Pexip client REST API

This guide describes the Pexip client REST API. It is designed for use by non-web-based, third-party voice/video applications that want to initiate or connect to conferences hosted on the Pexip Infinity platform.

We strongly recommend that web-based applications use the PexRTC JavaScript client API instead.

This API specification is regularly evolving between versions of the Pexip Infinity platform. While we will attempt to maintain backward compatibility, there may be significant changes between versions.

Using the API

The prefix for all conference-related API calls is:

https://<node_address>/api/client/v2/conferences/<conference_alias>/

where <node_address> is the address of a Conferencing Node and <conference_alias> is an alias of the conference you are connecting to. Under this API path comes a sequence of response API calls, for example:

https://10.0.0.1/api/client/v2/conferences/meet_alice/request_token

All commands in the client API are authenticated with a token, which is presented by the Pexip Conferencing Node. The token has a validity lifetime, before the end of which it must be refreshed. The token is presented in a HTTP header entitled "token" on every HTTP request, except for the initial request_token request.

Unless otherwise specified, all payloads of requests and responses are JSON objects, Content-Type: application/json.

The responses have two fields, status and result:

  • status is "success" if the command has been processed by Pexip, or "failure" if the command could not be processed. Note that this does not mean that the end result is success, only that the request has been received and processed.
  • the result field indicates if the request was successful.

Summary of API requests and events

This section summarizes the requests and server-sent events that may be used, which are then described in more detail.

Client control requests

These REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/<request>

Request GET/POST Description
request_token POST Requests a new token from the Pexip Conferencing Node.
refresh_token POST Refreshes a token to get a new one.
release_token POST Releases the token (effectively a disconnect for the participant).

Conference control functions

These REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/<request>

Request GET/POST Description
dial POST Dials out from the conference to a target endpoint.
conference_status GET Provides the status of the conference.
lock / unlock POST Locks / unlocks the conference.
start_conference POST Starts a conference and allows Guests in the "waiting room" to join the meeting.
muteguests / unmuteguests POST Mutes / unmutes all Guests on a conference.
disconnect POST Disconnects all conference participants, including the participant calling the function.
message POST Sends a message to all participants in the conference.
participants GET Returns the full participant list of the conference.
transform_layout POST Changes the conference layout, controls streaming content, and enables/disables indicators and overlay text.
clearallbuzz POST Lower all raised hands.

Participant functions

These participant REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/participants/<participant_uuid>/<request>

Request GET/POST Description
disconnect POST Disconnects a participant.
mute / unmute POST Mutes / unmutes a participant.
allowrxpresentation / denyrxpresentation POST Enables or disables a participant from receiving the presentation stream.
spotlighton / spotlightoff POST Enables or disables the "spotlight" on a participant.
unlock POST Lets a specified participant into the conference from the waiting room of a locked conference.
dtmf POST Sends DTMF digits to the participant.
calls POST Upgrades this connection to have a WebRTC or RTMP audio / video call element.
overlaytext POST Changes the participant name overlay text.
role POST Changes the role of the participant.
fecc POST Send Far End Camera Control messaging to the participant.
buzz POST Raise a participant's hand.
clearbuzz POST Lower a participant's hand.
transfer POST Transfers a participant to another conference.
avatar.jpg GET Obtains the image to display to represent a conference participant or directory contact.

Call functions

These call REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/participants/<participant_uuid>/calls/<call_uuid>/<request>

Request GET/POST Description
ack POST Starts media for the specified call (WebRTC calls only).
disconnect POST Disconnects the specified call.
dtmf POST Sends DTMF digits to the specified participant.

Server-sent events

To subscribe, open an HTTP connection to:

https://<node_address>/api/client/v2/conferences/<conference_alias>/events?token=<token_id>

Event Description
presentation_start Marks the start of a presentation, and includes the information on which participant is presenting.
presentation_stop The presentation has finished.
presentation_frame A new presentation frame is available.
participant_create A new participant has joined the conference.
participant_update A participant's properties have changed.
participant_delete A participant has left the conference.
participant_sync_begin / participant_sync_end These two messages start and end the sending of the complete participant list.
conference_update Conference properties have been updated.
layout

The stage layout has changed.

message_received A chat message has been broadcast to the conference.
stage An update to the "stage layout" is available. This declares the order of active speakers, and their voice activity.
call_disconnected Sent when a child call has been disconnected.
disconnect Sent when the participant is being disconnected from the Pexip side.

