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RTMP stands for Real-Time Messaging Protocol. It is a protocol for exchanging timed audio, video, and data messages between peers over a reliable transport such as TCP. In live streaming, RTMP is commonly used to send a stream from an encoder to an ingest server; it is not a codec, an encoder, or the format every viewer necessarily receives.
What RTMP does—and what it does not
Adobe’s 2012 specification describes RTMP as a bidirectional message-multiplexing protocol intended to carry parallel audio, video, and data messages with timing information between communicating peers. Adobe’s RTMP specification is the source for this basic definition.
- It is a protocol: rules for how messages are exchanged between endpoints.
- It is not a codec: the codec determines how audio or video is represented and compressed. RTMP transports payloads; it does not itself encode, compress, or improve picture or sound quality.
- It is not the whole streaming path: an encoder may send a contribution stream to an ingest server using RTMP, after which the service can process and deliver it to viewers using other methods.
The protocol has a historical association with Adobe and Flash. That history does not mean RTMP is only relevant to Flash: a 2024 memo by Michael Thornburgh reports that it continues to be used in media-streaming products and workflows. That establishes continued use as of the memo’s publication, not that every streaming service accepts RTMP today. Thornburgh’s 2024 RTMP Errata and Addenda also notes that the December 2012 specification contains errors, omissions, and ambiguities.
How RTMP works
At a high level, an RTMP client and server begin with a handshake. After that initial exchange, RTMP Chunk Stream breaks messages into chunks and multiplexes them for transmission. The messages include information such as timestamps, message length, message type, and a message-stream identifier. Their payload can contain audio, compressed video, or other data. The receiver uses the protocol information to reconstruct and interpret the messages; the media format and higher-level application determine what the payload means. Adobe’s specification describes these mechanics.
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This division of responsibilities is useful when troubleshooting: RTMP describes message transport, while the encoder’s selected codec and settings describe the media being carried. A connection problem, an unsupported codec, and a playback or delivery problem can therefore occur at different stages and need different fixes.
Where RTMP fits in a live-streaming workflow
- Capture and encode: a camera, screen capture, or prerecorded media source is prepared by streaming software. The software encodes the audio and video according to its settings.
- Send to ingest: the encoder connects to a destination’s ingest endpoint. RTMP is a commonly encountered protocol for this contribution connection, if the destination supports it.
- Process and deliver: the streaming service receives the contribution and may package or distribute it using other delivery methods. RTMP does not, by itself, specify the full viewer-facing delivery path.
For a concrete setup, use the protocol and ingest details provided by the destination rather than assuming that a service accepts RTMP. An OBS Project discussion lists RTMP, RTMPS, SRT, and HLS among output protocol options in its context, but it is not a definitive support table for every current OBS version or streaming service. The OBS discussion also considers SRT in a particular low-latency context; that is not a universal ranking.
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RTMP compared with HLS, WebRTC, and SRT
These names describe different roles or choices in streaming workflows. They are not interchangeable labels for the same technology, and none is universally best without knowing the source, destination, viewer endpoints, and latency requirements.
| Option | Role established by the cited sources | What to check for your use case |
|---|---|---|
| RTMP | Bidirectional media-message multiplexing over a reliable stream transport such as TCP. Adobe specification | Whether the encoder and ingest destination support it, and whether it fits the contribution workflow. |
| HLS | Multimedia delivery organized around playlists and media segments; a master playlist can describe variant streams for different encodings. RFC 8216 | Whether playlist-and-segment delivery and variant streams suit the playback and packaging requirements. |
| WebRTC | An interactive real-time communication framework whose required media transport is RTP, with RTCP also required and used. RFC 8825 | Whether the endpoints, signaling, and interactive-communication requirements match the application. |
| SRT | Listed as an OBS output option in the cited project discussion, which describes it as closest to the low-latency use case raised there. OBS discussion | Whether both ends support it, and whether its behavior fits the network conditions and destination requirements. |
The sources cited here do not provide a current service-by-service compatibility matrix or a universal latency figure. Check the destination’s current documentation and test the actual sender-to-ingest path when compatibility or delay is critical.
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RTMP, RTMPS, and security terminology
RTMPS appears alongside RTMP in the cited OBS protocol discussion, but the sources cited here do not establish enough technical detail to make claims about RTMPS security properties. Do not treat the names as proof that a particular destination supports a given mode or security configuration. Verify the exact protocol, endpoint, and connection requirements in that service’s current setup instructions.
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If your goal is to send a prerecorded video or playlist as a continuous YouTube live stream, you need an encoder or service that can deliver the media to YouTube’s ingest endpoint using a currently supported setup. RTMP is the transport protocol in that contribution leg when selected and supported; it does not itself create the playlist, encode the video, or guarantee that YouTube will accept the stream. Follow YouTube’s current live-stream setup requirements and ensure you have the rights to the audio and video you broadcast.
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Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Common RTMP setup problems
- The destination will not connect: confirm that it currently accepts RTMP, then check the exact ingest address, stream key, and any required connection mode against its instructions. A protocol appearing in an encoder does not guarantee a given service accepts it.
- The connection works but playback fails: distinguish transport from media compatibility. Check that the destination accepts the encoder’s audio and video formats and settings; RTMP does not convert an unsupported payload into a compatible one.
- The stream drops or stutters: investigate the sender’s network path and the destination’s ingest status, then check the encoder’s output and reconnect behavior. The cited materials do not establish one cause or a universal latency target.
- You are choosing between RTMP and another protocol: start with the destination’s supported protocols and the role you need—contribution ingest, playlist-and-segment delivery, or interactive communication. Do not choose from protocol names alone.
What to remember
RTMP is a timed-message transport protocol often encountered when an encoder sends media to an ingest server. It carries media payloads but does not define their codec or the entire viewer delivery system. Its practical suitability depends on both endpoints and the job the protocol must do.
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