SipSettings interface

Settings for a SIP node.

A SIP node is configured with an account and can make any number of calls over its lifetime, one at a time. Everything here is fixed for the life of the node (to use a different account, create a different node); calls are what come and go — placed with SipNode.dial() (or on creation, see autoDial), ended with SipNode.hangup() or by the far end. The node, its id, its subscriptions and anything subscribed to it carry on across calls.

Signature:

export interface SipSettings extends ProcessorNodeSettings<SipNode>, StreamStatisticsMixin

Extends: ProcessorNodeSettings<SipNode>, StreamStatisticsMixin

Properties

Property Type Description

account

SipAccountSettings

autoDial?

boolean

(Optional)

call?

SipCallSettings

(Optional)

codecs?

SipCodecName[]

(Optional)

continuousOutput?

boolean

(Optional)

dialTimeoutMs?

number

(Optional)

inbound?

"reject" | "auto" | "manual"

(Optional)

jitterBuffer?

JitterBufferConfig

(Optional)

onConnectionStats?

(stats: SipConnectionStatistics) ⇒ void

(Optional)

onDialRejected?

(rejection: SipDialRejection) ⇒ void

(Optional)

onDisconnect?

(reason: SipDisconnectReason) ⇒ void

(Optional)

onDtmf?

(digit: string) ⇒ void

(Optional)

onIncomingCall?

(call: SipIncomingCall) ⇒ void

(Optional)

onNegotiatedCodec?

(codec: SipNegotiatedCodec) ⇒ void

(Optional)

onRegistration?

(registration: SipRegistration) ⇒ void

(Optional)

onStatus?

(status: SipStatus) ⇒ void

(Optional)

proxy?

string

(Optional)

publicAddr?

string

(Optional)

registrationExpiresSeconds?

number

(Optional)

rtpPortRange?

{ start: number; end: number; }

(Optional)

sampleRate?

8000 | 16000 | 48000

(Optional)

sessionExpiresSeconds?

number

(Optional)

silenceSuppression?

boolean

(Optional)

sipPort?

number

(Optional)

sourceName

string

srtp?

"disabled" | "optional" | "mandatory"

(Optional)

srtpRequireTls?

boolean

(Optional)

startupThreshold?

InputThresholdSettings

(Optional) when specified, drop the far end’s (inbound) audio until it is arriving at realtime, so a burst at call setup does not set the media T0

statsIntervalMs?

number

(Optional)

stereo?

boolean

(Optional)

tls?

{ certFile?: string; privateKeyFile?: string; password?: string; caFile?: string; verifyServer?: boolean; }

(Optional)