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Which One is the Best Protocol for Live Streaming: RTMP vs HLS vs WebRTC

Which One is the Best Protocol for Live Streaming: RTMP vs HLS vs WebRTC

Live streaming is the process of broadcasting video and audio content across the internet within a real-time scenario. However, live streaming does depend on the protocols that determine rules and standards to transmit data within different devices in a network. The most common live-streaming protocols are RTMP, HLS, and WebRTC. This article will discuss important information on RTMP vs. HLS vs. WebRTC latency.

What is Video Streaming Protocol?

A video streaming protocol is a set of standards that determines how data is transmitted from one device to another through the internet. The protocol type uses segmentation in its processes, which means the video stream is divided into small chunks and then delivered to your viewers. The protocol then reassembles the chunks to generate original video content.

This happens because most of the video files aren’t designed for video streaming in the first place. Thus, converting them into streamable files is important, which involves breaking videos into chunks. Finally, the videos are sequentially transmitted for playback. The most popular video streaming protocol includes rivalry within them, through HLS vs. RTMP OBS vs. WebRTC protocol.

Meanwhile, the protocol is responsible for ensuring that the video data is securely, efficiently, and reliably transmitted without signal loss or buffering issues. The video is first encoded and stored in a server, after which the video streaming service uses a protocol for displaying video on the user’s device.

What is RTMP?

RTMP, a real-time messaging protocol, is a powerful video streaming protocol that provides instant communication support within the client and streaming server. This is done by providing live audio and video streaming from webcams, microphones, and other sources. Not only live streaming but the recorded content is also supported.

In addition, this TCP-based protocol is developed for maintaining smooth, persistent, and extended communications with low latency. With a TCP data transport facility, RTMP can encapsulate data with several container formats, including MP4 and MLV.

The protocol is ideal for delivering encoded data on media servers, social media platforms, and streaming platforms. Moreover, there are different individual formats or variations within the RTMP protocol, including RTMFP, RTMPE, RTMPT, and RTMPTE.

Furthermore, this protocol supports both audio and video codecs. Within audio, you will experience codecs like HE-AAC+ v1 & v2, AAC, Speex, AC-LC, and MP3. On the other hand, video codecs include Sorenson Spark®, H.264, VP8, and Screen Video v1 & v2.

Meanwhile, today RTMP protocol uses an RTMP encoder for ingesting live streams. This means when your video is sent to the streaming platform through encoding setup, the video transfers through the RTMP protocol.

What is HLS?

HLS is an HTTP live streaming protocol for delivering video and audio media to viewers over the internet. Both on-demand and live video content can be transmitted using an HLS protocol, and the capturing devices include microphones, cameras, etc. HLS is a scalable protocol that can deliver live streams across (CDNs) while using web servers.

Meanwhile, the content is transmitted in an optimized way where the video content from the capturing device is sent to the live video encoder. Afterward, the encoder transmits the data through HTTP on different video hosting platforms. By utilizing the HLS ingest, the video hosting platform will then transfer video content to the HTML5 video player.

Moreover, the HLS streaming protocol is fully compatible with various firewalls and devices, which is unavailable in most other protocols. The latency falls in the 15-30 second range during the live streams. Not to forget, the HLS protocol can encode live streams within multiple quality settings. Thus, users can choose the best streaming option based on available network bandwidths.

What is WebRTC?

WebRTC (Web Real-Time Communication) is a popular streaming platform that was developed by Google. This is an open-source project, but it provides peer-to-peer communication between different browsers and applications. Hence, you certainly don’t need third-party plugins. Moreover, WebRTC protocol utilizes technologies like JavaScript, HTML 5, and other codecs like Opus and VP8 for video, text, and audio encoding purposes.

Generally, WebRTC provides real-time communication in low WebRTC latency, which is extremely useful in video calling, conferencing, and online gaming. To minimize latency, WebRTC uses adaptive bit rate technology that adjusts the video quality based on available network bandwidths. Thus, you’ll experience high-quality video calls, streams, or broadcasts without lags or buffering.

Meanwhile, WebRTC protocol incorporates security encryptions of Secure Real Time Protocol (SRTP) and many other security standards. This means that your important data is transmitted through a fully secure network without risks of data breaches. Moreover, users can share files in different formats using WebRTC protocol without requiring audio or video connections.

Comparing RTMP vs. HLS vs. WebRTC

With technological advancement, meaningful digital connections are gradually increasing with available streaming services. Therefore, people worldwide can effectively and efficiently communicate, irrespective of geographical location. Concurrently, different protocols back the streaming services. This table compares RTMP vs. HLS, RTMP vs. WebRTC, and HLS vs. WebRTC.

Protocol TypeStreamingStreamingReal-time communication
Latency2 – 5 seconds20 – 30 seconds<500 milliseconds
SecurityNo encryption by defaultAES-128 encryption for HLS mediaSecurity features and encryption
Some Supported CodecsH.264, AAC, MP3, VP8, VP6H.264, HEVC, AAC, FLAC, Apple LosslessVP8, VP9, H.264, Opus
ScaleIt needs a special RTMP proxy to scale.Viewers in millionsLess or equal to 10,000 viewers
Adaptive BitrateNoYesYes
Browser SupportYou need a Flash Player plugin for this.Most modern browsers support this protocol.Most modern browsers support this protocol.

How to Choose the Most Suitable Live Streaming Protocols?

Now that we already have some idea about the differences between the three protocols: HLS, RTMP, and WebRTC. Then, how to choose the most suitable live streaming protocols for your scenario. When choosing a live streaming protocol, you should consider several factors, such as audience size, network conditions, and device compatibility.

Here are a few things to keep in mind for the best protocols for video streaming:

1. Audience size: If you expect a large audience, you may want to use a streaming protocol that can handle a high volume of traffic, such as RTMP.

2. Network conditions: If you expect viewers to be watching from different types of networks (e.g., Wi-Fi, cellular), you may want to choose a suitable protocol that can adapt to varying network conditions, such as HLS.

3. Device compatibility: If you want to reach a wide range of devices, you may want to choose a protocol that is widely supported, such as RTMP or HLS.

4. Latency: If low latency is important (e.g., for real-time interactions or gaming), you may want to choose a protocol that can achieve low latency, such as RTMP or WebRTC.

Ultimately, the best live streaming protocol for your needs will depend on your specific requirements and goals. It’s a good idea to test different protocols and see which one works best for your use case.

Build Live Streaming with ZEGOCLOUD Live Streaming SDK

Let me mention that ZEGOCLOUD offers a premium live streaming solution that delivers outstanding live audio and video quality with ultra-low latency of less than 300 milliseconds. With ZEGOCLOUD API/SDK, integrate customizable live streaming into your applications with minimal hassles and without coding from scratch.

Furthermore, the delivered video streaming has an ultra-low latency of 600ms per second. Secondly, ZEGOCLOUD supports different communication protocols, including RTMP, HLS, WebRTC, and more. Thus, users can broadcast, video call, or live stream across the internet without disruptions.

ZEGOCLOUD video streaming provides various features like screen sharing, real-time messaging, recording, quality monitoring, virtual avatar, and more. Meanwhile, the network supports stable streaming using a 99% stutter-free connection. In addition, ZEGOCLOUD comes with cross-platform support within different programming frameworks.


In this modern world, the geographical differences among people are gradually rising. However, the availability of technology has significantly reduced these barriers because of effective streaming communication. Different communication platforms use different streaming protocols, and the most used protocols are RTMP, HLS, and WebRTC. In this article, you discovered the details and comparison between all these protocols through comparison within RTMP vs. HLS vs. WebRTC latency.

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