HLS vs RTMP vs WebRTC

Introduction

Every single time you watch a video on demand or live stream, streaming protocols are used to deliver data over the internet. These protocols can sit in the session layers, presentation, and application. A streaming protocol is a particular method that is used to deliver multimedia throughout the web. To put it simply, it is a set of rules that controls how data is transferred, and how to deal with any errors that may occur in the course of this process.

Streaming protocols like RTMP, WebRTC, and HLS are used to deliver audio, videos online. HTTP-based protocols (HLS) depend on regular web servers to optimize the viewing experience and quickly scale. Streaming protocols like RTMP (Real-Time Messaging Protocol) enable fast video delivery using dedicated streaming servers. Whereas WebRTC (Web Real-Time Communications) supports audio, video, and data transfers and can be used for web calling, video chat, online messaging, and screen sharing all from within a web page or mobile app.

HLS

Based on MPEG2-TS, HLS (HTTP Live Streaming) is Apple’s patented streaming format. Support for the protocol is widespread in media players, web browsers, mobile devices, and streaming media servers including Linux, Microsoft Edge, Google Chrome browsers, Safari, Android, macOS, etc.

Like MPEG-DASH, HLS also works by splitting the overall stream into a series of small HTTP-based file downloads, each downloading one small piece of an overall potentially boundless transport stream. By using an extended M3U8 playlist, a list of available streams, encoded at different bit rates, is sent to the client.

Using HTTPS, HLS comprises a standard encryption mechanism and secure-key distribution, which altogether provide a simple DRM system. The biggest advantage of HLS is that it has the capability to deliver ABR (adaptive bitrate streaming) where the quality of the content to be expected at a high pixel rate. Also, it can be served from an ordinary web server or CDNs with a general configuration.

Pros:

Pocket-friendly protocol
Compatible on all the platforms
Supports HTML5 video player
Safest protocol to scale up the content to a massive base audience

Cons:

High latency compared to RTMP
Does not support Multicast
performance of HLS on other platforms is minimal compared to iOS

RTMP

RTMP or Real-Time Messaging Protocol was originally developed by Macromedia (now Adobe) for high-performance transmission of audio, video, and data over the Internet. RTMP delivers data, audio, and video in the open F4V, FLV, AMF, and SWF formats compatible with Adobe Flash Player.

It is a TCP-based protocol that maintains and allows low-latency communication and steady connections. As it supports adaptive bitrate streaming (ABS), it is the most appropriate choice for OTT services. Also, it is a cross-platform protocol that works on Linux, Mac, Windows, iOS, Android, Windows mobile, etc.

Pros:

Multicast support
Low buffering
Wide platform support
Can transmit video & audio on any rendition & formats

Cons:

Old codecs
Somewhat low security
Does not have native support in iOS
RTMP does not work in HTML5

WebRTC

WebRTC (Web Real-Time Communications) is a very powerful, exciting, leading-edge technology and standard. Being a free, open-source project, it offers mobile applications and web browsers with real-time communication through simple APIs (Application Programming Interfaces). It allows video and audio communication to work inside web pages by eliminating the need to download native apps or install plugins and allowing direct peer-to-peer communication. Supported by Opera, Mozilla, Google, Microsoft, and Apple, WebRTC is being standardized through the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C).

The factor that makes WebRTC unique is that it enables person-to-person communication. This means that WebRTC handles all the details of directly connecting two devices and transmitting the audio and video data in real-time. Supporting audio, video, and data transfers, WebRTC allows for web calling, video chat, online messaging, and screen sharing all from within a web page or mobile app. WebRTC enables better user experiences and simplifies client administration by eliminating vendor-specific, browser-specific plug-ins. The real-time communication capabilities make WebRTC an ideal technology for just about anything that requires real-time conversation.

Pros:

Better sound and video quality
Secure and stable
Supports all web browsers and operating systems
Requires no plug-ins
Open source technology
Ultra-low latency
Supports new Codecs

Cons:

Resource intensive
Generally supports only up to 50 members
Very expensive for scaling