Web Call Server REST hooks (1.0.0)

Download OpenAPI specification:

This document describes REST Hooks. All hooks are called via POST with a JSON body.

Hook URLs can be configured in WCS_HOME/conf/database.yml.

Example hook:

http://localhost:8081/apps/DefaultApp/connect

This is the very first hook called when a user attempts to connect to the server, for example via the WebSDK API:

Flashphoner.createSession()

Example hook:

http://localhost:8081/apps/DefaultApp/publishStream

This hook is called within the client's session and indicates that the client is attempting to publish a stream.

Hook triggers:

Client calls — for example connect, publishStream, playStream.
Events on the WCS side — for example StreamStatusEvent (stream status changed), CallStatusEvent (call status changed).
Periodic calls initiated by WCS itself — for example StreamKeepAliveEvent.

Each hook has a behavior badge — a coloured label that summarises how the Backend response affects the operation — and belongs to a scope visible in the left sidebar (Connection / Stream / SIP).

Hooks behavior

Confirm — WCS calls the hook before continuing the client operation. A 2xx response lets WCS proceed; a 4xx/5xx response rejects the operation when restOnError=FAIL (connect, playHLS, and playRTSP always fail on hook errors). If the 2xx response body contains fields, they override the corresponding operation parameters when restPolicy=OVERWRITE and restOverwrite are configured; connect always uses returned connection data.

Notify — reports a state change or completed action after the fact. Hook errors do not roll back the server action; returned fields are ignored except where explicitly documented (for example, the ESTABLISHED state of ConnectionStatusEvent).

Control — periodically asks the Backend whether an ongoing operation may continue. A 2xx response lets it keep running; a 4xx/5xx response tells WCS to stop it. For example, a 2xx for StreamKeepAliveEvent keeps the stream alive; a 4xx stops the stream.

Configuring hook behavior

The default per-hook behavior described above can be customised per Connection via the restClientConfig field returned by the connect hook. The field is a map keyed by hook name; each entry tunes how WCS invokes that hook for the rest of the session:

restPolicy — DISABLE (skip the hook entirely), NOTIFY (fire-and-forget, ignore Backend response), or OVERWRITE (apply restOverwrite fields from the response back to the operation).
restExclude — comma-separated list of fields to strip from the request body before sending to the Backend.
restOverwrite — comma-separated list of fields whose returned values are applied back to the operation (effective only with restPolicy=OVERWRITE).
restOnError — LOG (continue with the original payload) or FAIL (reject the operation).

WCS stores the returned restClientConfig on the session and applies it to every subsequent hook of the matching name.

Example fragment of the connect hook response body:

{ "restClientConfig": { "publishStream": { "restExclude": "hasVideo,hasAudio", "restOverwrite": "createDate", "restOnError": "LOG", "restPolicy": "OVERWRITE" } } }

restClientConfig flow

Hooks by scope

Connection — client transport connections, sessions, in-session data, and diagnostics.

connect — Client opens a transport connection (WebSocket, RTMP/Flash, CDN).
OnDataEvent — Client sends an arbitrary data payload over the session.
sessionDebug — Client requests session debug data collection.
submitBugReport — Client submits a bug report.
pushLogs — Client uploads diagnostic logs.
BugReportStatusEvent — Bug report processing result.
ConnectionStatusEvent — Connection state changed (ESTABLISHED, DISCONNECTED, FAILED).
DataStatusEvent — Delivery status of a client data payload.
ErrorStatusEvent — Server-side error notification tied to the Connection.
SessionDebugStatusEvent — Session debug data collection result.

Stream — media streams: publish, play, snapshot, recording, transcoding, and stream events.

publishStream — Client starts publishing a media stream.
snapshot — Client requests a snapshot of a stream.
sendStreamEvent — Client sends a custom application event on a stream.
unPublishStream — Client stops publishing a media stream.
playStream — Client starts playing a WebRTC stream.
stopStream — Client stops a playback stream.
playRTSP — WCS pulls an RTSP stream for republishing.
playHLS — Client requests HLS playback.
availableStream — Client queries whether a stream is available for playback.
RecordingStatusEvent — Recording state changed (STARTED, STOPPED, FAILED).
StreamEvent — Payload-bearing stream event delivered to subscribers (mute/unmute, snapshot status, custom data, resize, bandwidth); originated by the publisher via sendStreamEvent or by WCS itself.
StreamStatusEvent — Stream state changed (PUBLISHING, PLAYING, STOPPED, FAILED).
StreamTranscodingGroupEvent — Transcoding group state changed.
StreamKeepAliveEvent — Periodic stream liveness probe.

SIP — SIP signaling: calls, transfers, registration, DTMF.

call — Client initiates an outgoing SIP/WebRTC call.
answer — Client accepts an incoming call.
sendDtmf — Client sends a DTMF tone during a call.
hold — Client puts a call on hold.
unhold — Client resumes a held call.
transfer — Client initiates a SIP transfer (REFER) to another party.
hangup — Client terminates an active call.
OnCallEvent — Inbound SIP call is ringing.
OnTransferEvent — Inbound transfer (REFER) request.
CallStatusEvent — Call state changed (RING, SESSION_PROGRESS, ESTABLISHED, FINISH, FAILED).
RegistrationStatusEvent — SIP registration state changed (REGISTERED, UNREGISTERED, FAILED).
TransferStatusEvent — Transfer attempt outcome.

DefaultApp

/apps/DefaultApp

By default, all hooks use /apps/DefaultApp as part of the path. Examples:

http://localhost:8081/apps/DefaultApp/call
http://localhost:8081/apps/DefaultApp/answer
...

To use a custom path, configure it in WCS_HOME/conf/database.yml, for example https://mycompany.com/mycompanyapi. WCS will then call the hooks at the custom URL:

https://mycompany.com/mycompanyapi/call
https://mycompany.com/mycompanyapi/answer
...

Connection

Category only. Groups REST hooks fired on events of a specific client Connection itself — opening the transport session, in-session data exchange, diagnostics.

connect Confirm

/apps/DefaultApp/connect

WCS calls this hook when a client or agent opens a transport connection, for example WebSocket, RTMP/Flash, or CDN. The hook is invoked before the connection is fully accepted, so the Backend can validate the request, attach custom session attributes, or reject the connection. The request body contains the client login, session identifiers, app key, custom data, and transport details. The Backend may return the same payload, return a modified payload to override fields visible to WCS, or reply with a 4xx/5xx status to refuse the connection. If the response is non-2xx, WCS closes the connection and the client receives a FAILED connection status.

connect hook flow

Request Body schema: application/json

One of

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
width	integer <int32> Default video width in pixels set by the client at connection time. Streams typically override this with their own resolution. Primarily used for SIP call media sessions
height	integer <int32> Default video height in pixels set by the client at connection time. Streams typically override this with their own resolution. Primarily used for SIP call media sessions
supportedResolutions	string Client resolution in WxH format
useDTLS	boolean Whether the connection uses DTLS for secure media transport. Always true for WebRTC connections, false for RTMP
authToken	string Session token for reconnection. If the WebSocket connection is lost, the client can reconnect with this token to rejoin the same session instead of creating a new one
status	string Enum: "NEW" "PENDING" "ESTABLISHED" "DISCONNECTED" "FAILED" "REGISTERED" "UNREGISTERED" Connection status: NEW (created), PENDING (waiting), ESTABLISHED (connected — WebSocket, RTMP, or other), DISCONNECTED (connection lost, session may be preserved with keepAlive), FAILED (error). SIP-specific: REGISTERED (SIP registration on PBX successful), UNREGISTERED (SIP registration removed)
	object Map of REST hook configurations. Explanation of the fields: • restExclude — fields that should be excluded from the REST request • clientExclude — fields that should be excluded from the REST response • restOverwrite — fields whose values must be overwritten • restOnError — behavior on error: LOG or FAIL • restPolicy — application policy: DISABLE, NOTIFY, or OVERWRITE Map structure: { "publishStream": RestMethodConfig }
logLevel	string Per-connection logging level. Controls the log verbosity for this specific client without affecting other connections or global server logs
keepAlive	boolean Whether the session is preserved after WebSocket disconnect. When true, the client can reconnect with authToken to rejoin the same session. When false, the session is removed immediately on disconnect
mediaProviders	Array of strings unique List of media providers supported by the connected client (e.g. WebRTC, Flash for RTMP clients like OBS, MSE, WSPlayer)
clientVersion	string Version string reported by the client at connection time (e.g. 2.0)
clientOSVersion	string Client platform string reported at connection time (e.g. navigator.appVersion in browser)
clientBrowserVersion	string Client User-Agent string reported at connection time
msePacketizationVersion	integer <int32> MSE packetization format version. Version 1 is the default legacy format, Version 2 is an optimized format. Only applicable to MSE media provider