Other requests

Request Description
/api/client/v2/status Check whether a Conferencing Node is in maintenance mode.

Client control requests

This section describes in detail the requests that may be used to initiate and manage a connection to a Conferencing Node.

These REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/<request>

request_token

This POST requests a new token from the Pexip Conferencing Node.

Request example:

{"display_name": "Alice"}

Request fields:

display_name The name by which this participant should be known.

Response example:

{"status": "success", "result": 
{"token": "SE9TVAltZ...etc...zNiZjlmNjFhMTlmMTJiYTE%3D", 
"expires": "120", 
"participant_uuid": "2c34f35f-1060-438c-9e87-6c2dffbc9980",
"display_name": "Alice", 
"stun": [{"url": "stun:stun.l.google.com:19302"}], 
"analytics_enabled": true, 
"version": {"pseudo_version": "25010.0.0", "version_id": "10"},
"role": "HOST", 
"service_type": "conference", 
"chat_enabled": true, 
"current_service_type": "conference"}}

This result contains the token (abridged in the above example) to use to authenticate all future requests, and an expiry time (in seconds) after which this token becomes invalid. The full list of fields in the result is as follows:

token The authentication token for future requests.
expires Validity lifetime in seconds. Use refresh_token to obtain an updated token.
participant_uuid The uuid associated with this newly created participant. It is used to identify this participant in the participant list.
version The version of the Pexip server being communicated with.
role Whether the participant is connecting as a "HOST" or a "GUEST".
chat_enabled true = chat is enabled; false = chat is not enabled.
service_type

Either "conference", "gateway" or “test_call” depending on whether this is a VMR, gateway or Test Call Service respectively.

stun STUN server configuration from the Pexip Conferencing Node.
display_name Echoes the display name in the request.
analytics_enabled Whether the Automatically send deployment and usage statistics to Pexip global setting has been enabled on the Pexip installation.
current_service_type The service type this user is connecting into. May be "conference", "gateway" or “test_call” as for service_type if directly connecting in. May also be "waiting_room" if waiting to be allowed to join a locked conference, or "ivr" if on the PIN entry screen.

PIN protected conferences

If the conference is PIN-protected, the PIN must be specified in a "pin" HTTP header. If the PIN is required but is incorrect or missing, a "403 Forbidden" error is returned. The "pin" field in the response specifies whether a PIN is required for Hosts, and a "guest_pin" field in the response specifies whether a PIN is required for Guests.

Virtual Receptions

If the conference is a Virtual Reception, a "403 Forbidden" error is returned, with a "conference_extension" field. This field is either "standard" for a standard Virtual Reception, or "mssip" for a Virtual Reception towards Microsoft Skype for Business / Lync conferences. To join the target room, a second request_token request must be made, but with a conference_extension field in the request JSON, which contains the alias of the target conference.

refresh_token

This POST request refreshes a token to get a new one.

Request: empty.

Example response:

{"status": "success", "result": 
{"token": "SE9TVAltZ...etc...jQ4YTVmMzM3MDMwNDFlNjI%3D", 
"expires": "120"}}

Fields are:

token The new authentication token for future requests.
expires Validity lifetime in seconds.

release_token

This POST request releases the token (effectively a disconnect for the participant).

Request: empty.

Response: should be ignored.

Conference control functions

This section describes in detail the requests that may be used to manage an existing conference.

These REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/<request>

dial

This POST request dials out from the conference to a target endpoint. This function is only available to conference Hosts.

Request example:

{"role": "GUEST", "destination": "bob@example.com", "protocol": "sip", "source_display_name": "Alice"}

Request fields:

role

The level of privileges the participant has in the conference:

  • "HOST": the participant has Host privileges
  • "GUEST": the participant has Guest privileges
destination The target address to call.
protocol

The protocol to use to place the outgoing call:

  • "sip"
  • "h323"
  • "rtmp"
  • "mssip" (for calls to Microsoft Skype for Business / Lync)
  • "auto" (to use Call Routing Rules)
presentation_url

This additional parameter can be specified for RTMP calls to send the presentation stream to a separate RTMP destination.

