To convert a live RTMP feed to HLS, ingest the RTMP stream with FFmpeg or an RTMP server such as SRS, then publish an HLS playlist and its media segments over HTTP(S). If the incoming audio and video codecs are compatible with your target players, you may be able to package them without re-encoding; transcoding is a separate choice for compatibility or adaptive-bitrate renditions.
What RTMP-to-HLS conversion does
RTMP is commonly used to send a live contribution feed to a server. HLS packages video into media segments and a playlist that a player requests over HTTP(S). Conversion changes how the stream is packaged and delivered; it does not automatically improve picture quality or create multiple quality levels. For an overview of HLS output structure, see the AWS Elemental MediaLive tutorial.
There are two processing choices. Stream copy (also called transmuxing) repackages compatible audio and video without encoding them again. Transcoding decodes and re-encodes media, which may be needed if the source codecs are unsuitable for the intended players, or if you need resized or lower-bitrate renditions. A copied single source does not become adaptive bitrate by itself.
Choose a conversion route
| Route | Best suited to | What to account for |
|---|---|---|
| FFmpeg HLS muxer | Operators who want direct control of ingest, packaging, and output files. | You manage the process, writable storage, HTTP(S) serving, cleanup, and monitoring. Stream copy works only when the source is compatible with the intended output. |
| SRS with HLS enabled | Self-hosted setups that want an RTMP server to generate HLS as part of its service. | Check the exact SRS version and configuration. The cited version 7.0 documentation labels that version unstable and shows HLS disabled by default. |
| AWS Elemental MediaLive and MediaPackage | Managed workflows that need an encoding and packaging pipeline, with a CDN distribution path available. | More services and configuration are involved. Verify current AWS console steps and service behavior; the published workflow example dates to 2019 and describes a contribution-input redundancy limitation. |
For a straightforward self-hosted conversion, start with FFmpeg. If you need managed operations, multiple renditions, or a broader distribution architecture, evaluate the SRS and AWS options against your latency, scaling, and resilience requirements.
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Convert a live RTMP feed with FFmpeg
1. Check the source and prepare the output directory
Confirm that the RTMP URL is reachable and that it contains the audio and video streams you expect. Identify their codecs before choosing stream copy or encoders. Create an output directory that the FFmpeg process can write to, and configure your web server to serve its playlist and segment files over HTTP(S). Make sure the web server has suitable MIME types and access controls for your use case.
2. Start with a live HLS packaging command
This is an illustrative starting pattern, not a tested command or universal preset. Replace the example host, application, stream name, and output paths with values for your environment:
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ffmpeg -i "rtmp://HOST/APP/STREAM"
-c:v copy -c:a copy
-f hls
-hls_time 4
-hls_list_size 6
-hls_flags delete_segments+temp_file
-hls_segment_filename "/var/www/hls/stream_%06d.ts"
"/var/www/hls/stream.m3u8"The command copies video and audio into HLS segments without re-encoding, if FFmpeg accepts the incoming streams and the codecs suit your target players. If that is not true, select the needed input streams with explicit maps and configure appropriate encoders. For adaptive bitrate, define the encodes and rendition outputs explicitly and generate a master playlist; one copied source does not create those renditions.
3. Set segment length and live-window behavior deliberately
-hls_time 4 sets a target duration, not an exact cut timer. FFmpeg cuts at the next keyframe after the target has passed, so keyframe cadence affects actual segment length. FFmpeg recommends a closed GOP and a GOP size that fits the segment constraint. If you control the encoder upstream, coordinate its keyframe interval with your packaging target; otherwise, inspect the source cadence and resulting segment durations. The FFmpeg Formats Documentation explains the HLS muxer options.
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-hls_list_size 6 limits the number of entries in the live playlist. That is separate from how long unreferenced segment files remain on disk when segment deletion is enabled. Set playlist length and deletion behavior with both viewer delay and storage in mind: an old segment should not be removed before a viewer that is still downloading it can finish. Consult the installed FFmpeg build’s documentation for available options and deletion-threshold behavior.
4. Serve and check the HLS output
Open the playlist URL through your HTTP(S) server, not just from the local filesystem. Check that the playlist updates while the RTMP input is live, and that every segment path it references returns successfully. Keep the playlist and segment paths consistent, confirm file permissions allow the web server to read them, and test with the player and network conditions your audience will use.
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Use SRS to receive RTMP and generate HLS
SRS documents built-in HLS support. Its example starts SRS with HLS configuration, publishes to rtmp://localhost/live/livestream, and plays the generated playlist at http://localhost:8080/live/livestream.m3u8. Use those addresses as documentation examples, not as public endpoints for your deployment. Review the SRS HLS documentation for the version and configuration you will actually run.
