etc. WebRTC eliminates the need for plugins or additional software by building these capabilities right into the browser.
HLS is a HTTP-based media streaming protocol developed by Apple. It works by breaking the overall stream into smaller fragments which are delivered over HTTP. HLS is supported across various platforms like desktop browsers, mobile devices, smart TVs, etc. It allows adapting streaming quality as per changing network conditions.
As consumers demand richer interactive experiences and businesses enable more remote collaboration capabilities, there is a need to deliver low latency and real-time communication. This is driving the adoption of technologies like WebRTC and HLS. While both aim to deliver high-quality streaming, they have intrinsic differences that make them suitable for certain use cases over others. Analyzing WebRTC vs HLS allows selecting the right technology for delivering next-generation services.
WebRTC Overview
WebRTC enables real-time communication capabilities like voice calling, video calling, file sharing, and data channels directly in websites and apps. Open source projects and standards like WebRTC, ORTC, WebSockets, etc power it.
A key advantage of WebRTC is its use of peer-to-peer connections between end-user devices which minimizes latency. Media streams are encrypted end-to-end for security. As it leverages browser capabilities, there is no need for additional plugins or software downloads.
Some key features of WebRTC include:
- High-quality voice and video calling between peers
- Native file sharing and data channels
- Low latency by establishing direct connections
- End-to-end media encryption for security
- Screen sharing for improved collaboration
- Browser API access to microphone, camera etc
- Supported natively in modern web browsers
With WebRTC being supported across billions of devices like desktops, smartphones and tablets, it unlocks powerful real-time engagement opportunities across various use cases. The technology provides a secure and scalable foundation for adding live communications in any app or website.
HLS Overview
HLS (HTTP Live Streaming) is a media streaming protocol developed by Apple. It is commonly used to stream and playback live and on-demand video content on the web.
HLS works by splitting the overall video stream into smaller file segments of varying qualities. These segments are delivered using conventional HTTP requests and playlists. As HLS uses the ubiquitous HTTP protocol, it is compatible with a wide range of devices like desktops, mobiles, smart TVs, and set-top boxes.
Some key aspects of HLS include:
- Splitting video into segments of varying quality levels
- Serving segments over HTTP for broad compatibility
- Adaptive bitrate streaming by switching quality based on network conditions
- Requires a streaming server, cannot establish direct peer connections like WebRTC
- Primarily used for video delivery from server to client
- Typical latency is higher compared to WebRTC’s optimized low-latency streams
While HLS provides a solid foundation for delivering adaptive streaming video to devices, it lacks WebRTC’s innate support for ultra low-latency interactivity. The latency and infrastructure requirements of HLS make it less ideal for applications like video conferencing, live collaboration, etc.
Main Parameters for Comparison
WebRTC to HLS have some inherent differences that make them suitable for certain applications over others. Let’s compare them across key parameters:
- Latency and Lag
- Peer-to-peer vs Client-Server
- Platform and Browser Support
- Encryption and Security
- Scalability
- Cost & Infrastructure Requirements
Latency and Lag
| Parameter |
WebRTC |
HLS |
| Overall Latency |
Optimized for lowest latency |
Higher latency |
| Architecture |
Leverages direct P2P connections between endpoints |
Relies on client-server model |
| Typical Latency |
<150ms |
>250ms |
| Real-time communication |
Near real-time peer communication possible due to minimal lag |
Delays and lags noticeable in live streaming due to higher latency |
| Reasons for low latency |
- Eliminates intermediary network hops
- Media streamed directly between peers
|
- Video routed through servers
- Buffering induces lags
|
| Use cases |
Suited for applications like:
- Video conferencing
- Remote collaboration
- Telehealth
- Live chat
|
Better for:
- Streaming live events
- Video on demand
- Live linear video
|
WebRTC is designed to achieve lowest possible latency by streaming media directly between peers. This allows near real-time communication with a lag of less than 150ms. The direct peer-to-peer model avoids additional network hops and minimizes delays. This makes WebRTC ideal for use cases like video conferencing, telehealth and live collaboration which require real-time interactivity.
In comparison, HLS has higher inherent latency due to its client-server architecture. The video is routed through servers and buffered at the client side which induces lags. The latency typically ranges from 250ms and above. This makes HLS unsuitable for applications needing real-time engagement. However, the higher latency is acceptable for streaming live/VOD video to passive viewers.
