IrisRtcSession

IrisRtcSession

new IrisRtcSession()

Constructor for IrisRtcSession.
This class maintains APIs required for creating session for video call, audio call. Handles session related events and callbacks.

Methods

addPSTNParticipant(roomId, toTN, toRoutingId)

This API allows user to add a participant to the ongoing PSTN call

Parameters:
Name Type Description
roomId string

(MANDATORY) Unique id for the room
toTN E.164

(MANDATORY) Telephone number of callee in e.164 format
toRoutingId string

(MANDATORY) Routing id of the callee

addStream(localStream)

Add Stream to webrtc based on the call

Parameters:
Name Type Description
localStream object

Local stream to be added to conference

audioMuteToggle(roomId)

Mute or Unmute local audio

Parameters:
Name Type Description
roomId string

Unique Id for participants in a room

createChatSession(config, connection)

This API creates new Iris chat only session, where number of participants share chat messages over a conference.

Parameters:
Name Type Description
config json

(MANDATORY) Session config params requied to create a chat session
connection object

(MANDATORY) Rtc connection object

createSession(sessionConfig, connection, irisRtcStream)

Creating new session, A session represents a real time audio or video session where number of participants are allowed to share their video or audio with each other to create a conference. It can be a one-to-one conversation or a multiparty conference.

There are two types of session, Anonymous and Non-Anonymous session. In case of anonymous session, a participant may not be a registered user with Iris Platform, he uses a random id as his unique identfier known as routing Id.

Incase of a Non-anonymous session, a participant must be a registered user with Iris Platform, Iris will assign a routing id to the user.

A session created with this API can be Audio, Video or Screen Share

This API takes three attributes a sessionConfig JSON object as described below, a connection object which is an instance of IrisRtcConnection and a irisRtcStream object which is an instance of IrisRtcStream.

Parameters:
Name Type Description
sessionConfig json

(MANDATORY) A json object having all parameters required to create a session

Properties
Name Type Description
type string

(MANDATORY) Call type 'chat', 'video' or 'audio'
roomId string

Unique id of the room : (MANDATORY) for non-anonymous calls (MANDATORY) for anonymous calls if roomname is not passed
irisToken string

(MANDATORY) Iris JWT token
routingId string

(MANDATORY) Routing id of the caller
useAnonymousLogin boolean

(MANDATORY) true for anonymous user calls
useBridge boolean

(MANDATORY) true for videobridge calls
roomName string

(MANDATORY) if roomID for anonymous call is not passed
stream string

(OPTIONAL) 'sendrecv' - Local participant share his media and receives media from other participants as well
'sendonly' - Local particpant will not receive any media from other participants, but share his media.
'recvonly' - Local participant receives media from other participant but he can't share his own media
'nostream' - Local participant will neither sends nor recieves any media.
traceId string

(OPTIONAL) Unique time based UUID to identify call uniquely
sessionType string

(OPTIONAL) Session type 'create'
name string

(OPTIONAL) User name of the caller
publicId string

(OPTIONAL) Public id of the caller
maxParticipants integer

(OPTIONAL) A integer value to limit number of participants in an anonymous room only if roomname is provided
toDomain string

(OPTIONAL)
useUnifiedPlan boolean

(OPTIONAL) true for using unified-plan in sdp or false for using plan-b in sdp
connection object

(MANDATORY) IrisRtcConnection object
irisRtcStream object

(MANDATORY) Instance of IrisRtcStream after stream is created.

createSessionWithTN(sessionConfig, connection, irisRtcStream)

Creating new PSTN session, A real time audio session with a PSTN participant of outside world or another iris participant.

To create a PSTN session, user must be a registered with Iris and must have a valid Telephone number associated with the account.

PSTN session supports features like call hold/unhold, swap and merge.

This API takes three attributes a sessionConfig JSON object as described below, a connection object which is an instance of IrisRtcConnection and a irisRtcStream object which is an instance of IrisRtcStream where local audio stream is created.

