Schemas

Last updated August 14, 2026

AMDParams

Field	Type		Description
`initial_silence_timeout`	integer	optional	Max milliseconds of silence before declaring no_speech `2500`
`greeting_duration`	integer	optional	Speech duration threshold (ms) above which answerer is classified as machine `1500`
`after_greeting_silence`	integer	optional	Silence duration (ms) after initial speech to declare human `800`
`total_analysis_time`	integer	optional	Max analysis window in milliseconds. A threshold longer than this window suppresses that verdict; a window shorter than all of initial_silence_timeout, greeting_duration and after_greeting_silence is rejected, since the call could only end not_sure. `5000`
`minimum_word_length`	integer	optional	Minimum speech burst duration (ms) to count as a word `100`
`beep_timeout`	integer	optional	Max time (ms) to wait for the voicemail beep after machine detection. 0 or omitted = disabled. `0`

AddLegRequest

Field	Type		Description
`leg_id`	string	required	ID of the leg to add
`mute`	boolean	optional	If set, apply this mute state to the leg atomically before it joins the mixer (no race where un-muted audio enters the mix). Omit to leave current state untouched (useful when moving between rooms).
`deaf`	boolean	optional	If set, apply this deaf state to the leg atomically before it joins the mixer. Omit to leave current state untouched.
`accept_dtmf`	boolean	optional	If set, control whether this leg receives DTMF digits broadcast from other legs in the same room. Omit to leave current state untouched (default for new legs is true).
`role`	string	optional	If set, apply this routing role to the leg atomically before it joins the mixer. The room's routing matrix (see PUT /v1/rooms/{id}/routing) decides which other legs this leg hears and is heard by based on roles. Pass "" to clear the role (full mesh). Omit to leave the current role untouched.
`streams`	array[object]	optional	Additional audio streams of the leg to mix into this room, each with its own routing role. Omit to add only the leg's primary stream. A stream already mixed elsewhere is moved here.

AddLegStreamRequest

Field	Type		Description
`direction`	enum	optional	Media direction for the new stream, from this server's point of view. Defaults to sendrecv. `"sendrecv"` Values: `sendrecv`, `sendonly`, `recvonly`, `inactive`
`lang`	string	optional	BCP 47 language tag advertised as a=lang (RFC 8866), e.g. "es-ES" for a Spanish translation feed.
`content`	enum	optional	Value advertised as a=content (RFC 4796). Use "main" for original audio and "alt" for an alternative feed such as a translation. Values: `main`, `alt`, `speaker`, `slides`, `sl`
`label`	string	optional	Value advertised as a=label (RFC 4574), for correlating the stream with external metadata.
`room_id`	string	optional	If set, attach the new stream to this room once it is negotiated.
`role`	string	optional	Routing role to apply when room_id is set.

AddRoomStream

Field	Type		Description
`stream_id`	string	required	Stream identifier from GET /v1/legs/{id}/streams. The primary stream is not addressable here — it joins with the leg itself.
`role`	string	optional	Routing role for this stream inside the room.

AgentMessageRequest

Field	Type		Description
`message`	string	required	Context or instruction to inject into the running agent session

AnswerLegRequest

Field	Type		Description
`speech_detection`	boolean	optional	If true, emit speaking.started and speaking.stopped events for this leg. If false, suppress them. Omit to use the server default (SPEECH_DETECTION_ENABLED env var, default false).
`codec`	enum	optional	Explicit codec for the answer SDP. Must appear in the remote offer's offered_codecs list. Omit to use the server's default preference order. Values: `PCMU`, `PCMA`, `G722`, `opus`, `AMR-WB`, `AMR-NB`
`streams`	array[object]	optional	Rooms for the caller's additional audio streams, applied once the answer is negotiated. Positional: entry i addresses the i-th accepted stream beyond the primary, in m-line order — the caller's offer decides how many exist, so an entry with no matching stream is ignored. Use POST /v1/legs/{id}/streams/{streamId}/room to re-route a stream later.

AnswerLegStream

Field	Type		Description
`room_id`	string	optional	Room to mix this stream into once the answer is negotiated. May differ from the room the leg itself joins.
`role`	string	optional	Routing role for this stream inside its room.

AttachStreamRoomRequest

Field	Type		Description
`room_id`	string	required	Room to mix this stream into. May differ from the leg's own room.
`role`	string	optional	Routing role for the stream inside that room. The room's routing matrix decides who hears it.

