With the announcement that Adobe will stop supporting Flash, people started looking for an alternative solution to RTMP. With the shrinkage of time, the question of moving to which solution from RTMP has gained importance. Experts strongly recommend RTMP to WebRTC migration as an answer.
Flash And RTMP are Dying
Adobe Communications Team made an announcement in 2017.
“Given this progress, and in collaboration with several of our technology partners – including Apple, Facebook, Google, Microsoft and Mozilla – Adobe is planning to end-of-life Flash. Specifically, we will stop updating and distributing the Flash Player at the end of 2020 and encourage content creators to migrate any existing Flash content to these new open formats.”
Adobe Communications Team
July 25, 2017
So as 2020 comes to a halt, so does the life of Flash Player and also RTMP. Adobe will stop updating and distributing the Flash Player.
For a while, many Flash video streaming applications have been replaced by HTML5 solutions such as HTTP Live Streaming (HLS) and DASH (most of them only in the last 5 years when HTML5 browsers have finally offered media source and encrypted media extensions). But during this time, WebRTC was and still is the only option for ultra-low latency streaming. Even only this makes it a great alternative to RTMP.
As an HTML5-based solution, WebRTC does not require any browser plug-ins for playback and can utilize mapping techniques to transfer data between connected sessions. Moreover, WebRTC offers the quickest method for transporting live video across the internet.
If you’re using Flash for low-latency/real-time streaming, you have a really short time for RTMP to WebRTC migration.
Let’s discuss what RTMP and WebRTC are.
RTMP
RTMP is a protocol, created by Macromedia and now owned by Adobe, that provides high-performance transmission of video, audio and data between dedicated streaming servers and Adobe Flash Player across the internet.
Definition of RTMP by Adobe:
“Adobe’s Real Time Messaging Protocol (RTMP) provides a bidirectional message multiplex service over a reliable stream transport, such as TCP [RFC0793], intended to carry parallel streams of video, audio, and data messages, with associated timing information, between a pair of communicating peers.”
RTMP is created for high-performance transmission of media such as audio and video data.
WebRTC
One of the biggest advantages of WebRTC is that it converts millions of browsers into streaming terminals without the need to install any additional plugins. What’s more, WebRTC supports sub-second latency, which means no more delay! Finally, the protocol uses an adaptive bitrate technology that allows it to automatically adjust video quality and avoid any interruptions. Sounds good right?