Expand description
A SIP telephony library with an event-driven API.
Provides registration lifecycle, call state management, media pipeline, and mock types for testing consumer code without a real SIP transport.
Re-exports§
pub use call::Call;pub use call::VideoUpgradeRequest;pub use config::Config;pub use config::DialOptions;pub use config::DtmfMode;pub use config::PhoneBuilder;pub use error::Error;pub use error::Result;pub use phone::Phone;pub use sip::conn::TlsConfig;pub use subscription::SubId;pub use types::CallState;pub use types::Codec;pub use types::Direction;pub use types::EndReason;pub use types::ExtensionState;pub use types::ExtensionStatus;pub use types::NotifyEvent;pub use types::PhoneState;pub use types::SipMessage;pub use types::SubState;pub use types::VideoCodec;pub use types::VideoFrame;pub use types::VoicemailStatus;
Modules§
- call
- codec
- config
- dialog
- dialog_
info - Parser for
application/dialog-info+xml(RFC 4235). - dtmf
- error
- ice
- ICE-Lite implementation (RFC 8445 §2.2).
- jitter
- media
- mock
- mwi
- MWI (Message Waiting Indicator) via SIP SUBSCRIBE/NOTIFY (RFC 3842).
- phone
- registry
- rtcp
- Basic RTCP (RFC 3550) — Sender/Receiver Reports for trunk compatibility.
- sdp
- sip
- srtp
- SRTP (Secure RTP) implementation per RFC 3711 with SDES key exchange (RFC 4568).
- stun
- STUN Binding client (RFC 5389) and shared STUN primitives.
- subscription
- Generic SUBSCRIBE/NOTIFY subscription manager (RFC 6665).
- transport
- turn
- TURN client (RFC 5766).
- types
- video