SIP

Makes a SIP connection with the specified provider. From this connection it consumes a remote RTP stream with G711 A-law audio and publishes through this connection all audio tracks from the Room.

Read more about SIP here.

Compatibility

• WebRTC
• HLS
• RTSP
• File
• Recording

Requirements

• The specified SIP provider needs to support G711 A-law as it will be a format in which audio from SIP Endpoint will be published.

Configuration options

Required

• registrarCredentials (object) - registrar credentials which allows to make a connection with a SIP provider. For the exact credential structure see Configuration API.

Env variables

• FJ_COMPONENTS_USED - has to contain sip if SIP components will be used.
  By default set to an empty string (SIP components are disabled).
  Refer here for more info.

• FJ_SIP_IP - the IP address on which SIP server will listen. It must be a public IP and reachable from the outside network.