To convert a live RTMP feed to HLS, package the incoming stream as an HLS playlist and media segments, then serve those files over HTTP or HTTPS. With FFmpeg, you can often do this without re-encoding if the source codecs are compatible with your target players. Use an RTMP server such as SRS for built-in HLS output, or a managed pipeline such as AWS Elemental MediaLive and MediaPackage when you need cloud operations, adaptive-bitrate outputs, and distribution infrastructure.
What RTMP-to-HLS conversion changes
RTMP is commonly used to send a live feed to an ingest server. HLS packages video and audio into media segments and provides a playlist—usually an .m3u8 file—that tells a player which segments to request over HTTP(S). The player fetches the playlist and its referenced segments. See the FFmpeg HLS muxer documentation and the AWS Elemental MediaLive tutorial.
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Changing the delivery format does not automatically require encoding. If the incoming codecs and stream characteristics are supported by your HLS packaging and playback targets, FFmpeg can copy the encoded audio and video into HLS segments. Transcode when the codecs are incompatible, when you need to resize or change bitrates, or when you need multiple adaptive-bitrate renditions. Copying a single source does not create adaptive bitrate by itself.
Choose a conversion route
| Route | Best fit | Key considerations |
|---|---|---|
| FFmpeg HLS muxer | Self-hosted conversion with direct control over packaging. | Run FFmpeg where it can reach the RTMP source and write to a directory served over HTTP(S). Set playlist length and segment retention deliberately; configure encoding and maps explicitly for multiple renditions. |
| SRS with HLS enabled | Self-hosted RTMP ingest with HLS generated by the server. | Confirm the version and configuration you deploy. The cited SRS v7.0 documentation identifies that version as unstable and shows HLS disabled by default in its example configuration. Playback may not start until the first segment is generated. See SRS HLS documentation. |
| AWS MediaLive and MediaPackage | Managed cloud processing and downstream HLS packaging and distribution. | MediaLive accepts RTMP push and pull inputs in its documented input reference, but does not support RTMPS inputs there. The AWS workflow example uses an RTMP contribution stream into MediaLive and MediaPackage downstream; its FFmpeg input feeds only one of MediaLive’s two pipelines, a redundancy limitation. Verify current service behavior and console steps because the workflow PDF dates from 2019. See MediaLive input types and the AWS workflow example. |
Choose based on who will operate the system, whether stream copy is sufficient, latency and player requirements, rendition needs, CDN integration, and the redundancy and monitoring you need. HLS can work well for HTTP and CDN delivery, but SRS warns that its latency can be higher than RTMP or HTTP-FLV; actual end-to-end delay depends on segment length, playlist behavior, player buffering, network conditions, and configuration. Treat vendor latency estimates as estimates, not service guarantees. See SRS documentation.
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Convert a live RTMP feed with FFmpeg
This is an illustrative starting point, not a tested command or universal preset. Replace the input URL and output paths, and confirm that the codecs work for your intended players. The output directory must be writable by FFmpeg and served over HTTP(S).
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"- Check the source. Confirm that the RTMP URL is reachable and carries the expected video and audio streams. Check the codecs before relying on
-c:v copy -c:a copy. - Prepare the output directory. Create the HLS directory, grant FFmpeg write access, and configure your HTTP(S) server to read the playlist and segment files. Keep the playlist’s referenced segment paths consistent with the served paths.
- Set the packaging options. In the example,
-hls_time 4requests a target segment duration,-hls_list_size 6limits the live playlist to six entries, anddelete_segmentsremoves old segments according to FFmpeg’s retention behavior.temp_fileis also enabled. Review the options supported by your installed build in the FFmpeg formats documentation. - Start FFmpeg and inspect its output. Confirm that the process stays connected to the input and that the playlist and segment files are being created and updated.
- Test delivery over HTTP(S). Open the served
.m3u8URL in a compatible HLS player and check that each referenced segment can be fetched. Configure suitable MIME types, TLS at the delivery edge, and access controls for your use case.