BridgeView

Field	Type		Description
`id`	string	required	Bridge identifier
`room_id`	string	required	The peer room joined to the room in the path
`direction`	enum	required	Audio flow relative to the room in the path: bidirectional, send, receive, or none. Values: `bidirectional`, `send`, `receive`, `none`
`sample_rate`	integer	required	Shared mixer sample rate in Hz (both rooms must match).

ChallengeRequest

Field	Type
`realm`	string	required
`username`	string	optional
`password`	string	optional
`ha1`	string	optional
`algorithm`	string	optional
`qop`	array[string]	optional
`max_expires`	integer	optional

CreateLegRequest

Field	Type		Description
`type`	enum	required	Leg type Values: `sip`, `whatsapp`, `websocket`, `livekit_room`
`to`	string	optional	Destination. For sip legs, a SIP URI (e.g. "sip:[email protected]"). For whatsapp legs, an E.164 phone number (with or without '+').
`uri`	string	optional	Deprecated alias for `to` (sip legs only). Prefer `to`.
`from`	string	optional	Caller ID. A bare user-part (e.g. "+15551234567", "alice") sets the user of the SIP From header. A full SIP URI (e.g. "sip:[email protected]") sets both the user and the host; otherwise the host comes from the matched trunk's AOR realm, falling back to SIP_DOMAIN.
`privacy`	string	optional	SIP Privacy header value (e.g. "id", "none")
`ring_timeout`	integer	optional	Seconds to wait for answer; 0 = no timeout `0`
`max_duration`	integer	optional	Maximum call duration in seconds after connect. Automatically hung up when reached. 0 or omitted = no limit. `0`
`codecs`	array[enum]	optional	Codec preference order (sip legs only)
`headers`	object	optional	Custom headers to include in the outbound INVITE (sip/whatsapp) or the WebSocket upgrade request (websocket)
`room_id`	string	optional	Room ID to auto-add the leg to once media is ready (early_media or connected). If the room does not exist, it is automatically created.
`auth`	any	optional	Digest auth credentials. Required for whatsapp legs (Meta-issued password; username defaults to `from` with '+' stripped). Optional for sip legs (sipgo retries on 401/407 challenge).
`webhook_url`	string(uri)	optional	Route all events for this leg exclusively to this URL instead of global webhooks.
`webhook_secret`	string	optional	HMAC-SHA256 signing secret for the per-leg webhook.
`amd`	any	optional	Enable Answering Machine Detection on outbound calls. Include the object (even empty) to enable with defaults; omit to disable.
`accept_dtmf`	boolean	optional	If false, this leg will not receive DTMF digits broadcast from other legs in the same room. Defaults to true. `true`
`app_id`	string	optional	Application identifier. Carried through to all events for this leg. Use to filter the WebSocket event stream by app.
`speech_detection`	boolean	optional	If true, emit speaking.started and speaking.stopped events for this leg. If false, suppress them. Omit to use the server default (SPEECH_DETECTION_ENABLED env var, default false).
`rtt`	boolean	optional	For sip legs: offer Real-Time Text (ITU-T T.140 over RTP per RFC 4103) alongside audio. For websocket legs: enable the bidirectional text-message channel. Default: false. `false`
`streams`	array[object]	optional	SIP outbound only. Extra m=audio sections to offer alongside the call's primary bidirectional audio, so a multi-stream call is established by the first INVITE instead of a follow-up re-INVITE. Each entry binds its own RTP port and may be mixed into its own room. To add a stream to a call that is already up, use POST /v1/legs/{id}/streams instead.
`url`	string(uri)	optional	WebSocket target URL (ws:// or wss://) for outbound websocket legs. Required when type=websocket.
`sample_rate`	enum	optional	PCM sample rate for websocket legs. The room's mixer automatically resamples between this and the room rate. `16000` Values: `8000`, `16000`, `24000`, `48000`
`wire_format`	enum	optional	Audio framing for websocket legs. `binary` ships raw PCM as WebSocket binary frames; `json_base64` wraps PCM as `{"type":"audio","audio":"<base64>"}` text frames (browser-friendly). `"binary"` Values: `binary`, `json_base64`
`sample_format`	enum	optional	On-the-wire PCM sample encoding for websocket legs. v1 only supports `s16le`. `"s16le"` Values: `s16le`
`livekit`	any	optional	LiveKit room join parameters (only used when type=livekit_room).

