Mixing of SIP calls and WebRTC calls in Conference

[Arif Butt]

Hi there,

Is there a possibility of Mixing of SIP calls and WebRTC calls in Conference? kindly assist...

Regards
AB

 

[Max]

Good day.
It is possible to capture SIP call to a stream published on WCS server. Then the stream can be added to MCU mixer, if MCU based conference is used, or played by conference participants if RoomApi is used.
Please read how to capture stream from SIP call:
SIP as stream
SIP as RTMP
Incoming SIP call as RTMP
For example, let's suppose the following scenario:
1. SIP participants join to SIP conference arranged on SIP PBX by calling a certain callee number, for example 10000
2. WCS make SIP as stream call to this conference using /call/startup REST API query

POST /rest-api/call/startup HTTP/1.1
HOST: wcs:8081
content-type: application/json
 
{
    "callId":"123456789",
    "callee":"10000",
    "toStream":"stream1",
    "hasAudio":"true",
    "hasVideo":"false",
    "sipLogin":"10009",
    "sipAuthenticationName":"10009",
    "sipPassword":"1234",
    "sipDomain":"domain",
    "sipOutboundProxy":"domain",
    "sipPort":"5060",
    "appKey":"defaultApp",
    "sipRegisterRequired":"false",
    "visibleName":"WCS_10009"
}

SIP call stream stream1 will be published on WCS
3. WebRTC participants join to MCU mixer based conference. The mixer has name mixer1
4. SIP call stream stream1 is added to MCU mixer based conference
5. The mixer output stream without SIP call audio (to prevent echo) mixer1-stream1 is injected to SIP call using /call/inject_stream/startup REST API query

POST /rest-api/call/inject_stream/startup HTTP/1.1
Host: wcs:8081
Content-Type: application/json;charset=UTF-8
 
{
    "callId":"123456789",
    "streamName":"mixer1-stream1"
}

6. All the participants can hear each other