Struct MockPhone 

pub struct MockPhone { /* private fields */ }

Implementations

impl MockPhone

pub fn new(cmd_tx: Sender<(String, String)>) -> Self

Trait Implementations

impl Phone for MockPhone

fn register(&self, _aor: &str, regint: u32)

fn dial(&self, number: &str)

fn hangup(&self)

fn hangup_all(&self)

fn accept(&self)

fn hold(&self)

fn resume(&self)

fn mute(&self)

fn silence_alert(&self)

Stop the alert tone that is playing right now, without touching the call it belongs to. The caller keeps hearing ringback and the call can still be answered — this silences your side only, and only until the next alert. Default: no-op (mocks).

fn send_dtmf(&self, digit: char)

fn switch_line(&self, line: usize)

fn transfer(&self, uri: &str)

fn attended_transfer_start(&self, uri: &str)

fn attended_transfer_exec(&self)

fn attended_transfer_abort(&self)

fn add_header(&self, key: &str, value: &str)

fn rm_header(&self, key: &str)

fn set_audio_source(&self, spec: &str)

Switch the audio source at runtime, e.g. ausine,440 to send a tone, aufile,/path.wav to play a file, or aubridge,default to go silent. Applies to the active call.

fn arm_invite_response(&self, scode: u16, reason: &str, headers: Vec<String>)

Arm a custom SIP response for the next inbound INVITE(s): answer with scode/reason and the extra headers (each a full header line like Contact: <sip:…>, no trailing CRLF) instead of accepting the call. Sticky until Phone::disarm_invite_response. Arm it before the call arrives for deterministic behaviour.

fn disarm_invite_response(&self)

Clear any armed response — subsequent inbound INVITEs are accepted again.

fn select_call(&self, call_id: &str)

Make the call with SIP call_id the UA’s current call (baresip call_set_current), so subsequent hold/resume/hangup/mute target it. No-op if the id isn’t found. Default: no-op (mocks).

fn media_stats(&self) -> Option<MediaStats>

RTP media stats (jitter/loss/RTT + estimated MOS) for the active call, or the last finished call’s snapshot. None if unavailable (no call yet, or before the first RTCP report). Default: None (mocks).

fn audio_codec(&self) -> Option<CodecInfo>

The negotiated audio codec on the active call. None if there’s no call or it isn’t negotiated yet. Default: None (mocks).

fn received_dtmf(&self) -> String

DTMF digits received on the active/last call so far, in order (e.g. "1234#"). Default: empty (mocks).

fn inbound_headers(&self, call_id: &str) -> Vec<(String, String)>

All SIP headers received on the INVITE of the call with call_id, in order, consuming them from the store. Empty if none / unknown call. Default: empty (mocks).

fn deflect_incoming(&self, contact: &str, diversion: Option<&str>)

Deflect inbound calls to contact with a 302 Moved Temporarily (plus an RFC 5806 Diversion header when diversion is set). Thin wrapper over Phone::arm_invite_response.