Playback may fail if a viewer requests the playlist before SRS has generated its first HLS segment. Allow for initial segment creation before testing playback. The cited SRS documentation identifies version 7.0 as unstable and shows HLS disabled by default in its example configuration, so verify both version status and the effective configuration rather than assuming HLS is already enabled.
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Build a managed AWS pipeline
A documented AWS architecture sends an RTMP contribution feed to MediaLive, produces HLS output, and uses MediaPackage downstream. AWS lists RTMP push and RTMP pull as MediaLive input types; its cited input reference says MediaLive does not support RTMPS inputs. Check the current MediaLive input reference when selecting an upstream protocol.
The AWS RTMP-to-MediaLive-to-MediaPackage workflow example was published in 2019. It is useful for understanding the broad architecture and sequence, but verify current console steps and service behavior before deployment. In that example, the FFmpeg contribution stream feeds only one of MediaLive’s two pipelines, creating a redundancy limitation if the input or that pipeline fails. The workflow recommends putting a CDN such as CloudFront in front of MediaPackage for production distribution.
In a separate AWS tutorial, an HLS output consists of a parent manifest, rendition manifests, and media segments; its worked example uses one H.264 video encode and one audio encode. That tutorial uses RTP rather than RTMP as its input, so it illustrates the output structure, not an RTMP-specific setup.
Plan for latency, compatibility, and production operations
HLS is useful when HTTP delivery and CDN distribution matter, but it can introduce more delay than RTMP or HTTP-FLV. SRS describes HLS latency as higher than those alternatives; its latency figures are vendor documentation estimates, not a universal guarantee. Segment duration, playlist behavior, player buffering, network conditions, and configuration all affect the result. Measure end-to-end latency using the actual source, delivery path, and player.
- Player and codec support: Test the source codecs in your intended players. Do not assume stream copy will be accepted everywhere.
- Adaptive bitrate: Decide whether viewers need multiple renditions. If so, configure separate encodes and a master playlist rather than treating a single HLS output as adaptive.
- Distribution: For geographically broad delivery, assess CDN integration, origin behavior, and access controls.
- Resilience and monitoring: Plan for ingest or pipeline failure, process restarts, alerts, and recovery. A single contribution path can remain a single point of failure even when the downstream service has multiple pipelines.
- Storage and security: Monitor disk usage and segment cleanup, serve over TLS at the delivery edge where appropriate, and check permissions and access rules.
Troubleshoot common RTMP-to-HLS failures
| Symptom | Likely cause | What to check or change |
|---|---|---|
| No playlist or segments appear | The RTMP source is unreachable, the process stopped, or the output path is not writable. | Check the RTMP URL, FFmpeg or server logs, input stream presence, and output-directory permissions. |
| The playlist exists but playback fails | Referenced segments are missing, inaccessible, or incompatible with the player. | Request the playlist over HTTP(S), inspect its segment paths, test each referenced segment, and verify web-server read access and MIME configuration. |
| Segments are longer than the configured target | The next keyframe occurs after the target duration. | Inspect the input keyframe/GOP cadence. Coordinate a suitable closed GOP and GOP size with the segment constraint if you control encoding. |
| Playback starts late or initially fails in SRS | The player requests HLS before the first segment is generated, or it is buffering the live window. | Wait for initial segment creation and then reload or retest the playlist; account for player buffering when assessing startup. |
| AWS MediaLive does not connect to the upstream | The input protocol may be unsupported or the sender may start in the wrong sequence for the chosen workflow. | Confirm RTMP rather than RTMPS against the current AWS input reference. In the documented example, start MediaLive before the upstream sender. |
| Old segments disappear while viewers need them | Segment deletion is too aggressive for the live window or viewer download time. | Review playlist size and the deletion threshold as separate settings; retain unreferenced segments long enough for active clients to fetch them. |
Or let it run in the cloud
If your goal is to keep uploaded recordings looping as a 24/7 YouTube live stream, rather than convert an incoming RTMP feed into HLS for HTTP delivery, StreamNeo is a different kind of service. Upload a recording or build a playlist, add your YouTube stream key, and go live; StreamNeo loops the uploaded videos from the cloud, so your computer and home connection do not have to stay on. It supports any uploaded quality up to 4K 60fps at one flat price per slot, includes automatic recovery if YouTube drops the stream, and the first day is free with no card. Monthly billing is $9.99 per month. This is for uploaded video streaming to YouTube, not an RTMP-to-HLS converter or a camera-based live stream. Start the free day with StreamNeo.
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