Documentation

CreateSIPParticipantRequest extends Message
in package

Application

A SIP Participant is a singular SIP session connected to a LiveKit room via a SIP Trunk into a SIP DispatchRule

Generated from protobuf message livekit.CreateSIPParticipantRequest

Properties

$destination : mixed: NEXT ID: 23
$display_name : mixed: Optional display name for the 'From' SIP header.
$dtmf : mixed: Optionally send following DTMF digits (extension codes) when making a call.
$hide_phone_number : mixed: By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.
$include_headers : mixed: Map SIP headers from 200 OK to sip.h.* participant attributes automatically.
$krisp_enabled : mixed: Enable voice isolation for the callee.
$max_call_duration : mixed: Max call duration.
$media_encryption : mixed: Generated from protobuf field <code>.livekit.SIPMediaEncryption media_encryption = 18;</code>
$participant_identity : mixed: Optional identity of the participant in LiveKit room
$participant_metadata : mixed: Optional user-defined metadata. Will be attached to a created Participant in the room.
$participant_name : mixed: Optional name of the participant in LiveKit room
$play_dialtone : mixed: Generated from protobuf field <code>bool play_dialtone = 13;</code>
$play_ringtone : mixed: Optionally play dialtone in the room as an audible indicator for existing participants. The `play_ringtone` option is deprectated but has the same effect.
$ringing_timeout : mixed: Max time for the callee to answer the call.
$room_name : mixed: What LiveKit room should this participant be connected too
$sip_call_to : mixed: What number should be dialed via SIP
$sip_number : mixed: Optional SIP From number to use. If empty, trunk number is used.
$sip_trunk_id : mixed: What SIP Trunk should be used to dial the user
$trunk : mixed: Generated from protobuf field <code>.livekit.SIPOutboundConfig trunk = 20;</code>
$wait_until_answered : mixed: Wait for the answer for the call before returning.
$headers : mixed: These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint.
$participant_attributes : mixed: Optional user-defined attributes. Will be attached to a created Participant in the room.

Methods

__construct() : mixed: Constructor.
clearDestination() : mixed
clearDisplayName() : mixed
clearMaxCallDuration() : mixed
clearRingingTimeout() : mixed
clearTrunk() : mixed
getDestination() : Destination|null: NEXT ID: 23
getDisplayName() : string: Optional display name for the 'From' SIP header.
getDtmf() : string: Optionally send following DTMF digits (extension codes) when making a call.
getHeaders() : MapField: These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint.
getHidePhoneNumber() : bool: By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.
getIncludeHeaders() : int: Map SIP headers from 200 OK to sip.h.* participant attributes automatically.
getKrispEnabled() : bool: Enable voice isolation for the callee.
getMaxCallDuration() : Duration|null: Max call duration.
getMediaEncryption() : int: Generated from protobuf field <code>.livekit.SIPMediaEncryption media_encryption = 18;</code>
getParticipantAttributes() : MapField: Optional user-defined attributes. Will be attached to a created Participant in the room.
getParticipantIdentity() : string: Optional identity of the participant in LiveKit room
getParticipantMetadata() : string: Optional user-defined metadata. Will be attached to a created Participant in the room.
getParticipantName() : string: Optional name of the participant in LiveKit room
getPlayDialtone() : bool: Generated from protobuf field <code>bool play_dialtone = 13;</code>
getPlayRingtone() : bool: Optionally play dialtone in the room as an audible indicator for existing participants. The `play_ringtone` option is deprectated but has the same effect.
getRingingTimeout() : Duration|null: Max time for the callee to answer the call.
getRoomName() : string: What LiveKit room should this participant be connected too
getSipCallTo() : string: What number should be dialed via SIP
getSipNumber() : string: Optional SIP From number to use. If empty, trunk number is used.
getSipTrunkId() : string: What SIP Trunk should be used to dial the user
getTrunk() : SIPOutboundConfig|null: Generated from protobuf field <code>.livekit.SIPOutboundConfig trunk = 20;</code>
getWaitUntilAnswered() : bool: Wait for the answer for the call before returning.
hasDestination() : mixed
hasDisplayName() : mixed
hasMaxCallDuration() : mixed
hasRingingTimeout() : mixed
hasTrunk() : mixed
setDestination() : $this: NEXT ID: 23
setDisplayName() : $this: Optional display name for the 'From' SIP header.
setDtmf() : $this: Optionally send following DTMF digits (extension codes) when making a call.
setHeaders() : $this: These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint.
setHidePhoneNumber() : $this: By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.
setIncludeHeaders() : $this: Map SIP headers from 200 OK to sip.h.* participant attributes automatically.
setKrispEnabled() : $this: Enable voice isolation for the callee.
setMaxCallDuration() : $this: Max call duration.
setMediaEncryption() : $this: Generated from protobuf field <code>.livekit.SIPMediaEncryption media_encryption = 18;</code>
setParticipantAttributes() : $this: Optional user-defined attributes. Will be attached to a created Participant in the room.
setParticipantIdentity() : $this: Optional identity of the participant in LiveKit room
setParticipantMetadata() : $this: Optional user-defined metadata. Will be attached to a created Participant in the room.
setParticipantName() : $this: Optional name of the participant in LiveKit room
setPlayDialtone() : $this: Generated from protobuf field <code>bool play_dialtone = 13;</code>
setPlayRingtone() : $this: Optionally play dialtone in the room as an audible indicator for existing participants. The `play_ringtone` option is deprectated but has the same effect.
setRingingTimeout() : $this: Max time for the callee to answer the call.
setRoomName() : $this: What LiveKit room should this participant be connected too
setSipCallTo() : $this: What number should be dialed via SIP
setSipNumber() : $this: Optional SIP From number to use. If empty, trunk number is used.
setSipTrunkId() : $this: What SIP Trunk should be used to dial the user
setTrunk() : $this: Generated from protobuf field <code>.livekit.SIPOutboundConfig trunk = 20;</code>
setWaitUntilAnswered() : $this: Wait for the answer for the call before returning.