Understand segment timing and the live window
-hls_time is a target, not a timer that guarantees exact cuts. FFmpeg cuts a segment at the next keyframe after the target has passed. The FFmpeg documentation puts it this way: “Segment will be cut on the next key frame after this time has passed.” A four-second target can therefore produce longer segments if keyframes are farther apart. For predictable segmentation, align keyframe cadence with the segment constraint and use a closed GOP; check the installed build’s HLS options.
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The playlist length and segment deletion policy are related but distinct choices. A short playlist limits the visible live window, while retention determines how long unreferenced files remain on disk. Configure cleanup so a viewer still fetching an older segment has time to finish; account for storage use as well as playback behavior.
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If a target player cannot decode the source codecs, or if viewers need different resolutions or bitrates, replace stream copy with appropriate encoding and explicit stream maps. For adaptive-bitrate HLS, produce multiple renditions and a master playlist that points to their individual playlists. A single copied input yields a single rendition unless you build a more elaborate pipeline.
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Use SRS to generate HLS from RTMP
SRS documents a built-in HLS route: start SRS with HLS configured, publish RTMP to rtmp://localhost/live/livestream, then request the generated playlist at http://localhost:8080/live/livestream.m3u8. Follow the configuration instructions for the exact SRS version you deploy at SRS HLS documentation.
Do not expect playback before the first HLS segment exists. The SRS documentation specifically cautions that playback can fail if a player starts too early. Its v7.0 page identifies that version as unstable and shows HLS disabled by default in the displayed configuration, so check both version status and configuration rather than assuming HLS is active.
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Use AWS MediaLive and MediaPackage
A managed architecture can receive an RTMP contribution feed in MediaLive, generate HLS output, and send it to MediaPackage for downstream packaging and delivery. AWS’s workflow example uses FFmpeg as the RTMP contribution encoder, MediaLive for HLS adaptive-bitrate output, and MediaPackage downstream. For production distribution, that example recommends a CDN such as CloudFront in front of MediaPackage. The workflow PDF is from 2019, so treat it as architecture guidance and verify current console steps and service behavior.
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Check the input protocol before configuring the sender: AWS’s current MediaLive input reference lists RTMP push and pull and says RTMPS inputs are not supported in the cited reference. In the workflow example, MediaLive is started before the upstream sender. The example’s FFmpeg contribution stream feeds only one of MediaLive’s two pipelines, limiting redundancy if that input or pipeline fails. See AWS MediaLive input types and the AWS RTMP-to-MediaLive workflow.
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An AWS HLS output can include a parent manifest, rendition manifests, and .ts media segments. The AWS tutorial demonstrates that output structure with one H.264 video encode and one audio encode; its worked example uses RTP as input, rather than RTMP.
Troubleshoot RTMP-to-HLS playback
| Symptom | What to check | Likely action |
|---|---|---|
| No playlist or segments appear | RTMP reachability, source streams, FFmpeg logs, and write permissions. | Verify the input URL and expected audio/video tracks; fix output-directory permissions and confirm the process remains connected. |
| The playlist loads but playback fails | Whether the playlist’s segment URLs are correct and whether each segment can be fetched over HTTP(S). | Align filesystem and served paths, check read permissions, MIME configuration, access controls, and codec support in the player. |
| Segments are longer than the target | Keyframe/GOP cadence. | Remember that FFmpeg cuts at a keyframe after the target has passed; adjust keyframe cadence and GOP configuration if your encoding workflow allows it. |
| SRS playback fails immediately after publishing | Whether the first HLS segment has been generated. | Wait for initial segment creation, then reload or start the player. |
| AWS input will not connect | RTMP versus RTMPS and the order in which services and sender start. | Use an input protocol supported by the configured MediaLive input; the cited AWS reference does not support RTMPS inputs. In the documented workflow, start MediaLive before the upstream sender. |
| Old segments disappear while viewers are fetching them, or storage grows unexpectedly | Playlist size and segment-retention/deletion settings separately. | Increase retention if clients need longer to fetch older files, and monitor disk use as the live process continues. |
For production, also plan monitoring, cleanup, access control, TLS at the HTTP delivery edge, CDN behavior, and recovery from source or pipeline failures. Test latency with the actual player and network path; a segment-duration setting alone does not determine viewer-perceived delay.
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