I. Core Concept Analysis

1.1 Differences Between Protocols and Formats

In streaming media technology, Protocol and Format are two core yet distinct concepts:

Protocol: The rules and procedures that define how data is transmitted over networks, such as RTMP, RTSP, and WebRTC.

Format: Defines how audio and video data are organized and encapsulated, including container formats (e.g., FLV, MP4) and encoding formats (e.g., H.264, AAC).

1.2 Overview of Mainstream Protocols

Real-Time Streaming Protocol

Primarily used for real-time video communication, such as surveillance cameras and video conferencing systems. RTSP handles the control, while actual data transmission is typically accomplished via the RTP protocol.

RTMP (Real-Time Messaging Protocol)

A legacy product from Adobe Flash's heyday, it remains the primary protocol for live streaming today. Built on TCP, it delivers low-latency and stable connections.

HLS（HTTP Live Streaming）

Apple's HTTP-based adaptive streaming protocol, utilizing m3u8 index files and TS fragments, delivers excellent compatibility.

FLV（Flash Video）

A video container format that was once tightly integrated with Flash Player and is still used in HTTP-FLV live streaming solutions.

WebRTC (Web Real-Time Communication)

The W3C standard enables real-time audio and video communication between browsers without plugins, with ultra-low latency (<1 second).

II. Analysis of Technical Characteristics

2.1 RTSP (Real-Time Streaming Protocol)

RTSP is a network control protocol designed to establish and manage media sessions. It does not transmit data directly but instead creates channels for RTP/RTCP data streams.

flow of work ：

1.The client sends a DESCRIBE request to the RTSP server.

2.The server returns an SDP description that specifies media information

3.The client sends a SETUP request to establish a transmission channel.

4.The client sends a PLAY request to start playback

5.Media data is transmitted via the RTP protocol.

2.2 RTMP (Real-Time Messaging Protocol)

RTMP, a proprietary protocol developed by Adobe, is designed to transmit audio and video data between Flash players and servers. Built on TCP, it ensures stable connections with minimal latency.

Key features:

Low latency (typically 2-5 seconds)

Supports AMF (Action Message Format) data serialization

Using port 1935 by default may be blocked by the firewall

It remains widely used in live streaming.

2.3 HLS（HTTP Live Streaming）

HLS is an adaptive streaming protocol developed by Apple, operating on HTTP. It splits video streams into smaller files (typically in TS format) for download and playback via HTTP.

Core components:

m3u8 file: a playlist file containing media clip information

TS fragment: actual media data segment

Multi-bitrate adaptation: Supports switching between resolutions at different bandwidths

2.4 FLV（Flash Video）

FLV is a container format designed to encapsulate audio and video data. With Flash's obsolescence, browsers no longer natively support FLV, though the HTTP-FLV protocol still enables its use in live streaming.

2.5 WebRTC (Web Real-Time Communication)

WebRTC is an open standard that enables real-time peer-to-peer communication between browsers. It features multiple APIs and protocols to deliver low-latency audio and video transmission.

core technology ：

• MediaStream: Get audio and video streams

• RTCPeerConnection: handles P2P connections

• RTC Data Channel: Transmits any data

• Use UDP transmission with SRTP for security

III. Comparison and Selection Guide of Agreements

3.1 Comparison of Technical Parameters

3.2 Application Scenario Selection Guide

On-demand videos (e.g., Bilibili, YouTube):

We recommend using HLS or MPEG-DASH for optimal adaptive performance and compatibility.

Standard Live (High concurrency, High compatibility):

The streaming server employs RTMP, while the distribution server utilizes HLS to balance latency and compatibility.

Ultra-low latency live streaming (e.g., e-commerce live streaming, interactive live streaming):

Consider using HTTP-FLV or WebRTC to achieve latency within seconds or even sub-second.

Real-time communication (e.g., video conferencing, online education):

We recommend WebRTC, specifically designed for low-latency peer-to-peer communication.

security and protection monitoring ：

Traditional domains still predominantly use RTSP, which ensures excellent compatibility with hardware devices.

IV. Modern Streaming Media Architecture Example

4.1 Typical Live Streaming Workflow

1. Streaming Collection: Streamers use tools like OBS to push audio and video streams to the source server via the RTMP protocol.

2. Cloud processing: The server receives RTMP streams, performs transcoding and slicing, and generates multi-bitrate adaptive streams (HLS/DASH).

3 Content Delivery: The processed content is distributed via CDN to global edge nodes

4 Client playback: The user retrieves the stream from the nearest node and automatically selects the best resolution based on network conditions

4.2 Technological Development Trends

Current developments in streaming media technology are showing the following trends:

Low-latency: From HLS's 10+ second latency to WebRTC's sub-second latency

Standardization: MPEG-DASH as an international standard gains broader support

Enhanced encoding efficiency: New standards like AV1 and H.266 deliver higher compression rates.

Intelligence: AI technology for content understanding, image quality enhancement, and bandwidth prediction