streaming

Identifies the dialed participant as a streaming or recording device:

  • "yes": streaming/recording participant
  • "no": not a streaming/recording participant

Default: "no"

dtmf_sequence

An optional DTMF sequence to transmit after the call to the dialed participant starts.

source_display_name Optional field that specifies what the calling display name should be.
source Optional field that specifies the source URI (must be a valid URI for the conference).
call_type

Optional field that limits the media content of the call:

  • "video": main video plus presentation
  • "video-only": main video only
  • "audio": audio-only

Default: "video"

keep_conference_alive

Determines whether the conference continues when all other non-ADP participants have disconnected:

  • "keep_conference_alive": the conference continues to run until this participant disconnects (applies to Hosts only).
  • "keep_conference_alive_if_multiple": the conference continues to run as long as there are two or more "keep_conference_alive_if_multiple" participants and at least one of them is a Host.
  • "keep_conference_alive_never": the conference terminates automatically if this is the only remaining participant.

Default: "keep_conference_alive" for non-streaming participants, and "keep_conference_alive_never" for streaming participants.

remote_display_name

An optional friendly name for this participant. This may be used instead of the participant's alias in participant lists and as a text overlay in some layout configurations.

text

Optional text to use instead of remote_display_name as the participant name overlay text.

Response example:

{"status": "success", "result": ["977fcd1c-8e3c-4dcf-af45-e536b77af088"]}

The response is an array of UUIDs of new participants, if dial-out was successfully initiated. In most cases the dial-out will only generate a single call and thus a single UUID in this array, however if Pexip Infinity forks the call there may end up being multiple UUIDs. Only one of these will be answered, however, and the rest will be disconnected.

The call UUIDs will appear as new participants immediately, with a "service_type" of "connecting". If the call is answered, the participant will be updated with a new "service_type", typically being "conference". The participant may also be deleted if the receiver rejects the call, or the call attempt times out in 30 seconds if not answered.

conference_status

This GET request provides the status of the conference. Currently, the only conference properties available are the lock status of the conference, whether Guests are muted, and if the conference has been started. For example:

{"status": "success", "result": 
{"locked": false, "guests_muted": false, "started": true}}

lock / unlock

These POST requests are used to lock or unlock the conference. When a conference is locked, participants waiting to join are held at a "Waiting for Host" screen. These settings are only available to conference Hosts.

Request: empty.

Response: the result is true if successful, false otherwise.

start_conference

If the only user with Host rights is connected to the conference without media (as a presentation and control-only participant), Guests will remain in the “Waiting for Host” screen. This POST request starts the conference and any Guests in the "waiting room" will join the meeting. This is only available to conference Hosts.

Request: empty.

Response: the result is true if successful, false otherwise.

muteguests / unmuteguests

These POST requests are used to mute or unmute all Guests on a conference. When muted, no Guest participants can speak unless they are explicitly unmuted. When unmuted, all Guests on a conference can speak. These settings are only available to conference Hosts.

Request: empty.

Response: the result is true if successful, false otherwise.

disconnect

This POST request disconnects all conference participants, including the participant calling the function. This setting is only available to conference Hosts.

Request: empty.

Response: the result is true if successful, false otherwise.

message

This POST request sends a message to all participants in the conference.

Request example:

{"type": "text/plain", "payload": "Hello World"}

Request fields:

type The MIME Content-Type, such as "text/plain".
payload The contents of the message.

Response: the result is true if successful, false otherwise.

participants

This GET request returns the full participant list of the conference. See the description of the participant_create EventSource for more information.

transform_layout

This POST request changes the conference layout, controls streaming content, and enables/disables indicators and overlay text.

Request fields:

transforms

This is an object containing any of the following optional parameters:

layout

host_layout

guest_layout

In VMRs the layout for Hosts and Guests is controlled by the layout parameter.

In Virtual Auditoriums the Host layout is controlled by the host_layout parameter and the Guest layout is controlled by the guest_layout parameter.