CreateLegStream

Field	Type		Description
`direction`	enum	optional	Media direction for this stream, from this server's point of view. Defaults to sendrecv. `"sendrecv"` Values: `sendrecv`, `sendonly`, `recvonly`, `inactive`
`lang`	string	optional	BCP 47 language tag advertised as a=lang (RFC 8866), e.g. "es-ES" for a Spanish translation feed.
`content`	enum	optional	Value advertised as a=content (RFC 4796). Use "alt" for an alternative feed such as a translation. Values: `main`, `alt`, `speaker`, `slides`, `sl`
`label`	string	optional	Value advertised as a=label (RFC 4574), for correlating the stream with external metadata.
`room_id`	string	optional	Room to mix this stream into once the call connects. May differ from the leg's own room_id, which governs the primary stream.
`role`	string	optional	Routing role for this stream inside its room.

CreateRoomBridgeRequest

Field	Type		Description
`id`	string	optional	Custom bridge ID (auto-generated UUID if omitted)
`room_id`	string	required	The other room to join. Must use the same sample rate as the room in the path.
`direction`	enum	optional	Audio flow relative to the room in the path: bidirectional (both hear each other), send (path room → other only), receive (other → path room only), none (allocated but silent). Default: bidirectional. `"bidirectional"` Values: `bidirectional`, `send`, `receive`, `none`

CreateTrunkRequest

Field	Type		Description
`type`	enum	required	Trunk type discriminator. Only `sip_register` is implemented today; `ip_ip` is reserved and returns 501. Values: `sip_register`, `ip_ip`
`app_id`	string	optional	Application identifier carried through to every event emitted by this trunk.
`sip_register`	any	optional	Required when type == "sip_register". Configures the outbound REGISTER (registrar URI, AOR, digest credentials, expiry).
`ip_ip`	any	optional	Reserved for static-IP peering (no REGISTER). Not yet implemented; supplying this returns 501.

CreateTrunkResponse

Field	Type
`id`	string	required
`type`	string	required
`status`	string	required

DTMFRequest

Field	Type		Description
`digits`	string	required	DTMF digits to send (0-9, *, #)

DeepgramAgentRequest

Field	Type		Description
`settings`	object	optional	Full Deepgram agent settings object (agent.listen, agent.think, agent.speak, etc.). When omitted, sensible defaults are used (nova-3 STT, gpt-4o-mini LLM, aura-2-asteria-en TTS).
`greeting`	string	optional	Agent greeting message
`language`	string	optional	Language code (e.g. "en", "es")
`api_key`	string	optional	API key override (falls back to DEEPGRAM_API_KEY env var)

DeleteLegRequest

Field	Type		Description
`reason`	enum	optional	Disconnect reason. Only honored for unanswered SIP inbound legs (state `ringing` or `early_media`); on connected legs the body is ignored and the leg is hung up with the legacy `api_hangup` reason. The value flows through to `leg.disconnected`'s `cdr.reason` and selects the SIP final response: `busy`→486, `declined`/`rejected`→603, `unavailable`→480, `not_found`→404, `forbidden`→403, `server_error`→500. Values: `busy`, `declined`, `rejected`, `unavailable`, `not_found`, `forbidden`, `server_error`

EarlyMediaLegRequest

Field	Type		Description
`codec`	enum	optional	Explicit codec for the 183 Session Progress SDP. Must appear in the remote offer's offered_codecs list. Omit to use the server's default preference order. Values: `PCMU`, `PCMA`, `G722`, `opus`, `AMR-WB`, `AMR-NB`

ElevenLabsAgentRequest

Field	Type		Description
`agent_id`	string	required	ElevenLabs agent ID
`first_message`	string	optional	Override the agent's first message
`language`	string	optional	Language code (e.g. "en", "es")
`dynamic_variables`	object	optional	Key-value pairs passed to the agent as dynamic variables
`api_key`	string	optional	API key override (falls back to ELEVENLABS_API_KEY env var)

Error

Field	Type		Description
`instance_id`	string	optional	Instance identifier
`error`	string	required	Error message

ICECandidateInit

Field	Type		Description
`candidate`	string	required	ICE candidate string
`sdpMid`	string	optional	Media stream identification tag
`sdpMLineIndex`	integer	optional	Index of the media description

IPIPTrunkSpec

Field	Type		Description
`peer_uri`	string	optional	Static peer SIP URI for IP-IP peering. Reserved; not yet implemented.

