Understand RTMP, HLS, and WebRTC in one practical guide—compare streaming protocols, latency, scalability, and device compatibility to choose the right live streaming technology for your platform. Continue reading
Live streaming has become a fundamental part of digital media, information broadcast and entertainment. Everything right from gaming to corporate to education to crucial information delivery, live streaming has a very crucial role. The applications of live streaming are unlimited. However, what goes on behind the scenes is the selection of the right live streaming protocol. A protocol or a set of defined rules determines how data travels to the viewers. Popular live streaming protocols include RTMP, HLS, WebRTC, and more.
This blog serves as a guide on what exactly these live streaming protocols are and will also give a detailed breakdown of their strengths and weaknesses so that you can make an informed decision about which protocol suits your requirements.
A streaming protocol is a set of rules and standards that determine how audio and video data are transmitted from the source, likely a camera or encoder, to the end user device. Protocols manage crucial transmission tasks such as video compression, its transmission, buffering management—where the buffer is crucial for smooth playback, and managing data flow during streaming—metadata, playback synchronization, and latency control.
Different protocols are optimized to take care of different aspects in a more efficient way, as required by the broadcaster. Some focus on scalability, others might favour low latency, and many simply prioritize easy compatibility across multiple devices. For example, HLS can easily scale to millions of viewers using traditional web servers and CDNs.
Let’s understand various streaming protocols one by one in this article.
RTMP stands for real-time streaming protocol. It was originally developed by Adobe to support streaming between a media server and Flash player. Although Flash is no longer widely used, RTMP is still used not only in live streaming but also for streaming recorded content. Agreed its not a complete streaming protocol now, it is still widely used to ingest media from the encoder to the RTMP-enabled streaming platform.
RTMP is a TCP-based protocol that maintains a persistent connection between the broadcaster and the streaming server. Unlike newer streaming protocols that rely on HTTP delivery, RTMP establishes a direct communication channel between the encoder and the RTMP server.
Streaming starts with an RTMP encoder, which can be software encoders such as OBS Studio or hardware encoders like TeraDek. These encoders then send the processed streams to the RTMP servers or streaming platforms such as Muvi Live.
The content further goes to RTMP servers such as the Nginx RTMP module or live streaming solutions such as Muvi Live.
To publish a stream, broadcasters use:
RTMP URL – the server address
Stream Key – a unique identifier for the stream
There are 3 variants of the RTMP protocols that can be used as per the needs.
RTMP Standard: This is the original protocol that uses TCP port 1935 and is commonly used for stream ingestion.
RTMPS: Also known as RTMP secure, an additional SSL encryption layer is added to secure the content and improve enterprise security.
RTMPT: In an event where RTMP ports are blocked, they are tunneled through HTTP requests, allowing you to bypass firewalls.
RTMP (Real-Time Messaging Protocol) remains widely used in modern streaming workflows because of its reliable and low-latency ingest capabilities. It maintains a persistent TCP connection between the encoder and the RTMP server, allowing video and audio data to be transmitted continuously with minimal delay. This makes RTMP an ideal choice for sending live streams from broadcasting software like OBS, Wirecast, or vMix to streaming platforms. Another major advantage is its broad compatibility with encoders and media servers, as nearly all professional streaming tools support RTMP as the standard method for pushing live streams.
Despite its strengths, RTMP has several limitations in modern streaming environments. The biggest drawback is that RTMP is no longer supported for playback in modern browsers, as it was originally designed to work with Adobe Flash, which has been discontinued. Because of this, RTMP streams must usually be transcoded into delivery protocols like HLS or DASH before viewers can watch them on web browsers, mobile devices, or smart TVs.
HLS or HTTP Live Streaming (HLS) is one of the most widely used protocols for delivering video content over the internet. HTTP Live Streaming (HLS) is an HTTP-based adaptive bitrate streaming communications protocol developed by Apple Inc. and released in 2009. Again developed by Apple, HLS has become the industry standard for live streaming and video-on-demand delivery. Thanks to its universal compatibility with mobile devices, smart TVs, OTT platforms, and web browsers, it is a widely accepted solution for live streaming requirements.
Unlike RTMP, as we discussed earlier, HLS streaming, as the name suggests, works on a standard HTTP server, which is the same protocol that is used to deliver web pages. This enables it to be highly scalable and compatible with global browsers and CDNs. HLS supports both live and on-demand streaming, and is compatible with platforms such as Android, iOS, Windows (Microsoft), and AppleTV, as well as through Media Source Extensions on other operating systems.
HLS is also an adaptive bitrate streaming-enabled protocol that breaks a video file into several small segments, which are easier to transmit. HLS works by breaking the overall stream into a sequence of small HTTP-based file downloads, each downloading one short chunk of an overall potentially unbounded transport stream. Instead of sending one continuous video stream, HLS divides the video into short chunks (usually 2–10 seconds) and provides a playlist that tells the video player which segments to load and when.
This architecture allows video players to adjust video quality dynamically based on the viewer’s internet speed, ensuring smooth playback with minimal buffering. The original video is encoded in several distinct quality levels to enable a player to adapt to the bandwidth of the network. An HTTP server creates an M3U8 playlist file to index these segments, listing the order of segments and allowing for multiple streams at different bitrates. When users want to play HLS streams, the media player reads the playlist and loads the appropriate segments for playback.
