diagox

Dialog Go Exchange

diagox is modern approach for simple Back To Back VOIP solution built on top of sipgo and diago library.

It can be seen like Ingress Service for SIP/RTP, to allow you to scale API/VOIP services with addition of call monitoring, rate limiting and more.

Main Features:

• Call Bridging with inbound/outbound routing and media proxy.
• Call history with full SIP traffic and voice quality monitoring
• Optional: running in multi node for scaling and HA.
• API and GUI for checking History and Other

Project is free to use, but although used and tested in many scenarios, it is considered still experimental.

If interested to use this tool/service and supporting development, you can get latest updates, roadmap, code details. For all this you can contact me on mail

Features

• Call bridging with no audio transcoding
• Integrated Registrar to allow user registering and discovery
• Call Rate Limiting / Inbound + Outbound
• Simple routing and fast matching
• Fallback routing based on SIP Response
• Endpoint IP/Auth identification
• Call History with SIP Trace, RTCP metrics and Quality calculations
• Call Recording to WAV format / Mixed streams
• Integrated monitoring with statsviz (only for debuging)
• Configuration validation
• Structured logging

Install

Application is single binary, you can just download from latest release and run it.

Running:

wget https://github.com/emiago/diagox/releases/latest/download/diagox
chmox +x diagox
./diagox

It needs diagox.yaml for your routing configuration. Example below.

nano diagox.yaml

Configuration

Configuration is simple based on yaml. This offers higher automation and easier managing.

diagox.yaml

version: "2.4"


# transports for SIP
transports: 
  udp:
    transport: "udp"
    bind: 0.0.0.0
    port: 5060
    external_host: my.domain.com # Use env SIP_EXTERNAL_HOST to set for all transports
    external_media_ip: 1.2.3.4 # Use env SIP_EXTERNAL_MEDIA_IP to set for all transports
  tcp:
    transport: "tcp"
    bind: 0.0.0.0
    port: 5060
  udp_local:
    transport: "udp"
    bind: 127.0.0.1
    port: 5099
  
# routes allow you to customize which endpoint your number should reach
# - `default` route context is special one which is always used unless incoming endpoint did not override
# - you can create more routes and split logic as you like
# - keeping route context small can improve call route matching
routes:
  # Default context goes every call from all endpoints
  # If not created or overided it has this configuration
  default:
    - id: ""
      match: "any" # Match any number
      use_registry: true # Check registry. Any register user will be reached

  incoming: 
    # Order here matters 
    # Required fields are: id and endpoint

    # Example with strict number matching
    - id: "4912345678" 
      endpoint: carrier_internal # On which endpoint to send

    # Example with prefix matching
    - id: "381"
      match: prefix
      endpoint: carrier_internal

    # Example with prefix matching and passing SIP headers
    - id: "121"
      match: prefix
      endpoint: carrier_internal
      sip_headers_pass: ["X-Myheader", "X-Account-ID"] # This headers are copied from incoming call to outgoing
      sip_headers: # This headers are added on outgoing channel.
        X-Fixed-Header: "Call121"

  # Example of outgoing where you want to have your internal endpoint route calls out
  outgoing: 
    - id: "987" 
      match: "prefix"
      endpoint: carrier_external

    # Example where to match all if did not match any of previous
    # Instead of endpoint we just want to hangup call. This is where hangup module is provided
    - id: ""
      match: "any" #Match all
      hangup:
        code: 404
  
  # Example when you want to route call but also do some fallback if initial endpoint/carrier failed
  with_fallback:
    - id: 49
      match: "prefix"
      endpoint: carrier_primary
      fallback: 
        codes: [401, 404, 487]
        endpoints: ["carrier_second", "carrier_third"]   

# Endpoints define way to identify your incoming SIP traffic and which route to use.
# Identifing is done with `match`. Supported values are: ip, user
# - `ip` is your incoming source IP
# - `user` matches by SIP From header and USER property  ex. From: <sip:[email protected]>
# Match order is: user, ip
endpoints:
  carrier_external:
    route: incoming
    match: 
      type: "ip" 
      values: ["182.168.0.0/24"]
    
    # Auth will identify incoming SIP INVITE/REGISTER additionally with Digest authentication
    auth:
      username: "test" 
      password: "test123"

    # Uri is for your outgoing or how to reach this carrier. 
    # If you use this endpoint in routes, this SIP uri will be used to reach. User part is replaced by caller ID.
    uri: "sip:carrier.external.com:5080"
    transport: tcp # You can force which transport to use. This is id of transport defined in transports
  
  carrier_internal:
    # When matched call will be sent to outgoing route
    route: outgoing

    # Incoming options
    match: 
      type: "ip"
      values: ["127.0.0.1/8"]

  alice: # -> [email protected]
    # routing: default # If not defined `default`` route is used
    match: 
      type: "user" 
    auth:
      username: "alice" 
      password: "test123"    

  bob: # -> [email protected]
    match: 
      type: "user" 
    auth:
      username: "carrier" 
      password: "test123" 
    

Global configuration

More features are offered using global configuration or toogling some features.

// Log formating.
string  "LOG_LEVEL" envDefault:"info" // debug, info, warn, error
string  "LOG_FORMAT" envDefault:"console" // json, console
// Rate limiting for incoming
bool    "RATE_LIMITER_IN_ENABLED" envDefault:"false"
int64   "RATE_LIMITER_IN_DIALOG_RPS"
int64   "RATE_LIMITER_IN_DIALOG_MAX"
// Rate limiting for outgoing
bool    "RATE_LIMITER_OUT_ENABLED" envDefault:"false"
int64   "RATE_LIMITER_OUT_DIALOG_RPS"
// Default outbound dial uri if not defined by endpoint
string   "OUTBOUND_DIAL_URI" envDefault:"sip:[email protected]:5066"
string   "CONF_FILE" envDefault:"diagox.yaml"
bool     "CDR_ENABLE" envDefault:"true"
string   "RECORDINGS_PATH" envDefault:"recordings"
bool     "FRONTEND_ENABLE" envDefault:"false"
string   "SIP_BIND_IP" envDefault:""     // Useful for pods/nodes that have dedicated IP
string   "SIP_EXTERNAL_IP" envDefault:"" // Useful for pods/nodes that have dedicated IP
// SIP_HOSTNAME identifies that message are matching this hostname. Used in registrar for example
string SIPHostname   `env:"SIP_HOSTNAME" envDefault:""`

Frontend (Work In Progress)

Sharing just a peaks.

NOTE: Frontend is not active by default

History

Call history with VOIP metrics and data

Multi node running

Dependencies

• MySQL - CDR/config storing
• Redis - Registry/Dialog caching

For more interest, will be shared...