An Endpoint responsible for:

• registering at a SIP provider,
• dialing a phone number,
• receiving audio from the callee and forwarding it to other Endpoints,
• mixing audio tracks from other Endpoints and sending them to the callee.

Limitations

• Incoming calls are unsupported,
• only the G.711 A-law audio codec is supported (SDP negotiation will fail if the SIP provider doesn't support it).

Setup

All SIP Endpoints share a single UDP socket for SIP signaling messages. By default, it is opened on 0.0.0.0:5060; this can be changed by adding the following line to your config.exs:

config :membrane_rtc_engine_sip, sip_address: "1.2.3.4", sip_port: 5061

The range of UDP ports (available to all SIP Endpoints) used for RTP media stream exchange can be modified by adding the following line to your config.exs:

config :membrane_rtc_engine_sip, port_range: {from, to}     # (both ends inclusive)

Bin options

Passed via struct Membrane.RTC.Engine.Endpoint.SIP.t/0

• rtc_engine

  pid()

  Required
  PID of parent Engine

• registrar_credentials

  RegistrarCredentials.t()

  Required
  Credentials needed to connect with the SIP registrar server

• external_ip

  String.t()

  Required
  External IPv4 address of the machine running the Endpoint, required for SDP negotiation

• register_interval_ms

  non_neg_integer()

  Default value: 45000
  Interval (in ms) in which keep-alive (keep-registered) REGISTER messages will be sent to the SIP registrar server

• disconnect_if_alone

  boolean()

  Default value: true
  Whether the Endpoint should disconnect from the call when all incoming tracks are removed, i.e. when all other Endpoints publishing audio are removed from the Engine

Pads

:input

Accepted formats:

Membrane.RTP

:output

Accepted formats:

Membrane.RTP