Implementing RTMP streaming from a mobile device

Implementing RTMP streaming from a mobile device

When developing a mobile app with a live broadcast feature to YouTube, Twitch, or Facebook Live, you quickly hit a limitation: neither iOS nor Android provide a built-in RTMP API. You have to integrate third-party libraries, which means configuring hardware encoding via VideoToolbox or MediaCodec, managing the capture session, handling connection drops, and adapting bitrate to the network. Our team has completed dozens of RTMP streaming integrations—from fitness apps to online learning platforms. A typical client request: the user needs to broadcast the screen or camera in real time with minimal latency, while the stream should work correctly in the background and recover after network loss.

The RTMP protocol runs over TCP, ensuring reliable data delivery, but requires proper timeout and reconnection settings. For ultra-low latency (<500 ms), WebRTC is used, but popular platforms don't accept it directly—a server-side re-streamer is needed. So RTMP remains the most practical choice for mobile streaming.

Why RTMP stays the standard for mobile streaming

RTMP uses TCP, provides reliable data delivery, and works through most corporate firewalls (port 1935). Latency is 1-3 seconds, acceptable for streaming to platforms. WebRTC gives latency <500 ms, but YouTube and Twitch don't accept it directly—a server re-streamer is required. SRT runs over UDP but also needs additional server software. Protocol comparison:

How to avoid black screen and connection loss

Three typical problems of mobile RTMP streaming and their solutions:

Startup delay. The first few seconds are unstable—SPS/PPS NAL units aren't cached on the server yet. On iOS solution: set kVTCompressionPropertyKey_AllowFrameReordering: false — B-frames are disabled, latency reduces by 50%.

Connection loss. On network interruption, automatic reconnection is needed. On iOS, HaishinKit has a built-in reconnect mechanism. On Android, handle onConnectionFailedRtmp with repeated startStream() calls using exponential backoff (initial interval 2 s, increase up to 30 s).

Black screen. Occurs if startStream() is called before capture initialization. Order: attachCamera → startPreview → connect → publish. On iOS, also configure AVCaptureSession before starting.

iOS: HaishinKit

The best Swift-native option. No FFmpeg dependency, hardware encoding via VideoToolbox. HaishinKit starts streaming 2x faster than FFmpeg-based libraries.

Minimal setup:

import HaishinKit

let rtmpConnection = RTMPConnection()
let rtmpStream = RTMPStream(connection: rtmpConnection)

rtmpStream.videoSettings = VideoCodecSettings(
    videoSize: CGSize(width: 1280, height: 720),
    bitRate: 2_000_000,
    profileLevel: kVTProfileLevel_H264_High_AutoLevel as String
)
rtmpStream.audioSettings = AudioCodecSettings(bitRate: 128_000)

// Attach camera
let camera = AVCaptureDevice.default(.builtInWideAngleCamera, for: .video, position: .back)
try rtmpStream.attachCamera(camera)
try rtmpStream.attachAudio(AVCaptureDevice.default(for: .audio))

// Preview
let hkView = MTHKView(frame: previewView.bounds)
hkView.videoGravity = .resizeAspectFill
rtmpStream.addOutput(hkView)
previewView.addSubview(hkView)

// Start
rtmpConnection.connect("rtmp://live.example.com/live")
rtmpStream.publish("stream_key")

Reconnection on drop: subscribe to RTMPConnection.Event.rtmpStatus, on code == RTMPStatusCode.connectClosed — reconnectDelay(3) and repeat connect(). HaishinKit has a built-in reconnect mechanism.

Monitoring: rtmpStream.info.byteCount and RTMPStream.currentFPS — watch actual FPS. If it falls below 20, it's a sign of poor connection.

Android: rtmp-rtsp-stream-client-java

Library by Pedro Vicente. Supports Camera1, Camera2, CameraX, Screen capture. Hardware encoding via MediaCodec.

val rtmpCamera = RtmpCamera2(binding.surfaceView, object : ConnectCheckerRtmp {
    override fun onConnectionSuccessRtmp() { /* update UI */ }
    override fun onConnectionFailedRtmp(reason: String) {
        rtmpCamera.stopStream()
        retryConnection()
    }
    override fun onDisconnectRtmp() { retryConnection() }
    override fun onAuthErrorRtmp() { /* show error */ }
    override fun onAuthSuccessRtmp() {}
    override fun onNewBitrateRtmp(bitrate: Long) { updateBitrateUI(bitrate) }
})

// Prepare (resolution, bitrate, FPS, audio)
rtmpCamera.prepareVideo(1280, 720, 30, 2_000_000) && rtmpCamera.prepareAudio(128_000, 44100, true)
rtmpCamera.startStream("rtmp://live.example.com/live/stream_key")

RtmpCamera2 accepts SurfaceView or TextureView. For Jetpack Compose: AndroidView { RtmpCamera2(context, ...) }.

