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WCS can work as a WebRTC-SIP gateway. In this case, audio and video stream of a SIP call made through WCS can be captured and played in a browser or republished to another server.

Typical use case

A video call is established between WCS and a SIP device (SIP MCU, conference server or a SIP softphone)
WCS receives audio and video data from this SIP device
The WCS server redirects the received audio and video traffic to an RTMP server or another device capable of receiving and processing an RTMP stream

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REST-methods and response statuses

REST-method

Example of REST-query

Example of REST-response body

Response status

/call/startup

Code Block

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{  
   "callId":"123456711",
   "callee":"10000",
   "toStream":"stream1",
   "rtmpUrl":"rtmp://localhost:1935/live/",
   "rtmpStream":"rtmp_stream1",
   "hasAudio":"true",
   "hasVideo":"true",
   "sipLogin":"10009",  "sipAuthenticationName":"10009",
 "sipPassword":"1234",
"sipDomain":"226.226.225.226",
"sipOutboundProxy":"226.226.225.226",
   "sipPort":"5060",
"appKey":"defaultApp",
"sipRegisterRequired":"false"
}

{}

200 - The call is accepted for processing

409 - Conflict with an existing RTMP URL

/call/find

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{"status" : "ESTABLISHED"}

or

{"callId":"R08NQya-5NMe5v7q-JNkboaS-CGMlFi"}

Code Block

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[{
"custom": {},
"nodeId": null,
"appKey": null,
"sessionId": null,
"callId": "R08NQya-5NMe5v7q-JNkboaS-CGMlFi",
"parentCallId": null,
"incoming": false,
"status": "ESTABLISHED",
"sipStatus": 200,
"rtmpUrl": null,
"rtmpStream": null,
"streamName": null,
"rtmpStreamStatus": null,
"caller": "001",
"callee": "002",
"hasAudio": true,
"hasVideo": false,
"sdp": null,
"visibleName": "001",
"inviteParameters": null,
"mediaProvider": "Flash",
"sipMessageRaw": null,
"isMsrp": false,
"target": null,
"holdForTransfer": false
}]

200 - call is found
404 - call with the given parameters is not found

/call/find_all

{}

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[{
"custom": {},
"nodeId": null,
"appKey": null,
"sessionId": null,
"callId": "R08NQya-5NMe5v7q-JNkboaS-CGMlFi",
"parentCallId": null,
"incoming": false,
"status": "ESTABLISHED",
"sipStatus": 200,
"rtmpUrl": null,
"rtmpStream": null,
"streamName": null,
"rtmpStreamStatus": null,
"caller": "001",
"callee": "002",
"hasAudio": true,
"hasVideo": false,
"sdp": null,
"visibleName": "001",
"inviteParameters": null,
"mediaProvider": "Flash",
"sipMessageRaw": null,
"isMsrp": false,
"target": null,
"holdForTransfer": false
}]

200 - calls are found
404 - calls are not found

/call/terminate

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{"callId" : "becee2c0-13b4-11e7-b817-c1649197cae8"}

200 - call is terminated
404 - call is not found

/call/send_dtmf

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{
"callId" : "52173e00-13b6-11e7-b817-c1649197cae8",
"dtmf":"9",
"type":"RFC2833"
}

200 - DTMF is sent
404 - call is not found

/call/inject_stream

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{
"callId": "ad5ac8a0-b518-11e7-82c7-999b45e427ba",
"streamName": "mixer1_stream"
}

200 - audio stream is added to the call
404 - call is not found

Parameters

Parameter name	Description	Example
callId	SIP Call ID - a unique identifier string	Xq2tlLcX89tTjaji
callee	SIP callee	10001
toStream	Name of the stream on the WCS server the call is published to	call_stream1
rtmpUrl	RTMP URL - address of the RTMP server	rtmp://rtmp-server.flashphoner.com:1935/live Here live - is the name of the RTMP application. Also, in RTMP URL the instance name and the query string can be specified, for example: rtmp://rtmp-server.flashphoner.com:1935/live/_definst_?param1=value1&param2=value2
rtmpStream	Name of the RTMP stream on the RTMP server	streamName2
hasAudio	If true, SDP will have the 'sendrecv' parameter in audio. If false, it gets 'recvonly'.	true
hasVideo	If true, SDP will have the 'sendrecv' parameter in video. If false, it gets 'recvonly'.	true
status	Call status on the WCS server	ESTABLISHED The complete list of statuses is available in the Call Flow (see the CallStatusEvent method).
sipStatus	Associated SIP-status	200
rtmpStreamStatus	Status of the RTMP stream	RTMP_STREAM_ACTIVE RTMP_STREAM_WAIT - RTMP-stream is initializing RTMP_STREAM_ACTIVE - RTMP-stream has initialized and connection is established RTMP_CONNECTION_LOST - RTMP-connection is lost RTMP_CONNECTION_FAILED - RTMP-connection was not established
caller	SIP caller
visibleName	Displayed name of the caller
mediaProvider	NOT USED	NOT USED

SDP parameters recvonly and sendrecv

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This address can be changed via WCS CLI. Se the description of the command line interface to get more information about application management in WCS.

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When the 'production' mode is on, each REST/HTTPS or REST/HTTP query requires HTTP Basic Authentication.
Standard username and password are admin:admin.
You can change the password in WCS CLI. (See more about Command Line Interface here.)

In the REST Console, you can add authorization as follows

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Versions Compared

Old Version 3

New Version 4

Key

Typical use case

REST-methods and response statuses

Parameters

SDP parameters recvonly and sendrecv