Imagine this: a visitor presses the intercom button, and your phone instantly rings, showing the camera video. With one tap, you open the door. Behind this simplicity lies a complex engineering puzzle: WebRTC or SIP stack, ONVIF protocol, push notifications with sub-second latency even when the app is closed, lock relay control via the device API, and event recording. We develop mobile apps for intercoms on a turnkey basis, from architecture selection to publishing on the App Store and Google Play. A typical project takes 4 to 12 weeks and costs between $15,000 and $50,000, depending on hardware compatibility and multi-apartment requirements.
Consider a typical scenario: in an apartment building, a single app for residents routes a call from the intercom of a specific entrance to the correct apartment. It must also support video recording, guest QR codes, and integration with the existing access control system. Our team has 10+ years of experience in embedded and mobile solutions, so we guarantee tight deadlines and transparent reporting.
Why Protocol Choice Matters
The first question in design is the type of intercom hardware. It determines which video communication protocol to use.
Ready-made IP Intercoms with SIP — Mobile App Development
Devices like Hikvision DS-KV6113, Grandstream GDS3710, Beward DS06M support SIP and ONVIF. The phone registers as a SIP client on Asterisk/FreeSWITCH. Incoming call from intercom → SIP INVITE → server → push to phone → CallKit (iOS) / ConnectionService (Android).
Custom Intercom on a Single-Board Computer
We use Raspberry Pi, ESP32-S3, or NXP i.MX — we choose the stack ourselves. WebRTC via Pion (Go) or aiortc (Python) on the device, signaling via WebSocket to our server, then the mobile client.
Cloud Intercom
Proprietary solutions (Ring, Dahua, Hikvision EZVIZ) provide P2P SDKs. Integration is fast but leads to vendor lock-in.
How to Ensure Call Delivery in Fractions of a Second
Notification delay is critical: a guest is at the door. On iOS, the right path is APNs VoIP push with PKPushRegistry and PKPushType.voIP. The system wakes the app immediately, even from Force Quit, and shows the native call interface via CXProvider. According to CallKit documentation, this is the standard approach for VoIP apps.
let update = CXCallUpdate()
update.remoteHandle = CXHandle(type: .generic, value: "Door entrance")
update.hasVideo = true
update.localizedCallerName = "Intercom"
provider.reportNewIncomingCall(with: callUUID, update: update) { error in ... }
The APNs VoIP certificate is separate from the push certificate. Don't forget voip in UIBackgroundModes in Info.plist.
On Android, we use ConnectionService + FCM with high priority (priority: high). TelecomManager.addNewIncomingCall() shows the system call. Problem: on Xiaomi, Huawei, OPPO aggressive battery optimizations kill the FCM connection. Solution: JobScheduler keepalive + instruct the user to add the app to exceptions. For Huawei, we use HMS Push.
For cross-platform projects (Flutter), we use flutter_callkit_incoming and firebase_messaging — native bindings to CallKit and ConnectionService.
WebRTC Video Communication
After answering the call, a WebRTC peer connection is established. Signaling: SDP exchange via our WebSocket server (or Asterisk with WebSocket transport for SIP/WebRTC). ICE candidates are gathered via STUN; for NAT traversal, we use TURN (Coturn). Video from the intercom camera is rendered in RTCMTLVideoView (Metal, iOS) or SurfaceViewRenderer (Android). Audio — RTCAudioTrack. AEC (echo cancellation) is built into WebRTC — critical to prevent echo through the intercom speaker. Video latency: 150-400 ms on Wi-Fi, 400-800 ms on 4G — acceptable for deciding to open or not.
Protocol comparison: SIP stack is suitable for standard IP intercoms, but WebRTC provides 3-5 times lower video latency due to direct P2P connection.
Lock Control
HTTP or MQTT request to the device or server: POST /api/unlock or publish("home/door/unlock", "1"). Confirmation of opening via door sensor (optional): the OPEN event is displayed in the app. The "Open" button is active only during the call — after ending, it disappears. Event log: each call, opening, and rejection is written to the database with timestamp and userId.
Video Event Recording
Video recording for each call (ringback recording): a media server (Janus record plugin, Ant Media) writes the stream to WebM/MP4. Storage in S3/MinIO with retention of the last 30 events or 7 days (lifecycle rule). The mobile client shows history: a timeline with preview of the first frame, duration, and a play button using AVPlayer / ExoPlayer from S3 presigned URL.
Multi-Apartment Building
Multiple entrances, multiple apartments. Routing: the intercom at entrance 3 calls only residents of apartments in entrance 3. In Asterisk: dialplan with Dial(SIP/apartment_${EXTEN}). Each apartment is a separate SIP account. In the app, the user is linked to their apartment account. Guest access: the owner issues a temporary QR code with a limited validity period.
How routing works in a multi-apartment building
Each entrance intercom has a unique SIP number. When the call button is pressed, the intercom sends an INVITE with the apartment number (DTMF). Asterisk converts the DTMF to the apartment's SIP account number. If the call is not answered, it is forwarded to the mobile app via VoIP push. On failure, video is recorded and a notification is sent.
Step-by-Step Intercom Integration Process
- Hardware audit — analysis of intercom protocols, ONVIF/SIP support, relay output capabilities.
- Architecture design — stack selection (SIP vs WebRTC), server side, DBMS.
- Server-side development — Asterisk/WebRTC server, REST API for the app, MQTT broker.
- Mobile client development — iOS (Swift + CallKit) and Android (Kotlin + ConnectionService) with video, lock control, history.
- Integration and testing — test bench with a real intercom, latency checks, error scenarios.
- Publishing to App Store and Google Play — provisioning profiles, code signing, push certificates.
Development Phases
| Phase |
Content |
Timeline |
| Hardware audit and architecture |
Device protocols, stack selection |
3-5 days |
| Server side |
Asterisk/WebRTC server, API |
1-2 weeks |
| Mobile client iOS + Android |
CallKit, video, lock control |
2-3 weeks |
| Event recording and history |
S3, player, log |
1 week |
| Testing on hardware |
QA on real intercom |
1 week |
Total from 1 to 3 months depending on complexity and multi-apartment requirements.
Hardware Comparison
| Type |
Protocol |
Complexity |
Video Latency |
Vendor Lock-in |
| SIP intercom |
SIP + ONVIF |
Medium |
200-500 ms |
No |
| Custom (RPi) |
WebRTC |
High |
150-400 ms |
No |
| Cloud (Ring/Dahua) |
P2P SDK |
Low |
300-600 ms |
Yes |
What's Included in the Work
Upon project completion, you receive:
- Source code for the mobile app (iOS/Android) with comments
- Built IPA/APK and publishing configurations
- Server side (if required) in Docker containers
- API documentation and deployment instructions
- 1 month of technical support after delivery
- Code in a private Git repository
If you want to evaluate your project, contact us — we will prepare a preliminary architecture and estimate within 2 business 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).
| 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.