IPIPTrunkView

Field	Type		Description
`peer_uri`	string	optional

Leg

Field	Type		Description
`instance_id`	string	optional	Instance identifier
`id`	string	required	Unique leg identifier (UUID)
`type`	enum	required	Leg type Values: `sip_inbound`, `sip_outbound`, `webrtc`, `whatsapp_in`, `whatsapp_out`, `websocket_in`, `websocket_out`, `moq_in`, `livekit_publish`, `livekit_participant`
`state`	enum	required	Leg state Values: `ringing`, `early_media`, `connected`, `held`, `hung_up`
`room_id`	string	optional	Room ID if the leg is in a room, empty otherwise
`muted`	boolean	required	Whether the leg is muted (cannot be heard by others)
`deaf`	boolean	required	Whether the leg is deaf (cannot hear others)
`accept_dtmf`	boolean	required	Whether the leg receives DTMF digits broadcast from other legs in the same room. Defaults to true.
`held`	boolean	required	Whether the call is on hold (SIP legs only)
`role`	string	optional	Routing role used by the room's audio routing matrix (e.g. "customer", "agent", "supervisor"). Empty string means unroled (full mesh).
`app_id`	string	optional	Application identifier for event stream filtering.
`sip_headers`	object	optional	Deprecated: X-* headers from the inbound INVITE. Only present on sip_inbound legs. Use `headers` for new code; it carries the same map plus surfaces handshake headers for websocket legs.
`headers`	object	optional	Custom protocol headers exposed by the leg's transport — X-/P- headers from a SIP INVITE, the WebSocket upgrade request, or supplied at outbound dial time.

LegStreamView

Field	Type		Description
`id`	string	required	Stream identifier, stable for the life of the dialog. The primary stream is always "0".
`mid`	string	optional	The stream's SDP a=mid token (RFC 5888), used to correlate it across offer/answer.
`index`	integer	required	Position of the stream's m= line in the SDP. Fixed for the life of the dialog (RFC 3264 §8).
`primary`	boolean	required	True for the call's main bidirectional audio stream, which cannot be removed or attached to a room independently.
`state`	enum	required	Negotiation state. Values: `pending`, `active`, `removed`
`direction`	enum	required	Negotiated media direction from this server's point of view. Values: `sendrecv`, `sendonly`, `recvonly`, `inactive`
`desired_direction`	enum	optional	Direction requested by the application. Survives hold/unhold, unlike the negotiated direction. Values: `sendrecv`, `sendonly`, `recvonly`, `inactive`
`codec`	string	optional	Codec negotiated for this stream. Streams on one leg may use different codecs.
`sample_rate`	integer	optional	Native sample rate of the stream's codec, in Hz.
`local_port`	integer	optional	Local RTP port. Each stream binds its own port; a shared transport is undefined without BUNDLE (RFC 9143).
`remote_addr`	string	optional	Remote RTP address media is currently sent to.
`label`	string	optional	The stream's a=label value (RFC 4574), for correlating it with external metadata.
`content`	string	optional	The stream's a=content value (RFC 4796), e.g. "main" for original audio and "alt" for a translated feed.
`lang`	string	optional	The stream's a=lang value (RFC 8866): a BCP 47 language tag such as "en" or "es-ES".
`room_id`	string	optional	Room this stream's audio is mixed into. A secondary stream may sit in a different room than its leg.
`role`	string	optional	Routing role of this stream within its room. Streams carry their own role, independent of their leg's.

LiveKitParams

Field	Type		Description
`url`	string(uri)	optional	LiveKit server endpoint (wss://...). Overrides LIVEKIT_URL.
`token`	string	optional	Pre-signed LiveKit JWT. Mutually exclusive with `room`/`identity` (mint mode); if both are present the token wins.
`room`	string	optional	LiveKit room name. Required when minting (i.e. `token` is empty AND LIVEKIT_TOKEN_SIGNING_ENABLED=true).
`identity`	string	optional	LiveKit participant identity. Required when minting.
`participant_name`	string	optional	Display name for the participant; surfaces in LK Room UIs.
`permissions`	any	optional	LiveKit grant flags. Nil pointers default to publish=true, subscribe=true, data=false, admin=false.
`token_ttl`	string	optional	Go duration string (e.g. "30m", "6h"). Used only when minting. Defaults to LIVEKIT_DEFAULT_TOKEN_TTL (6h).
`opus_bitrate`	integer	optional	Override LIVEKIT_OPUS_BITRATE for this leg. 6000..510000.

LiveKitPermissions

Field	Type		Description
`can_publish`	boolean	optional	Allow publishing tracks. Default true.
`can_subscribe`	boolean	optional	Allow subscribing to remote tracks. Default true.
`can_publish_data`	boolean	optional	Allow publishing data channel messages. Default false (audio bridge does not use data).
`room_admin`	boolean	optional	Grant admin actions on the room (e.g., server-side MuteTrack of remote participants). Default false.