Parameters:
Name Type Description
sessionConfig json

(MANDATORY) A json object having all parameters required to create a room

Properties
Name Type Description
type string

(MANDATORY) Call type must be pstn
irisToken string

(MANDATORY) Iris JWT token
routingId string

(MANDATORY) Routing id of the caller
useAnonymousLogin boolean

(MANDATORY) false
useBridge boolean

(MANDATORY) true, PSTN Call will be a videobridge call
toTN string

(MANDATORY) Telephone number of caller in E.164 fromat (Mandatory for pstn calls)
fromTN string

(MANDATORY) Telephone number of callee in E.164 fromat (Mandatory for pstn calls)
traceId string

(OPTIONAL) Unique time based UUID to identify call uniquely
toDomain string

(OPTIONAL)
sessionType string

(OPTIONAL) Session type 'create'
publicId string

(OPTIONAL) Public id of the caller
connection object

(MANDATORY) IrisRtcConnection object
irisRtcStream object

(MANDATORY) IrisRtcStream instance after the stream is created.

downgradeToChat(downgradeConfig, notificationPayload)

Downgrade existing video/audio session to chat session

Parameters:
Name Type Description
downgradeConfig json

(MANDATORY) user config for downgrading a session to chat
notificationPayload json

(MANDATORY) notification data for joining a chat session

endSession()

API to end the session

getRealTimeStats(roomId)

This API gets real time media stats

Parameters:
Name Type Description
roomId string

(MANDATORY) Unique Id for participants in a room

joinChatSession(config, connection, notificationPayload)

This API allows to join a chat only session when participant gets an invitation for chat session.

Parameters:
Name Type Description
config json

(MANDATORY) A json object having all parameters required to create a room
connection object

(MANDATORY) IrisRtcConnection object
notificationPayload json

(MANDATORY) Notification payload having roomid, roomtoken and roomtokenexpirytime

joinSession(sessionConfig, Connection, irisRtcStream, notificationPayload)

This API is called to join a Iris Rtc session incase of non-anonymous call. For incoming calls client should pass notification information having required parameters to join session.
notification payload sent to this API must have roomid, roomtoken and roomtokenexpirytime.

Parameters:
Name Type Description
sessionConfig json

(MANDATORY) A json object having all parameters required to create a room

Properties
Name Type Description
type string

(MANDATORY) Call type must be pstn
irisToken string

(MANDATORY) Iris JWT token
routingId string

(MANDATORY) Routing id of the caller
useAnonymousLogin boolean

(MANDATORY) false, join session is called for non anonymous calls only
useBridge boolean

(MANDATORY) true for videobridge calls
Connection object

(MANDATORY) IrisRtcConnection object
irisRtcStream object

(MANDATORY) Instance of IrisRtcStream after stream is created.
notificationPayload json

(MANDATORY) Notification payload having roomid, roomtoken and roomtokenexpirytime

muteParticipantAudio(roomId, jid, mute)

Mute remote participant's audio

Parameters:
Name Type Description
roomId string

(MANDATORY) Unique Id for participants in a room
jid string

(MANDATORY) Remote participant's id
mute boolean

(MANDATORY) true -> mute, false -> unmute

muteParticipantVideo(roomId, jid, mute)

Mute remote participant's video

Parameters:
Name Type Description
roomId string

(MANDATORY) Unique Id for participants in a room
jid string

(MANDATORY) Remote participant's id
mute boolean

(MANDATORY) true -> mute, false -> unmute

onChatAck(roomId, chatAckJson)

Acknowledgement API for chat messages sent

Parameters:
Name Type Description
roomId string

Unique to participants
chatAckJson json

Unique id for the each chat message sent

Properties
Name Type Description
statusCode string

Status code for sent message
statusMessage string

Status message for the message sent
id string

Unique Id of the message sent
roomId string

Room id
rootNodeId string

Root node id for the message - If message is sent
childNodeId string

Child node id for the meesage - If message is sent

onChatMessage(roomId, chatMsgJson)

API to receive the chat messages from other participants

Parameters:
Name Type Description
roomId string

Unique to participants
chatMsgJson json

Chat message json from the other participant

Properties
Name Type Description
message string

Chat message from participant
from string

Remote participant's unique id
roomId string

Unique to participants
rootNodeId UUIDv1

Root node id for the message
childNodeId UUIDv1

Child node id for the meesage

onChatState(roomId, participantJid, chatState)

This callback is called when chat state of a participant is changed

Parameters:
Name Type Description
roomId string

Room ID
participantJid string

Participant Jid
chatState string

chat state indicator

onDataChannelMessage()

When a text message is received through datachannel

onMessage(roomId, colibriClass, obj)

Parameters:
Name Type Description
roomId string

RoomId
colibriClass string

Data channel colibriclass
obj json

info

onRemoteStream(roomId, stream)

Parameters:
Name Type Description
roomId string

Room Id of current session.
stream object

Media stream of remote participant

onRemoteStreamRemove(roomId, participantJid)

Callback when a stream is removed from the session.