Peer-to-peer vs Client-Server
| Parameter |
WebRTC |
HLS |
| Architecture |
Peer-to-Peer |
Client-Server |
| Media handling |
Media is transmitted directly between end-user browsers |
Media is routed through intermediary streaming servers |
| Latency |
Very low latency as media streamed directly between peers |
Higher latency as media streams pass through servers |
| Scalability |
P2P model distributes load across peers |
Adding viewers increases load on streaming servers |
| Bandwidth usage |
Efficient bandwidth utilization as media not replicated through servers |
Can lead to bandwidth congestion on server uplinks |
| Real-time engagement |
Enables true real-time communication with minimal lags |
Delays noticeable due to network hops and buffering |
| Use cases |
Video conferencing, remote collaboration, interactive live streaming, etc |
On-demand video streaming, live event broadcasting, linear TV distribution |
A defining characteristic of WebRTC is its ability to create direct peer-to-peer connections between web browsers and devices. This removes the need for an intermediary streaming server. Media is transmitted directly between endpoints, minimizing latency, server loads and network congestion. True real-time communication enables use cases like high-quality video conferencing.
In contrast, HLS relies completely on a client-server architecture for delivering media streams. While this allows reaching scale, it increases latency, and network hops, and can create bandwidth bottlenecks on the server uplinks. Scaling requires continually expanding server-side resources. The inherent latency makes HLS unsuitable for interactive applications.
Platform and Browser Support
| Parameter |
WebRTC |
HLS |
| Browser Support |
Supported natively in Chrome, Firefox, Edge, Opera, etc |
No native browser support, needs client libraries |
| Mobile Support |
Works seamlessly on both Android and iOS platforms |
iOS has native support. Android needs integration |
| Desktop Support |
Supported on Windows, Mac and Linux |
Needs browser plugins or app integration |
| Ease of Integration |
Can be added with simple JavaScript without additional libraries |
Requires integrating platform-specific SDKs and players |
| Reliable Support |
Consistent support across most modern browsers and devices |
Robust support limited mainly to Apple ecosystem |
| Reason |
Standards-based approach using ubiquitous web APIs |
Proprietary protocol tied to specific platforms |
One of WebRTC’s major edge is native support across billions of modern web browsers like Chrome, Firefox, Edge etc on both desktop and mobile platforms. It can be easily integrated into any web or mobile app using simple JavaScript APIs without needing additional libraries.
In comparison, HLS has fragmented support across platforms and browsers. There is no native browser support and it requires integrating platform-specific players and SDKs into apps. While iOS has robust native HLS support, Android needs integration of third-party libraries. Reliable support is limited mainly to Apple devices and browsers. The proprietary nature of HLS causes this ecosystem lock-in.
Encryption and Security
| Parameter |
WebRTC |
HLS |
| Encryption |
Mandates DTLS/SRTP for all sessions |
Encryption optional, streams can be unencrypted |
| Encryption Level |
End-to-end at media level |
HTTPS provides server-client encryption |
| Data Protection |
Full protection of confidential media and data |
Limited protection, media not encrypted |
| Tampering |
Hard to tamper as media encrypted from source to destination |
Unencrypted streams easy to tamper |
| Eavesdropping |
End-to-end encryption prevents eavesdropping |
Can sniff unencrypted streams |
| DRM Piracy |
Strong DRM prevents stream ripping |
Unencrypted streams prone to piracy |
| Use Case Impact |
Secure for transmitting sensitive data |
Poor security makes it risky for confidential use cases |
WebRTC mandates the use of strong encryption protocols like DTLS and SRTP for all sessions. This provides secure end-to-end protection for confidential media and data transmitted between peers. Tampering and eavesdropping on confidential streams is very difficult.
In contrast, HLS streams can be unencrypted over plain HTTP. While HTTPS encryption secures the stream delivery from server to client, the actual media itself remains unprotected. This makes HLS streams prone to hijacking, tampering and DRM piracy. The poor security posture makes HLS unsuitable for use cases involving sensitive data.
