Table of Contents

VoipClient

VoipClient is the single public entry point. It owns the SIP stack, the media pipeline and the audio devices, and exposes everything else through typed sub-facades. It implements IDisposable — construct one per application (or per tenant) and dispose it on shutdown.

using var client = new VoipClient(new VoipConfiguration { /* … */ });

Sub-facades

Property Type Responsibility
Lines PhoneLineManager Registration and the registered lines
Calls CallManager The active calls
Media MediaManager Receivers, senders, connectors, recording/playback
Modules ModuleRegistry The module extension point (Get<T>/TryGet<T>)
DeviceManager DeviceManager Audio device enumeration and runtime control
QualityManager QualityManager RTCP quality snapshots
PolicyManager PolicyManager Runtime policy (e.g. inbound acceptance)
SessionManager SessionManager Lifecycle of underlying sessions
TelemetryManager TelemetryManager Diagnostic sinks/traces
RecordingManager / PlaybackManager module facades Convenience recording/playback

Convenience methods

The client also carries the high-level verbs most apps need directly:

  • ConnectAsync(SipAccount) — register and return the line
  • DialAndWaitUntilConnectedAsync(line, uri) — dial and await answer
  • AttachDefaultAudioAsync(call) / DetachDefaultAudioAsync(call)
  • OnIncomingCall(handler) — subscribe to inbound calls with an IDisposable
  • Audio device control: GetAvailableInputAudioDevices(), SwitchAudioInputDevice(id), SetAudioInputVolume(v), SetAudioInputMuted(b), UpdateAudioFormat(format) (and the matching output variants) — see Audio devices

Configuration

VoipConfiguration is an immutable (init-only) options record:

Property Default Meaning
UserAgent "CalloraVoipSdk/1.0" SIP User-Agent header
LoggerFactory null ILoggerFactory for diagnostics
DefaultTransport Udp Default outbound SIP transport: UDP / TCP / TLS / WS / WSS
Tls null TLS settings for the TLS/WSS transports
SrtpPolicy Optional Disabled / Optional / Required — see SRTP/SRTCP
OfferDtlsSrtp false Offer DTLS-SRTP keying (RFC 5763) instead of SDES
RequireSecureSignalingForSdes false Refuse SDES keying over insecure signalling (UDP/TCP/WS)
DtlsCertificate null Certificate for the DTLS-SRTP handshake; self-signed when unset
PreferredAudioCodecs null Ordered codec preference for offers/answers
BridgeAudioFormat Passthrough Wire format when bridging two calls (Passthrough or Pcmu)
EnableVideo false Negotiate a video m-line (transport-only)
PreferredVideoCodecs null Ordered video codec preference
AudioDevice SilenceAudioDevice Default audio device when none is attached
EnableAutomaticAudioDeviceSelection true Load the platform audio package by reflection
MaxConcurrentCallsPerLine 10 Guard against runaway call fan-out
InboundMediaTimeout 15 s Drop inbound calls with no media
HangupHeldCallOnMediaSilence false Also apply the media timeout to held calls
Ice IceConfiguration Full ICE (RFC 8445/7675) — opt-in, off by default, unproven in production trunks
Services null IServiceProvider used to resolve ports (DI composition)

Lifecycle

Dispose() unregisters lines and makes a best-effort BYE for still-active calls. Details and graceful-shutdown guidance: Lifecycle & dispose.

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