Parameters:
Name Type Description
roomId string
participantJid string

onRoomLocked(roomId, lock, rejoin)

Parameters:
Name Type Description
roomId string

Room Id of the session locked/unlocked
lock boolean

lock/unlock status of the room, true - locked, false - unlocked
rejoin boolean

true - room is locked with rejoin as true
false - room is locked with rejoin as false

onSessionAdministratorJoined(roomId, adminJid, lock, rejoin)

This callback is triggered when room Admin is joined.

Parameters:
Name Type Description
roomId string

Room Id of the current session
adminJid string

Jid of the Administrator
lock boolean

lock/unlock status of the room, true - locked, false - unlocked
rejoin boolean

true - room is locked with rejoin as true
false - room is locked with rejoin as false

onSessionAdministratorLeft(roomId, adminJid)

This is callback is triggered when a room admin leaves, and session will be ended as soon as Admin leaves the room.

Parameters:
Name Type Description
roomId string

Room Id of the session where admin is left the room
adminJid string

Jid of the admin

onSessionConnected(roomId)

Callback to inform session is connected successfully

Parameters:
Name Type Description
roomId string

Room Id

onSessionCreated(roomId, traceId, msg)

Callback for succeffully created session

Parameters:
Name Type Description
roomId string

roomId created
traceId string

Identifies each session uniquely
msg string

A message string for any warning from backend, ex. Max participants, if max participants value is already set by someone, it can't be overrriden but session is still created.

onSessionDataChannelOpen(roomId)

This callback will be sent when data channel is opened with video bridge

Parameters:
Name Type Description
roomId string

onSessionEnd(roomId)

Callback to inform the session id for the session ended

Parameters:
Name Type Description
roomId string

Room Id unique to participants

onSessionError(roomId, errorCode, errorMessage)

Callback to inform about the session relarted errors

Parameters:
Name Type Description
roomId string

Room ID
errorCode string

Error code
errorMessage string

Error message

onSessionJoined(roomId, myJid)

onSessionJoined is called when caller joins session.

Parameters:
Name Type Description
roomId string

RoomId to which user has joined
myJid string

Jid of the caller

onSessionParticipantAudioMuted(roomId, jid, audioMute)

Called when participant's auido is muted or unmuted

Parameters:
Name Type Description
roomId string

Room Id of curret session
jid string

Unique jid of the participant
audioMute string

Status of audio. True - Muted. False - Not muted

onSessionParticipantJoined(roomId, participantJid)

Callback to inform remote participant joined successfully

Parameters:
Name Type Description
roomId string

Room name of the participant joined
participantJid string

Unique Jid of the remote participant

onSessionParticipantLeft(roomId, participantJid, closeSession)

Callback to inform remote participant has left the session

Parameters:
Name Type Description
roomId string

Room name of the participant joined
participantJid string

Unique Jid of the remote participant
closeSession boolean

Boolean value to close the session if all remote participants leave room

onSessionParticipantNotResponding(roomId, participantJid)

Callback for notifying client of about a not responding participant in the room

Parameters:
Name Type Description
roomId string

Room id
participantJid string

Unique Jid of the remote participant

onSessionParticipantVideoMuted(roomId, jid, videoMute)

Called when participant's video is muted or unmuted

Parameters:
Name Type Description
roomId string

Unique Id for participants
jid string

Unique jid of the participant
videoMute string

Status of video. True - Muted. False - Not muted

onSessionRejoin(roomId)

Called with session is trying to rejoin incase of black screen

Parameters:
Name Type Description
roomId string

Room id of the current session

onSessionRestartFailed(roomId)

This is called when black screen is seen and tried peer connection restart to bring back video but still if we see black screen then report via this callback