Responses

Request samples

Payload

Content type

application/json

Example

WebRTC

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"width": 640,
"height": 480,
"supportedResolutions": "720x480",
"useDTLS": true,
"authToken": "d62440e7-0e38-4831-9fc1-7d3beb2a0d31",
"status": "ESTABLISHED",
"restClientConfig": {"publishStream": {"restExclude": "hasVideo,hasAudio",
"clientExclude": "sdp",
"restOverwrite": "createDate",
"restOnError": "LOG",
"restPolicy": "DISABLE"
}
},
"logLevel": "DEBUG",
"keepAlive": true,
"mediaProviders": ["WebRTC"
],
"clientVersion": "2.0",
"clientOSVersion": "5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"clientBrowserVersion": "Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"msePacketizationVersion": 2
}

Response samples

Content type

application/json

Example

WebRTC

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"width": 640,
"height": 480,
"supportedResolutions": "720x480",
"useDTLS": true,
"authToken": "d62440e7-0e38-4831-9fc1-7d3beb2a0d31",
"status": "ESTABLISHED",
"restClientConfig": {"publishStream": {"restExclude": "hasVideo,hasAudio",
"clientExclude": "sdp",
"restOverwrite": "createDate",
"restOnError": "LOG",
"restPolicy": "DISABLE"
}
},
"logLevel": "DEBUG",
"keepAlive": true,
"mediaProviders": ["WebRTC"
],
"clientVersion": "2.0",
"clientOSVersion": "5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"clientBrowserVersion": "Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"msePacketizationVersion": 2
}

OnDataEvent Confirm

/apps/DefaultApp/OnDataEvent

WCS calls this hook when a client sends an arbitrary data payload to the server via the WebSDK session.sendData(data) call. The request body contains the application-defined payload and the originating session identifiers.sendData(data) (status ACCEPTED). A hook failure rejects the data operation when restOnError=FAIL and the same Promise is rejected (status FAILED); otherwise WCS logs the failure and continues with the original payload.

OnDataEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
operationId	string Unique identifier (UUID) of this data operation, generated by the client. The client uses it to correlate the asynchronous response — delivered via DataStatusEvent — back to the originating sendData call.
payload	object Application-defined JSON payload that the client sent through the data channel; the contents are opaque to WCS and are forwarded as-is
status	string Outcome of the data operation. Populated by WCS after the sendData flow completes and delivered to the client via the WebSDK SEND_DATA_STATUS event and to the Backend via DataStatusEvent. Possible values: ACCEPTED (operation succeeded; the payload was forwarded to the target session(s)) or FAILED (operation failed; the failure reason is in the info field). In the OnDataEvent request body this field is typically empty (the operation has not yet been processed).
info	string Additional information
sessionIds	Array of strings Explicit list of signaling session IDs (Connection IDs) that should receive this data payload; when empty the payload is broadcast to every session in the originating app
allSessionIds	Array of strings Effective list of signaling session IDs (Connection IDs) that will receive this data payload, including the originating session and any explicitly added sessionIds

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"operationId": "2497358d-dd6e-4601-926c-cb2b6b12391d",
"payload": {"appKey": "defaultApp",
"body": "Hello alice",
"contentType": "application/imdn+xml",
"custom": {"property1": { },
"property2": { }
},
"from": "<tel:+447912345678;tag=ab12cd34ef56>\n",
"id": "a3f9c2b1-ff00-4e5d-9c77-112233445566",
"isImdnRequired": true,
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@127.0.0.1",
"origin": "http://localhost:8081",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"sipMessageRaw": "INVITE sip:...",
"status": "IMDN_NOTIFICATION_SENT",
"to": "<tel:+447912345678;tag=ab12cd34ef56>\n"
},
"status": "ACCEPTED",
"info": "msg: null",
"sessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
],
"allSessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"operationId": "2497358d-dd6e-4601-926c-cb2b6b12391d",
"payload": {"appKey": "defaultApp",
"body": "Hello alice",
"contentType": "application/imdn+xml",
"custom": {"property1": { },
"property2": { }
},
"from": "<tel:+447912345678;tag=ab12cd34ef56>\n",
"id": "a3f9c2b1-ff00-4e5d-9c77-112233445566",
"isImdnRequired": true,
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@127.0.0.1",
"origin": "http://localhost:8081",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"sipMessageRaw": "INVITE sip:...",
"status": "IMDN_NOTIFICATION_SENT",
"to": "<tel:+447912345678;tag=ab12cd34ef56>\n"
},
"status": "ACCEPTED",
"info": "msg: null",
"sessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
],
"allSessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
]
}

sessionDebug Confirm

/apps/DefaultApp/sessionDebug

WCS calls this hook when a client starts or stops session debugging, for example by calling Session.startDebug() or Session.stopDebug() in WebSDK. The feature can be disabled by the sessionDebugEnabled system property. The request body contains the session identifiers and the debug command.

sessionDebug hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
status	string Enum: "STARTED" "STOPPED" "FAILED" "PASSED" Session debug status
command	string Command. Supported commands: "start", "stop".

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"status": "STARTED",
"command": "start"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"status": "STARTED",
"command": "start"
}

submitBugReport Confirm

/apps/DefaultApp/submitBugReport

WCS calls this hook when a client submits a bug report through the session, for example by calling Session.submitBugReport() with a JSON report object. The request body contains report text, report type, and session identifiers.

submitBugReport hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
text	string Free-form description of the issue, supplied by the client when calling the WebSDK Session.submitBugReport({text, type}). On the server it is appended to the report file together with system info (login, OS, browser, log folder). Present in the submitBugReport hook payload; not used by BugReportStatusEvent.
type	string Optional bug report category supplied by the client when calling Session.submitBugReport({text, type}). When set to no_media (and client_dump_level < 2), WCS additionally starts a tcpdump capture of the current SIP call's media ports for no_media_dump_interval milliseconds (default 15000). Other values are accepted but trigger no special server-side behavior. Present in the submitBugReport hook payload; not used by BugReportStatusEvent.
file	string Filename of the report saved by WCS under WCS_HOME/logs/bugreports/{yyyy-MM-dd}/. Format: client-{sipLogin}-{yyyy.MM.dd.HH.mm.ss}-{epochMillis}.report. Populated by the server when delivering BugReportStatusEvent; not supplied by the client.

