To convert a live RTMP feed to HLS, receive the feed with an HLS-capable server or FFmpeg, then publish an HLS playlist and its media segments over HTTP(S). If the input codecs already suit your target players, this can be a packaging-only operation—often called transmuxing—with no video or audio re-encoding. Transcoding is a separate choice for codec compatibility, resizing, or multiple adaptive-bitrate renditions.
What RTMP-to-HLS conversion produces
HLS delivery consists of a playlist, usually an .m3u8 file, and media segments referenced by that playlist. A compatible player fetches the playlist and segments over HTTP(S). Converting the delivery format does not, by itself, improve picture quality or create additional resolutions.
Choose between copying the incoming audio and video streams and encoding them anew. Stream copy avoids an encode when the input is accepted by the HLS packager and intended players. Transcoding is needed when codecs or stream characteristics are incompatible, or when you need resized or differently encoded renditions. A single copied source does not become adaptive bitrate automatically.
Route 1: package a live RTMP feed with FFmpeg
FFmpeg’s HLS muxer can package a live RTMP input. The following is an illustrative starting pattern, not a tested command or universal preset. Replace the input URL and output paths, and verify that the installed FFmpeg build supports the options used.
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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"- Make the output directory. Create
/var/www/hls/, or your chosen directory, and ensure the FFmpeg process can write to it. Your web server must be able to read the playlist and segments. - Set the input. Replace
rtmp://HOST/APP/STREAMwith the reachable RTMP publishing URL. Confirm the source is live and includes the audio and video streams you expect. - Choose copy or encode. The example uses
-c:v copy -c:a copy. If stream copy is incompatible with your packaging target or players, select suitable encoders and stream mappings explicitly. For adaptive bitrate, configure multiple encoded renditions and a master playlist; the example does not do this. - Set the segment target and playlist window.
-hls_time 4sets a target duration, while-hls_list_size 6limits the playlist to six entries. These values are examples, not universal settings. - Serve the result over HTTP(S). Configure the web server to serve the playlist and segment paths with suitable MIME types, permissions, TLS at the delivery edge, and access controls appropriate to your stream.
- Test the live URL. Open the
.m3u8URL in an HLS-capable player and check that every referenced segment can be fetched successfully.
Segment duration, keyframes, and retention
FFmpeg documents hls_time as a target: “Segment will be cut on the next key frame after this time has passed.” As a result, setting a four-second target does not guarantee segments exactly four seconds long. The incoming keyframe cadence affects the actual cuts. FFmpeg recommends a closed GOP and a GOP size that fits the segment constraint; consult the FFmpeg Formats Documentation for the supported options in your installed build.
A bounded playlist and segment deletion are related but separate operational choices. With deletion enabled, configure how many unreferenced segments remain so a viewer still fetching an older segment has time to finish. Playlist length, segment duration, and retention affect the live window, disk use, and playback behavior. Check the FFmpeg HLS muxer options for the deletion threshold and tune it for your player and delivery conditions.
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Route 2: use an RTMP server with HLS enabled
SRS documents built-in HLS support, so an SRS server can receive RTMP and generate HLS without a separate FFmpeg process in the ingest path. Its example starts SRS with HLS configuration, publishes to rtmp://localhost/live/livestream, and opens the generated playlist at http://localhost:8080/live/livestream.m3u8. Follow the configuration for the version you actually deploy: the cited SRS 7.0 HLS documentation describes that version as unstable and shows HLS disabled by default in its configuration.
Do not expect playback before the first HLS segment exists. SRS specifically cautions that playback can fail if it is started too early. SRS also lists HLS compatibility and CDN distribution among its strengths, while warning that HLS latency is higher than RTMP/HTTP-FLV. That is a product-documentation comparison, not a universal latency guarantee: actual end-to-end delay depends on segment duration, playlist behavior, player buffering, network conditions, and configuration. Test with the player and delivery setup you intend to use. See SRS’s HLS documentation.
