Charon is a software client for SIP telephone services. It can be a client of a SIP server as an internal account (extension) as well as terminating a sip-trunk with the propagation of telephone numbers inside it - it does not require any additional services to implement such a configuration.

This component is also responsible for handling the DTMF (Dual tone multi-frequency) signals that come when users press number keys on their phones. Such signals are forwarded to Automate as digits in plain text. RFC-4733 standard is supported for DTMF over SIP and sent in RTP by using 101 telephone-event codes.

API

Default REST API port: 8126.

Healthcheck endpoint: /healthCheck on port 8558

Other ports used: 2552, 9095

Databases

SQL

The component has its own SQL Database for storing information mapping of phone numbers with SIP Trunks, along with other necessary configuration.

Communication

This service communicates with external SIP servers and internal applications like Qpido and Dialog-sentione

Security

Endpoints are secured with static API key defined in config. Communication is only internal.

Configuration

The service configuration is located in the application.conf file, below is an example file with important values:

voice.gateway {
  pbx.registrars: [
    {
      host: 1.2.3.4
      port: 5060
      name = default
      username = "<USER>"
      password = "<PASS>"
      external-local-host: 1.2.3.4 // external/public IP on which Charon can be accessed
      local-sip-port: 5060
      destination-number-patterns: [] // those regex patterns are checked, when there is redirect from bot to agent number. If this number matches any of the regular expressions in list then this SIP Trunk is used. 
    }
  ]

  charon {
    rtp {
      jitter-buffer-size: 7
      first-packet-timeout: 5s
    }

    database {
      username = vg-admin
      password = ${DB_PASS}
      url = "jdbc:postgresql://example.org:5001/voice-gateway"
      driver = org.postgresql.Driver
      show-sql = false
      persistence = "postgres_pu"
      schema = charon
      liquibase.enabled = false
      db-properties {
        hibernate.generate_statistics = true
        hibernate.jmx.enabled = true
        hibernate.hikari.registerMbeans = true
        hibernate.hikari.poolName = charonHikariCP
      }
   }
    scheduler {                                                                                        base-url: "http://qpido:8124"
       api-key: "<QPIDO_API_KEY>" // Set in config for Qpido
       type: QPIDO
    }
    api {
      base-url: "http://0.0.0.0:8126"
      api-key: "<API_KEY>"
    }

    local-transfer-patterns: ["[0-9]{1,12}"]
  }


  pbx.sound.rtp.port {
    min: 20000
    max: 65529
  }

  network.interface: eth0 // interface to be used by SIP
  rtp.network.interface: ens256 // interface to be used for RTP communication
  sip.limit-messages-per-second: 1000

  sip {
    port.protocol {
      receive: [udp]
      send: udp
    }
    nat {
      stunEnabled: false
      stunServerAddress: 1.2.3.4
      stunServerPort: 3478
      stunConnectionTimeout: 1s
    }
  }

}

Field description:

• pbx-registrars - defining basic network parameters to create a SIP Trunk route, along with the ability to identify each of them by the 'name' parameter
• rtp - defining the rtp transmission parameters
• database - defining access parameters to the Postgresql database
• api - defining access parameters to the prepared REST API for the service
• local-transfer-patterns - a regular expression specifying connections to be made in a way other than SIP REFER
• pbx.sound.port - definition of the port range for audio stream communication
• rtp.network.interface - used when there is a configuration requirement to send the audio stream via an interface other than the default one
• sip.port.protocol - definition of the SIP protocol communication type