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How the RTMP-to-WebRTC workflow works
This is a server-side workflow, not a browser setting. An encoder or IP camera publishes RTMP to a Wowza live application; Wowza then serves the stream to WebRTC playback clients. The current Wowza setup guide lists Wowza Streaming Engine 4.11 or later with the latest configuration and a live application with TLS configured as prerequisites. Encryption is mandatory. Wowza’s WebRTC setup guide was updated September 30, 2026.
Configure Wowza for secure WebRTC playback
1. Configure TLS and signaling
In WSE Manager, configure the SSL/TLS host port for WebRTC WebSocket signaling, choose the WebRTC implementation, save, and restart the virtual host. Wowza’s current guide identifies v2 as the implementation selection. Follow the instructions for the installed version if its interface or configuration differs. The guide also describes the VHost.xml alternative.
WebRTC media uses DTLS-SRTP, while signaling must use HTTPS or WSS. Do not expose an insecure signaling endpoint as a substitute for TLS.
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2. Enable WebRTC on the live application
In the application’s WebRTC setup, enable “Play WebRTC from Wowza Streaming Engine,” then save and restart the application. Enable publishing only if the application also needs to ingest WebRTC sources; it is not required just to play the RTMP input over WebRTC. WHIP/WHEP can be enabled for standards-based HTTP endpoints if your workflow needs them. As an XML alternative, set the application’s <WebRTC><Enable> element to true.
3. Publish the RTMP source
Configure your encoder or camera to publish through the normal RTMP workflow to the Wowza live application. Confirm the incoming stream is active in Wowza before troubleshooting WebRTC playback. The source may be an encoder or an IP camera; Wowza’s tutorial describes both as possible RTMP inputs.
Choose passthrough or transcoding
Whether Wowza can pass media through depends on the source audio codec, video codec, and video encoding profile relative to the WebRTC output and clients. Wowza says transcoding is required when one of these differs. Check what the encoder is actually sending rather than relying only on its preset name.
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When to transcode
Wowza’s RTMP-to-WebRTC example starts with H.264 High-profile video and AAC audio, then converts video to H.264 Main profile and audio to Opus. The example uses an application transcoder and a custom template. It recommends no B-frames, 720p at 30 fps for compatibility, and 48 kHz stereo audio. These are Wowza recommendations for compatibility, not universal guarantees for every browser, device, or network.
Transcoding can make an otherwise incompatible source usable, but it adds processing and can increase startup time and latency compared with a video-only passthrough workflow. If low latency is the priority, first determine whether the source already matches what the intended clients support.
WebRTC codec support
Wowza’s workflow reference lists HEVC/H.265, H.264, VP8, and VP9 for WebRTC video, and Opus (recommended), PCMU, and PCMA for audio. Browser support varies, so verify support for the exact codec and profile on the target clients before choosing an encoder or transcoder output. Wowza says its WebRTC workflow supports audio and video, but does not implement data channels for uses such as text chat. See Wowza’s WebRTC workflow reference, updated August 25, 2026.
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Test playback securely
Use the Wowza test page
Wowza’s hosted test page expects a secure signaling URL in this form, with your certificate domain, application, and stream names substituted:
wss://[ssl-certificate-domain-name]/webrtc-session.json
Enter the matching application and stream names, then test from a client that can reach the server’s signaling and media endpoints. If playback fails, confirm the RTMP stream is active first, then check the URL, certificate, signaling settings, application enablement, codecs, and network path.
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Prepare a production playback page
Wowza says a production playback page must be hosted on a server using SSL/TLS. Its current setup guide names the Wowza Flowplayer Real-Time Streaming (WebRTC) plugin as a production playback option. Use the current setup instructions for the plugin and your installed Wowza version rather than assuming the test page is a production player.
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Wowza’s workflow article lists these ports for its documented configuration. Confirm the actual configured media address and firewall rules for your deployment; port requirements can depend on configuration and network topology.
| Purpose | Port or range | Protocol |
|---|---|---|
| WSS signaling | 443 | TCP |
| Default media range | 6970–9999 | UDP |
| Fallback-related configuration | 1935 | TCP |
Clients must be able to reach the signaling endpoint and the configured media address. Wowza also warns that endpoint-security deep packet inspection can cause stuttering; where applicable, configure bypasses for the documented ports.
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Direct WebRTC delivery is suited to one-to-few and few-to-few use, but performance can suffer as connections grow and each delivery consumes bandwidth. For one-to-many distribution, Wowza recommends considering transcoding followed by HLS or MPEG-DASH, which it says uses significantly less bandwidth than direct WebRTC delivery. Stream Targets can send streams to CDNs or Wowza Video.
Troubleshoot common failures
- The stream does not appear in the player: Check that the RTMP source is publishing to the intended live application and that Wowza shows it as active. Verify the application and stream names entered in the test page.
- Signaling fails before playback starts: Confirm TLS is configured, the signaling URL uses
wss://, TCP 443 is reachable, and the virtual host and application were restarted after configuration changes. - Playback starts but has no audio or video: Check the incoming codecs and video profile against the selected WebRTC output and client support. Transcode where a source codec or profile is incompatible; Wowza’s example maps AAC to Opus and H.264 High to Main.
- Playback stutters or drops: Check that clients can reach the configured UDP media range and address, and investigate firewall or endpoint-security inspection. Also consider whether direct WebRTC fan-out exceeds the intended one-to-few or few-to-few use.
- Playback works on the test page but not the production site: Ensure the production page is served over HTTPS and that it uses the correct secure signaling URL and stream identifiers.
- Text chat is expected to travel with the stream: Wowza’s documented WebRTC workflow does not implement data channels; provide chat separately or select a system that supports the needed data-channel behavior.
Copyright and YouTube live rules are separate concerns
This Wowza workflow explains how to deliver an RTMP source to WebRTC clients; it does not grant rights to the source material or determine whether a separate YouTube broadcast complies with YouTube’s policies. If your production also streams to YouTube, confirm that you have the necessary rights and that the content meets YouTube’s current rules, including rules relevant to reused content. Do not treat successful RTMP ingest or WebRTC playback as proof of permission.
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