$destination

NEXT ID: 23


        protected
            mixed
    $destination
     = null

Generated from protobuf field optional .livekit.Destination destination = 22;

$display_name

Optional display name for the 'From' SIP header.


        protected
            mixed
    $display_name
     = null

Cases:

Unspecified: Use legacy behavior - display name will be set to be the caller's number.
Empty string: Do not send a display name, which will result in a CNAM lookup downstream.
Non-empty: Use the specified value as the display name.

Generated from protobuf field optional string display_name = 21 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

$dtmf

Optionally send following DTMF digits (extension codes) when making a call.


        protected
            mixed
    $dtmf
     = ''

Character 'w' can be used to add a 0.5 sec delay.

Generated from protobuf field string dtmf = 5;

$hide_phone_number

By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.


        protected
            mixed
    $hide_phone_number
     = false

If true, a random value for identity will be used and numbers will be omitted from attributes.

Generated from protobuf field bool hide_phone_number = 10;

$include_headers

Map SIP headers from 200 OK to sip.h.* participant attributes automatically.


        protected
            mixed
    $include_headers
     = 0

When the names of required headers is known, using headers_to_attributes is strongly recommended. When mapping 200 OK headers to follow-up request headers with attributes_to_headers map, lowercase header names should be used, for example: sip.h.x-custom-header.

Generated from protobuf field .livekit.SIPHeaderOptions include_headers = 17;

$krisp_enabled

Enable voice isolation for the callee.


        protected
            mixed
    $krisp_enabled
     = false

Generated from protobuf field bool krisp_enabled = 14;

$max_call_duration

Max call duration.


        protected
            mixed
    $max_call_duration
     = null

Generated from protobuf field .google.protobuf.Duration max_call_duration = 12;

$media_encryption

Generated from protobuf field <code>.livekit.SIPMediaEncryption media_encryption = 18;</code>


        protected
            mixed
    $media_encryption
     = 0

$participant_identity

Optional identity of the participant in LiveKit room


        protected
            mixed
    $participant_identity
     = ''

Generated from protobuf field string participant_identity = 4;

$participant_metadata

Optional user-defined metadata. Will be attached to a created Participant in the room.


        protected
            mixed
    $participant_metadata
     = ''

Generated from protobuf field string participant_metadata = 8 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

$participant_name

Optional name of the participant in LiveKit room


        protected
            mixed
    $participant_name
     = ''

