Whatever may be the platform you use for live streaming, for a seamless live streaming experience the technology and protocols that is supported in the streaming platform is much important than any other features,
So here we are going to take a deep dig into the two major technological features that are used in the top live streaming platforms, WebRTC and RTMP.
What are they, why are we using them and much more, hang on and read further to know them all.
Table of Contents
What is the Streaming Protocol?
Before we go into details, both WebRTC and RTMP are streaming protocols, that are used in live streaming a video, so what is a streaming protocol,
Streaming protocols are methods of data transmission where a media file ( video or audio) is transferred from source to streaming devices.
Simply put, video is cut down into chunks of video bits and delivered into different streaming devices.
What is WebRTC?
WebRTC is a real-time streaming protocol supported by most of the browsers and mobile applications to transfer media files through application programming interfaces.
It is coded in C++, JavaScript.
What is RTMP?
The streaming protocol known as Real-Time Messaging Protocol (RTMP) used to send video files to Adobe Flash players as a single source and this protocol helps to broadcast video with little latency.
In the initial stages of RTMP streaming, encoding and ingestion takes place whereas The protocol is then bundled with a compatible one WebRTC or HLS.
What are the Key Differences between WebRTC vs. RTMP
Since live streaming is based on Real-time communication, it is supported by both RTMP (Real-Time Messaging Protocol) and WebRTC (Web Real-Time Communication), some of its differences are mentioned below,
WebRTC vs RTMP
Plugins or other applications are not required for direct peer-to-peer communication between web browsers thanks to a current technology called WebRTC. It works well for data sharing as well as voice and video calls.
It enables encryption for safe connection and employs the UDP protocol for speedier transfer. It’s also readily available because it’s included into the majority of contemporary web browsers.
However, RTMP is a protocol that is frequently used for live broadcasting. Its foundation is the TCP protocol, which guarantees dependable data delivery but may cause some latency. Content producers frequently use RTMP to webcast live events, seminars.
When it comes to advantages, WebRTC excels in real-time communication and low latency. For audio and video calls where rapid communication is essential, it’s ideal. Additionally, it has file transmission and screen sharing capabilities, which make it adaptable to a variety of communication requirements.
However, RTMP works well for live streaming to a wider audience. Better control over the streaming process is made possible by it, as demonstrated by features like adaptive streaming, which modifies the quality according to the viewer’s internet connection. Additionally, it works with a variety of streaming services and encoding programmes, providing more alternatives for content producers.
On a final note, WebRTC works best for in-the-moment communication such as phone and video chats, and RTMP works better for live streaming events to a huge set of viewers.
Which Is Better For Streaming Protocol? WebRTC or RTMP
On a comparison metrics for a better live streaming, we have listed out how both WebRTC and RTMP benefits and leads to a good live streaming experience.
Latency speed
RTMP works on Transmission Control Protocol (TCP), and data transmission and with a reliable network connection it often has an latency of 0.5 seconds or more latency .
Whereas WebRTC comes with a latency report of less than 0.1s. It is better for video conferencing or real-time device control during live streaming.
Scalability
RTMP can provide live streaming to thousands or even millions of audiences since it is designed in such a way that it caters to a huge number of users.
In contrast, WebRTC provides live streaming to a smaller number of audiences, normally within the thousand limit. Over here RTMP is the better choice.
Encoder/Player and Browser Support
Most of the encoder software and video players support RTMP due to adobe Flash player, but on the dark side it also started to lose support by modern browsers.
WebRTC is highly supported by modern browsers with built-in API support and there is no need for installing any software or external plugins.
Cloud Assistance
The majority of cloud service providers, including AWS and Alibaba Cloud, offer video streaming capabilities that are easily connected with RTMP.
On the other hand, in order to use WebRTC for video streaming, we must install on-premise streaming servers in the cloud, like Jitsi Meet.
API Assistance
Selecting your technology based on API support is crucial. Since the majority of contemporary browsers feature native APIs that are integrated with WebRTC and can be called directly with Javascript, WebRTC is leading the way in API support.
However, in order to use RTMP, we will need to make use of open source libraries or already-existing software solutions.
Supported Streaming Protocols by OnTheFly
OnTheFly supports RTMP with low latency and high storage capacity to maximize their productivity and the live streaming opportunities.
The core features of OnTheFly is so flexible and supportive even beginners can work with it in ease.
To know more how it works click here
Conclusion
In conclusion, popular technologies like RTMP and WebRTC can be utilized to create custom video streaming services. When it comes to cloud vendor integration and video player support, RTMP is superior.
However, WebRTC provides near real-time latency and quicker streaming. The choice lies in the usage and the necessity of the protocol for the user. Make a wise choice by deciding on the above factors,
for more information to reach out to us in the comment section.