Bitrate adaptation: rtmpCamera.setVideoBitrateOnFly(newBitrate) — change bitrate without restarting the stream.

Library comparison:

How we do it: stack and configuration

We use the described libraries with additional adaptive bitrate configuration. On iOS, we monitor CPU load and reduce bitrate by 25% if FPS drops below 20. On Android, we use onNewBitrateRtmp for dynamic adjustment. Traffic savings up to 40% without quality loss. For authentication, we use stream key in URL or RTMP auth. We support rtmps for YouTube Live.

Our engineers tune encoding parameters for specific scenarios: for screen streaming — lower bitrate, for camera — higher. We guarantee stable operation on devices with iOS 15+ and Android 10+. Contact us to evaluate your project.

Work process

1. Analytics — select protocol (RTMP/WebRTC), server, platform. Define latency and quality requirements.
2. Design — architecture of streaming module, background mode handling, reconnection, monitoring.
3. Implementation — integrate library, configure encoding, test on real devices.
4. Testing — check latency, video quality, behavior during network drops, publish to TestFlight/Google Play.
5. Deployment — prepare for release, documentation, support.

What is included

• RTMP streaming integration on iOS and/or Android
• Hardware encoding and adaptive bitrate setup
• Reconnection and error handling implementation
• TLS (rtmps) support
• Documentation for server and streaming platform configuration
• Assistance with App Store and Google Play review (background mode and privacy requirements)
• Compatibility guarantee with latest OS versions

Why choose us

We have been in mobile development for over 5 years and have completed more than 50 video projects. Our developers are certified for iOS and Android. We guarantee code quality and adherence to deadlines. Experience with RTMP streaming integrations allows us to avoid typical mistakes and deliver projects on time.

If you need RTMP streaming in your mobile app, contact us for a project evaluation. We will calculate the cost and timeline individually. Order a turnkey RTMP streaming integration and get a ready module in 2-3 days.

How to Choose a Camera Approach on Mobile Platforms?

Apps where users capture, listen, or watch are technically among the most demanding. We deal with this every day. Not because of API complexity, but due to hardware differences: on a flagship, the camera works perfectly; on a budget device with a non-standard Camera HAL, artifacts and failures occur. On iOS, stabilization differs between generations. Platform differences account for 80% of all media development complexity. Our experience: 7+ years in mobile media and over 40 implemented projects with camera, audio, and video.

What are the Differences Between CameraX, Camera2, and AVFoundation?

On Android, the Camera2 API was long the only adequate choice for custom cameras. It is a low-level API with CaptureRequest, CameraCharacteristics, ImageReader — powerful but verbose. Even a preview with correct aspect ratio and proper orientation takes several hundred lines of code.

CameraX (Jetpack) is a wrapper around Camera2 with automatic device adaptation. Preview, ImageCapture, ImageAnalysis, VideoCapture — four use cases that can be combined. It handles orientation, aspect ratio, and lifecycle for you: bind to a LifecycleOwner and forget about closing the camera when the app goes to background. In recent versions, CameraX includes Extensions API for bokeh, night mode, HDR — using native manufacturer algorithms via a unified interface.

When is Camera2 needed directly?: RAW capture via ImageFormat.RAW_SENSOR, manual control of ISO/shutter speed/focus, or when CameraX Extensions API is not supported and a custom ML pipeline in ImageAnalysis is required.

On iOS, AVFoundation is the only path for a custom camera. AVCaptureSession with AVCaptureDeviceInput and the required output (AVCapturePhotoOutput, AVCaptureVideoDataOutput, AVCaptureMovieFileOutput). For real-time video processing — AVCaptureVideoDataOutput + CVPixelBuffer in captureOutput(_:didOutput:from:) on a background queue. This is where CoreML models receive frames for inference.

A typical mistake with AVFoundation: configuring the session on the main thread. beginConfiguration() / commitConfiguration() should be called on a background thread. Otherwise, the preview freezes, and the user sees a frozen UI. This mistake appears in 70% of the projects we have audited.

Why is AudioFocus Critical for Android Apps?