Generated from protobuf field string participant_name = 7 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

$play_dialtone

Generated from protobuf field <code>bool play_dialtone = 13;</code>


        protected
            mixed
    $play_dialtone
     = false

$play_ringtone

Optionally play dialtone in the room as an audible indicator for existing participants. The `play_ringtone` option is deprectated but has the same effect.


        protected
            mixed
    $play_ringtone
     = false

Generated from protobuf field bool play_ringtone = 6 [deprecated = true];

$ringing_timeout

Max time for the callee to answer the call.


        protected
            mixed
    $ringing_timeout
     = null

Generated from protobuf field .google.protobuf.Duration ringing_timeout = 11;

$room_name

What LiveKit room should this participant be connected too


        protected
            mixed
    $room_name
     = ''

Generated from protobuf field string room_name = 3;

$sip_call_to

What number should be dialed via SIP


        protected
            mixed
    $sip_call_to
     = ''

Generated from protobuf field string sip_call_to = 2;

$sip_number

Optional SIP From number to use. If empty, trunk number is used.


        protected
            mixed
    $sip_number
     = ''

Generated from protobuf field string sip_number = 15;

$sip_trunk_id

What SIP Trunk should be used to dial the user


        protected
            mixed
    $sip_trunk_id
     = ''

Generated from protobuf field string sip_trunk_id = 1;

$trunk

Generated from protobuf field <code>.livekit.SIPOutboundConfig trunk = 20;</code>


        protected
            mixed
    $trunk
     = null

$wait_until_answered

Wait for the answer for the call before returning.


        protected
            mixed
    $wait_until_answered
     = false

Generated from protobuf field bool wait_until_answered = 19;

$headers

These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint.


        private
            mixed
    $headers

Generated from protobuf field map<string, string> headers = 16 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

$participant_attributes

Optional user-defined attributes. Will be attached to a created Participant in the room.


        private
            mixed
    $participant_attributes

Generated from protobuf field map<string, string> participant_attributes = 9 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

__construct()

Constructor.


    public
                    __construct([array<string|int, mixed> $data = null ]) : mixed

Parameters

$data : array<string|int, mixed> = null

{ Optional. Data for populating the Message object.

@type string $sip_trunk_id
      What SIP Trunk should be used to dial the user
@type \Livekit\SIPOutboundConfig $trunk
@type string $sip_call_to
      What number should be dialed via SIP
@type string $sip_number
      Optional SIP From number to use. If empty, trunk number is used.
@type string $room_name
      What LiveKit room should this participant be connected too
@type string $participant_identity
      Optional identity of the participant in LiveKit room
@type string $participant_name
      Optional name of the participant in LiveKit room
@type string $participant_metadata
      Optional user-defined metadata. Will be attached to a created Participant in the room.
@type array|\Google\Protobuf\Internal\MapField $participant_attributes
      Optional user-defined attributes. Will be attached to a created Participant in the room.
@type string $dtmf
      Optionally send following DTMF digits (extension codes) when making a call.
      Character 'w' can be used to add a 0.5 sec delay.
@type bool $play_ringtone
      Optionally play dialtone in the room as an audible indicator for existing participants. The `play_ringtone` option is deprectated but has the same effect.
@type bool $play_dialtone
@type bool $hide_phone_number
      By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.
      If true, a random value for identity will be used and numbers will be omitted from attributes.
@type array|\Google\Protobuf\Internal\MapField $headers
      These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint.
@type int $include_headers
      Map SIP headers from 200 OK to sip.h.* participant attributes automatically.
      When the names of required headers is known, using headers_to_attributes is strongly recommended.
      When mapping 200 OK headers to follow-up request headers with attributes_to_headers map,
      lowercase header names should be used, for example: sip.h.x-custom-header.
@type \Google\Protobuf\Duration $ringing_timeout
      Max time for the callee to answer the call.
@type \Google\Protobuf\Duration $max_call_duration
      Max call duration.
@type bool $krisp_enabled
      Enable voice isolation for the callee.
@type int $media_encryption
@type bool $wait_until_answered
      Wait for the answer for the call before returning.
@type string $display_name
      Optional display name for the 'From' SIP header.
      Cases:
      1) Unspecified: Use legacy behavior - display name will be set to be the caller's number.
      2) Empty string: Do not send a display name, which will result in a CNAM lookup downstream.
      3) Non-empty: Use the specified value as the display name.
@type \Livekit\Destination $destination
      NEXT ID: 23