Parameters:
Name Type Description
roomId string

RoomId of the session

onSessionScreenShareStopped(roomId, message)

Called when moderator rights are revoked and screen share is stopped

Parameters:
Name Type Description
roomId string

Room Id of the session
message string

Information on why screen share is stopped.

onSessionSIPStatus(roomId, jid, status)

Different states of a PSTN Call

Parameters:
Name Type Description
roomId string

Unique Id for room
jid string

Unique jid of user
status string

status of pstn call. Possible states initializing, connecting, ringing, connected, hold and disconnected

onSessionStreamReceiveComplete(roomId, toJid)

Called when remote participants video is received and answer is sent

Parameters:
Name Type Description
roomId string

Room Id of current session.
toJid string

toJid

onSessionTypeChange(roomId, participantJid, type)

This is called when a participant changes type of the call ex. From chat > video or video > chat, callType will be "video" for a video call "audio" for a audio call "chat" for a chat session

Parameters:
Name Type Description
roomId string

Unique to participants in the room
participantJid string

Jid of the participant whose session type has changed
type string

Session type changed to ex. video, chat

onUserProfileChange(roomId, jid, profileJson)

Called when there is a change in user profile. ex. Dispaly name

Parameters:
Name Type Description
roomId string

Unique Id for room
jid string

Unique jid of the user
profileJson json

Json with user profile

Properties
Name Type Description
displayName string

Name of the participant
pstnStatus string

Status of pstn call
participantRole string

Role of the participant in room, he can be a moderator or a participant

pstnHold(roomId, participantJid)

This API allows user put a PSTN call on hold

Parameters:
Name Type Description
roomId string

room id
participantJid string

Jid of the pstn participant

pstnMerge(roomId, firstParticipantJid, secondSession, secondParticipantJid)

This API allows user to merge two pstn calls

Parameters:
Name Type Description
roomId string

(MANDATORY) Room Id
firstParticipantJid string

(MANDATORY) Jid of the participant in first call
secondSession object

(MANDATORY) IrisRtcSession of the second participant
secondParticipantJid string

(MANDATORY) Jid of the participant in second call

pstnUnHold(roomId, participantJid)

This API allows user to unhold a PSTN call

Parameters:
Name Type Description
roomId string

(MANDATORY) Room id
participantJid string

(MANDATORY) Jid of the pstn participant

removeStream(localStream)

Removes streams from conference

Parameters:
Name Type Description
localStream object

Stream to be removed from the conference

screenShare(roomId, irisRtcStream, streamConfig, useGetDisplayMedia)

This API allows user to share desktop screen. Example for streamConfig to be sent while sharing screen

Example
var streamConfig = {

constraints: {
      audio: false,
      video: {
          mandatory: {
              chromeMediaSource: "desktop",
              chromeMediaSourceId: streamId,
              maxWidth: window.screen.width,
              maxHeight: window.screen.height,
              minFrameRate: 15,
              maxFrameRate: 15,
          },
          optional: []
      }
  },
  screenShare: true,
  useGetDisplayMedia: false
 }
Parameters:
Name Type Description
roomId string

Room Id
irisRtcStream object

(MANDATORY) IrisRtcStream object
streamConfig json

(MANDATORY) Stream config json example as mentioned above

Properties
Name Type Description
constraints json

(MANDATORY) Media Constraints

Properties
Name Type Description
audio string

(MANDATORY) Media constrainsts for audio
video string

(MANDATORY) Media constrainsts for video
screenShare boolean

(MANDATORY) True if it is a screen share call
useGetDisplayMedia boolean

(OPTIONAL) true if user wants to use new getDisplayMedia of chrome for screen share

selectParticipant(roomId, participantId)

Elects the participant with the given id to be the selected participant in order to receive higher video quality.

Parameters:
Name Type Description
roomId string

(MANDATORY) Room ID
participantId string

(MANDATORY) the identifier of the participant

sendChatMessage(roomId, id, message, topic)

This API is called to send a chat message.

