RTMP Server

Turn your Android phone or tablet into a live streaming server: publish RTMP, RTSP, SRT or WebRTC from OBS, ffmpeg or any encoder; HLS, FLV or WebRTC plays on any device on your Wi-Fi. One tap starts it — no dedicated hardware, no NAS, nothing to rent.

Sub-second playback in a browser, and recordings you can scrub like any local file.

No stream source yet? Open Streams and tap Play a sample stream — the app plays a built-in sample through the same pipeline, so you can watch it on your phone before setting anything up.

START IN THREE STEPS
1. Tap Start. The app shows the exact URLs to use.
2. Point OBS, ffmpeg or your encoder at the RTMP, RTSP or SRT address.
3. Scan the QR code, or open the player page in a browser, and watch.

PUBLISH TO IT
• RTMP — OBS, ffmpeg and most encoders
• RTSP — IP cameras and NVRs
• SRT — low latency, tolerant of packet loss
• WebRTC / WHIP — OBS 30 and newer

WATCH FROM IT
• WebRTC / WHEP — sub-second latency in a browser, no plugin
• HTTP-FLV — low latency, and audio for AAC sources
• HLS (m3u8) — the widest compatibility
• Built-in player page with a QR code: scan and watch

RECORD WHILE YOU STREAM
• One independent recording task per stream, so each source records on its own
• Segments are remuxed into standard MP4 you can seek and scrub
• Set an end time, or keep the few seconds from before you tapped record
• Play recordings in the app, browse and download them from any browser
• Disk guard pauses recording before storage runs out, and never deletes your files
• Files stay in private storage until you export them

PULL AND RELAY
• Pull an RTSP, RTMP, SRT or HTTP-FLV source and republish it locally — record and distribute it like any other stream

TOOLS BUILT IN
• Live preview of every incoming stream
• Bitrate, active stream count and received bytes in real time
• Copy buttons and QR codes for every push and pull URL
• A viewer page at the phone's address: every stream in a browser, no app needed
• Event log that follows the app language
• Screen protection with 8 styles for long sessions
• 17 languages, dark Material 3 interface

HONEST LIMITS
• WebRTC carries H.264 video and Opus audio. Streams arriving with AAC audio play silently in a browser — switch the player page to HTTP-FLV to hear them.
• Publishing over WHIP from a browser requires HTTPS, so use OBS 30 or newer on a local network.
• RTMPS and HTTPS switch on after you import a certificate; without one the app stays on plain HTTP inside your LAN.
• The server runs while the app is in the foreground. Keep the screen on for long sessions; the app includes a screen protection mode for exactly that.