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Typedef

Static Public Summary
public

The current browser's infos

public

Audio codec presets to use for

public

Video codec presets to use for

public

communicationQuality presets to use for

public

The events supported by Invite#on

public

The events supported by Local#on.

public

The events supported by Reach#on

public

The events supported by Room#on

public

The events supported by Remote#on.

public

TURN/STUN server configuration

public

The available stream types

Static Public

public Browser: Object source

The current browser's infos

Properties:

NameTypeAttributeDescription
name string

current browser's name

version number
  • optional

current browser's version

minVersion number
  • optional

current browser's minimum supported version

compatible boolean

is current browser supported ?

public Codec/audio: Object source

Audio codec presets to use for

Properties:

NameTypeAttributeDescription
OPUS RegExp

Opus audio codec http://opus-codec.org/. Only choice for high-quality audio.

G722 RegExp

G.722 audio codec

G711 RegExp

G.711 audio codec

ISAC RegExp

iSAC audio codec. Good for voice data, but not suitable for high-quality audio streams.

ISAC_16 RegExp

iSAC audio codec (16kHz)

ISAC_32 RegExp

iSAC audio codec (32kHz)

ILBC RegExp

iLBC audio codec. For bad channels & low bandwidth.

ILBC RegExp

iLBC audio codec. For bad channels & low bandwidth.

public Codec/video: Object source

Video codec presets to use for

Properties:

NameTypeAttributeDescription
VP8 RegExp

VP8 is the only video codec officially supported by WebRTC

VP9 RegExp

VP8's successor.

H264 RegExp

MPEG-4 part 10. Only Firefox.

VP10 RegExp

Suited for UHD video. No support yet

H265 RegExp

H.264's successor. No support yet

public CommunicationQuality/bitrate: Object source

communicationQuality presets to use for

Properties:

NameTypeAttributeDescription
BAD RegExp

video bitrate: 256; audio bitrate: 16

LOW RegExp

video bitrate: 512; audio bitrate: 32

HIGH RegExp

video bitrate: 2048; audio bitrate: 128

public Events/Invite: Object source

The events supported by Invite#on

Properties:

NameTypeAttributeDescription
ACCEPTED string

Fired when the invite has been accepted

REJECTED string

Fired when the invite has been accepted

CANCELED string

Fired when the invite has been canceled

public Events/Local: Object source

The events supported by Local#on.

Properties:

NameTypeAttributeDescription
SUBSCRIBED string

Fired when the is subcribed by remote

WEBRTC_ERROR string

Fired when an error is raised on webrtc call.

public Events/Reach: Object source

The events supported by Reach#on

Properties:

NameTypeAttributeDescription
USER_ADDED string

Fired when a new user is registered

USER_CHANGED string

Fired when an existing user logs in or out or changes is status

USER_REMOVED string

Fired when a user is unregistered

ROOM_ADDED string

Fired when a room is created

ROOM_CHANGED string

Fired when a room status is changing

ROOM_REMOVED string

Fired when a room is closed definitely

INVITE_ADDED string

Fired when an invite is received

INVITE_CHANGED string

Fired when an invite status is modified

public Events/Room: Object source

The events supported by Room#on

Properties:

NameTypeAttributeDescription
PARTICIPANT_ADDED string

Fired when a new participant is added to the room. Does not mean he's connected but that he's invited to

PARTICIPANT_CHANGED string

Fired when a participant changes is status (enter/leaves the room)

PARTICIPANT_REMOVED string

Fired when a user leaves definitely or is banned

MESSAGE_ADDED string

Fired when a new instant message is sent to the room

MESSAGE_CHANGED string

Fired when an instant message is edited

MESSAGE_REMOVED string

Fired when an instant message is removed

STREAM_PUBLISHED string

Fired when a participant publishes a stream

STREAM_UNPUBLISHED string

Fired when a participant stops the publishing of his stream

public Events/Stream: Object source

The events supported by Remote#on.

Properties:

NameTypeAttributeDescription
MUTE string

Fired when the mute status of the stream changes

SIZE string

Fired when the size of the stream changes

WEBRTC_ERROR string

Fired when an error is raised on webrtc call.

public ICEServer: Object source

TURN/STUN server configuration

Properties:

NameTypeAttributeDescription
urls string | string[]

The server url. (stun:<ip|hostname>:<port> or turn:<ip|hostname>:<port?>)

username string
  • optional

The user name for this server

credential string
  • optional

The password for this server

public StreamTypes: Object source

The available stream types

Properties:

NameTypeAttributeDescription
AUDIO string
  • default: 'audio'

Audio communication service

VIDEO string
  • default: 'video'

Video communication service

AUDIO_VIDEO string
  • default: 'audio-video'

Audio and video communication service

SCREEN_SHARING string
  • default: 'screen-sharing'

Screen-sharing communication service