The streaming process begins with cameras and other video production equipment capturing content and then getting it encoded into various codecs. For example, H.264 or H.265 for video and AAC for audio. This prepares the stream for processing by the streaming server.
Once the video reaches the streaming server, it is divided into small media segments, with each segment being typically around 2 seconds. Breaking the video into small segments allows the player to load gradually and provides a seamless viewing experience that is uninterrupted and of very high quality.
Post segmentation, an M3U8 playlist file is generated. It contains essential metadata about the next available video segment. M3U* file tells the video player:
Which segment to lead
Duration of segment
Order of playback
A combination of M3U8 playlist and segments is then transferred using HTTP servers or CDNs, which works easily through firewalls and offers seamless integration with existing web infrastructure.
When a viewer starts a stream:
The video player downloads the M3U8 playlist
The player requests the first video segment
Segments are downloaded sequentially
The player buffers upcoming segments for smooth playback
This process continues until the stream ends.
Due to the segmentation process, HLS often encounters latency issues, which can hamper the live streaming experience as delay can sometimes reach up to 10 seconds. To counter this, Apple introduces LL-HLS, which reduces delay by the following:
Delivering small partial segments
Using HTTP/2
Reducing segment duration
According to Apple’s low latency HLS specification, Low-Latency HLS (LL-HLS) reduces latency to under 2 to 6 seconds for time-sensitive events, ensuring better performance and compatibility with industry standards.
HLS has become the industry standard for video delivery because of its universal compatibility. Since it is HTTP-based, it works with all types of content delivery networks and enables streaming platforms to distribute content globally at a very large scale. Support for adaptive bitrate streaming adds an edge, making streams automatically adjust to internet speeds and varying network conditions. HLS provides mechanisms for players to adapt to unreliable network conditions without causing user-visible playback stalling. For example, HLS allows the player to use a lower quality video on an unreliable wireless network, thus reducing bandwidth usage. The original video in HLS is encoded in several distinct quality levels, allowing the player to choose between variant streams during playback. The video source is encoded and split into small segments (e.g., MPEG-2 Transport Stream files, often labeled as .ts). HLS is also highly compatible with modern devices and platforms, including smartphones, tablets, smart TVs, and web browsers. Additionally, HLS can traverse any firewall or proxy server that lets through standard HTTP traffic, unlike UDP-based protocols such as RTP. HLS offers robust security, supporting AES-128 encryption to protect content. It also supports dynamic ad insertion using splice information based on the SCTE-35 specification. Subtitles can be included alongside audio and video tracks to enhance accessibility and user experience. The ability to deliver live content is a key advantage, as it helps engage audiences through streaming events and connects people globally.
Now talking about its few disadvantages, HLS streaming faces latency issues. Standard HLS can introduce higher latency (typically 10-15 seconds) compared to traditional protocols like RTMP. Traditional HLS delivers chunks, which introduces delays that can go as high as 30 seconds. This makes it unsuitable for applications that require real-time interaction,s such as video conferencing, eAuctions, etc.
WebRTC (Web Real-Time Communication) is a modern communication technology that enables real-time audio, video, and data streaming directly between browsers and devices without requiring plugins or additional software. Live streaming gives you a way to connect with your employees, customers, and community. Live streaming is one of the best ways to connect in a meaningful and authentic way with your community. You can engage followers, customers, or employees around the world, wherever they watch. You can easily stream from any device across social media platforms or your own website. Live streaming allows you to create engaging experiences that resonate with audiences across all your important moments. Creating lasting impressions through genuine interactions can bring people together. Developed by Google and later standardized by the W3C and IETF, WebRTC has become the foundation for many real-time apps such as video conferencing, live collaboration tools, online gaming, telemedicine, and interactive live streaming.
The WebRTC API is the foundation for building real-time peer-to-peer communication apps, enabling features like screen sharing, live media streaming, and direct exchange of media data. Unlike traditional streaming protocols like RTMP or HLS streaming, which rely on servers to distribute media, WebRTC is designed to establish direct peer-to-peer communication between users. This architecture allows WebRTC to achieve extremely low latency, often under 500 milliseconds, making it ideal for applications where immediate interaction is critical. In WebRTC, objects such as connections, media tracks, and data channels manage the state and transmission of media data and messages. Developers can access underlying transport protocols and security features, and use RTCSessionDescription objects to define the parameters and description of each session during offer/answer negotiation. The user agent can request identity assertions for secure communication, and packets are exchanged to maintain signaling and data transfer. Implementation often involves writing code and using libraries like adapter.js to ensure browser compatibility.
One of the most advantageous factors of WebRTC is that it works natively inside all modern browsers, such as Chrome, Firefox, Safari, etc.
The first step remains the same for all. Audio and video are captured from multiple devices using the getUserMedia API, resulting in media data that is combined into a media stream object. Once captured, this media stream object is ready to be transmitted to the other user.
Before two devices can communicate directly, they must exchange information about how to connect. This process is called signaling.
