RTMP vs WebRTC: Choose the right protocol for your live streaming 

Video streaming has advanced and evolved over time and giving a seamless experience to the viewers is extremely significant. Streaming protocols play a major role in ensuring the video asset reaches your audience from the source and all the delivery points in between.  

RTMP and WebRTC are two such streaming protocols that enable video streaming over the internet. Although HTTP can be used as a standard protocol for transmitting media files over the internet, protocols like WebRTC and RTMP are more reliable and faster.  

Let us understand and compare WebRTC and RTMP, and further discuss what is the desirable protocol to choose while deploying your video asset.  


Web Real Time Communication (WebRTC) is an open-source technology for video and audio communication that works directly between browsers and applications. Since it is an open-source technology, many developers continuously contribute it and it is an evolving protocol. Originally a Google endeavor, it was standardized and developed by the W3C and IETF and has been widely used for video conferencing, live streaming, and online gaming.  

WebRTC is a combination of protocols, JavaScript APIs and standards that enable real-time communication. Moreover, WebRTC functions in a peer-to-peer architecture, where video and media assets are transferred directly from one user to another, without the need for a server.  


Real-Time Messaging Protocol was developed by Adobe back in the 90s and was used to transfer audio, video, and other media assets between Adobe Flash Player clients and the streaming servers. Since Flash died long ago, RTMP also lost its position in the developer community. Although, RTMP has evolved into being widely used in ingesting video assets and encoding.  

RTMP faces difficulties in getting past firewalls and does not work well with Adaptive Bitrate Streaming. Even after being pulled down and losing the scope of functioning in the area of client to user transmissions, RTMP is highly capable of low-latency streaming. This makes, it still a go-to and reliable streaming protocol.  

RTMP is widely used in the first few processes of streaming, including ingestion and encoding. Later, the protocol is packaged into a compatible one like HLS or WebRTC. 

RTMP vs WebRTC: Key features 

Now that we have understood what each protocol stands for and how they help in streaming your video assets, let us dive more into the capabilities and compare the two.  


RTMP is a proprietary protocol, which is optimized for low latency streaming. RTMP’s average latency is around 2-5 seconds. Whereas, WebRTC has a latency range of less than 1 second, which makes it better for real-time streaming and video conferencing. In spite of the lesser average latency range of RTMP, it can be optimized to a much lesser latency rate for low-latency streaming. 


Since RTMP is used for first-mile contribution in streaming, it has limited scalability. RTMP can be transcoded into other protocols like HLS, which has extremely good scalability.  

Similarly, WebRTC also have poor scalability since it consumes a lot of bandwidth in the process. The peer-to-peer nature of WebRTC prevents it from streaming to a larger audience pack.

Encoder/Browser support 

RTMP is supported by most of the traditional browsers, but it has very less compatibility with new browsers once it fell after the Flash player’s retirement. Unlike RTMP, WebRTC is compatible with a wide number of browsers and encoders and can be played without plugins and extra app installations. WebRTC also has a built-in API support in HTML3, the standard markup language.


RTMP ensures high-quality video streaming with support for Full HD (1080p) and 4K resolution. It also supports different codecs, such as H.264 and VP8, for video compression and quality. 

WebRTC provides high-quality video streaming, but it depends on the network conditions. If the network has a high-speed and low latency, WebRTC can provide high-quality video. However, if the network has low-speed or high latency, the video quality can be compromised. 


RTMP uses a variation called RTMPS, which adds an extra security encryption layer to prevent unauthorized access. This extra layer encryption can be TSL (Trasport layer Security) or SSL (Secure Socket Layer). As long as your desirable broadcast system supports it, you can use RTMPS in place of RTMP.  

WebRTC has mandated a security layer with SRTP (Secure Real Time Protocol encryption. And according to the TF (Internet Engineering Task Force) standards, creating non-encrypted WebRTC connections are prohibited. Further, popular browsers like Edge, Chrome, and Firefox employ browser-level security while using WebRTC. They require a secured HTTP or HTTPS connection and the users should provide permission to the websites to access microphone and camera to protect the device information.  

Let’s wrap it up

To conclude, both RTMP and WebRTC streaming protocols have their own advantages and disadvantages. While choosing a protocol, it is necessary for it to be compatible with the whole streaming set up including the device and encoders. RTMP is a popular technology and can be leveraged to stream your live video content to hundreds and thousands of users at a time. Once you use RTMP for the video ingestion and encoding, it can be changed into HLS (HTTPS Live Streaming) for optimized and seamless delivery.  

With Videograph, you can quickly set up a live channel, stream content using the RTMP protocol, and generate an HLS URL with ABR for your visitors. Learn more about creating live streams in Videograph. 

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