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Route 3: use managed AWS media services
A managed pipeline is an option when you need cloud-based processing and distribution rather than operating the packaging server yourself. AWS documents MediaLive input types and lists RTMP push and RTMP pull; its cited input reference says MediaLive does not support RTMPS inputs. Confirm the current service behavior and configuration before deployment in the MediaLive input reference.
An AWS workflow example uses FFmpeg as an RTMP contribution encoder, MediaLive to produce HLS adaptive-bitrate output, and MediaPackage downstream. It recommends a CDN such as CloudFront in front of MediaPackage for production distribution. The example was published in 2019, so use it for the broad architecture and sequencing, not as a guarantee that current console steps or service behavior are unchanged. It also describes a redundancy limitation: the FFmpeg contribution stream feeds only one of MediaLive’s two pipelines, so failure of that input or pipeline can undermine redundancy. Review the workflow’s failure assumptions before treating the pipeline as resilient. See the AWS workflow example.
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A separate AWS Elemental MediaLive tutorial shows the shape of HLS output: a parent manifest, rendition manifests, and media segments. Its worked input is RTP rather than RTMP, so it illustrates the output structure, not an RTMP-specific setup.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Choose a route for the delivery you need
| Route | Best fit | Key considerations |
|---|---|---|
| FFmpeg HLS muxer | Direct control over packaging on a server you operate | Manage process supervision, writable storage, web serving, cleanup, monitoring, and any encoding or rendition configuration. |
| SRS with HLS enabled | Receiving RTMP and generating HLS in an RTMP server | Check the deployed version and configuration; account for initial segment generation, delivery setup, and HLS latency. |
| MediaLive and MediaPackage | Managed cloud processing and an HLS distribution pipeline | Verify current AWS behavior, input protocol compatibility, pipeline resilience, CDN design, and operational costs for your workload. |
Compare the routes against the codecs you receive, whether stream copy is sufficient, the need for adaptive-bitrate renditions, target latency, player behavior, CDN and geographic scale, and how you will monitor and recover from failures. None has a universally correct segment duration or latency setting.
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Troubleshoot RTMP-to-HLS playback
- No playlist or segments appear: Confirm the RTMP URL is reachable and live, inspect FFmpeg or server logs, and verify that the output directory exists and is writable.
- The playlist loads but playback fails: Check that each segment path referenced in the playlist matches the files on disk and that the web server can read and serve them. Verify HTTP(S) access, MIME handling, and any access controls.
- Audio or video is missing: Inspect the incoming streams and codecs, then check FFmpeg stream selection and logs. If the input is not accepted by the target packager or player, configure explicit maps and encoding rather than assuming stream copy will work.
- Segments are longer than the target: Inspect the source keyframe/GOP cadence. The muxer cuts at a suitable keyframe after the target has passed, so changing
hls_timealone may not produce the segment duration you expect. - Playback starts late on SRS: Allow time for the first HLS segment to be generated before opening the playlist.
- MediaLive does not connect: Check whether the sender and MediaLive input agree on RTMP versus RTMPS. The AWS workflow example also sequences starting MediaLive before the upstream sender; verify the current AWS procedure for your configuration.
- Older segments disappear before viewers can fetch them: Review playlist length and the HLS muxer’s segment deletion threshold. Keep unreferenced segments long enough for clients already downloading them.
- Production delivery is fragile or slow: Review monitoring, process or service recovery, storage cleanup, access control, TLS at the HTTP delivery edge, CDN behavior, and redundancy. Measure end-to-end latency using the actual player and network conditions.
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StreamNeo is for a different use case: it loops uploaded videos to a YouTube channel as a 24/7 live stream. It is not an RTMP-to-HLS converter and does not output HLS. To use it, upload a recording or build a playlist, add your YouTube stream key, and go live. The cloud keeps the stream running without a computer or home connection staying on. It supports any uploaded quality up to 4K 60fps at one flat price per slot, with automatic recovery if YouTube drops the stream. The first day is free with no card.
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