PipecatAgentRequest

Field	Type		Description
`websocket_url`	string(uri)	required	WebSocket URL of the Pipecat bot (e.g. ws://my-bot:8765)

PlaybackRequest

Field	Type		Description
`url`	string(uri)	required	URL of the audio file (mutually exclusive with tone)
`tone`	string	required	Built-in telephone tone name. Format: {country}_{type} or bare {type} (defaults to US). Types: ringback, busy, dial, congestion. Countries: us, gb, de, fr, au, jp, it, in, br, pl, ru. Examples: us_ringback, gb_busy, dial.
`mime_type`	string	required	MIME type (e.g. audio/wav). Required when using url.
`repeat`	integer	required	Number of times to repeat playback (url only) `0`
`volume`	integer	required	Volume adjustment in dB (-8 to 8) `0`

RTTRequest

Field	Type		Description
`text`	string	required	UTF-8 text to send. May be one or more characters and may include T.140 control codes (e.g. backspace U+0008, CR/LF).

RecordingRequest

Field	Type		Description
`storage`	enum	required	"file" (default) — local disk, "s3" — upload to S3 after recording stops, "gcs" — upload to Google Cloud Storage via the native GCS API (Application Default Credentials / Workload Identity) Values: `file`, `s3`, `gcs`
`multi_channel`	boolean	required	When true, record each participant to a separate mono WAV file in addition to the full mix. Only applies to room recordings. `false`
`s3_bucket`	string	required	S3 bucket name. Overrides S3_BUCKET env var. Required if env var is not set.
`s3_region`	string	required	AWS region. Overrides S3_REGION env var. Default us-east-1.
`s3_endpoint`	string	required	Custom S3 endpoint (MinIO, etc.). Overrides S3_ENDPOINT env var.
`s3_prefix`	string	required	Key prefix (e.g. recordings/). Overrides S3_PREFIX env var.
`s3_access_key`	string	required	AWS access key ID. Overrides default credential chain.
`s3_secret_key`	string	required	AWS secret access key. Must be set together with s3_access_key.
`gcs_bucket`	string	required	GCS bucket name. Overrides GCS_BUCKET env var. Required if env var is not set when storage=gcs.
`gcs_object_name_prefix`	string	required	Object name prefix (e.g. recordings or recordings/). Overrides GCS_OBJECT_NAME_PREFIX env var. A trailing slash is added automatically when missing.
`filename`	string	required	Optional output basename for the WAV file. A .wav suffix is added when missing. Must be a single path segment (no directories). Dots inside the name are preserved (only a trailing .wav is treated as the extension). Rejected with 409 if the file already exists or another recording is using the same name. When omitted, a timestamped name is generated.

RegistrationAcceptRequest

Field	Type		Description
`max_expires`	integer	optional

RegistrationRejectRequest

Field	Type		Description
`code`	integer	optional
`reason`	string	optional

RegistrationView

Field	Type
`aor`	string	required
`contact`	string	required
`socket`	string	required
`transport`	string	required
`user_agent`	string	optional
`call_id`	string	optional
`app_id`	string	optional
`created_at`	string	required
`last_refresh`	string	required
`expires_at`	string	required
`granted_expires_seconds`	integer	required

RegistrationsResponse

Field	Type		Description
`bindings`	array[object]	required

Room

Field	Type		Description
`instance_id`	string	optional	Instance identifier
`id`	string	required	Room identifier
`app_id`	string	optional	Application identifier for event stream filtering.
`sample_rate`	integer	required	Mixer sample rate in Hz (8000, 16000, or 48000).
`participants`	array[object]	required	Legs currently in this room

RoomCreateRequest

Field	Type		Description
`id`	string	required	Custom room ID (auto-generated UUID if omitted)
`webhook_url`	string(uri)	optional	Route all events for this room exclusively to this URL instead of global webhooks.
`webhook_secret`	string	optional	HMAC-SHA256 signing secret for the per-room webhook.
`app_id`	string	optional	Application identifier. Carried through to all events for this room. Use to filter the WebSocket event stream by app.
`sample_rate`	enum	optional	Mixer sample rate in Hz. Allowed values: 8000, 16000, 48000. Default: 16000. `16000` Values: `8000`, `16000`, `48000`

RoomRoutingRequest

Field	Type		Description
`matrix`	object	required	Listener-role → list of allowed source roles. Omitted listener roles default to full mesh. Empty list = hears nothing.