}

clearDestination()


    public
                    clearDestination() : mixed

clearDisplayName()


    public
                    clearDisplayName() : mixed

clearMaxCallDuration()


    public
                    clearMaxCallDuration() : mixed

clearRingingTimeout()


    public
                    clearRingingTimeout() : mixed

clearTrunk()


    public
                    clearTrunk() : mixed

getDestination()

NEXT ID: 23


    public
                    getDestination() : Destination|null

Generated from protobuf field optional .livekit.Destination destination = 22;

Return values

Destination|null

getDisplayName()

Optional display name for the 'From' SIP header.


    public
                    getDisplayName() : string

Cases:

Unspecified: Use legacy behavior - display name will be set to be the caller's number.
Empty string: Do not send a display name, which will result in a CNAM lookup downstream.
Non-empty: Use the specified value as the display name.

Generated from protobuf field optional string display_name = 21 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Return values

string

getDtmf()

Optionally send following DTMF digits (extension codes) when making a call.


    public
                    getDtmf() : string

Character 'w' can be used to add a 0.5 sec delay.

Generated from protobuf field string dtmf = 5;

Return values

string

getHeaders()

These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint.


    public
                    getHeaders() : MapField

Generated from protobuf field map<string, string> headers = 16 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Return values

MapField

getHidePhoneNumber()

By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.


    public
                    getHidePhoneNumber() : bool

If true, a random value for identity will be used and numbers will be omitted from attributes.

Generated from protobuf field bool hide_phone_number = 10;

Return values

bool

getIncludeHeaders()

Map SIP headers from 200 OK to sip.h.* participant attributes automatically.


    public
                    getIncludeHeaders() : int

Generated from protobuf field .livekit.SIPHeaderOptions include_headers = 17;

Return values

int

getKrispEnabled()

Enable voice isolation for the callee.


    public
                    getKrispEnabled() : bool

Generated from protobuf field bool krisp_enabled = 14;

Return values

bool

getMaxCallDuration()

Max call duration.


    public
                    getMaxCallDuration() : Duration|null

Generated from protobuf field .google.protobuf.Duration max_call_duration = 12;

Return values

Duration|null

getMediaEncryption()

Generated from protobuf field <code>.livekit.SIPMediaEncryption media_encryption = 18;</code>


    public
                    getMediaEncryption() : int

Return values

int

getParticipantAttributes()

Optional user-defined attributes. Will be attached to a created Participant in the room.


    public
                    getParticipantAttributes() : MapField

Generated from protobuf field map<string, string> participant_attributes = 9 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Return values

MapField

getParticipantIdentity()

Optional identity of the participant in LiveKit room


    public
                    getParticipantIdentity() : string

Generated from protobuf field string participant_identity = 4;

Return values

string

getParticipantMetadata()

Optional user-defined metadata. Will be attached to a created Participant in the room.


    public
                    getParticipantMetadata() : string

Generated from protobuf field string participant_metadata = 8 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Return values

string

getParticipantName()

Optional name of the participant in LiveKit room


    public
                    getParticipantName() : string

Generated from protobuf field string participant_name = 7 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Return values

string

getPlayDialtone()

Generated from protobuf field <code>bool play_dialtone = 13;</code>


    public
                    getPlayDialtone() : bool

Return values

bool

getPlayRingtone()

Optionally play dialtone in the room as an audible indicator for existing participants. The `play_ringtone` option is deprectated but has the same effect.


    public
                    getPlayRingtone() : bool

Generated from protobuf field bool play_ringtone = 6 [deprecated = true];

Return values

bool

getRingingTimeout()

Max time for the callee to answer the call.


    public
                    getRingingTimeout() : Duration|null

Generated from protobuf field .google.protobuf.Duration ringing_timeout = 11;

Return values

Duration|null

getRoomName()

What LiveKit room should this participant be connected too


    public
                    getRoomName() : string

Generated from protobuf field string room_name = 3;

