Model Structure
The unique identifier of the stream
The status of the stream. Possible values:
finished, broadcasting, created, preparing, error, failedThe status of the playlist. Only applicable when type is
playlist. Possible values: finished, broadcasting, created, preparing, error, failedThe type of the stream. Possible values:
liveStream, ipCamera, streamSource, VoD, playlistThe publish type of the stream. Read-only field updated on the server side. Possible values:
WebRTC, RTMP, Pull, SRTThe name of the stream
The description of the stream
Video filter controlled by the user. Default value is true
The date when record was created in milliseconds (unix timestamp)
Planned start date in unix timestamp (seconds). Stream will only accept publishing when current time is greater than this value. Applicable for RTMP and WebRTC streams
Planned end date in unix timestamp (seconds). Stream will only accept publishing when current time is less than this value. Applicable for RTMP and WebRTC streams
The duration of the stream in milliseconds
The list of endpoints such as Facebook, Twitter or custom RTMP endpoints. See Endpoint model
The list of broadcasts in the playlist. This list has values when the broadcast type is
playlistIdentifier of whether stream is public or not
Identifier of whether stream is 360 degree video or not
The URL that will be notified when stream is published, ended and muxing finished. Receives POST requests with parameters:
id, action (liveStreamStarted, liveStreamEnded, vodReady), vodName, streamName, categoryThe category of the stream
The IP address of the IP camera or publisher
The username of the IP camera
The password of the IP camera
The quality of the incoming stream during publishing
The speed of the incoming stream. For better quality and performance it should be around 1.00
The stream URL for fetching stream. Should be defined for IP cameras or cloud streams
The origin address server broadcasting
MP4 muxing enabled status. Values:
1 (enabled), -1 (disabled), 0 (no settings for the stream)WebM muxing enabled status. Values:
1 (enabled), -1 (disabled), 0 (no settings for the stream)Initial time to start playing in milliseconds. Can be used in VoD files or stream sources with seek support
Number of subtracks allowed to be created for the broadcast. Useful for limiting conference attendees. Value of
-1 means no limitThe expire time in milliseconds. If this value is 10000, the broadcast should be started within 10 seconds after creation. Value of
0 means stream will never expireThe RTMP URL where to publish live stream to
True when a broadcast is created directly through streaming without being created earlier through REST service. False by default
Viewer Statistics
The number of HLS viewers of the stream
The number of DASH viewers of the stream
The number of WebRTC viewers of the stream
The number of RTMP viewers of the stream
Viewer Limits
Number of allowed maximum WebRTC viewers for the broadcast. Value of
-1 means no limitNumber of allowed maximum HLS viewers for the broadcast. Value of
-1 means no limitNumber of allowed maximum DASH viewers for the broadcast. Value of
-1 means no limitStream Quality Metrics
Publishing start time of the stream in unix timestamp milliseconds
The total bytes received until now
The received bitrate per second
Width of the incoming stream in pixels
Height of the incoming stream in pixels
Number of packets ingested and waiting for processing. Generally applies to RTMP, SRT ingest and Stream Source pull. This number should be low (less than 10)
Number of frames ingested and waiting for encoding. This number should be low (less than 10)
Number of dropped packets in total while ingesting. Packets can be dropped if server is loaded or WebRTC connectivity is not healthy. Should be zero in perfect scenario
Number of dropped frames in total while transcoding. If there is adaptive bitrate and server is loaded, encoder may not transcode in real-time and frames can be dropped to prevent memory crash
WebRTC Quality Metrics
Lost packets’ ratio in WebRTC ingest. This value should be around 0.01 (1%)
Number of packets lost in WebRTC ingest
Jitter in milliseconds for WebRTC ingest. This value should be less than 50ms. Lower is better
Round Trip Time in milliseconds for WebRTC ingest. This value should be less than 50ms. Lower is better
Additional Metadata
User-Agent of the publisher
Remote IP address of the stream publisher
Latitude of the broadcasting location
Longitude of the broadcasting location
Altitude of the broadcasting location
Multi-track Support
If this broadcast is a track of a WebRTC stream, this variable holds the ID of that main stream
Absolute start time in milliseconds (unix timestamp). Used for measuring absolute latency
Playlist Configuration
Current playing index for playlist types
Identifier of playlist loop status. If true, playlist loops infinitely. If false, playlist plays once and finishes
Advanced Configuration
Name of the subfolder that will contain stream files
Metadata field for custom usage
Broadcast role for selective playback
The HLS parameters of the broadcast
Identifier of whether stream should start/stop automatically. Effective for Stream Sources/IP Cameras. If there is no viewer after certain amount of seconds, it will stop. If a user wants to watch, it will start automatically
The list of encoder settings for adaptive bitrate streaming
Maximum idle time in seconds without updating the broadcast. If broadcast is not updated for this duration, webhook is fired
Example JSON
Related API Endpoints
- Create Broadcast -
POST /v2/broadcasts - Get Broadcast -
GET /v2/broadcasts/{id} - Update Broadcast -
PUT /v2/broadcasts/{id} - Delete Broadcast -
DELETE /v2/broadcasts/{id} - List Broadcasts -
GET /v2/broadcasts/list/{offset}/{size} - Get Broadcast Statistics -
GET /v2/broadcasts/{id}/stats
