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What an RTMP-to-WebRTC bridge does
RTMP is commonly used to publish a live feed to a server. WebRTC is used to deliver interactive, low-latency playback to viewers. A media server sits between them: it receives the RTMP stream and makes a WebRTC playback route available. SRS explicitly describes RTMP-to-WebRTC conversion, and OvenMediaEngine documents RTMP input with WebRTC output. SRS documentation · OvenMediaEngine documentation
“Convert” does not always mean decoding and re-encoding every frame. If the incoming audio and video formats are compatible with the server’s WebRTC output, forwarding may be sufficient. If a codec, profile, or output requirement is incompatible, use a supported transcoder or a separate processing step. OvenMediaEngine documents an embedded transcoder; MediaMTX points to FFmpeg when transcoding, filtering, or handling an unsupported protocol is needed. MediaMTX forwarding documentation
Choose a bridge for your stream
| Option | What its documentation establishes | Consider it when |
|---|---|---|
| OvenMediaEngine | RTMP input, WebRTC output, embedded transcoding, and other streaming outputs. Official guide | You need a self-hosted media server and want an integrated transcoding option; check codec support, playback/signaling configuration, and scaling needs. |
| SRS | Its official LTS v6 overview explicitly names RTMP-to-WebRTC conversion. Official overview | You need the documented protocol path and are comfortable selecting a release and configuring a media server. |
| MediaMTX with WHIP and, if needed, FFmpeg | Documents forwarding through WHIP and recommends FFmpeg when transcoding, filtering, or unsupported protocol handling is needed. Official guide | Forwarding is enough, or you can add a processing step and the destination supports WHIP. |
| Managed service | OvenMedia Labs presents OvenMedia Cloud as a managed streaming service based on OvenMediaEngine. The cited product page does not state commercial terms. OvenMedia Labs | You want the provider to take on some server operations; confirm regions, security, scale, features, and pricing directly. |
No one option is best for every stream. Compare source and destination protocol support, codec compatibility, latency goals, expected publisher and viewer scale, WebRTC network reachability, monitoring and recovery, and who will operate the service.
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Set up the RTMP-to-WebRTC path
- Select and install the bridge. Evaluate SRS, OvenMediaEngine, or MediaMTX against the protocols and media processing your stream needs. Follow the guide for the exact product and release you deploy; configuration keys and network requirements differ. SRS LTS v6 · OvenMediaEngine · MediaMTX
- Configure RTMP ingest. Set the encoder or upstream source to publish to the server’s RTMP ingest address and stream name. Keep the publishing key private. Use encrypted transport when the server supports it and the deployment requires it.
- Enable WebRTC delivery. Configure the server’s WebRTC listener and signaling path. Make sure the network candidates it advertises are reachable by actual viewers; depending on the deployment, this may involve firewall/NAT configuration or relay services. Do not copy a generic port recipe: use the matching server and deployment documentation.
- Check audio and video compatibility. Confirm that the selected server can deliver the source’s codecs and profiles to WebRTC clients. If it cannot, enable a supported server transcoder or add an FFmpeg processing step where appropriate. MediaMTX identifies FFmpeg for cases requiring transcoding, filtering, or unsupported protocol handling. MediaMTX forwarding guide
- Test from the viewer’s network. Confirm that the server receives the RTMP feed, a real target network can establish WebRTC playback, and audio and video render. Also check reconnect behavior and whether playback remains acceptable when viewer bandwidth changes.
Amazon IVS real-time stages are a specific case, not a general converter
Amazon IVS real-time stages document RTMP/RTMPS and WHIP as ingest choices. That does not by itself establish that IVS will take an arbitrary existing RTMP feed and republish it as WebRTC. For a general RTMP-to-WebRTC bridge, use a service whose documentation explicitly supports that input-to-output path.
IVS ingest requirements
For IVS real-time stages, AWS specifies H.264 Baseline video, a one- or two-second IDR interval, 10–60 fps, zero B-frames, 160p–720p, and a maximum video bitrate of 8.5 Mbps. AWS specifies AAC-LC audio for RTMP and Opus for WHIP. These are IVS requirements, not universal WebRTC limits. AWS IVS stream ingest documentation
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AWS warns that RTMP encoders with B-frames may be disconnected and recommends veryfast and zerolatency settings for H.264 in this workflow. Its RTMP publishing guide describes creating a stage and ingest configuration, associating the configuration with a stage, and publishing with the resulting endpoint and stream key. AWS recommends RTMPS unless a verified use case requires plain RTMP. Its OBS example recommends one- or two-second keyframes, disabling B-frames, and staying below 2.5 Mbps because that described workflow lacks simulcast. AWS also says RTMP streams must contain both audio and video. These settings and limits apply to the documented IVS workflow only. AWS IVS RTMP publishing guide
Latency, network reachability, and recovery
OvenMediaEngine describes its WebRTC output as sub-second, but that is a product capability description, not an independently measured guarantee for a particular deployment. Viewer-perceived latency depends on the entire path: source encoding and ingest, server processing, network conditions, and playback. OvenMediaEngine guide
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WebRTC playback also depends on viewers being able to reach the server’s advertised network candidates. Test from the networks your audience actually uses, not just from the server’s local network. The exact listener, signaling, firewall, NAT, and relay setup depends on the product and deployment, so use that product’s current guide rather than assuming one universal configuration.
Troubleshoot common failures
- The server does not receive the stream: Check the RTMP ingest address, stream name, publishing key, and encoder output configuration. Confirm the server’s ingest endpoint is enabled and reachable.
- RTMP connects but WebRTC playback does not: Verify WebRTC listener/signaling configuration and whether advertised network candidates are reachable through the deployment’s firewall or NAT. Test from an external viewer network.
- Video plays but audio is missing, or vice versa: Check both audio and video codec compatibility. Do not assume an RTMP-compatible audio codec will meet the requirements of every WebRTC destination; the IVS RTMP and WHIP audio requirements, for example, differ.
- The bridge rejects the media or playback fails on a codec/profile: Confirm the destination’s supported formats. If direct forwarding cannot meet them, configure the server’s transcoder or an FFmpeg processing step where supported.
- IVS disconnects an RTMP encoder: For the IVS real-time-stage workflow only, check the AWS requirements for B-frames, H.264 settings, keyframe interval, resolution, frame rate, bitrate, and the required audio/video tracks.
- Latency is higher than expected: Check the complete source-to-viewer path, including encoding, server processing, routing, and client playback. A vendor’s latency description is not a guarantee for every network or configuration.
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