Return values

string

getSipCallTo()

What number should be dialed via SIP


    public
                    getSipCallTo() : string

Generated from protobuf field string sip_call_to = 2;

Return values

string

getSipNumber()

Optional SIP From number to use. If empty, trunk number is used.


    public
                    getSipNumber() : string

Generated from protobuf field string sip_number = 15;

Return values

string

getSipTrunkId()

What SIP Trunk should be used to dial the user


    public
                    getSipTrunkId() : string

Generated from protobuf field string sip_trunk_id = 1;

Return values

string

getTrunk()

Generated from protobuf field <code>.livekit.SIPOutboundConfig trunk = 20;</code>


    public
                    getTrunk() : SIPOutboundConfig|null

Return values

SIPOutboundConfig|null

getWaitUntilAnswered()

Wait for the answer for the call before returning.


    public
                    getWaitUntilAnswered() : bool

Generated from protobuf field bool wait_until_answered = 19;

Return values

bool

hasDestination()


    public
                    hasDestination() : mixed

hasDisplayName()


    public
                    hasDisplayName() : mixed

hasMaxCallDuration()


    public
                    hasMaxCallDuration() : mixed

hasRingingTimeout()


    public
                    hasRingingTimeout() : mixed

hasTrunk()


    public
                    hasTrunk() : mixed

setDestination()

NEXT ID: 23


    public
                    setDestination(Destination $var) : $this

Generated from protobuf field optional .livekit.Destination destination = 22;

Parameters

$var : Destination

Return values

$this

setDisplayName()

Optional display name for the 'From' SIP header.


    public
                    setDisplayName(string $var) : $this

Cases:

Unspecified: Use legacy behavior - display name will be set to be the caller's number.
Empty string: Do not send a display name, which will result in a CNAM lookup downstream.
Non-empty: Use the specified value as the display name.

Generated from protobuf field optional string display_name = 21 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Parameters

$var : string

Return values

$this

setDtmf()

Optionally send following DTMF digits (extension codes) when making a call.


    public
                    setDtmf(string $var) : $this

Character 'w' can be used to add a 0.5 sec delay.

Generated from protobuf field string dtmf = 5;

Parameters

$var : string

Return values

$this

setHeaders()

These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint.


    public
                    setHeaders(array<string|int, mixed>|MapField $var) : $this

Generated from protobuf field map<string, string> headers = 16 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Parameters

$var : array<string|int, mixed>|MapField

Return values

$this

setHidePhoneNumber()

By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.


    public
                    setHidePhoneNumber(bool $var) : $this

If true, a random value for identity will be used and numbers will be omitted from attributes.

Generated from protobuf field bool hide_phone_number = 10;

Parameters

$var : bool

Return values

$this

setIncludeHeaders()

Map SIP headers from 200 OK to sip.h.* participant attributes automatically.


    public
                    setIncludeHeaders(int $var) : $this

Generated from protobuf field .livekit.SIPHeaderOptions include_headers = 17;

Parameters

$var : int

Return values

$this

setKrispEnabled()

Enable voice isolation for the callee.


    public
                    setKrispEnabled(bool $var) : $this

Generated from protobuf field bool krisp_enabled = 14;

Parameters

$var : bool

Return values

$this

setMaxCallDuration()

Max call duration.


    public
                    setMaxCallDuration(Duration $var) : $this

Generated from protobuf field .google.protobuf.Duration max_call_duration = 12;

Parameters

$var : Duration

Return values

$this

setMediaEncryption()

Generated from protobuf field <code>.livekit.SIPMediaEncryption media_encryption = 18;</code>


    public
                    setMediaEncryption(int $var) : $this

Parameters

$var : int

Return values

$this

setParticipantAttributes()

Optional user-defined attributes. Will be attached to a created Participant in the room.


    public
                    setParticipantAttributes(array<string|int, mixed>|MapField $var) : $this

Generated from protobuf field map<string, string> participant_attributes = 9 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Parameters

$var : array<string|int, mixed>|MapField

Return values

$this

setParticipantIdentity()

Optional identity of the participant in LiveKit room


    public
                    setParticipantIdentity(string $var) : $this

Generated from protobuf field string participant_identity = 4;