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"text": "Connection issue",
"type": "no_media",
"file": "client-sipLogin-2025.11.21.18.28.35-1763735315290.report"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"text": "Connection issue",
"type": "no_media",
"file": "client-sipLogin-2025.11.21.18.28.35-1763735315290.report"
}

pushLogs Confirm

/apps/DefaultApp/pushLogs

WCS calls this hook when session-scoped client log records are uploaded during a debug session and before the logs are stored. The request body contains the originating session identifiers and the log entries.

pushLogs hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
logs	string Log data

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"logs": "Log line"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"logs": "Log line"
}

BugReportStatusEvent Notify

/apps/DefaultApp/BugReportStatusEvent

WCS sends this notification hook after the submitBugReport flow has successfully saved the bug report file to WCS_HOME/logs/bugreports/{yyyy-MM-dd}/. The hook is not invoked if the file save fails. The request body contains the saved file name and the originating session identifiers. After the hook returns, WCS sends the result as a notifyBugReport WebSocket message to the client; the standard WebSDK does not expose this message to user-level handlers.

BugReportStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
text	string Free-form description of the issue, supplied by the client when calling the WebSDK Session.submitBugReport({text, type}). On the server it is appended to the report file together with system info (login, OS, browser, log folder). Present in the submitBugReport hook payload; not used by BugReportStatusEvent.
type	string Optional bug report category supplied by the client when calling Session.submitBugReport({text, type}). When set to no_media (and client_dump_level < 2), WCS additionally starts a tcpdump capture of the current SIP call's media ports for no_media_dump_interval milliseconds (default 15000). Other values are accepted but trigger no special server-side behavior. Present in the submitBugReport hook payload; not used by BugReportStatusEvent.
file	string Filename of the report saved by WCS under WCS_HOME/logs/bugreports/{yyyy-MM-dd}/. Format: client-{sipLogin}-{yyyy.MM.dd.HH.mm.ss}-{epochMillis}.report. Populated by the server when delivering BugReportStatusEvent; not supplied by the client.

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"text": "Connection issue",
"type": "no_media",
"file": "client-sipLogin-2025.11.21.18.28.35-1763735315290.report"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"text": "Connection issue",
"type": "no_media",
"file": "client-sipLogin-2025.11.21.18.28.35-1763735315290.report"
}

ConnectionStatusEvent Notify

/apps/DefaultApp/ConnectionStatusEvent

WCS sends this notification hook every time a client connection changes its state — for example when the session is ESTABLISHED, DISCONNECTED, or FAILED. The request body contains the connection identifiers, the new status, and fields specific to the connection type: WebRTC, Flash, or SIP. For the ESTABLISHED notification, fields returned by the Backend may be used in the client getUserData response.

ConnectionStatusEvent hook flow

Request Body schema: application/json

One of

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
width	integer <int32> Default video width in pixels set by the client at connection time. Streams typically override this with their own resolution. Primarily used for SIP call media sessions
height	integer <int32> Default video height in pixels set by the client at connection time. Streams typically override this with their own resolution. Primarily used for SIP call media sessions
supportedResolutions	string Client resolution in WxH format
useDTLS	boolean Whether the connection uses DTLS for secure media transport. Always true for WebRTC connections, false for RTMP
authToken	string Session token for reconnection. If the WebSocket connection is lost, the client can reconnect with this token to rejoin the same session instead of creating a new one
status	string Enum: "NEW" "PENDING" "ESTABLISHED" "DISCONNECTED" "FAILED" "REGISTERED" "UNREGISTERED" Connection status: NEW (created), PENDING (waiting), ESTABLISHED (connected — WebSocket, RTMP, or other), DISCONNECTED (connection lost, session may be preserved with keepAlive), FAILED (error). SIP-specific: REGISTERED (SIP registration on PBX successful), UNREGISTERED (SIP registration removed)
	object Map of REST hook configurations. Explanation of the fields: • restExclude — fields that should be excluded from the REST request • clientExclude — fields that should be excluded from the REST response • restOverwrite — fields whose values must be overwritten • restOnError — behavior on error: LOG or FAIL • restPolicy — application policy: DISABLE, NOTIFY, or OVERWRITE Map structure: { "publishStream": RestMethodConfig }
logLevel	string Per-connection logging level. Controls the log verbosity for this specific client without affecting other connections or global server logs
keepAlive	boolean Whether the session is preserved after WebSocket disconnect. When true, the client can reconnect with authToken to rejoin the same session. When false, the session is removed immediately on disconnect
mediaProviders	Array of strings unique List of media providers supported by the connected client (e.g. WebRTC, Flash for RTMP clients like OBS, MSE, WSPlayer)
clientVersion	string Version string reported by the client at connection time (e.g. 2.0)
clientOSVersion	string Client platform string reported at connection time (e.g. navigator.appVersion in browser)
clientBrowserVersion	string Client User-Agent string reported at connection time
	object (RTCMetricsServerDescription) Client-side instructions for collecting and shipping WebRTC RTC metrics: ingest endpoint, authorization header, per-stat metric whitelist, batching window, and optional compressor
msePacketizationVersion	integer <int32> MSE packetization format version. Version 1 is the default legacy format, Version 2 is an optimized format. Only applicable to MSE media provider

Responses

Request samples

Payload

Content type

application/json

Example

WebRTC

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"width": 640,
"height": 480,
"supportedResolutions": "720x480",
"useDTLS": true,
"authToken": "d62440e7-0e38-4831-9fc1-7d3beb2a0d31",
"status": "ESTABLISHED",
"restClientConfig": {"publishStream": {"restExclude": "hasVideo,hasAudio",
"clientExclude": "sdp",
"restOverwrite": "createDate",
"restOnError": "LOG",
"restPolicy": "DISABLE"
}
},
"logLevel": "DEBUG",
"keepAlive": true,
"mediaProviders": ["WebRTC"
],
"clientVersion": "2.0",
"clientOSVersion": "5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"clientBrowserVersion": "Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"webRTCMetricsServerDescription": {"ingestPoint": "http://example.com",
"authorization": "Bearer eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXV...",
"types": {"codec": {"metrics": "timestamp,payloadType,mimeType,clockRate,channels"
},
"inbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsReceived,packetsReceivedWithEct1,packetsReceivedWithCe,packetsReportedAsLost,packetsReportedAsLostButRecovered,packetsLost,jitter,trackIdentifier,mid,remoteId,framesDecoded,keyFramesDecoded,framesRendered,framesDropped,frameWidth,frameHeight,framesPerSecond,qpSum,totalDecodeTime,totalInterFrameDelay,totalSquaredInterFrameDelay,pauseCount,totalPausesDuration,freezeCount,totalFreezesDuration,lastPacketReceivedTimestamp,headerBytesReceived,packetsDiscarded,fecBytesReceived,fecPacketsReceived,fecPacketsDiscarded,bytesReceived,nackCount,firCount,pliCount,totalProcessingDelay,estimatedPlayoutTimestamp,jitterBufferDelay,jitterBufferTargetDelay,jitterBufferEmittedCount,jitterBufferMinimumDelay,totalSamplesReceived,concealedSamples,silentConcealedSamples,concealmentEvents,insertedSamplesForDeceleration,removedSamplesForAcceleration,audioLevel,totalAudioEnergy,totalSamplesDuration,framesReceived,decoderImplementation,playoutId,powerEfficientDecoder,framesAssembledFromMultiplePackets,totalAssemblyTime,retransmittedPacketsReceived,retransmittedBytesReceived,rtxSsrc,fecSsrc,totalCorruptionProbability,totalSquaredCorruptionProbability,corruptionMeasurements"
},
"outbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsSent,bytesSent,mid,mediaSourceId,remoteId,rid,encodingIndex,headerBytesSent,retransmittedPacketsSent,retransmittedBytesSent,rtxSsrc,targetBitrate,totalEncodedBytesTarget,frameWidth,frameHeight,framesPerSecond,framesSent,hugeFramesSent,framesEncoded,keyFramesEncoded,qpSum,psnrSum,psnrMeasurements,totalEncodeTime,totalPacketSendDelay,qualityLimitationReason,qualityLimitationDurations,qualityLimitationResolutionChanges,nackCount,firCount,pliCount,encoderImplementation,powerEfficientEncoder,active,scalabilityMode,packetsSentWithEct1"
},
"remote-inbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsReceived,packetsReceivedWithEct1,packetsReceivedWithCe,packetsReportedAsLost,packetsReportedAsLostButRecovered,packetsLost,jitter,localId,roundTripTime,totalRoundTripTime,fractionLost,roundTripTimeMeasurements,packetsWithBleachedEct1Marking"
},
"remote-outbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsSent,bytesSent,localId,remoteTimestamp,reportsSent,roundTripTime,totalRoundTripTime,roundTripTimeMeasurements"
},
"media-source": {"metrics": "timestamp,trackIdentifier,kind,audioLevel,totalAudioEnergy,totalSamplesDuration,echoReturnLoss,echoReturnLossEnhancement,width,height,frames,framesPerSecond"
}
},
"sampling": 500,
"batchSize": 10,
"collect": "on",
"compression": ["none"
]
},
"msePacketizationVersion": 2
}