The layout options are:

  • "1:0": main speaker only
  • "1:7": main speaker and up to 7 previous speakers
  • "1:21": main speaker and up to 21 previous speakers
  • "2:21": two main speakers and up to 21 previous speakers
  • "4:0": equal size layout, up to a maximum of 4 speakers

Note that the layout parameter is an alias for host_layout, and that an attempt to set guest_layout in a service that is not a Virtual Auditorium will return a "400 Bad Request" error.

streaming_indicator Determines whether the streaming indicator icon is disabled (false) or enabled (true).
recording_indicator Determines whether the recording indicator icon is disabled (false) or enabled (true).
enable_overlay_text Determines whether participant names overlay text is disabled (false) or enabled (true).
streaming

This can be used to specifically control the content sent to a streaming participant, for example to send a different layout to the stream from that which is seen by standard participants.

It is an object containing any of the following optional parameters:

layout

Sets the layout seen by the streaming participant (regardless of Host or Guest role). The options are:

  • "1:0": main speaker only
  • "1:7": main speaker and up to 7 previous speakers
  • "1:21": main speaker and up to 21 previous speakers
  • "2:21": two main speakers and up to 21 previous speakers
  • "4:0": equal size layout, up to a maximum of 4 speakers

If waiting_screen_enabled is true, the layout parameter is ignored, as the only possible layout for a splash screen is 1:0.

waiting_screen_enabled

Determines whether the stream_waiting splash screen is displayed (true) or not (false) i.e. the standard conference content is streamed. This can be used as a "holding" screen while you wait for people in the conference to get ready to start. Note that:

  • When the waiting screen is enabled, all in-conference indicators, the plus n indicator, and text overlay are disabled.
  • The watermark is included by default.
  • If the conference is using a v1 style theme, the v2 stream_waiting splash screen is still used (with its default background, icon and text). When using v2 themes the entire stream_waiting splash screen content is customizable as usual.

Default: false

plus_n_pip_enabled

Controls whether the "plus n indicator" i.e. the number of participants present in the conference, is displayed (true) or not (false).

Default: true

indicators_enabled

Controls whether the in-conference indicators (audio participant indicators, conference locked indicators etc.) are displayed (true) or not (false).

Default: true

free_form_overlay_text

Allows you to customize (override) the overlay text. It contains a list of strings to apply to each participant in the order in which the participants are displayed.

The number of strings to supply depends upon the layout, for example you should specify up to 8 strings in a 1:7 layout. If you specify fewer strings than there are displayed participants, the remaining participants get no overlay text. You can specify a zero length string to miss a position in the order e.g. ["alice", "", "bob"]. The strings are applied to the position, not the participant i.e. if a participant switches in or out or moves within the layout, the strings stay fixed to their original positions.

If free_form_overlay_text is specified:

  • the specified strings are always displayed, regardless of whether enable_overlay_text is set or not (either via VMR configuration or as a transforms parameter)
  • it will replace all of the actual participant names if enable_overlay_text is set.

If you make multiple requests for the same conference, the changes are applied cumulatively (with the most recent settings taking precedence). If required, you can use {"transforms":{}} to reset all parameters.

Response: the result is true if successful, false otherwise.

Here are some example requests:

  • To change a Virtual Meeting Room layout: {"transforms":{"layout": "2:21"}}
  • To change a Virtual Auditorium layout: {"transforms":{"guest_layout":"2:21", "host_layout": "4:0"}}
  • To enable streaming and recording indicators: {"transforms":{"streaming_indicator": true, "recording_indicator": true}}
  • To enable overlay text: {"transforms":{"enable_overlay_text": true}}
  • To set custom overlay text for a 4:0 layout: {"transforms":{"layout": "4:0", "free_form_overlay_text":["Top left", "top right", "bottom left", "bottom right"]}}
  • To set 2:21 layout for normal participants, and 1:0 for streaming participants: {"transforms":{"layout": "2:21", "streaming": {"layout": "1:0"}}}
  • To enable the holding screen for streaming participants: {"transforms":{"streaming": {"waiting_screen_enabled": true}}}

clearallbuzz

This POST request lowers all raised hands.

Request: empty.

Response: the result is true if successful, false otherwise.

Participant functions

Within a conference, operations can be performed on participants, if the client has Host privileges.

These participant REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/participants/<participant_uuid>/<request>

where <node_address> is the Conferencing Node, <conference_alias> is an alias of the conference, and <participant_uuid> is the uuid of the participant you are controlling. Under this path comes the request, for example:

https://10.0.0.1/api/client/v2/conferences/meet_alice/participants/7f8bdd7f-2d39-4c3f-9236-3e95b21f21a8/disconnect

disconnect

This POST request disconnects a participant.

