pub struct CreateSIPParticipantOptions {Show 17 fields
pub participant_identity: String,
pub participant_name: Option<String>,
pub participant_metadata: Option<String>,
pub participant_attributes: Option<HashMap<String, String>>,
pub display_name: Option<String>,
pub sip_number: Option<String>,
pub dtmf: Option<String>,
pub wait_until_answered: Option<bool>,
pub play_dialtone: Option<bool>,
pub hide_phone_number: Option<bool>,
pub ringing_timeout: Option<Duration>,
pub max_call_duration: Option<Duration>,
pub enable_krisp: Option<bool>,
pub headers: Option<HashMap<String, String>>,
pub include_headers: Option<SipHeaderOptions>,
pub media_encryption: Option<SipMediaEncryption>,
pub timeout: Option<Duration>,
}Fields§
§participant_identity: StringOptional identity of the participant in LiveKit room
participant_name: Option<String>Optionally set the name of the participant in a LiveKit room
participant_metadata: Option<String>Optionally set the free-form metadata of the participant in a LiveKit room
participant_attributes: Option<HashMap<String, String>>§display_name: Option<String>Optional custom caller ID shown to the callee. Requires SIP provider support. If unset, the phone number is used; set it to an empty string to trigger a CNAM lookup on providers that support it.
sip_number: Option<String>§dtmf: Option<String>Optionally send following DTMF digits (extension codes) when making a call. Character ‘w’ can be used to add a 0.5 sec delay.
wait_until_answered: Option<bool>Wait for the call to be answered before returning.
When true, the request blocks until the call is answered or fails,
and returns SIP error codes (e.g., 486 Busy, 603 Decline) on failure.
When false (default), returns immediately while the call is still dialing.
play_dialtone: Option<bool>Optionally play dialtone in the room as an audible indicator for existing participants
hide_phone_number: Option<bool>§ringing_timeout: Option<Duration>§max_call_duration: Option<Duration>§enable_krisp: Option<bool>§headers: Option<HashMap<String, String>>SIP headers sent as-is on the INVITE; may help the SIP endpoint identify the call as coming from LiveKit.
include_headers: Option<SipHeaderOptions>Which SIP response headers to map to sip.h.* participant attributes.
media_encryption: Option<SipMediaEncryption>Media encryption policy for the call.
timeout: Option<Duration>Per-request timeout override. Defaults to a longer value when
wait_until_answered is set (dialing takes time), otherwise the client
default. Raised, if needed, to stay above ringing_timeout.
Trait Implementations§
Source§impl Clone for CreateSIPParticipantOptions
impl Clone for CreateSIPParticipantOptions
Source§fn clone(&self) -> CreateSIPParticipantOptions
fn clone(&self) -> CreateSIPParticipantOptions
1.0.0 (const: unstable) · Source§fn clone_from(&mut self, source: &Self)
fn clone_from(&mut self, source: &Self)
source. Read more