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UA Configuration Parameters

JsSIP.UA requires a configuration object with mandatory and optional parameters. Example:

var configuration = {
  'ws_servers':         'ws://sip-ws.example.com',
  'uri':                'sip:alice@example.com',
  'password':           'superpassword'
};

var coolPhone = new JsSIP.UA(configuration);

Full list of configuration parameters below:

Mandatory parameters

uri

SIP URI associated to the User Agent (String). This is a SIP address given to you by your provider.

uri: "sip:alice@example.com"

ws_servers

Set of WebSocket URIs to connect to. If not specified ports 80 and 443 are taken for non secure WebSocket connections and connections over SSL/TLS, respectively. This parameter can be expressed in multiple ways:

  • Single String to define a single WebSocket URI.
  • Array of Strings to define multiple WebSocket URIs.
  • Array of Objects to define multiple WebSocket URIs with weight. URIs with higher weight value are used prior to those with lower value.
ws_servers: "ws://sip-ws.example.com"

ws_servers: "wss://sip-ws.example.com:8443/sip?KEY=1234"

ws_servers: [
  "ws://sip-ws-1.example.com",
  "ws://sip-ws-2.example.com",
]

ws_servers: [
  {ws_uri: "ws://sip-ws-1.example.com", weight: 10},
  {ws_uri: "ws://sip-ws-2.example.com", weight: 1}
]

Optional parameters

display_name

Descriptive name (String) to be shown to the called party when calling or sending IM messages. It must NOT be enclosed between double quotes even if the given name contains multi-byte symbols (JsSIP will always enclose the display_name value between double quotes).

display_name: "Alice ¶€ĸøĸø"

password

SIP Authentication password (String).

password: "1234"

authorization_user

Username (String) to use when generating authentication credentials. If not defined the value in uri parameter is used.

authorization_user: "alice123"

register

Indicate if JsSIP User Agent should register automatically when starting. Valid values are true and false (Boolean). Default value is true.

register: false

register_expires

Registration expiry time (in seconds) (Integer). Default value is 600.

register_expires: 300

registrar_server

Set the SIP registrar URI. Valid value is a SIP URI without username. Default value is null which means that the registrar URI is taken from the uri parameter (by removing the username).

registrar_server: 'sip:registrar.mydomain.com'

no_answer_timeout

Time (in seconds) (Integer) after which an incoming call is rejected if not answered. Default value is 60.

no_answer_timeout: 120

trace_sip

Indicate whether incoming and outgoing SIP request/responses must be logged in the browser console (Boolean). Default value is false.

trace_sip: true

stun_servers

String or Array of Strings indicating the STUN server(s) to use for IP address discovery. Values must include “stun:” or “stuns:” schema. Default value is ["stun:stun.l.google.com:19302"].

stun_servers: "stun:example.org"

stun_servers: ["stun:example.org", "stuns:example.org"]

stun_servers: ["stun:example.org:8000"]

turn_servers

Object or Array of Objects indicating the TURN server(s) and corresponding username and password to be used for media relay (in case no peer-to-peer media is possible). ‘urls’ (a String or an Array of String) can include “turn:” or “turns” schema. No TURN server is set by default.

turn_servers: { urls:"turn:example.org", username:"turnuser", credential:"turnpassword"}

turn_servers: { urls:["turn:example.org", "turn:example2.org"], username:"turnuser", credential:"turnpassword"}

turn_servers: [{ urls:"turn:example.org", username:"turnuser", credential:"turnpassword"}]

turn_servers: [
  { urls:"turn:example.org", username:"turnuser", credential:"turnpassword" },
  { urls:"turn:example.org?transport=udp", username:"turnuser2", credential:"turnpassword2"}
]

use_preloaded_route

If set to true every SIP initial request sent by JsSIP includes a Route header with the SIP URI associated to the WebSocket server as value. Some SIP Outbound Proxies require such a header. Valid values are true and false (Boolean). Default value is false.

ws_servers: "ws://example.org/sip-ws"
use_preloaded_route: true

The Route header will look like Route: <sip:example.org;lr;transport=ws>

ws_servers: "wss://example.org:8443"
use_preloaded_route: true

The Route header will look like Route: <sip:example.org:8443;lr;transport=ws>

connection_recovery_min_interval

Minimum interval (Number) in seconds between WebSocket reconnection attempts. Default value is 2.

connection_recovery_min_interval: 4

connection_recovery_max_interval

Maximum interval (Number) in seconds between WebSocket reconnection attemps. Default value is 30.

connection_recovery_max_interval: 60

hack_via_tcp

Set Via transport parameter in outgoing SIP requests to “TCP”. Useful when traversing SIP nodes that are not ready to parse Via headers with “WS” or “WSS” value in a Via header. Valid values are true and false (Boolean). Default value is false.

hack_via_tcp: true

hack_ip_in_contact

Set a random IP address as the host value in the Contact header field and Via sent-by parameter. Useful for SIP registrars not allowing domain names in the Contact URI. Valid values are true and false (Boolean). Default value is a false.

hack_ip_in_contact: true

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