- Quick deployment and testing of the server
- WCS update to 5.2
- Installing and testing
- Removing
- Working with the server
- Architecture
- Core settings
- Settings file flashphoner.properties
- Settings file loadbalancing.xml
- Settings file log4j.properties
- Settings file watchdog.properties
- Settings file watchdog.log4j.properties
- Settings file rtsp.auth
- Key store wss.jks
- File flashphoner.serverid
- Certificate myflashphoner-ca
- WCS.version
- Settings file wcs-core.properties
- SDP settings files
- Settings file database.yml
- SSL certificates management
- Logging
- Monitoring
- Using SSH tunnel
- Connecting from JConsole
- Connecting from Visual VM
- Connecting from Java Mission Control
- WCS integration to Prometheus
- WCS to Zabbix integration
- Load and resource usage information
- Information about errors and configuration parameters
- Stream parameters monitoring with REST API
- Server health check
- Centralized stream statistics and CDN events collection to MySQL DB
- Centralized server data logging to ClickHouse DB
- Command line interface
- CLI v 2
- Network traffic analysis
- Server domain name support as external address
- HTTP interfaces access restriction
- Websocket connections restriction by domain
- Stream publishing and playback restriction by name
- Websocket client URI configuration
- Media ports management
- Server ports configuration to accept client connections
- Thread pools tuning
- Diagnostics and troubleshooting
- Server tuning recommendations
- Memory management in Java
- Server performance testing
- Before moving to production
- Server functions
- Streaming video functions
- Stream capturing and publishing to the server
- From a web camera in a browser via WebRTC
- From the computer screen (screen sharing) in a browser via WebRTC
- From an HTML5 Canvas element (whiteboard) in a browser via WebRTC
- By means of Flash Player via RTMP
- Using RTMP encoder
- From an IP camera via RTSP
- From another server via RTMP
- RTMP stream capturing by re-publishing from another RTMP server
- From another WCS server via WebRTC
- From an Android mobile app via WebRTC
- From an iOS mobile app via WebRTC
- Capturing VOD from a file
- RTP stream publishing via RTSP
- MPEG-TS RTP stream publishing
- WebRTC publishing via WHIP
- Automatic streams capture on server start
- RTSP-interleaved stream capture from dump file
- Managing camera and microphone
- Bitrate management when capturing WebRTC stream in browser
- Key frames management while capturing WebRTC in browser
- Published stream normalizing
- Jitter buffer and frames collection in stream published
- Captured stream management
- Stream recording
- Stopping the video stream on the server side
- Taking a PNG snapshot of the stream
- Stream decoding
- Stream transcoding
- Stream watermarking
- FPS filter
- Using AAC codecs
- WebRTC stream picture rotation
- Minimal publishing bitrate control
- Decoded frames interception and handling
- Decoded frames interception and hangling with OpenCV
- Server audio processing
- Injecting one stream into another
- Playing a video stream from the server
- In a browser via WebRTC
- In a browser using Flash Player via RTMP
- In a browser via MSE
- In a browser via Websocket + Canvas, WSPlayer
- In a browser via HLS
- In an Android mobile application via WebRTC
- In an iOS mobile application via WebRTC
- In a player via RTSP
- In a player via RTMP
- In a browser with Delight Player
- In a browser via WebRTC ABR
- Stream availability for playback
- Publishing and playing stream via WebRTC over TCP
- Publisher and player channel quality control
- WebRTC traffic encryption hardware acceleration
- DTLS support for WebRTC streaming
- WebRTC RTP bundle support
- IPv6 support for WebRTC
- Websocket traffic proxying for WebRTC publishing/playing
- H264 encoding profiles management
- Stream event passing to subscribers
- Republishing a video stream
- Working with chat rooms
- Stream capturing and publishing to the server
- Stream mixer functions
- SFU functions with Simulcast
- Streaming video CDN functions
- Streaming video and SIP integration functions
- Stream capturing from a SIP call
- Redirecting a SIP call to a stream (SIP as Stream function)
- Republishing a SIP call to an RTMP stream to the given server (SIP as RTMP function)
- Republishing incoming SIP call to a stream
- Redirecting a stream to a SIP call using /call/inject_stream
- Redirecting an audio file to a SIP call using /call/inject_sound
- SIP call stream raw audio recording
- WebRTC-SIP gateway functions
- Hardware acceleration support for video transcoding
- Working through Firewall
- Load testing
- Load balancing
- Web SDK
- SFU SDK
- iOS SDK
- Android SDK
- Raw WebSocket API
- REST API
- REST Hooks
- REST Methods
- Invoking a REST method
- Authorization on backend
- Four types of REST methods
- The list of methods and their parameters
- restClientConfig object description
- Controlling REST methods
- The match between client invocations and REST methods
- REST methods object fields
- Event statuses
- Data exchange - OnDataEvent
- Error handling
- Sending custom error message to a client
- Using REST hook to authorize user by domain
- WCS in Amazon EC2
- WCS on Digital Ocean
- WCS on Google Cloud Platform
- WCS in Yandex.Cloud
- WCS in Docker
- WCS in Equinix Metal (ex Packet.Net)
- WCS in WSL 2
- Billing
- Technical support
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