Stable Live Video from Mobile via SRT Protocol
Your mobile device streams live video from the scene, but the LTE network is unstable—packet loss reaches 20%. RTMP streams break, viewers see a black screen. We solve this with the SRT Protocol. Unlike RTMP, SRT works over UDP with built-in error correction (ARQ/FEC), tolerates up to 30% packet loss, and automatically recovers the stream without restarting. In fact, SRT is 5 times more reliable than RTMP under similar packet loss conditions. For mobile streaming over unreliable LTE or public Wi-Fi, this is a game-changer. Our team has over 5 years of mobile development experience and has delivered over 20 video streaming projects. We guarantee stable SRT connections in challenging network conditions with certified developers. A typical project with both iOS and Android integration starts at $1000, with savings on server costs compared to RTMP. Contact us for a consultation on your project. Integration starts from $500 for basic setup on one platform.
Why SRT Outperforms RTMP on Mobile
At just 5% packet loss, RTMP starts buffering and breaking; SRT with a latency=500ms buffer remains stable at 20-25% loss. SRT provides up to 10 times lower latency (200–2000 ms vs 5–10 s) and uses jitter buffers and FEC to maintain quality. For reporter-style streaming from events, remote video production, and field broadcasts, SRT has become the professional standard—native support in OBS, vMix, and Blackmagic. Latency is adjustable via the latency parameter: 200 ms minimum for good networks, 500-2000 ms for mobile networks with jitter. Larger buffer = more resilience, but higher delay. Infrastructure savings: SRT doesn't require expensive servers or TCP accelerators—it runs over UDP, reducing operational costs.
Ensuring a Stable Connection Under Weak Signal
SRT supports automatic reconnection within connectionTimeout. While basic parameter tuning helps, we go further: we monitor pktSndLoss and when it exceeds 5%, we automatically reduce videoBitrate and increase latency. This requires a reconnection but prevents a total stream failure.
iOS: HaishinKit with SRT
HaishinKit has supported SRT since version 2.x through SRTConnection:
import HaishinKit
let srtConnection = SRTConnection()
let srtStream = SRTStream(connection: srtConnection)
srtStream.videoSettings = VideoCodecSettings(
videoSize: CGSize(width: 1920, height: 1080),
bitRate: 5_000_000,
profileLevel: kVTProfileLevel_H264_High_AutoLevel as String
)
srtStream.audioSettings = AudioCodecSettings(bitRate: 192_000)
try srtStream.attachCamera(AVCaptureDevice.default(.builtInWideAngleCamera, for: .video, position: .back))
try srtStream.attachAudio(AVCaptureDevice.default(for: .audio))
// Caller mode: we initiate the connection
srtConnection.connect("srt://192.168.1.100:9998?latency=500&streamid=live/stream1")
srtStream.publish()
The streamid identifies the stream on the server (MediaMTX, SRT Live Server use it for routing). latency is in milliseconds. Reconnection on disconnect: SRT automatically reconnects within connectionTimeout. HaishinKit exposes SRTConnection.Event.connect with a reason—handle in your delegate.
Android: rtmp-rtsp-stream-client-java with SRT
The rtmp-rtsp-stream-client-java library supports SRT via SrtCamera2:
val srtCamera = SrtCamera2(binding.surfaceView, connectChecker)
srtCamera.prepareVideo(
width = 1920, height = 1080, fps = 30,
bitrate = 5_000_000,
rotation = 0,
profile = CodecUtil.H264_BASELINE
)
srtCamera.prepareAudio(bitrate = 192_000, sampleRate = 44100, isStereo = true)
srtCamera.startStream("srt://192.168.1.100:9998?streamid=live/stream1&latency=500")
Under the hood: native libsrt (C++) compiled for Android ABIs (arm64-v8a, armeabi-v7a, x86_64).
Alternative: FFmpegKit with libsrt
Build FFmpegKit with libsrt and use it directly:
-f avfoundation -i 0:0 -c:v h264_videotoolbox -b:v 5M -f mpegts "srt://server:9998?latency=500&streamid=live/test"
Advantage: H.265 support (hevc_videotoolbox), more control. Con: FFmpegKit doesn't manage the camera directly—you need a separate capture via AVCaptureVideoDataOutput + pipe.