Request: empty

Response: the result is true if successful, false otherwise.

mute / unmute

These POST requests are used to mute or unmute a participant.

Request: empty.

Response: the result is true if successful, false otherwise.

allowrxpresentation / denyrxpresentation

These POST requests are used to enable or disable a participant from receiving the presentation stream. (Participants are enabled by default.)

Request: empty.

Response: the result is true if successful, false otherwise.

spotlighton / spotlightoff

These POST requests are used to enable or disable the "spotlight" on a participant.

The spotlight feature locks any spotlighted participants in the primary positions in the stage layout, ahead of any current speakers. When any participants have been spotlighted, the first one to be spotlighted has the main speaker position, the second one has the second position (leftmost small video, for example), and so on. All remaining participants are arranged by most recent voice activity, as is default. For more information, see Spotlighting a participant.

Request: empty.

Response: the result is true if successful, false otherwise.

unlock

This POST request lets a specified participant into the conference from the waiting room of a locked conference.

Request: empty.

Response: the result is true if successful, false otherwise.

dtmf

This POST request sends DTMF digits to the participant.

Request example:

{"digits": "1234"}

Request fields:

digits The DTMF digits to send.

Response: the result is true if successful, false otherwise.

calls

This POST request upgrades this connection to have an audio/video call element. There are two variants of this request, depending upon whether a WebRTC or RTMP call is to be established.

WebRTC

Request example to add a WebRTC element:

{"call_type": "WEBRTC", "sdp": "..."}

Request fields:

call_type "WEBRTC" for a WebRTC call.
sdp Contains the SDP of the sender.
present Optional field. Contains "send" or "receive" to act as a presentation stream rather than a main audio/video stream.

Response example (WebRTC):

{"status": "success", "result": {
"call_uuid": "50ed679d-c622-4c0e-b251-e217f2aa030b",
"sdp": "..."}}

The response contains the SDP of the Pexip node, and a call_uuid. This call_uuid is used to control the call. The ack function must be called on this call_uuid in order to start media after the SDP has been exchanged and ICE has been completed.

RTMP

Request example to add an RTMP element:

{"call_type": "RTMP"}

Request fields:

call_type "RTMP" for an RTMP call.
present Optional field. Contains "send" or "receive" to act as a presentation stream rather than a main audio / video stream.
streaming Optional field. Set to "true" if this is to be treated as a streaming participant for recording purposes.
bandwidth Optional field. If supplied it provides a maximum incoming / outgoing bandwidth in kbps.

Response example (RTMP):

{"status": "success", "result": {
"call_uuid": "50ed679d-c622-4c0e-b251-e217f2aa030b", 
"url": "rtmp://10.0.0.1:40002/pexip/50ed679d-c622-4c0e-b251-e217f2aa030b", 
"secure_url": "rtmps://hostname.domain:40003/pexip/50ed679d-c622-4c0e-b251-e217f2aa030b"}}

The response contains RTMP URLs that can be connected to by the client – both an insecure (rtmp://) and secure (rtmps://) variant. The RTMPS URL is only returned if a SIP TLS FQDN is configured for the Conferencing Node, and requires a valid TLS certificate to be installed on the Conferencing Node.

overlaytext

Changes the participant name overlay text. The text is only applied if overlay text is enabled on a VMR. It can also change the text of an audio-only participant.

Request example:

{"text": "The Dude"}

Request fields:

text Text to use as the participant name overlay text.

Response: the result is true if successful, false otherwise.

role

Changes the role of the participant.

Request example:

{"role": "chair"}

Request fields:

role "chair" = Host participant; "guest" = Guest participant

Response: the result is true if successful, false otherwise.

fecc

Send Far End Camera Control messaging to the participant.

Note that this does not send FECC to all participants; it can either be used in a gateway call or be sent to a specific participant identified by the target UUID (as seen in the participant list).

Request fields:

action Either "start", "stop", or "continue".
target UUID of the target participant (from the participant list). Leave undefined for a gateway call.
movement

An array of movements, consisting of:

axis Either "pan", "tilt", or "zoom".
direction Use "left", "right" for pan; "up", "down" for tilt; or "in", "out" for zoom.

