Top 6 Live Streaming Protocols Comparison: RTP、SRT、RTMP、RIST、ZiXi、WebRTC

Top 6 Live Streaming Protocols Comparison: RTP、SRT、RTMP、RIST、ZiXi、WebRTC



With the rapid development of digital media, live streaming services have become an indispensable part of daily life. From online concerts and sports events to video conferences, high quality, stable real-time video streaming technology is crucial for a seamless viewing experience.

This article will compare six common live streaming protocols: RTP, SRT, RTMP, RIST, ZiXi, and WebRTC. We will explore their technical characteristics, application scenarios, and differences to help customers who want to build live streaming services choose the solution that best suits their needs.

Introduction to Network Communication Protocols: TCP and UDP

Before diving into the various live streaming protocols, it's essential to understand the basics of TCP and UDP, the two types of network communication protocols.

What is a Network Communication Protocol?

A "network communication protocol" is a set of rules that determine how information is sent and received over a network. Think of it as the language or method of communication between different computers or devices, allowing them to understand and exchange data with each other.

Imagine you want to send a letter to a distant friend. You would follow certain steps: write the content, put it in an envelope, write the address, attach a stamp, and then drop it into a mailbox. Here, you are following the "protocol" of mailing a letter. Similarly, network communication protocols specify the format of data packets (like letters), how to send them, and how to ensure they are safely transmitted from one device to another.

These protocols ensure that information can flow smoothly between different devices and technologies, regardless of the operating systems or network technologies they use. This is why your phone, computer, or tablet can easily access and exchange data over the internet.

TCP (Transmission Control Protocol)

TCP establishes a network channel before starting to transmit data and strictly ensures that all transmitted data is correctly delivered and in the correct order. Think of it as a train where all data is placed in individual carriages, and the data in all carriages is delivered to the destination in sequence. Since the railway track is pre-established and the number of carriages is fixed, it is easy to detect and resend any missing carriage.

However, due to these rigorous requirements, the preparation time required before transmission, such as confirming the number of carriages or establishing the railway track, will be longer. In summary, TCP performs better in terms of accuracy but not as well as UDP in terms of live streaming latency.

UDP (User Datagram Protocol)

UDP (User Datagram Protocol) is a more flexible transmission method compared to TCP. It divides the data into several parts and transmits them separately without guaranteeing the delivery order. Think of it as dividing the data into several parts and giving them to several couriers to deliver separately. Each courier handles only the data in their hands, so no prior communication is needed, reducing the required preparation time.

However, the order of data delivery cannot be guaranteed, and if a courier loses data, it may not be recoverable. In summary, UDP performs better in reducing latency but not as well as TCP in terms of accuracy. This may result in issues such as audio and video desynchronization, fluctuating video quality, and brief blurring or distortion of the video.

With this basic understanding of these two transmission methods, we can delve deeper into the details of each live streaming protocol.

RTP (Real-time Transport Protocol)

RTP is a widely used protocol designed for real-time audio and video data transmission, using UDP structure for transmission. It supports time synchronization and sequential transmission of multimedia streams but does not provide any mechanism to ensure reliable data transmission or control congestion. It is usually used with RTCP (RTP Control Protocol) to provide quality monitoring. Due to its low latency, it is commonly used in scenarios where a certain degree of packet loss is acceptable but low latency is required, such as IP cameras or IoT scenarios.

RTMP (Real-time Messaging Protocol)

RTMP was originally developed by Macromedia to transmit streaming media audio, video, and data over the internet between a Flash player and a server and later adopted by Adobe for use with Flash technology. RTMP is currently the most popular live streaming protocol and is relatively old. In the official definition, it does not include support for newer transcoding/encoding technologies such as H.265 or AV1, relying on individual CDN providers to develop their own solutions. Additionally, RTMP is lacking in security encryption compared to newer protocols. In terms of latency, since RTMP is based on a TCP structure, although it ensures high stability, it results in relatively higher latency in live streaming.

Despite these shortcomings, due to its widespread use and stable technological accumulation, including support from various hardware and software devices, RTMP is still the most recommended live streaming protocol today. BlendVision One can also stably transmit based on this protocol, minimizing live streaming latency. It is suitable for scenarios that require high definition and stability or need to be transmitted through a CDN, such as concerts, sports events, or product demonstrations.

SRT (Secure Reliable Transport)

SRT is an open-source protocol designed to transmit high-quality video content over public networks. Compared to others, it is a newer protocol. Initially developed by Haivision and Wowza, SRT currently has more than 450 participating members. SRT is implemented using the UDP method at its core, but it also implements TCP-like transmission reliability, combining the advantages of low-latency transmission and high reliability. It is particularly suitable for unstable network environments, with the minimum live streaming latency as low as 120ms. Additionally, in terms of security, SRT ensures the secure transmission of video content through error correction mechanisms and encryption functions. While SRT appears to offer the advantages of both TCP and UDP transmission modes, its lower popularity compared to RTMP is mainly due to the lack of support from live streaming devices.

RIST (Reliable Internet Stream Transport)

RIST is another emerging open-source protocol relative to SRT, mainly driven by the VSF activity group. Like SRT, it is a live streaming protocol developed on top of UDP with added reliability. Its advantages and disadvantages are similar to SRT's: the advantage is that it combines low latency and high reliability, while the disadvantage is the current insufficient support from live streaming devices. SRT and RIST are still in the development stage, and their future performance can be anticipated.


Zixi is a commercial protocol provided by the Zixi company. Unlike SRT or RIST, which are open-source protocols, Zixi is a solution developed by a single company and has been in development for over 10 years. Zixi is also developed based on UDP. Through intelligent error correction mechanisms, it provides TCP-like high reliability on top of UDP's low latency. As a commercial service, ease of use and the completeness of the service itself are its biggest advantages. However, it is not a free open-source service and cannot be freely adjusted according to developers' needs.

WebRTC (Web Real-time Communication)

WebRTC differs from the aforementioned transmission methods. It was developed by Google and is essentially a protocol for mutual transmission between browsers, typically without passing through a central server. Instead, each browser transmits to each other, making it unsuitable for large-scale transmission via CDN. WebRTC is also based on UDP, so its latency is very low. Additionally, because it is not designed for high-quality video, it is not suitable for live streaming activities that require high video quality. Therefore, in general use, WebRTC is more suitable for small group real-time interactions, such as meetings or fan meet-and-greets.

Live Streaming Protocol Comparative Analysis

When comparing these live streaming protocols, we can evaluate their advantages and limitations based on the customer's live streaming needs and usage scenarios to find the most suitable protocol.

RTP and RTMP still dominate the early and current live streaming solutions, especially due to the widespread support from live streaming devices, making them difficult to replace by newcomers.

However, with technological advancements, protocols such as SRT, RIST, and ZiXi offer more efficient and reliable options, especially in scenarios where strict requirements for both transmission latency and quality are needed.

As for WebRTC, due to its particularity, it is the best choice for scenarios that do not require high video quality or large-scale dissemination but only real-time interaction.

Depending on the customer's specific needs, it may be necessary to use more than one live streaming mode in combination. BlendVision has helped many enterprise customers solve live streaming technical problems in the past and can assist customers in finding the best solution through professional consulting services.

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