RoomRoutingUpdateRequest

Field	Type		Description
`updates`	array[object]	required	Per-listener-role row replacements applied as a single atomic update.

RoomRoutingView

Field	Type		Description
`matrix`	object	required	Listener-role → list of allowed source roles. Roles absent from the matrix default to full mesh.

RoutingRowUpdate

Field	Type		Description
`listener_role`	string	required	The role whose row is being replaced.
`sources`	array[string]	required	New list of allowed source roles for this listener role. Pass null to clear the row (full mesh).

SIPAuth

Field	Type		Description
`username`	string	optional	Digest auth username. Optional for whatsapp legs (defaults to `from` with '+' stripped, per Meta's spec).
`password`	string	required	Digest auth password.

SIPRECParticipantView

Field	Type		Description
`participant_id`	string	required	The participant_id attribute from the recording metadata document.
`aor`	string	optional	The participant's address of record, e.g. "sip:[email protected]".
`name`	string	optional	The participant's display name, when the metadata carries one.

SIPRECSessionView

Field	Type		Description
`leg_id`	string	required	Leg carrying the recording session.
`session_id`	string	optional	Communication session being recorded, from the metadata's sessionrecordingassoc.
`data_mode`	enum	optional	Data mode of the most recently applied metadata document (RFC 7865 §6.1). Values: `complete`, `partial`
`room_id`	string	optional	Room this session's streams were attached to, when SIPREC_ROOM_MODE placed them in one.
`participants`	array[object]	required	Every party currently recorded by this session.
`streams`	array[object]	required	Every negotiated media stream, joined to the participant it carries.
`metadata`	string	optional	The raw rs-metadata XML document as most recently received.

SIPRECStreamView

Field	Type		Description
`leg_stream_id`	string	required	Identifier of the leg stream carrying this recorded media, as used by /v1/legs/{id}/streams.
`mid`	string	optional	The stream's SDP a=mid token (RFC 5888).
`label`	string	optional	The stream's a=label value (RFC 4574). This is the key that binds the m= section to the recording metadata.
`direction`	enum	optional	Negotiated media direction. Always recvonly or inactive: a recording server never transmits. Values: `recvonly`, `inactive`
`codec`	string	optional	Codec negotiated for this stream.
`room_id`	string	optional	Room this stream's audio is mixed into, when it has been attached to one.
`role`	string	optional	Routing role of this stream within its room. Defaults to the participant's identity.
`participant_id`	string	optional	Participant whose audio arrives on this stream, from the metadata's participantstreamassoc send binding.
`participant_aor`	string	optional	Address of record of the participant sending on this stream.
`participant_name`	string	optional	Display name of the participant sending on this stream.

SIPRegisterTrunkSpec

Field	Type		Description
`registrar_uri`	string	required	Upstream registrar SIP URI (e.g. "sip:pbx.example.com:5060" or "sips:pbx.example.com:5061;transport=tls").
`aor`	string	required	Address-of-record this trunk REGISTERs (e.g. "sip:[email protected]"). Becomes the From URI on outbound REGISTER, and the From / P-Asserted-Identity host on outbound INVITEs placed `from` this AOR.
`username`	string	optional	Digest auth username. Defaults to the AOR user-part when empty.
`password`	string	required	Digest auth password. Required. Never returned in any response.
`contact_user`	string	optional	Override the user-part of the Contact header sent in REGISTER. Defaults to the AOR user-part.
`expires_seconds`	integer	optional	Requested registration lifetime in seconds. Clamped to [SIP_OUTBOUND_REGISTRATION_MIN_EXPIRES_SECONDS, SIP_OUTBOUND_REGISTRATION_MAX_EXPIRES_SECONDS]. Default: SIP_OUTBOUND_REGISTRATION_DEFAULT_EXPIRES_SECONDS (3600).

SIPRegisterTrunkView

Field	Type
`registrar_uri`	string	required
`aor`	string	required
`username`	string	optional
`contact_uri`	string	optional
`requested_expires_seconds`	integer	required
`granted_expires_seconds`	integer	optional
`last_registered_at`	string	optional
`next_refresh_at`	string	optional
`call_id`	string	optional
`cseq`	integer	optional
`source_address`	string	optional

