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
ofStrings
to define multiple WebSocket URIs.Array
ofObjects
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
authorization_user
Username (String
) to use when generating authentication credentials. If not defined the value in uri
parameter is used.
authorization_user: "alice123"
connection_recovery_max_interval
Maximum interval (Number
) in seconds between WebSocket reconnection attemps. Default value is 30
.
connection_recovery_max_interval: 60
connection_recovery_min_interval
Minimum interval (Number
) in seconds between WebSocket reconnection attempts. Default value is 2
.
connection_recovery_min_interval: 4
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"
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
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_via_ws
When connecting to a secure WebSocket server (“wss://”) force “WS” as Via transport value instead of “WSS”. Useful when traversing HTTP/WebSocket proxies that act as TLS tunnels. Valid values are true
and false
(Boolean
). Default value is false
.
hack_via_ws: true
instance_id
String
indicating the UUID URI to be used as instance ID to identify the UA instance when using GRUU.
instance_id: "uuid:8f1fa16a-1165-4a96-8341-785b1ef24f12"
instance_id: "8f1fa16a-1165-4a96-8341-785b1ef24f12"
log
Object
providing the desired log behavior.
builtinEnabled
Boolean
indicating whether JsSIP should write log messages in the browser console. Default value is true
.
level
Number
or String
indicating the verbose level of the JsSIP log. Valid values are 3
, 2
, 1
, 0
or "debug"
, "log"
, "warn"
, "error"
respectively. Default value is 2
(or log
).
connector
User defined Function
which will be called everytime a log is generated, according to the enabled
and level
options.
The function is called with the following semantics:
/*
level: String representing the level of the log message ('debug', 'log', 'warn', 'error')
category: String representing the JsSIP instance class firing the log. ie: 'jssip.ua'
label: String indicating the 'identifier' of the class instance when the log level is '3' (debug). ie: transaction.id
content: String representing the log message
*/
connector(level, category, label, content);
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
password
SIP Authentication password (String
).
password: "1234"
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'
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"]
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
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>