Signaling includes exchanging:
Network addresses
Media capabilities
Encryption information
Session parameters and description objects (RTCSessionDescription), which define the parameters of the WebRTC session, including offer/answer negotiations and SDP (Session Description Protocol) details
Once signalling begins, the WebRTC protocol also attempts to determine the best route for communication using Interactive Connectivity Establishment (ICE). ICE collects potential network paths, each represented as an object called a candidate. These candidate objects include:
Local network addresses
Public IP addresses
Relay servers
During this process, network packets are exchanged between peers to test connectivity and determine which candidate object provides the most reliable path. Developers can access information about the selected transport path to monitor or optimize the connection.
The best connection path is then selected.
Once the connection is established, WebRTC transmits live media, including audio and video streams, as encrypted media data using SRTP (Secure Real-Time Transport Protocol).
WebRTC connections are encrypted by default, ensuring that live media and media data streams remain secure during transmission. Developers can access security features and underlying transport details, such as SCTP transport and encryption layers, to further manage and monitor the secure delivery of these streams. Like HLS, WebRTC is capable of adaptive bitrate streaming, so you can deliver multiple renditions of your stream with the optimal quality for each viewer.
WebRTC (Web Real-Time Communication) is designed to enable ultra-low latency communication, making it ideal for web and browser-based apps that require real-time interaction. Unlike traditional streaming protocols, WebRTC can achieve latency as low as 200–500 milliseconds, allowing viewers and broadcasters to interact almost instantly. It also works natively within modern web browsers without requiring plugins, simplifying deployment and improving accessibility across devices.
WebRTC empowers apps with advanced features such as screen sharing, peer-to-peer audio and video calls, and real-time data transfer. Implementing these features often involves writing code that utilizes APIs and may require using a library like adapter.js to ensure browser compatibility. For identity and security, the user agent handles requests for identity assertions and manages authentication processes, helping to secure peer connections.
All 3 streaming protocols have their own advantages and disadvantages. While RTMP is a great way to ingest streams, HLS offers excellent large-scale streaming capabilities, making it ideal for delivering live content that engages audiences and connects people globally. WebRTC, on the other hand, is effective for peer-to-peer communication, offering secure, real-time content delivery. Live streaming with these protocols not only enhances the streaming experience but also fosters connections within various communities, such as workplaces, religious groups, and social gatherings, by enabling authentic interactions and shared experiences. We encourage you to explore the potential of live streaming to engage your audience and create impactful content.
RTMP comes best when your streaming workflow requires sending live streams via an encoder to a streaming platform such as Muvi Live. While it is not a complete standalone streaming protocol owing to the discontinuation of Flash-based streaming, it is still a widely used component to assist in streaming.
For example, combining RTMP with HLS gives you an easy, cost-effective way to broadcast live events.
HLS streaming is the best choice for large-scale video distribution, supporting both live and on demand streaming.
Typical use cases include:
OTT platforms
Sports broadcasting (especially for live content to engage audiences)
News streaming
Corporate live streaming
Video-on-demand platforms
HLS can deliver subtitles as an additional media track, enhancing accessibility and user experience during media playback.
If your priority is scalability, device compatibility, and reliability, HLS is the preferred protocol.
WebRTC is ideal for interactive streaming requirements largely due to its peer-to-peer communication methodologies. It powers real-time communication features in web apps, enabling functionalities such as live media exchange directly between peers. This makes WebRTC a great option if you wish to host video conferences, live gaming interactions, online classrooms, auctions, and especially use cases like screen sharing for real-time collaboration and content sharing. If your viewers require instant communication, WebRTC is an optimal choice.
Each streaming protocol excels in different performance areas.
RTMP provides reliable low-latency ingestion from encoders to streaming servers. HLS streaming offers unmatched scalability and device compatibility for delivering video to large audiences. WebRTC enables ultra-low latency communication for interactive applications, supporting live media, screen sharing, and other real-time features in apps. Because of these differences, modern streaming platforms often combine all three protocols to balance performance, scalability, and real-time interaction.
Live streaming is about delivering high-quality streams across the globe, and that can be achieved with the use of the right live streaming solution. Muvi Live can help you launch and deliver uninterrupted, secure streamers to millions of viewers, including building live streaming apps for various platforms such as Android, iOS, and more.
Bonus Tip: Muvi Live also lets you live stream directly from your mobile device, no encoder needed, and supports Android devices natively.
So yes, live streams are everywhere. From hospitals, to educational institutions to betting shops to sports, live streaming finds its application in almost all industries that use video to pass information. Muvi Live can help you create apps, websites, and platforms that can livestream content.
When you choose Muvi Live as your live streaming solution provider, you get:
Muvi Live App for streaming directly via phone
VAST-based ad support, with the option to bring your own ad server
DRM-enabled secure streaming environment
SDK support to build live video functionality into the app of your choice
Get a free 14-day trial today to learn more.
Written by: Shivashish
Shivashish works as a content writer at Muvi. He has worked in domains like e-commerce, employee engagement, sports and entertainment. A poet by heart, Shivashish believes in creating quality content that is rich in information and easy to understand.
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