STTRequest

Field	Type		Description
`language`	string	required	Language code (e.g. "en", "es")
`partial`	boolean	required	Emit partial (non-final) transcripts `false`
`provider`	enum	optional	STT provider: "elevenlabs" (default), "deepgram" (/v1/listen), "deepgram_flux" (/v2/listen, conversational turn detection) or "azure" Values: `elevenlabs`, `deepgram`, `deepgram_flux`, `azure`
`api_key`	string	optional	API key override (falls back to ELEVENLABS_API_KEY, DEEPGRAM_API_KEY or AZURE_SPEECH_KEY env var depending on provider)
`model`	string	optional	Provider-specific model. Deepgram: default "nova-3". Deepgram Flux: "flux-general-en" (default) or "flux-general-multi".
`keyterms`	array[string]	optional	Terms to boost recognition of (Deepgram and Deepgram Flux).
`endpointing`	integer	optional	Deepgram only: milliseconds of silence before a segment is finalized. 0 disables endpointing.
`utterance_end_ms`	integer	optional	Deepgram only: milliseconds of silence after which an stt.turn event with event=utterance_end is emitted. Deepgram requires interim results for this, which are requested automatically and still suppressed unless partial is true.
`eager_eot_threshold`	number	optional	Deepgram Flux only: end-of-turn confidence that fires an eager_end_of_turn stt.turn event, enabling speculative generation. Must be between 0.3 and 0.9. When unset, no eager_end_of_turn or turn_resumed events are emitted at all.
`eot_threshold`	number	optional	Deepgram Flux only: end-of-turn confidence required to close a turn. Deepgram default 0.7.
`eot_timeout_ms`	integer	optional	Deepgram Flux only: milliseconds of silence after which a turn is closed regardless of confidence. Deepgram default 5000.
`language_hints`	array[string]	optional	Deepgram Flux only: candidate language codes for the "flux-general-multi" model.

SetLegRoleRequest

Field	Type		Description
`role`	string	required	New routing role for the leg. The room's routing matrix decides which other legs this leg hears and is heard by based on roles. Pass an empty string to clear the role (full mesh).

StartSIPRECRequest

Field	Type		Description
`srs_uri`	string	required	SIP URI of the session recording server, e.g. "sip:[email protected]:5060". A recording session carries the metadata document alongside the SDP and exceeds the UDP message limit, so the target should accept TCP.
`leg_ids`	array[string]	optional	Which participants to record. Each entry is either a leg ID (that leg's own audio) or "<legID>#<streamID>" for one of a leg's secondary audio streams mixed into the room. Empty or absent records every participant. An entry that is not in the room is a 404.
`session_id`	string	optional	Communication session identifier put in the recording metadata. Defaults to the room ID.
`app_id`	string	optional	Application identifier tagged onto the resulting leg and its events.
`auth_username`	string	optional	SIP digest username, when the recording server challenges the INVITE.
`auth_password`	string	optional	SIP digest password, when the recording server challenges the INVITE.
`headers`	object	optional	Extra SIP headers to include in the INVITE. Require: siprec is always sent.

StatusResponse

Field	Type		Description
`instance_id`	string	optional	Instance identifier
`status`	string	required

TTSRequest

Field	Type		Description
`text`	string	required	Text to synthesize
`voice`	string	required	Provider-specific voice identifier. ElevenLabs: voice name or ID. AWS Polly: voice ID (e.g. Joanna, Matthew). Google Cloud: voice name — either full format (e.g. en-US-Neural2-F) or short name for Gemini models (e.g. Achernar, Kore). Deepgram: model name (e.g. aura-2-asteria-en).
`model_id`	string	required	Provider-specific model/engine. ElevenLabs: model ID. AWS Polly: engine (standard, neural, long-form, generative; default neural). Google Cloud: model name (e.g. gemini-2.5-pro-tts, chirp3-hd).
`language`	string	optional	Language code (e.g. "en-US", "pl-pl"). Required for Google Gemini TTS voices that use short names (e.g. Achernar). Auto-extracted from full voice names like en-US-Neural2-F.
`prompt`	string	optional	Style/tone instruction for promptable voice models (Google Gemini TTS only). E.g. "Read aloud in a warm, welcoming tone."
`volume`	integer	required	Volume adjustment in dB (-8 to 8) `0`
`provider`	enum	optional	TTS provider: "elevenlabs" (default), "aws", "google", or "deepgram" Values: `elevenlabs`, `aws`, `google`, `deepgram`
`api_key`	string	optional	ElevenLabs: API key override (falls back to ELEVENLABS_API_KEY env var). AWS: optional ACCESS_KEY:SECRET_KEY override (falls back to default AWS credential chain). Google Cloud: optional API key override (falls back to Application Default Credentials). Deepgram: API key override (falls back to DEEPGRAM_API_KEY env var).