Parameters

$var : string

Return values

$this

setParticipantMetadata()

Optional user-defined metadata. Will be attached to a created Participant in the room.


    public
                    setParticipantMetadata(string $var) : $this

Generated from protobuf field string participant_metadata = 8 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Parameters

$var : string

Return values

$this

setParticipantName()

Optional name of the participant in LiveKit room


    public
                    setParticipantName(string $var) : $this

Generated from protobuf field string participant_name = 7 [(.logger.redact) = true, (.logger.redact_format) = "<redacted ({{ .Size }} bytes)>"];

Parameters

$var : string

Return values

$this

setPlayDialtone()

Generated from protobuf field <code>bool play_dialtone = 13;</code>


    public
                    setPlayDialtone(bool $var) : $this

Parameters

$var : bool

Return values

$this

setPlayRingtone()

Optionally play dialtone in the room as an audible indicator for existing participants. The `play_ringtone` option is deprectated but has the same effect.


    public
                    setPlayRingtone(bool $var) : $this

Generated from protobuf field bool play_ringtone = 6 [deprecated = true];

Parameters

$var : bool

Return values

$this

setRingingTimeout()

Max time for the callee to answer the call.


    public
                    setRingingTimeout(Duration $var) : $this

Generated from protobuf field .google.protobuf.Duration ringing_timeout = 11;

Parameters

$var : Duration

Return values

$this

setRoomName()

What LiveKit room should this participant be connected too


    public
                    setRoomName(string $var) : $this

Generated from protobuf field string room_name = 3;

Parameters

$var : string

Return values

$this

setSipCallTo()

What number should be dialed via SIP


    public
                    setSipCallTo(string $var) : $this

Generated from protobuf field string sip_call_to = 2;

Parameters

$var : string

Return values

$this

setSipNumber()

Optional SIP From number to use. If empty, trunk number is used.


    public
                    setSipNumber(string $var) : $this

Generated from protobuf field string sip_number = 15;

Parameters

$var : string

Return values

$this

setSipTrunkId()

What SIP Trunk should be used to dial the user


    public
                    setSipTrunkId(string $var) : $this

Generated from protobuf field string sip_trunk_id = 1;

Parameters

$var : string

Return values

$this

setTrunk()

Generated from protobuf field <code>.livekit.SIPOutboundConfig trunk = 20;</code>


    public
                    setTrunk(SIPOutboundConfig $var) : $this

Parameters

$var : SIPOutboundConfig

Return values

$this

setWaitUntilAnswered()

Wait for the answer for the call before returning.


    public
                    setWaitUntilAnswered(bool $var) : $this

Generated from protobuf field bool wait_until_answered = 19;

Parameters

$var : bool

Return values

$this

CreateSIPParticipantRequest extends Message in package Application

Table of Contents

Properties

Methods

Properties

$destination

$display_name

$dtmf

$hide_phone_number

$include_headers

$krisp_enabled

$max_call_duration

$media_encryption

$participant_identity

$participant_metadata

$participant_name

$play_dialtone

$play_ringtone

$ringing_timeout

$room_name

$sip_call_to

$sip_number

$sip_trunk_id

$trunk

$wait_until_answered

$headers

$participant_attributes

Methods

__construct()

Parameters

clearDestination()

clearDisplayName()

clearMaxCallDuration()

clearRingingTimeout()

clearTrunk()

getDestination()

Return values

getDisplayName()

Return values

getDtmf()

Return values

getHeaders()

Return values

getHidePhoneNumber()

Return values

getIncludeHeaders()

Return values

getKrispEnabled()

Return values

getMaxCallDuration()

Return values

getMediaEncryption()

Return values

getParticipantAttributes()

Return values

getParticipantIdentity()

Return values

getParticipantMetadata()

Return values

getParticipantName()

Return values

getPlayDialtone()

Return values

getPlayRingtone()

Return values

getRingingTimeout()

Return values

getRoomName()

Return values

getSipCallTo()

Return values

getSipNumber()

Return values

getSipTrunkId()

Return values

getTrunk()

Return values

getWaitUntilAnswered()

Return values

hasDestination()

CreateSIPParticipantRequest extends Message
in package

Application