Which means that you could, for example, send a command to pan, tilt and zoom at the same time.

timeout The duration for which to send the signal. Recommended values are 1000 (1 second) for initial "start" message; 200 for "continue" messages.

Request example:

{"action": "start", "movement": [{"axis": "pan", "direction": "left"}, {"axis": "zoom", "direction": "in"}], "timeout": 1000};

Response: the result is true if successful, false otherwise.

buzz

This POST request raises a participant's hand.

Request: empty

Response: the result is true if successful, false otherwise.

clearbuzz

This POST request lowers a participant's hand.

Request: empty

Response: the result is true if successful, false otherwise.

transfer

Transfers a participant to another conference.

The target conference is identified by the alias in "conference_alias", and they will have the specified "role". If the target is PIN-protected, the PIN for the target role must be specified in the "pin" field.

Request example:

{"role": "guest", "conference_alias": "meet@example.com", "pin": “1234"}

Request fields:

role Role can be "guest" or "chair" (Host).
conference_alias Target conference alias.
pin PIN code for the specified role at the specified conference, if required.

Response: the result is true if successful, false otherwise.

avatar.jpg

This GET request obtains the image to display to represent a conference participant or directory contact.

Call functions

Using the call_uuid, further operations can be undertaken on the calls as part of the nominated participant.

These call REST URIs take the format:

https://<node_address>/api/client/v2/conferences/<conference_alias>/participants/<participant_uuid>/calls/<call_uuid>/<request>

where <node_address> is the Conferencing Node, <conference_alias> is an alias of the conference, <participant_uuid> is the uuid of the participant, and <call_uuid> is the uuid of the call you are controlling. Under this path comes the request, for example:

https://10.0.0.1/api/client/v2/conferences/meet_alice/participants/7f8bdd7f-2d39-4c3f-9236-3e95b21f21a8/calls/c34f35f-1060-438c-9e87-6c2dffbc9980/disconnect

ack

This POST request starts media for the specified call (WebRTC calls only).

Request: empty.

Response: empty.

disconnect

This POST request disconnects the specified call.

Request: empty.

Response: the result is true if successful, false otherwise.

dtmf

This POST request sends DTMF digits to the specified participant.

Request example:

{"digits": "1234"}

Response: the result is true if successful, false otherwise.

Server-sent events

Clients can subscribe to an HTTP EventSource which feeds events from the conference as they occur.

To subscribe, open an HTTP connection to:

https://<node_address>/api/client/v2/conferences/<conference_alias>/events?token=<token_id>

where <node_address> is the Conferencing Node, <conference_alias> is an alias of the conference, and <token_id> is the session token, for example:

https://10.0.0.1/api/client/v2/conferences/meet_alice/events?token=123456

This allows the token to be specified on the URI, since custom headers cannot be added to Event Sources in browsers today. However, if headers can be added this will be accepted too, and the query parameter will not be required.

Each event contains an event name, and some events may contain a payload of data, which is a JSON object.

presentation_start

This marks the start of a presentation, and includes the information on which participant is presenting.

Example data:

{"presenter_name": "Bob", "presenter_uri": "bob@example.com"}

presentation_stop

The presentation has finished.

Data: none

presentation_frame

A new presentation frame is available at:

https://<node_address>/api/client/v2/conferences/<conference_alias>/presentation.jpeg

An alternative image at a higher resolution is also available at:

https://<node_address>/api/client/v2/conferences/<conference_alias>/presentation_high.jpeg

Note that these URLs require the token and the event ID of the presentation_frame event to be present as a header or a query parameter in order to download the presentation frame, for example:

https://10.0.0.1/api/client/v2/conferences/meet_alice/presentation.jpeg?id=MTAuNDQuOTkuMl8xOA==&token=b3duZXI9T...etc...2FmGzA%3D

Data: none

participant_create

A new participant has joined the conference.

