Difference Between WEBRTC & RTMP For Live Streaming

Live streaming is an exciting technology for businesses who want to connect with their audience over the internet. The basics of live streaming are simple but there are certain important technical aspects that require some research to understand and take a decision. One of these is choosing the right live streaming protocol.
This article will help you understand the different live streaming protocols and compare these to help you make an informed decision. It seems complicated but it really isn’t. Let’s get going.

What Is A Streaming Protocol?

A streaming protocol is a method used to deliver multimedia across the internet. A streaming protocol regulates data transfer and management of any errors during the transfer. We will be comparing two popular streaming protocols – RTMP and WEBRTC.

Using RTMP For Live Streaming

RTMP is Real-Time Messaging Protocol is the most commonly used protocol. It maintains a stable connection and allows for low-latency communication. Since RTMP has been around for nearly 2 decades, it is supported by most streaming platforms and software and has grown to be a mainstream protocol used by major companies like Facebook, YouTube, and Amazon.


  • RTMP cannot play on HTML5 players without a converter such as HLS.
  • Possibility of interruptions due to low bandwidth
  • Incompatible with HTTP without special servers like the Flash Media Server or other third party servers.
  • Standard stream delay is around five to 30 seconds
  • Uses old codec like H.264 video codec and AAC audio codec which do not provide the best quality.


  • The installation process is simple, effective, and reliable.
  • Provides low latency while transmitting live broadcasts even if the internet connection is unreliable.
  • Transmits audio and video on any formats and rendition.
  • Supported universally for live streaming.
  • Adaptable to allow viewers to skip and rewind parts of the feed or to join a live stream after it has begun.

Using WebRTC For Live Streaming

WebRTCS or Web Real-Time Communication is a free open-source standard for real-time communications supported by all browsers. It infuses Real-Time Communications (RTC) capabilities into browsers and mobile applications.


  • WebRTCS is still under development which means the code used now may undergo important changes in the near future.
  • Lacks the high standards of encryption and security features since it transfers data through browsers.
  • Not supported by Internet Explorer (Microsoft browser).


  • Transforms browsers into streaming terminals without any additional plugins.
  • Supported by all browsers like Safari, Google Chrome, Firefox, Mozilla, and Opera.
  • Supports sub-second latency, which means no more delays.
  • Uses an adaptable bit rate technology, which allows it to automatically adjust video quality and prevent any drops and interruptions.

Which One Is The Best For Live Streaming?

You should analyze streaming protocols based on scalability, latency, quality, use, playback support, proprietary vs. open source, and codec requirements.
If we analyse both RTMP and WebRTCS based on these factors, RTMP clearly wins. But it is also important to understand if it suits your objective.

RTMP is best if your live streaming

  • Has a global audience
  • Runs a business with real-time billing
  • Uses a mixer and broadcast streams
  • Broadcasts 24×7 from embedded devices
  • Uses a CDN Streams on Facebook and YouTube

WebRTCS is best if you

  • Want low latency one-to-one or many-to-many applications, like video conferencing.
  • Want to avoid the installation by your users of any 3rd party apps.
  • Want to implement streaming in your app very fast.

For successful live streaming, you have to invest some time in determining which streaming protocol will appeal to your viewers. If this is challenging for you, you should consider hiring live streaming solution providers who will take care of everything.

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