Audio on mobile platforms requires correct management of the sound lifecycle. AudioFocus is a coordination mechanism between apps. AudioManager.requestAudioFocus() with OnAudioFocusChangeListener. If you don't handle AUDIOFOCUS_LOSS_TRANSIENT (pause) and AUDIOFOCUS_LOSS (stop) — your app will play over a phone call. That guarantees a bad review on Google Play. Android Developer Guide: AudioFocus

On iOS, AudioSession categories define behavior: playback — for players (continues playing when screen is locked), record — for recording, muting other sources, playAndRecord — for voice messages. Wrong category — the app mutes the user's background music on start.

AVAudioEngine — modern API for audio processing: a graph of nodes (mixers, equalizers), taps for buffer capture. For real-time speech — SFSpeechRecognizer + inputNode.installTap.

On Android for recording with noise suppression — NoiseSuppressor.isAvailable() + create(audioRecord.audioSessionId). Works not on all devices, need a fallback.

Video: Playback and Streaming

ExoPlayer (Media3) — standard for Android. Supports HLS, DASH, SmoothStreaming, progressive playback. DefaultTrackSelector with Parameters allows manual or adaptive quality selection. DRM via DefaultDrmSessionManager with Widevine L1/L3.

Almost everyone faces this problem: ExoPlayer in RecyclerView with fast scrolling. Need a PlayerPool — a pool of reusable players. Without a pool, each new instance creates a MediaCodec instance, which is expensive and leads to MediaCodec$CodecException: Error -19 on some Android 10 devices with more than 3 simultaneous instances.

AVPlayer / AVPlayerViewController on iOS — for playback. For custom UI — AVPlayerLayer + custom controls. HLS works natively via AVPlayer(url:) with m3u8. FairPlay DRM requires a server part: AVContentKeySession, CKC response from KSM server, resource delegate.

For Flutter — video_player as a base layer, chewie for UI. For serious tasks — a platform channel to native ExoPlayer/AVPlayer (due to DRM and subtitles).

WebRTC on mobile — via native frameworks or flutter_webrtc. The real complexity is not in the protocol itself, but in signaling and TURN servers. Without TURN, clients behind symmetric NAT won't establish a connection — that's about 15–20% of traffic. Coturn is the standard open-source server.

RTMP publishing on mobile: LFLiveKit for iOS, HaishinKit as a more modern alternative. On Android — rtmp-rtsp-stream-client-java or via FFmpeg with JNI. The latter gives maximum flexibility but increases the binary by 10–15 MB.

Media Processing: Compression and Transcoding

ProRes video can take up to 6 GB/minute. Compression is needed before upload. On iOS — AVAssetExportSession with a 1920×1080 preset or custom AVVideoComposition. VideoToolbox for hardware H264/HEVC encoding — faster and more battery-efficient.

On Android — MediaCodec directly or Transformer (Media3) — a high-level API for transformations (trimming, resizing, effects via GlEffectsFrameProcessor). For images — BitmapFactory.Options.inSampleSize for downsampling, Glide / Coil for caching. Coil on Coroutines fits well with Compose. Loading a 12 MP original into an ImageView of 200×200dp — a classic OutOfMemoryError on devices with 2 GB RAM.

How to Implement Streaming on Mobile Devices: Step-by-Step Plan

1. Define requirements: target latency, number of concurrent users, need for P2P.
2. Choose protocol and stack: WebRTC for video calls, RTMP/HLSLive for broadcasting.
3. Set up signaling (SIP, WebSocket, MQTT) and TURN server.
4. Implement publishing/viewing via native API or cross-platform plugin.
5. Test on real devices with different cameras and network conditions.
6. Optimize bitrate and resolution based on bandwidth.

What is Included in the Work

Development Process for Media Functionality

Complexity is non-linear: basic video playback — 1–2 days, custom camera with frame processing and streaming — 3–5 weeks. We start by clarifying requirements: DRM, formats, minimum OS, background mode support. Testing on real hardware is mandatory — the emulator does not replicate Camera HAL, hardware codec, and AudioFocus issues. Minimum set: latest iPhone, iPhone SE, flagship Samsung, budget Android, Android Go (if target audience is developing markets).

Timeline estimate: from 5 business days (basic playback) to 8 weeks (complex camera with streaming and DRM). Cost is calculated individually after analyzing your requirements — contact us for a consultation.

Our service: "Mobile Media Integration" — this is our expertise. Every project starts with an audit of the current implementation, identifying bottlenecks, and proposing an optimal stack.

Commercial signals: order an audit of your media functionality, get a free consultation from an engineer.