Native SIP clients in production
IP telephony apps on PJSIP for iOS and Android. Registration, INVITE, REFER, DTMF, attended transfer, conferences, and NAT traversal with ICE, TURN, and STUN. Integration with Asterisk, FreeSWITCH, and Kamailio PBXs.
- PJSIP
- sip.js
- Asterisk
- Kamailio
- Opus
Native WebRTC for voice and video
PeerConnection on the official WebRTC SDK for iOS and Android. SDP negotiation, ICE restart, simulcast, and SVC with Opus, H.264, VP8, and VP9 codecs. In-house or managed SFUs (mediasoup, Janus, LiveKit) for group calls.
- WebRTC
- mediasoup
- LiveKit
- H.264
- VP9
CallKit and ConnectionService integrated
Calls that appear on the lock screen, in the system history, and on Bluetooth car kits. CallKit and PushKit on iOS, ConnectionService and Telecom on Android. Background calls, push-to-call, and answering from AirPods or the watch.
- CallKit
- PushKit
- ConnectionService
- Telecom
Audio routing and network resilience
AVAudioSession and AudioManager configured for real calls: headset switching, interruption handling, hands-free mode, echo, and noise suppression. Adaptive jitter buffer, FEC, PLC, and transparent reconnection over unstable mobile networks.
- AVAudioSession
- AudioManager
- AEC
- Jitter buffer
Real calls with production-grade metrics.