Engineering Deep Dive
Under the Hood
A detailed look at how Meow delivers native performance, crystal-clear audio, and rock-solid SIP connectivity, all in a lightweight package.
Layered Architecture
Five cleanly separated layers with one-way dependencies. Each layer only talks to the one directly below it.
┌───────────────────────────────────────────────────┐
│ QML UI Layer │
│ Main.qml · Views · Components │
├───────────────────────────────────────────────────┤
│ QmlBridge (C++) │
│ Context properties · Signal/Slot │
├───────────────────────────────────────────────────┤
│ Core Controllers │
│ CallManager · SettingsManager · LicenseManager │
│ ContactModel · CallStateMachine · Database │
├───────────────────────────────────────────────────┤
│ SIP Engine │
│ PJSIP · PjAccount · PjCall · SRTP │
├───────────────────────────────────────────────────┤
│ Audio Backend │
│ PulseAudioDev (Linux) · WASAPI/WMME (Windows) │
└───────────────────────────────────────────────────┘Audio Pipeline
Custom audio backend built for VoIP. Not a generic media player repurposed for calls.
Custom PulseAudio Backend
Direct PulseAudio integration via dlopen(libpulse-simple.so.0). No dependency on system-compiled audio backends. Works everywhere PulseAudio runs.
VoIP-Optimized Buffers
80ms target playback latency, 20ms capture fragments. Tuned for voice clarity over raw throughput.
Audio Hot-plug
Background pactl subscribe process with 500ms debounce. Plug in a headset mid-call and it just works.
Conference Bridge
PJSIP conference bridge at 8000 Hz clock rate. Native G.711 match eliminates unnecessary resampling.
// PulseAudio buffer configuration
pa_buffer_attr playback_attrs = {
.maxlength = (uint32_t)-1,
.tlength = 80 * frame_size, // 80ms
.prebuf = (uint32_t)-1,
.minreq = (uint32_t)-1,
.fragsize = 0
};
pa_buffer_attr capture_attrs = {
.maxlength = (uint32_t)-1,
.tlength = 0,
.prebuf = 0,
.minreq = 0,
.fragsize = 20 * frame_size // 20ms
};
// Conference bridge — native G.711
epCfg.medConfig.clockRate = 8000;SIP & Call Management
State-machine-driven call handling with thread-safe PJSIP integration.
QStateMachine FSM
Single source of truth for call state. Transitions are atomic and thread-safe via Qt signal/slot.
Thread-Safe PJSIP
All PJSIP callbacks marshal to the Qt thread via QMetaObject::invokeMethod(Qt::QueuedConnection).
DTMF (RFC 2833)
In-band DTMF tone generation for IVR navigation. Compatible with all major PBX systems.
Contact Enrichment
Incoming calls are matched against contacts using phone number normalization with country code awareness.
Security & Licensing
Cryptographically sound licensing with hardware-bound activation.
Hardware Fingerprinting
SHA-256 composite of CPU model, RAM (rounded GB), first non-loopback MAC address, and machine-id. Unique but stable across reboots.
Ed25519 Signatures
License tokens are Ed25519-signed by the server. The client verifies offline with a hardcoded 32-byte public key. No network needed for validation.
Anti-Reset Protection
Server fuzzy-matches 3 of 4 hardware factor hashes. Reinstalling the OS or clearing local storage won't reset your trial. The server remembers.
Performance
Native code means native performance. No runtime overhead, no garbage collector pauses.
Memory Comparison (typical idle call)
Platform Support
Truly native on each platform. Not a cross-platform wrapper with platform quirks.
Linux
PrimaryWindows
SupportedStandards & Compatibility
Built on PJSIP, the most widely deployed open-source SIP stack. Standards-first, vendor-agnostic.
SIP: Session Initiation Protocol
RTP: Real-Time Transport Protocol
DTMF Tones via RTP Payloads
SRTP: Secure Real-Time Transport
SIP: Locating SIP Servers
SIP Outbound Connections
Tested PBX Systems
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