Listener vs Caller Mode
SRT supports two modes:
- Caller — the mobile device initiates the connection to a server. Most common.
- Listener — the device listens for incoming connections. Useful for p2p without a server or when the server is behind NAT.
- Rendezvous — both ends initiate simultaneously. For p2p through NAT without relay.
For mobile streaming, we use Caller. Listener on mobile is rare (IoT device without public IP).
Encryption
SRT has built-in AES-128/256 encryption via passphrase and pbkeylen in the URL. This is a differentiator from RTMP—encryption is protocol-level, not a TLS wrapper.
srt://server:9998?passphrase=MySuperSecret&pbkeylen=32
Quality Monitoring
SRT API provides statistics via srt_bistats(): pktSndLoss, pktRcvLoss, pktRetrans, mbpsSendRate, msRTT. HaishinKit exposes some stats in SRTStream.info. We display bitrate, RTT, and packet loss in the UI—professional users value this. When pktSndLoss exceeds 5%, we automatically reduce videoBitrate and increase latency (requires reconnection).
SRT vs RTMP Comparison
| Parameter |
SRT |
RTMP |
| Protocol |
UDP with ARQ+FEC |
TCP |
| Latency |
200–2000 ms (configurable) |
5–10 s |
| Packet loss tolerance |
up to 30% |
up to 5% |
| Encryption |
Built-in AES-128/256 |
via TLS |
| Mobile support |
HaishinKit, FFmpegKit |
RTMP SDK |
Server Compatibility
| Server |
SRT Support |
| MediaMTX |
Native, no extra config |
| Nginx + nginx-srt-module |
Yes |
| Ant Media Server |
Yes |
| Wowza Streaming Engine |
Yes (4.8+) |
| OBS Studio |
As receiver via srt://listen |
| FFmpeg |
libsrt option when compiling |
Solution Architecture
Integration of SRT streaming includes configuring latency, passphrase, streamid, and connection mode. For iOS we use HaishinKit, for Android - rtmp-rtsp-stream-client-java or FFmpegKit. Connection statistics are collected via API and displayed in UI for quality control. When network degrades, bitrate is automatically lowered and buffer increased.
What's Included in the Work
- Requirements analysis and library selection.
- Integration of SRT streaming on iOS and/or Android.
- Configuration of
latency, passphrase, streamid parameters.
- Implementation of quality monitoring (RTT, loss, bitrate).
- Documentation and training for your team.
- Post-launch support.
Timeline: integration on one platform — 2–4 business days. Cost starts from $500 per platform. To evaluate your project, contact us — we'll respond within 1 business day. We have successfully completed 20+ projects with guaranteed performance metrics.
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).
| Protocol |
Latency |
Application |
| RTMP |
2–5 sec |
Streaming to YouTube/Twitch |
| HLS |
6–30 sec |
VOD, broadcast |
| DASH |
6–30 sec |
VOD with adaptive bitrate |
| WebRTC |
< 500 ms |
Video calls, P2P |
| SRT |
1–4 sec |
Professional streaming |
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
- Define requirements: target latency, number of concurrent users, need for P2P.
- Choose protocol and stack: WebRTC for video calls, RTMP/HLSLive for broadcasting.
- Set up signaling (SIP, WebSocket, MQTT) and TURN server.
- Implement publishing/viewing via native API or cross-platform plugin.
- Test on real devices with different cameras and network conditions.
- Optimize bitrate and resolution based on bandwidth.
Typical Mistakes in Media Feature Development
- Configuring AVFoundation session on the main thread.
- Missing AudioFocus Loss handling on Android.
- Ignoring
MediaCodec limitations on cheap devices.
- Using emulator for camera tests — emulator does not replicate HAL issues.
- Memory leaks when recreating media players without a pool.
What is Included in the Work
| Deliverable |
Description |
| Requirements analysis |
Stack selection, priorities, test devices |
| Design |
Architecture, data flow diagrams, API selection |
| Implementation |
Code using chosen tools |
| Backend integration |
GraphQL/REST, DRM, WebRTC signaling |
| Testing |
On real devices (at least 5 models) |
| Documentation |
API documentation, build instructions |
| Post-release support |
1 month incident support, team training |
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.