The JSON object fields include:

buzz_time

A Unix timestamp of when this participant raised their hand, otherwise zero.

call_direction Either "in" or "out" as to whether this is an inbound or outbound call.
disconnect_supported Set to "YES" if the participant can be disconnected, "NO" if not.
display_name

The display name of the participant.

encryption "On" or "Off" as to whether this participant is connected via encrypted media.
external_node_uuid The UUID of an external node e.g. a Skype for Business / Lync meeting associated with an external participant. This allows grouping of external participants as the UUID will be the same for all participants associated with that external node.
fecc_supported Set to "YES" if this participant can be sent FECC messages; "NO" if not.
has_media Boolean indicating whether the user has media capabilities.
is_audio_only_call Set to "YES" if the call is audio only.
is_external

Boolean indicating if it is an external participant, e.g. coming in from a Skype for Business / Lync meeting.

is_muted Set to "YES" if the participant is administratively muted.
is_presenting Set to "YES" if the participant is the current presenter.
is_streaming_conference Boolean indicating whether this is a streaming/recording participant.
is_video_call Set to "YES" if the call has video capability.
local_alias

The calling or "from" alias. This is the alias that the recipient would use to return the call.

mute_supported Set to "YES" if the participant can be muted, "NO" if not.
overlay_text Text that may be used as an alternative to display_name as the participant name overlay text.
protocol

The call protocol.

Values: "api", "webrtc", "sip", "rtmp", "h323" or "mssip".

(Note that the protocol is always reported as "api" when an Infinity Connect client dials in to Pexip Infinity.)

role

The level of privileges the participant has in the conference:

  • "chair": the participant has Host privileges
  • "guest": the participant has Guest privileges
rx_presentation_policy Set to "ALLOW" if the participant is administratively allowed to receive presentation, or "DENY" if disallowed.
service_type

The service type:

  • "connecting": for a dial-out participant that has not been answered
  • "waiting_room": if waiting to be allowed to join a locked conference
  • "ivr": if on the PIN entry screen
  • "conference": if in a VMR
  • "lecture" if in a Virtual Auditorium
  • "gateway": if it is a gateway call
  • "test_call": if it is a Test Call Service
spotlight

A Unix timestamp of when this participant was spotlighted, if spotlight is used.

start_time

A Unix timestamp of when this participant joined (UTC).

transfer_supported Set to "YES" if this participant can be transferred into another VMR; "NO" if not.
uuid

The UUID of this participant, to use with other operations.

uri

The URI of the participant.

vendor The vendor identifier of the browser/endpoint with which the participant is connecting.

Example data:

{
"buzz_time": 0,
"call_direction": "in",
"disconnect_supported": "YES",
"display_name": "Alice",
"encryption": "On",
"external_node_uuid": "",
"fecc_supported": "NO",
"has_media": false,
"is_audio_only_call": "NO",
"is_external": false,
"is_muted": "NO",
"is_presenting": "NO",
"is_streaming_conference": false,
"is_video_call": "YES",
"local_alias": "meet.alice",
"mute_supported": "YES",
"overlay_text": "Alice",
"presentation_supported": "NO",
"protocol": "api",
"role": "chair",
"rx_presentation_policy": "ALLOW",
"service_type": "conference",
"spotlight": 0,
"start_time": 1441720992,
"transfer_supported": "YES",
"uri": "Infinity_Connect_10.44.21.35",
"uuid": "50b956c8-9a63-4711-8630-3810f8666b04",
"vendor": "Pexip Infinity Connect/2.0.0-25227.0.0 (Windows NT 6.1; WOW64) nwjs/0.12.2 Chrome/41.0.2272.76"
}

participant_update

A participant's properties have changed.

Data: a full JSON object is supplied, as for participant_create.

participant_delete

A participant has left the conference.

Data: the JSON object contains the UUID of the deleted participant, for example:

{"uuid": "65b4af2f-657a-4081-98a8-b17667628ce3"}

participant_sync_begin / participant_sync_end

At the start of the EventSource connection, these two messages start and end the sending of the complete participant list in the form of participant_create events. This allows a participant that has been temporarily disconnected to re-sync the participant list.

conference_update

Conference properties have been updated. Currently, the only conference properties available are the lock status of the conference, whether Guests are muted, and if the conference has been started. For example:

{"locked": false, "guests_muted": false, "started": true}

layout

The stage layout has changed.

Data: an object containing the following fields:

view

The layout currently seen by the participant, including:

  • "1:0": main speaker only
  • "1:7": main speaker and up to 7 previous speakers
  • "1:21": main speaker and up to 21 previous speakers
  • "2:21": two main speakers and up to 21 previous speakers
  • "4:0": equal size layout, up to a maximum of 4 speakers
participants

An array of UUIDs for the participants, in order, starting from the main speaker position.