Parameters:
Name Type Description
roomId string

(MANDATORY) Room ID
id string

(MANDATORY) Unique Id for each message sent.
message string

(MANDATORY) Chat message to be sent
topic string

(OPTIONAL) Notification topic that user subscribed for chat Format: appdomain/type, ex: abcd.comcast.com/chat

sendChatState(roomId, chatState)

This API is called to set chat state like active, composing, paused inactive, gone

Parameters:
Name Type Description
roomId string

(MANDATORY) Room Id
chatState string

(MANDATORY) Permitted chatState values are active, composing, paused, inactive or gone

sendDataChannelMessage(roomId, msg, to)

Send a message through data channel to a participant or to all participants in conference

Parameters:
Name Type Description
roomId string

(MANDATORY) Room ID
msg string

(MANDATORY) Message to be sent
to string

(MANDATORY) Participant jid or to send to all - ""

sendDTMFTone(roomId, tone, duration, interToneGap)

This API allows user to send DTMF tones

Parameters:
Name Type Description
roomId string

(MANDATORY) Room Id
tone string

(MANDATORY) DTMF tone
duration integer

(OPTIONAL) duration of the tone. Should be with in (70, 6000)
interToneGap integer

(OPTIONAL) inter tone gap. Should be more than 50

setDisplayName(roomId, name)

API to set dispaly name for the user

Parameters:
Name Type Description
roomId string

(MANDATORY) Room ID
name string

(MANDATORY) Name for the user

setLastN(roomId, lastN)

Selects a new value for "lastN". The requested amount of videos are going to be delivered after the value is in effect. Set to -1 for unlimited or all available videos.

Parameters:
Name Type Description
roomId string

(MANDATORY) Room ID
lastN integer

(MANDATORY) the new number of videos the user would like to receive.

stopScreenShare(roomId)

Call this API once screen share is stopped. This is API removes desktop stream from conference.

Parameters:
Name Type Description
roomId string

(MANDATORY) Room Id

switchAudioStream(streamConfig)

API allows users to switch their mic during the call

Parameters:
Name Type Description
streamConfig json

(MANDATORY) Stream config json example as mentioned above

Properties
Name Type Description
streamType string

(MANDATORY) Stream Type must be audio
constraints json

(MANDATORY) Media Constraints

Properties
Name Type Description
video boolean

(MANDATORY) video boolean value must be false
audio json

(MANDATORY) audio value must be contain deviceId value

Properties
Name Type Description
deviceId json

(MANDATORY) Object for selected audio Input device

Properties
Name Type Description
exact string

(MANDATORY) deviceId for selected audio Input device

switchStream(roomId, irisRtcStream, streamConfig)

This API allows user to switch the stream between the camera, this can used for screen share with the constraints having source id for desktop sourceid

Parameters:
Name Type Description
roomId string

(MANDATORY) Room Id
irisRtcStream object

(MANDATORY) IrisRtcStream object
streamConfig json

(MANDATORY) Stream config json example as mentioned above

Properties
Name Type Description
streamType string

(MANDATORY) Type of stream audio or video
resolution string

(MANDATORY) Resolution for the video
constraints json

(MANDATORY) Media Constraints

Properties
Name Type Description
audio string

(MANDATORY) Media constrainsts for audio
video string

(MANDATORY) Media constrainsts for video
screenShare string

(MANDATORY) True if it is a screen share call

updateVideoElementRef(roomId, videoElement)

Update with the remote video's html element reference This function will check if video is playing or not and restart peer connection if video doesn't show up

Parameters:
Name Type Description
roomId string

roomId created
videoElement object

Remote video element reference

upgradeToAudio(irisRtcStream, upgradeConfig, notificationPayload)

Move to a audio session from chat session

Parameters:
Name Type Description
irisRtcStream object

(MANDATORY) IrisRtcStream instance after stream is created
upgradeConfig json

(MANDATORY) user config for upgrading to a audio session
notificationPayload json

(MANDATORY) notification data for joining a chat session

upgradeToVideo(irisRtcStream, upgradeConfig, notificationPayload)

Move to a video session from chat session

Parameters:
Name Type Description
irisRtcStream object

(MANDATORY) IrisRtcStream instance after stream is created
upgradeConfig json

(MANDATORY) user config for upgrading to a video session
notificationPayload json

(MANDATORY) notification data for joining a chat session

videoMuteToggle(roomId)

Mute or unmute local video

Parameters:
Name Type Description
roomId string

Unique Id for participants in a room