TransferCompleteRequest

Field	Type
`success`	boolean	required
`status_code`	integer	optional
`reason`	string	optional

TransferDeclineRequest

Field	Type		Description
`code`	integer	optional
`reason`	string	optional

TransferProgressRequest

Field	Type		Description
`status_code`	integer	required
`reason`	string	optional

TransferRequest

Field	Type		Description
`target`	string	required	SIP URI to transfer the call to (e.g. "sip:[email protected]").
`replaces_leg_id`	string	optional	ID of an existing connected SIP leg whose dialog should be replaced (attended transfer). Omit for blind transfer.

TrunkView

Field	Type
`id`	string	required
`type`	string	required
`app_id`	string	optional
`status`	string	required
`last_error`	string	optional
`created_at`	string	required
`sip_register`	any	optional
`ip_ip`	any	optional

TrunksListResponse

Field	Type		Description
`trunks`	array[object]	required

UpdateLegStreamRequest

Field	Type		Description
`role`	string	optional	New routing role for this stream inside its room. The room's routing matrix decides who hears it. Pass an empty string to clear the role (full mesh). Omit to leave it untouched. Applied atomically — the room's allow-sets are recomputed in a single mixer-mutex acquisition, so no audio bleeds through mid-change.

UpdateRoomBridgeRequest

Field	Type		Description
`direction`	enum	required	New audio flow relative to the room in the path: bidirectional, send, receive, or none. Values: `bidirectional`, `send`, `receive`, `none`

VAPIAgentRequest

Field	Type		Description
`assistant_id`	string	required	VAPI assistant ID
`first_message`	string	optional	Override the agent's first message
`variable_values`	object	optional	Key-value pairs passed as VAPI variable values (assistantOverrides.variableValues)
`api_key`	string	optional	API key override (falls back to VAPI_API_KEY env var)

VolumeRequest

Field	Type		Description
`volume`	integer	required	Volume adjustment (-8 to 8, ~3dB per step, 0 = unchanged)

WebRTCCandidatesResult

Field	Type		Description
`candidates`	array[object]	required
`done`	boolean	required

WebRTCOfferRequest

Field	Type		Description
`sdp`	string	required	SDP offer from the browser
`app_id`	string	optional	Application identifier. Carried through to all events emitted for this leg, and matched against the VSI `app_id` filter.

WebRTCOfferResult

Field	Type		Description
`leg_id`	string	required
`sdp`	string	required

Event envelope delivered via HTTP POST to registered webhook URLs. Event-specific fields are flattened into the top-level object (no "data" wrapper). Includes X-Signature-256 header when a secret is configured, and an X-Event-Id header equal to the event_id field.

Field	Type		Description
`type`	enum	required	Values: `leg.ringing`, `leg.early_media`, `leg.connected`, `leg.disconnected`, `leg.joined_room`, `leg.left_room`, `leg.muted`, `leg.unmuted`, `leg.deaf`, `leg.undeaf`, `leg.hold`, `leg.unhold`, `leg.command_failed`, `leg.stream_added`, `leg.stream_removed`, `leg.stream_rejected`, `leg.stream_failed`, `leg.stream_room_changed`, `leg.stream_role_changed`, `siprec.session_started`, `siprec.session_ended`, `siprec.metadata_updated`, `siprec.participant_joined`, `siprec.participant_left`, `dtmf.received`, `rtt.received`, `speaking.started`, `speaking.stopped`, `playback.started`, `playback.finished`, `playback.error`, `tts.started`, `tts.finished`, `tts.error`, `tts.staged`, `tts.discarded`, `recording.started`, `recording.finished`, `recording.paused`, `recording.resumed`, `leg.transfer_initiated`, `leg.transfer_requested`, `leg.transfer_progress`, `leg.transfer_completed`, `leg.transfer_failed`, `room.created`, `room.deleted`, `room.bridged`, `room.bridge_updated`, `room.unbridged`, `room.routing_changed`, `leg.role_changed`, `stt.text`, `stt.turn`, `agent.connected`, `agent.disconnected`, `agent.user_transcript`, `agent.agent_response`, `amd.result`, `amd.beep`, `sip.registration_attempt`, `sip.registration_active`, `sip.registration_expired`, `sip.outbound_registration_active`, `sip.outbound_registration_failed`, `sip.outbound_registration_expired`
`timestamp`	string(date-time)	required
`event_id`	string(uuid)	optional	Stable per-event idempotency key; identical across delivery retries and across all subscribers of the event.
`instance_id`	string	optional	Instance identifier