Example data:

{"view": "1:7",
"participants": ["a0196175-b462-48a1-b95c-f322c3af57c1", "65b4af2f-657a-4081-98a8-b17667628ce3”]}

message_received

A chat message has been broadcast to the conference.

Data: an object containing the following fields:

origin Name of the sending participant.
uuid UUID of the sending participant.
type MIME content-type of the message, usually text/plain.
payload Message contents.

Example data:

{"origin": "Alice", 
"type": "text/plain", 
"payload": "Hello World", 
"uuid": "eca55900-274d-498c-beba-2169aad9ce1f"}

stage

An update to the "stage layout" is available. This declares the order of active speakers, and their voice activity.

Data: an array of objects per active participant. Each participant has the following fields:

participant_uuid The UUID of the participant.
stage_index

The index of the participant on the "stage". 0 is most recent speaker, 1 is the next most recent etc.

vad

Audio speaking indication. 0 = not speaking, 100 = speaking.

Example data:

[
{"stage_index": 0, 
"participant_uuid": "a0196175-b462-48a1-b95c-f322c3af57c1", 
"vad": 0}, 
{"stage_index": 1, 
"participant_uuid": "65b4af2f-657a-4081-98a8-b17667628ce3", 
"vad": 0}
]

call_disconnected

This is sent when a child call has been disconnected (e.g. when a screensharing child call has been closed if presentation has been stolen by another participant).

Data: contains both the UUID of the child call being disconnected, and the reason for the disconnection if available, e.g.:

{"call_uuid": "50ed679d-c622-4c0e-b251-e217f2aa030b", 
"reason": "API initiated participant disconnect"}

disconnect

This is sent when the participant is being disconnected from the Pexip side.

Data: the reason parameter contains a reason for this disconnection, if available, e.g.:

{"reason": "API initiated participant disconnect"}

Other miscellaneous requests

Conferencing Node status (maintenance mode)

Load balancers can use the https://<node_address>/api/client/v2/status REST API command to check whether a Conferencing Node is in maintenance mode, for example:

https://10.0.0.1/api/client/v2/status

If the node is not in maintenance mode, it returns a 200 OK with the following JSON:

{
  "status": "success",
  "result": "OK"
}

If the node is in maintenance mode, it returns a 503 with the following JSON:

{
  "status": "failed",
  "result": "Maintenance mode"
}

Changelog

Changes in version 22:

  • New participant functions: fecc, buzz, clearbuzz.
  • New conference control function: clearallbuzz.
  • New fields in participant_create event: buzz_time, transfer_supported, fecc_supported, mute_supported, disconnect_supported.
  • New streaming parameter in transform_layout conference control option.

Changes in version 21:

  • New /api/client/v2/status command to check whether a Conferencing Node is in maintenance mode.
  • New transform_layout conference control option.

Changes in version 20:

None.

Changes in version 19:

None.

Changes in version 18:

The participant_create event has a new external_node_uuid parameter.

Changes in version 17:

After a presentation_frame event, you can now fetch a higher-resolution image presentation_high.jpeg as an alternative to presentation.jpeg.

Changes in version 16:

None.

Changes in version 15:

None.

Changes in version 14:

  • New service_type of “test_call”.
  • The dial command has a new text parameter.
  • New avatar.jpg participant GET request.

Changes in version 13:

None.

Changes in version 12:

  • New start_conference conference control function that allows you to start a conference and allow Guests in the "waiting room" into the meeting.
  • New transfer participant function.
  • The dial command has two new parameters: keep_conference_alive and remote_display_name; the protocol parameter can be "auto".

Changes in version 11:

  • New role participant function.
  • New layout event.
  • The dial command has new dtmf_sequence, source_display_name, source and call_type fields.
  • The participant_create event has new is_audio_only_call, is_external and is_video_call parameters.

Changes in version 10:

  • WebRTC and Skype for Business / Lync multiple participant records for the same participant are no longer seen; you now receive a single record per participant, so there is longer a participant_id value.
  • DTMF tones can be sent to a specific participant.
  • The request_token can be used to access a Pexip Virtual Reception.
  • The dial out command now returns an array of UUIDs.
  • The dial out command includes a streaming participant indicator.

More information

For more information about using this API, contact your Pexip representative.