Response samples

Content type

application/json

Example

WebRTC

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"width": 640,
"height": 480,
"supportedResolutions": "720x480",
"useDTLS": true,
"authToken": "d62440e7-0e38-4831-9fc1-7d3beb2a0d31",
"status": "ESTABLISHED",
"restClientConfig": {"publishStream": {"restExclude": "hasVideo,hasAudio",
"clientExclude": "sdp",
"restOverwrite": "createDate",
"restOnError": "LOG",
"restPolicy": "DISABLE"
}
},
"logLevel": "DEBUG",
"keepAlive": true,
"mediaProviders": ["WebRTC"
],
"clientVersion": "2.0",
"clientOSVersion": "5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"clientBrowserVersion": "Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/142.0.0.0 Safari/537.36",
"webRTCMetricsServerDescription": {"ingestPoint": "http://example.com",
"authorization": "Bearer eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXV...",
"types": {"codec": {"metrics": "timestamp,payloadType,mimeType,clockRate,channels"
},
"inbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsReceived,packetsReceivedWithEct1,packetsReceivedWithCe,packetsReportedAsLost,packetsReportedAsLostButRecovered,packetsLost,jitter,trackIdentifier,mid,remoteId,framesDecoded,keyFramesDecoded,framesRendered,framesDropped,frameWidth,frameHeight,framesPerSecond,qpSum,totalDecodeTime,totalInterFrameDelay,totalSquaredInterFrameDelay,pauseCount,totalPausesDuration,freezeCount,totalFreezesDuration,lastPacketReceivedTimestamp,headerBytesReceived,packetsDiscarded,fecBytesReceived,fecPacketsReceived,fecPacketsDiscarded,bytesReceived,nackCount,firCount,pliCount,totalProcessingDelay,estimatedPlayoutTimestamp,jitterBufferDelay,jitterBufferTargetDelay,jitterBufferEmittedCount,jitterBufferMinimumDelay,totalSamplesReceived,concealedSamples,silentConcealedSamples,concealmentEvents,insertedSamplesForDeceleration,removedSamplesForAcceleration,audioLevel,totalAudioEnergy,totalSamplesDuration,framesReceived,decoderImplementation,playoutId,powerEfficientDecoder,framesAssembledFromMultiplePackets,totalAssemblyTime,retransmittedPacketsReceived,retransmittedBytesReceived,rtxSsrc,fecSsrc,totalCorruptionProbability,totalSquaredCorruptionProbability,corruptionMeasurements"
},
"outbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsSent,bytesSent,mid,mediaSourceId,remoteId,rid,encodingIndex,headerBytesSent,retransmittedPacketsSent,retransmittedBytesSent,rtxSsrc,targetBitrate,totalEncodedBytesTarget,frameWidth,frameHeight,framesPerSecond,framesSent,hugeFramesSent,framesEncoded,keyFramesEncoded,qpSum,psnrSum,psnrMeasurements,totalEncodeTime,totalPacketSendDelay,qualityLimitationReason,qualityLimitationDurations,qualityLimitationResolutionChanges,nackCount,firCount,pliCount,encoderImplementation,powerEfficientEncoder,active,scalabilityMode,packetsSentWithEct1"
},
"remote-inbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsReceived,packetsReceivedWithEct1,packetsReceivedWithCe,packetsReportedAsLost,packetsReportedAsLostButRecovered,packetsLost,jitter,localId,roundTripTime,totalRoundTripTime,fractionLost,roundTripTimeMeasurements,packetsWithBleachedEct1Marking"
},
"remote-outbound-rtp": {"metrics": "timestamp,ssrc,kind,packetsSent,bytesSent,localId,remoteTimestamp,reportsSent,roundTripTime,totalRoundTripTime,roundTripTimeMeasurements"
},
"media-source": {"metrics": "timestamp,trackIdentifier,kind,audioLevel,totalAudioEnergy,totalSamplesDuration,echoReturnLoss,echoReturnLossEnhancement,width,height,frames,framesPerSecond"
}
},
"sampling": 500,
"batchSize": 10,
"collect": "on",
"compression": ["none"
]
},
"msePacketizationVersion": 2
}

DataStatusEvent Notify

/apps/DefaultApp/DataStatusEvent

WCS calls this hook when a client sends a DataStatusEvent WebSocket message to acknowledge a data payload it received from WCS. Typical scenario: the Backend pushes a payload via REST POST /rest-api/v3/data/send, the receiving client gets the data as OnDataEvent and responds with DataStatusEvent to confirm processing — the standard WebSDK does not send this message, so the hook is meant for custom client integrations. The request body contains the originating session identifiers, the operationId that correlates with the original payload, and the delivery status.

DataStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
operationId	string Unique identifier (UUID) of this data operation, generated by the client. The client uses it to correlate the asynchronous response — delivered via DataStatusEvent — back to the originating sendData call.
payload	object Application-defined JSON payload that the client sent through the data channel; the contents are opaque to WCS and are forwarded as-is
status	string Outcome of the data operation. Populated by WCS after the sendData flow completes and delivered to the client via the WebSDK SEND_DATA_STATUS event and to the Backend via DataStatusEvent. Possible values: ACCEPTED (operation succeeded; the payload was forwarded to the target session(s)) or FAILED (operation failed; the failure reason is in the info field). In the OnDataEvent request body this field is typically empty (the operation has not yet been processed).
info	string Additional information
sessionIds	Array of strings Explicit list of signaling session IDs (Connection IDs) that should receive this data payload; when empty the payload is broadcast to every session in the originating app
allSessionIds	Array of strings Effective list of signaling session IDs (Connection IDs) that will receive this data payload, including the originating session and any explicitly added sessionIds

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"operationId": "2497358d-dd6e-4601-926c-cb2b6b12391d",
"payload": {"appKey": "defaultApp",
"body": "Hello alice",
"contentType": "application/imdn+xml",
"custom": {"property1": { },
"property2": { }
},
"from": "<tel:+447912345678;tag=ab12cd34ef56>\n",
"id": "a3f9c2b1-ff00-4e5d-9c77-112233445566",
"isImdnRequired": true,
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@127.0.0.1",
"origin": "http://localhost:8081",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"sipMessageRaw": "INVITE sip:...",
"status": "IMDN_NOTIFICATION_SENT",
"to": "<tel:+447912345678;tag=ab12cd34ef56>\n"
},
"status": "ACCEPTED",
"info": "msg: null",
"sessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
],
"allSessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"operationId": "2497358d-dd6e-4601-926c-cb2b6b12391d",
"payload": {"appKey": "defaultApp",
"body": "Hello alice",
"contentType": "application/imdn+xml",
"custom": {"property1": { },
"property2": { }
},
"from": "<tel:+447912345678;tag=ab12cd34ef56>\n",
"id": "a3f9c2b1-ff00-4e5d-9c77-112233445566",
"isImdnRequired": true,
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@127.0.0.1",
"origin": "http://localhost:8081",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"sipMessageRaw": "INVITE sip:...",
"status": "IMDN_NOTIFICATION_SENT",
"to": "<tel:+447912345678;tag=ab12cd34ef56>\n"
},
"status": "ACCEPTED",
"info": "msg: null",
"sessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
],
"allSessionIds": ["/192.168.1.102:13252/192.168.1.101:8443"
]
}

ErrorStatusEvent Notify

/apps/DefaultApp/ErrorStatusEvent

WCS sends this notification hook whenever an unexpected error related to the current session is raised while processing a SIP or WebRTC operation. The error is always tied to a specific client Connection. The request body contains the error identifiers, code, and message.

ErrorStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
status	string Status. Can be NEW, PENDING, ESTABLISHED, DISCONNECTED, FAILED, REGISTERED, UNREGISTERED, TRYING, RINGING, RING, RING_MEDIA, HOLD, BUSY, FINISH, SESSION_PROGRESS, WRONG_SIPPROVIDER_ADDRESS, TOO_MANY_REGISTER_ATTEMPTS, SIP_PORTS_BUSY, MEDIA_PORTS_BUSY, LICENSE_RESTRICTION, LICENSE_NOT_FOUND, AUTHENTICATION_FAIL, USER_NOT_AVAILABLE, INTERNAL_SIP_ERROR, CALLEE_NAME_IS_NULL, PAYMENT_REQUIRED and etc.
info	string Extra information
sipMessageRaw	string Raw SIP signaling message(s)
apiMethod	string Method, this field should indicate API method so client can fire right event
id	string Operation identifier. For instance: mediaSessionId
	object (Context) Target context

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"status": "NEW",
"info": "App not found",
"sipMessageRaw": "INVITE sip:...",
"apiMethod": "MessageStatusEvent",
"id": "62a497c7-07ee-4580-a477-0e3c8927b4fc",
"target": {"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443"
}
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"status": "NEW",
"info": "App not found",
"sipMessageRaw": "INVITE sip:...",
"apiMethod": "MessageStatusEvent",
"id": "62a497c7-07ee-4580-a477-0e3c8927b4fc",
"target": {"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443"
}
}

SessionDebugStatusEvent Notify

/apps/DefaultApp/SessionDebugStatusEvent

WCS sends this notification hook when a session debug exchange changes its state — STARTED after the client starts debug, STOPPED after it stops, then PASSED or FAILED depending on whether the captured publish/play streams completed successfully. The request body contains the report identifiers and the resulting status.

SessionDebugStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
status	string Enum: "STARTED" "STOPPED" "FAILED" "PASSED" Session debug status
command	string Command. Supported commands: "start", "stop".

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"status": "STARTED",
"command": "start"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"status": "STARTED",
"command": "start"
}

Stream

Category only. Groups REST hooks fired on media streams owned by a specific Connection — publish, play, recording, transcoding, snapshots.

publishStream Confirm

/apps/DefaultApp/publishStream

WCS calls this hook when a client starts publishing a stream — over WebRTC, RTMP, or any other supported protocol. The Backend can authorize publication, override stream parameters, or reject the publish operation. The request body contains the stream identifiers, media information, and publisher session.

publishStream hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

snapshot Confirm

/apps/DefaultApp/snapshot

WCS calls this hook when a client requests a server-side snapshot of a published stream via the WebSDK call stream.snapshot(). The request body contains the stream identifiers and snapshot parameters.

snapshot hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

sendStreamEvent Confirm

/apps/DefaultApp/sendStreamEvent

WCS calls this hook when a publisher pushes a custom event to subscribers of its stream. The Backend can authorize the broadcast, modify the payload, or reject the request. The request body contains the stream identifiers, the event type, and the payload.

The event is emitted by the WebSDK — via stream.muteAudio(), stream.unmuteAudio(), stream.muteVideo(), stream.unmuteVideo(), stream.sendData(payload).

sendStreamEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
type	string Enum: "data" "audioMuted" "audioUnmuted" "videoMuted" "videoUnmuted" "snapshotCompleted" "snapshotFailed" "resize" "notEnoughBandwidth" "addedToMixing" "removedFromMixing" "startedRecording" "stoppedByReInit" "stoppedRecording" "failedRecording" "addedToMultiRecording" "removedFromMultiRecording" "answer" "sdp" "offer" Event type
	object Free-form map carrying optional event metadata; the contents are forwarded to subscribers as-is

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"type": "audioMuted",
"payload": {"property1": { },
"property2": { }
}
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"type": "audioMuted",
"payload": {"property1": { },
"property2": { }
}
}

unPublishStream Confirm

/apps/DefaultApp/unPublishStream

WCS calls this hook when a publisher stops sending a previously published stream. The Backend can authorize or reject the unpublish operation before WCS terminates the publishing session. The request body contains the stream identifiers and the current publishing state.

unPublishStream hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

playStream Confirm

/apps/DefaultApp/playStream

WCS calls this hook when a client starts playing an existing stream. The Backend can authorize playback, override the played stream name, or reject the play operation. The request body contains the stream identifiers and the player session.

playStream hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

stopStream Confirm

/apps/DefaultApp/stopStream

WCS calls this hook when an active playback session is about to be stopped by a client command. Publishing sessions use unPublishStream. The request body contains the stream identifiers and the current playback state.

stopStream hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

playRTSP Confirm

/apps/DefaultApp/playRTSP

WCS calls this hook when an external RTSP client (for example VLC or ffplay) connects to WCS's built-in RTSP server and presents an auth token in the URL. WCS asks the Backend whether the token is valid before serving the RTSP session. The hook fires only when the rtsp_server_auth_enabled setting is true and a token is present in the URL. The request body contains the RTSP URL, user agent, stream identifiers, and the token as a custom field. The response does not rewrite the RTSP URL, stream name, or playback parameters.

2xx response — RTSP session is served:

playRTSP — auth granted

4xx/5xx response — RTSP session is rejected:

playRTSP — auth rejected

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
name	string Full RTSP URL of the remote source WCS pulls and exposes as an internal stream

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"name": "rtsp://localhost:554/stream_name?aclAuth=auth_token"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"name": "rtsp://localhost:554/stream_name?aclAuth=auth_token"
}

playHLS Confirm

/apps/DefaultApp/playHLS

WCS calls this hook when an HLS player (browser, VLC, ffplay) requests an HLS playlist or segment for a stream and presents a token in the URL. WCS asks the Backend whether the token is valid before serving the resource. The request body contains the stream name, the subscriber session identifier, and the token as a custom field. The response does not rewrite the stream URL, stream name, or playback parameters. The hook is fired once per unique token — the result is cached on the WCS node, so subsequent requests with the same token reuse the cached decision without firing the hook again.

playHLS hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
name required	string Stream name

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"name": "stream1"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"name": "stream1"
}

availableStream Confirm

/apps/DefaultApp/availableStream

WCS calls this hook when a client explicitly probes stream availability via the WebSDK call stream.available(). The request body contains the requested stream identifiers and the player session. After a 2xx response WCS performs the actual availability check — it verifies that the stream exists locally, has a valid CDN route, or satisfies composite-stream profiles; if none applies the WebSDK Promise rejects with reason STREAM_NOT_FOUND (or RESTRICTED_ACCESS on a CDN-restricted node). A hook failure rejects the probe when restOnError=FAIL.

availableStream hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

RecordingStatusEvent Notify

/apps/DefaultApp/RecordingStatusEvent

WCS sends this notification hook when a stream recording completes — either successfully (recorded file is finalised) or with an error. The hook is not fired on recording start. The request body contains the source stream identifiers and the resulting RecordReport (file name, duration, size, error details). After the hook returns, WCS sends a notifyRecordComplete WebSocket message to the client; the standard WebSDK does not expose this message to user-level handlers — the recording file name is instead available to the client via stream.getRecordInfo().

RecordingStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
	object (RecordReport) Audio recording report for a SIP call: file paths of incoming, outgoing, and mixed tracks, sampling rate, and the originating Call-ID

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"report": {"incomingFileName": "record/10001-10002-2025-12-04-12-34-56-5cbedcae-0c5c-455e-9c56-5d7c78a82b8d-10-in.wav",
"outgoingFileName": "record/10001-10002-2025-12-04-12-34-56-5cbedcae-0c5c-455e-9c56-5d7c78a82b8d-10-out.wav",
"mixedFilename": "record/10001-10002-2025-12-04-12-34-56-5cbedcae-0c5c-455e-9c56-5d7c78a82b8d-10.wav",
"samplingRate": 48000,
"callId": "6d62f660"
}
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"report": {"incomingFileName": "record/10001-10002-2025-12-04-12-34-56-5cbedcae-0c5c-455e-9c56-5d7c78a82b8d-10-in.wav",
"outgoingFileName": "record/10001-10002-2025-12-04-12-34-56-5cbedcae-0c5c-455e-9c56-5d7c78a82b8d-10-out.wav",
"mixedFilename": "record/10001-10002-2025-12-04-12-34-56-5cbedcae-0c5c-455e-9c56-5d7c78a82b8d-10.wav",
"samplingRate": 48000,
"callId": "6d62f660"
}
}

StreamEvent Notify

/apps/DefaultApp/StreamEvent

WCS sends this notification hook every time a stream-level event is dispatched to a subscriber of the stream. The event carries a type and a payload; the same event is also delivered to the subscriber's WebSDK STREAM_EVENT callback. The request body contains the stream identifiers, event type, and payload.

The event is emitted by one of the following sources:

the publisher's WebSDK — explicitly via sendStreamEvent or implicitly through the convenience methods muteAudio, unmuteAudio, muteVideo, unmuteVideo, sendData;
the REST endpoint POST /rest-api/v3/stream/send_event (server-initiated broadcast);
WCS itself, for example resize (video dimensions changed) or notEnoughBandwidth.

The hook is fired per subscriber: a stream with N active subscribers produces N hook calls per event. Plan Backend throughput accordingly, or set restPolicy=DISABLE on StreamEvent in restClientConfig if Backend notification is not needed.

StreamEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
type	string Enum: "data" "audioMuted" "audioUnmuted" "videoMuted" "videoUnmuted" "snapshotCompleted" "snapshotFailed" "resize" "notEnoughBandwidth" "addedToMixing" "removedFromMixing" "startedRecording" "stoppedByReInit" "stoppedRecording" "failedRecording" "addedToMultiRecording" "removedFromMultiRecording" "answer" "sdp" "offer" Event type
	object Free-form map carrying optional event metadata; the contents are forwarded to subscribers as-is

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"type": "audioMuted",
"payload": {"property1": { },
"property2": { }
}
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"type": "audioMuted",
"payload": {"property1": { },
"property2": { }
}
}

StreamStatusEvent Notify

/apps/DefaultApp/StreamStatusEvent

WCS sends this notification hook every time a stream changes its state. Possible values: PUBLISHING, PLAYING, PAUSED, UNPUBLISHED, STOPPED, FAILED. The request body contains the new stream status, identifiers, and media metadata.

StreamStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

StreamTranscodingGroupEvent Notify

/apps/DefaultApp/StreamTranscodingGroupEvent

WCS sends this notification hook when the set of audio or video transcoding groups for a stream changes — typically when a subscriber that requires a new codec or resolution variant joins or leaves the stream. The request body contains the source mediaSessionId, the current audioGroups and videoGroups sets (each describing one target codec/resolution variant and the subscribers consuming it), and optional proxyAudioGroup/proxyVideoGroup for unmodified pass-through.

StreamTranscodingGroupEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier in UUID format
	object (TranscodingGroupParams) Per-track configuration of a transcoding group; binds the source track type (audio or video), target codec and resolution, and the set of subscribers that consume the produced track
	object (TranscodingGroupParams) Per-track configuration of a transcoding group; binds the source track type (audio or video), target codec and resolution, and the set of subscribers that consume the produced track
	Array of objects (TranscodingGroupParams) unique Audio transcoding groups produced for the source stream; each group describes one target codec/sample-rate variant and the subscribers consuming it
	Array of objects (TranscodingGroupParams) unique Video transcoding groups produced for the source stream; each group describes one target codec/resolution variant and the subscribers consuming it
	Array of objects (TranscodingGroupParams) unique Convenience union of audioGroups and videoGroups
	Array of objects (TranscodingGroupParams) unique Convenience union of proxyAudioGroup and proxyVideoGroup, omitting any null entries

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "f0598830-d017-11f0-affd-2f622bc7bad6",
"proxyAudioGroup": {"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
},
"proxyVideoGroup": {"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
},
"audioGroups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
],
"videoGroups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
],
"groups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
],
"proxyGroups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "f0598830-d017-11f0-affd-2f622bc7bad6",
"proxyAudioGroup": {"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
},
"proxyVideoGroup": {"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
},
"audioGroups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
],
"videoGroups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
],
"groups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
],
"proxyGroups": [{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"codec": "H264",
"width": 720,
"height": 480,
"sampleRate": 48000,
"subscribersId": ["f0598830-d017-11f0-affd-2f622bc7bad6"
],
"audio": true
}
]
}

StreamKeepAliveEvent Control

/apps/DefaultApp/StreamKeepAliveEvent

WCS sends this control hook periodically for active published and played streams when streaming session keep-alive is enabled for the application key. It allows the Backend to keep the media session active or force WCS to stop the stream on the server side. The request body contains the current stream identifiers, media information, and the published flag that shows whether the checked media session is publishing or playing the stream. A 2xx Backend response keeps the stream active and WCS does not change the media session. Any non-2xx response, for example 403 Forbidden, or hook request failure makes WCS terminate the media session with the keep-alive termination reason and notify the client with a failed stream status.

2xx response — stream continues:

StreamKeepAliveEvent continue flow

4xx/5xx response — stream is terminated:

StreamKeepAliveEvent terminate flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
mediaSessionId	string Media session identifier within the WCS instance in UUID format
remoteMediaElementId	string Remote media session identifier in UUID format
name	string Stream name
published	boolean Whether this is a publish stream (true) or a play stream (false)
hasVideo	boolean Whether the stream has video
hasAudio	boolean Whether the stream has audio
status	string Enum: "NEW" "PENDING" "PUBLISHING" "PLAYING" "PAUSED" "UNPUBLISHED" "STOPPED" "FAILED" "RESIZE" "SNAPSHOT_COMPLETE" "NOT_ENOUGH_BANDWIDTH" Stream status
sdp	string SDP (Session Description Protocol) associated with the stream. For WebRTC this is the client's offer SDP, for other types a default SDP is used
audioCodec	string Audio codec name (e.g. opus, PCMA, PCMU, G729, AAC)
videoCodec	string Video codec name (e.g. H264, VP8, VP9, H265)
info	string Diagnostic information about the stream, typically the reason for stopping or an error description (e.g. Failed by RTP activity, Stopped by publisher stop, Failed by ICE error)
description	string Stream state description
record	boolean Whether the client requested recording on publish. Does not guarantee that recording is active
recordName	string Recording file name (MP4 or MKV). The file is located in the record_dir directory ({WCS_HOME}/records by default)
width	integer <int32> Video width in pixels
height	integer <int32> Video height in pixels
bitrate	integer <int64> Video bitrate in bps
minBitrate	integer <int64> Minimum bitrate in bps
maxBitrate	integer <int64> Maximum bitrate in bps
quality	integer <int32> Video quality as CRF (Constant Rate Factor) for the encoder. Range: 0 (lossless) to 60 (lowest quality). Lower values produce higher quality. For example, CRF 23 is a common default for H.264
rtmpUrl	string RTMP URL specified by the client on publish for re-streaming to an RTMP server (e.g. rtmp://server:1935/live/stream)
parentMediaSessionId	string Parent media session identifier in UUID format
timeShift	integer <int64> Time offset in milliseconds from the first publish in the room to this stream's publish. Used for multi-track recording synchronization — aligns streams that start at different times within the same room
gop	integer <int32> Group of Pictures — number of frames between keyframes. For example, with fps=30 and gop=60, a keyframe is produced every 2 seconds
fps	integer <int32> Frames per second
audioBitrate	integer <int64> Audio bitrate in bps
codecImpl	string Codec implementation name
transport	string Transport protocol used for media delivery: UDP, TCP, or SRT
profile	integer <int32> Video codec profile (e.g. 66 = Baseline, 77 = Main, 100 = High for H.264)
level	integer <int32> Video codec level (e.g. 31 = Level 3.1, 52 = Level 5.2)
preset	string Encoder preset (e.g. ultrafast, medium, slow)
cvoExtension	boolean Has CVO(Coordinated Video Orientation) RTP header extension or not
	Array of objects (VideoStreamProfile) Negotiated stream profiles for adaptive bitrate (ABR); each entry describes one transcoded resolution/bitrate variant
	Array of objects (AudioStreamProfile)
createDate	string <date-time> Stream creation timestamp in ISO 8601 format
endDate	string <date-time> Timestamp when the stream was finalised on WCS, in ISO 8601 format; null while the stream is still active
mediaType	string Stream type: publish, play, pull, push, pcap, mixer, transcoder, transcoder2, sfu-relay
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object (MediaState) Per-track media state describing whether audio or video is currently muted on the stream
	object Open map of locally collected media information for this stream, such as negotiated SDP attributes, codec parameters, and SSRCs
mediaProvider	string Media provider used by the client: WebRTC (Web SDK with WebRTC support), Flash (deprecated, legacy Flash player), MSE (Media Source Extensions playback), WSPlayer (WebSocket-based player), HLS (HLS playback), RTSP (RTSP pull), VOD (Video on Demand from file), MPEGTS (MPEG-TS ingest)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"mediaSessionId": "d1ddc1ce-90e7-40dd-93b7-fff0e4438a60",
"remoteMediaElementId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"name": "stream1",
"published": true,
"hasVideo": true,
"hasAudio": true,
"status": "NEW",
"sdp": "v=0\r\no=- 82374628 2 IN IP4 192.168.100.20\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 49170 RTP/AVP 0\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:0 PCMU/8000\r\na=sendrecv\r\nm=video 51372 RTP/AVP 96\r\nc=IN IP4 192.168.100.20\r\na=rtpmap:96 VP8/90000\r\na=sendrecv\r\n",
"audioCodec": "opus",
"videoCodec": "H264",
"info": "Failed by HLS writer error",
"description": "Video RTP activity",
"record": true,
"recordName": "s0.mp4",
"width": 720,
"height": 480,
"bitrate": 1000,
"minBitrate": 0,
"maxBitrate": 0,
"quality": 23,
"rtmpUrl": "rtmp://example.com:1935/live",
"parentMediaSessionId": "f1ddc2ce-90e7-40dd-93b7-fff0e4438a60",
"timeShift": 1000,
"gop": 60,
"fps": 30,
"audioBitrate": 1000,
"codecImpl": "OPENH264",
"transport": "UDP",
"profile": 66,
"level": 52,
"preset": "medium",
"cvoExtension": true,
"profiles": [{"width": 1280,
"height": 720,
"gop": 60,
"fps": 30,
"bitrate": 2500000,
"codec": "H264",
"codecImpl": "OPENH264",
"quality": 23,
"profile": 66,
"level": 31,
"preset": "veryfast",
"audioGroupId": "audio0"
}
],
"audioProfiles": [{"bitrate": 64000,
"codec": "opus",
"rate": 48000,
"channels": 2,
"groupId": "default",
"wrappedCodec": "opus"
}
],
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"mediaType": "play",
"audioState": {"muted": true
},
"videoState": {"muted": true
},
"localMediaInfo": {"property1": { },
"property2": { }
},
"mediaProvider": "WebRTC"
}

SIP

Category only. Groups REST hooks fired on SIP signaling within a specific Connection — calls, transfers, registration, DTMF.

call Confirm

/apps/DefaultApp/call

WCS calls this hook when an outgoing SIP/WebRTC call is initiated. The Backend can authorize the call, override caller/callee fields, or reject it before WCS sends SIP INVITE. The request body contains the call identifiers, parties, and SDP information.

call hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

answer Confirm

/apps/DefaultApp/answer

WCS calls this hook when an incoming call is answered and a 200 OK with the local SDP is about to be sent to the remote party. The Backend can inspect or override the SDP, or reject the answer. The request body contains the call identifiers and the answer SDP.

answer hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

sendDtmf Confirm

/apps/DefaultApp/sendDtmf

WCS calls this hook when a client asks to send a DTMF tone on an active call. The Backend can authorize the tone, override the DTMF transport (RFC2833/INFO), or reject the request. The request body contains the call identifiers, the digit, and the transport type.

sendDtmf hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the active call
dtmf	string DTMF digit to send (e.g. 0-9, *, #)
type	string Enum: "INFO" "INFO_RELAY" "RFC2833" DTMF transmission method

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "6d62f660",
"dtmf": "9",
"type": "INFO"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "6d62f660",
"dtmf": "9",
"type": "INFO"
}

hold Confirm

/apps/DefaultApp/hold

WCS calls this hook when a client asks to place an active call on hold and before WCS sends the SIP re-INVITE. The request body contains the held call identifiers, parties, and media details.

hold hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

unhold Confirm

/apps/DefaultApp/unhold

WCS calls this hook when a client asks to resume a held call and before WCS sends the SIP re-INVITE. The request body contains the resumed call identifiers, parties, and media details.

unhold hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

transfer Confirm

/apps/DefaultApp/transfer

WCS calls this hook when a client initiates a call transfer (blind or attended) and a SIP REFER is about to be sent. The Backend can authorize the transfer, override transfer targets, or reject it. The request body contains the source call identifiers and the target party.

transfer hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Call-ID identifier in uuid@host format
target	string Target Call-ID identifier in uuid@host format

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"target": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com"
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"target": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com"
}

hangup Confirm

/apps/DefaultApp/hangup

WCS calls this hook when a client asks to terminate an active call and before WCS sends the SIP BYE or CANCEL to the remote party. The request body contains the call identifiers and termination context.

hangup hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

OnCallEvent Confirm

/apps/DefaultApp/OnCallEvent

WCS calls this hook when an incoming SIP INVITE is received for a registered client and before the incoming call notification is delivered to the client. The request body contains the incoming call details: caller, callee, and SDP. Returned fields may be used in the client notification.

OnCallEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

OnTransferEvent Confirm

/apps/DefaultApp/OnTransferEvent

WCS calls this hook when an incoming SIP REFER is received on an active call and before the transfer request is delivered to the client. The request body contains the transfer parameters and the original call identifiers. Returned fields may be used in the client notification.

OnTransferEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

CallStatusEvent Notify

/apps/DefaultApp/CallStatusEvent

WCS sends this notification hook every time a call changes its state — for example RING, SESSION_PROGRESS, ESTABLISHED, FINISH, or FAILED. The request body contains the current call status, identifiers, parties, and signaling details.

CallStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

RegistrationStatusEvent Notify

/apps/DefaultApp/RegistrationStatusEvent

WCS sends this notification hook after a client connection triggers SIP registration and the registration state changes, for example to REGISTERED, UNREGISTERED, or AUTHENTICATION_FAIL. The request body contains the WCS node, application, session, and resulting SIP registration status. For the REGISTERED notification, fields returned by the Backend may be used in the client callback.

RegistrationStatusEvent hook flow

Request Body schema: application/json

nodeId	string ID representing this server
appKey	string Application identifier
sessionId	string Session identifier
status	string Enum: "REGISTERED" "UNREGISTERED" "AUTHENTICATION_FAIL" SIP registration status

Responses

Request samples

Payload

Content type

application/json

{"nodeId": "f02vkq9PbK9A8MrBnKatCK9clLbHbPyR@95.191.130.39",
"appKey": "defaultApp",
"sessionId": "/5.129.23.83:53866/95.191.130.39:8443-caab56ae-481a-484e-8ca6-ff8a47456080",
"status": "REGISTERED"
}

Response samples

Content type

application/json

{"nodeId": "f02vkq9PbK9A8MrBnKatCK9clLbHbPyR@95.191.130.39",
"appKey": "defaultApp",
"sessionId": "/5.129.23.83:53866/95.191.130.39:8443-caab56ae-481a-484e-8ca6-ff8a47456080",
"status": "REGISTERED"
}

TransferStatusEvent Notify

/apps/DefaultApp/TransferStatusEvent

WCS sends this notification hook after a previously started transfer completes — successfully, with a NOTIFY from the transferee, or with an error. The request body contains the final transfer status and the related call identifiers.

TransferStatusEvent hook flow

Request Body schema: application/json

	object Custom key-value parameters
nodeId	string Unique identifier of the WCS server instance. Auto-generated as {randomString}@{serverIP} and stored in {WCS_HOME}/conf/node_id
appKey	string REST hook application key. Determines the webhook URL for event notifications, as configured in database.yml
sessionId	string Signaling session ID (Connection ID). For WebSocket connections the format is /remoteIP:port/localIP:port. Format varies for other connection types
origin	string Origin URL of the client page (browser domain)
callId	string Unique SIP Call-ID of the call. Format: uuid@host. Sent in the Call-ID header of the SIP INVITE request
callUuid	string Unique internal call UUID assigned by server. Also used by iOS CallKit for VoIP calls
parentCallId	string Parent SIP Call-ID for transferred or chained calls; format uuid@host
incoming	boolean Call direction: true = incoming, false = outgoing
status	string Call status. SIP 180 → RING, SIP 183 → SESSION_PROGRESS, SIP 200 → ESTABLISHED. Other values: HOLD (call placed on hold), BUSY (e.g. SIP 486 Busy Here), FINISH (call ended)
caller	string Caller (From) peer identifier; value used in SIP From header
callee	string Callee (To) peer identifier; value used in SIP To header
localAudioCodec	string Audio codec negotiated by WCS on the local side of the SIP session
localVideoCodec	string Video codec negotiated by WCS on the local side of the SIP session
localWidth	integer <int32> Local video width in pixels
localHeight	integer <int32> Local video height in pixels
remoteAudioCodec	string Audio codec negotiated by WCS on the remote side of the SIP session
remoteVideoCodec	string Video codec negotiated by WCS on the remote side of the SIP session
remoteWidth	integer <int32> Remote video width in pixels
remoteHeight	integer <int32> Remote video height in pixels
createDate	string <date-time> Call creation timestamp in ISO 8601 format
endDate	string <date-time> Call end timestamp in ISO 8601 format; null while the call is still active
hasAudio	boolean Whether audio media is present
hasVideo	boolean Whether video media is present
sdp	string SIP SDP offer. Contains the local SDP generated by WCS for this call
visibleName	string Caller's display name used in the SIP From header
	object Additional SIP INVITE parameters supplied by the caller (e.g. custom header values forwarded into the INVITE)
mediaProvider	string Deprecated. Always set to Flash for REST API calls
sipStatus	integer <int32> Last SIP response status code from the remote endpoint (e.g. 180 Ringing, 183 Session Progress, 200 OK, 486 Busy Here, 408 Request Timeout)
rtmpUrl	string RTMP URL of the target server for re-streaming the RTP media received from the SIP endpoint (e.g. rtmp://server:1935/live/stream)
rtmpStream	string If provided, this value is appended to rtmpUrl to form the final RTMP publish target. Example: rtmpUrl=rtmp://localhost:1935/live + rtmpStream=test → rtmp://localhost:1935/live/test
streamName	string If provided, after the SIP session is established the incoming RTP media will be converted into a server-side internal stream with the specified name. The stream then becomes available for playback and further processing including HLS, transcoding, mixing, etc.
rtmpStreamStatus	string Enum: "RTMP_STREAM_WAIT" "RTMP_STREAM_ACTIVE" "RTMP_CONNECTION_FAILED" "RTMP_CONNECTION_LOST" Status of the RTMP re-streaming for this call: RTMP_STREAM_WAIT (SIP call established, waiting for RTMP connection), RTMP_STREAM_ACTIVE (RTMP stream active), RTMP_CONNECTION_FAILED, RTMP_CONNECTION_LOST
sipMessageRaw	string Last raw SIP message exchanged in this call (request or response from/to the remote SIP endpoint)
target	string SIP target
holdForTransfer	boolean Whether the call is on hold specifically for a subsequent transfer (SIP REFER). Regular hold without this flag does not allow transfer
sipSDP	Array of strings unique Extra SDP attribute lines forwarded together with the SIP offer/answer (a=... lines outside the main SDP)

Responses

Request samples

Payload

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}

Response samples

Content type

application/json

{"custom": {"example": 0
},
"nodeId": "7aoC1Q9IkTCn4zEsNAKvl1P1QMnXfnKE@192.168.100.1",
"appKey": "defaultApp",
"sessionId": "/192.168.1.102:13252/192.168.1.101:8443",
"origin": "https://demo.flashphoner.com:8443",
"callId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@flashphoner.com",
"callUuid": "4a64e860-c5cd-11f0-b763-ab6a382eac0f",
"parentCallId": "7f8a3c731f6f4f83a9b21c0f9e2c1234@parent.flashphoner.com",
"incoming": true,
"status": "RING",
"caller": "10001",
"callee": "10002",
"localAudioCodec": "opus",
"localVideoCodec": "H264",
"localWidth": 720,
"localHeight": 360,
"remoteAudioCodec": "opus",
"remoteVideoCodec": "H264",
"remoteWidth": 720,
"remoteHeight": 360,
"createDate": "2026-04-12 10:30:15.123000+00:00",
"endDate": "2026-04-12 10:35:42.456000+00:00",
"hasAudio": true,
"hasVideo": false,
"sdp": "v=0\\r\\no=Flashphoner 0 1759480746034...",
"visibleName": "10001",
"inviteParameters": {"property1": "string",
"property2": "string"
},
"mediaProvider": "Flash",
"sipStatus": 200,
"rtmpUrl": "rtmp://localhost:1935/live",
"rtmpStream": "test",
"streamName": "s0",
"rtmpStreamStatus": "RTMP_STREAM_WAIT",
"sipMessageRaw": "INVITE sip:...",
"target": "sip:alice@example.co...",
"holdForTransfer": true,
"sipSDP": ["string"
]
}