Scalability
| Parameter |
WebRTC |
HLS |
| Architecture |
Peer-to-peer |
Client-server |
| Scaling Model |
P2P distributes load by connecting peers directly |
Streaming servers handle entire audience load |
| Cloud Support |
Google, Mozilla provides cloud services like TURN servers |
Requires own streaming infrastructure |
| Load Balancing Needed? |
No, peers connect directly |
Yes, to distribute load across servers |
| CDN Requirement |
Typically not needed |
CDNs required to cache and distribute streams |
| Cost Impact |
Minimizes infrastructure cost as no media servers needed |
Scaling incurs significant server and CDN costs |
| Small Audiences |
Handles well, scales higher as more peers join |
Overkill for small audiences |
| Large Audiences |
Additional cloud services may be needed |
Requires huge investments into streaming infrastructure |
A key advantage of WebRTC’s peer-to-peer model is efficient scalability without dedicated streaming servers. The streaming load is distributed across peers. While Google and Mozilla provide cloud services like TURN if needed, WebRTC’s serverless nature minimizes infrastructure costs.
However, HLS scaling depends completely on streaming servers and CDNs to reach large audiences. Supporting millions of concurrent users requires huge investments into load balancing, geo-distributed CDN PoPs, server capacity, etc. This makes HLS inefficient for smaller audiences but important for massive scale.
Cost and Infrastructure Requirements
| Parameter |
WebRTC |
HLS |
| Architecture |
Peer-to-peer |
Client-server |
| Media Infrastructure |
Only STUN/TURN servers for signaling and fallback |
Video streaming servers, CDNs, load balancers etc. |
| Cost Impact |
Minimal infrastructure costs due to P2P streaming |
High infrastructure costs |
| Servers Needed |
Small number of servers for session establishment |
Many distributed streaming servers |
| CDNs Needed |
Typically not needed |
CDN costs for content distribution |
| Ongoing Costs |
Low operational costs |
High costs for server, CDN and bandwidth |
| Provider/Owner Impact |
Lower cost for service providers |
Increases costs for content owners and providers |
A major benefit of WebRTC is significantly lower infrastructure costs due to its peer-to-peer nature. Only a few servers are needed for session establishment and fallback. The actual streaming occurs directly between user devices.
HLS relies on a client-server architecture, requiring large investments in streaming servers, CDNs, high bandwidth capacity, load balancing etc. Video providers also incur ongoing costs for distributing streams to large audiences globally.
Thus, WebRTC lowers costs for service providers by minimizing infrastructure. HLS increases costs substantially for streaming video platforms and content owners due to its infrastructure requirements.
Tragofone: Changing the ecosystem of WebRTC-enabled VoIP Softphones
Tragofone is an innovative player that is disrupting the VoIP softphone market by leveraging WebRTC technology. The company provides white-labeled mobile and desktop softphones that enable browser-to-browser calling, instant messaging, video conferencing, and other collaboration capabilities.
Some key aspects of Tragofone’s offerings:
- Powered by WebRTC for frictionless in-browser communication
- Easy integration with existing telephony infrastructure
- Auto-provisioning for simplified deployment at scale
- White-labeling allows full brand customization
- Available across mobile and desktop platforms
- Rich features like screen sharing, presence, call forwarding etc
- Designed for security with end-to-end encryption
Tragofone unlocks the potential for high-quality voice and video calling from within a browser by harnessing WebRTC’s low-latency peer-to-peer media streaming. This removes friction in the adoption of VoIP softphones for remote calling and collaboration use cases.
As emerging players like Tragofone continue innovating, the future looks promising for WebRTC-based communication. WebRTC has the potential to completely change the VoIP softphone ecosystem by bringing frictionless in-browser calling and collaboration.
Endgame
WebRTC is engineered to deliver ultra-low latency communication through its lightweight peer-to-peer architecture. This makes it ideal for applications like video conferencing, remote collaboration, interactive live streaming, etc which require real-time engagement.
HLS is better suited for high-volume media distribution from a server to a large passive audience. Its support for adaptive bitrate streaming and broad device compatibility has made it the de facto standard for video-on-demand and live video playback.
The choice between WebRTC vs HLS eventually depends on the specific use case requirements around latency, scale, security, and device support.
For adding real-time communications capabilities to your apps, Tragofone’s innovative WebRTC-based softphones are worth considering. Avail a free trial today or request a demo to learn more.
