SIP configuration

Your PBX must support SIP over secure WebSocket and browser-compatible WebRTC audio. Authentication identity and the SIP address can differ.

import type { SipConfig } from "@azeer-ui-widget/react-phone";

// Store this in state after sign-in; keep its identity stable.
const makeConfig = (username: string, password: string): SipConfig => ({
  websocketUrl: "wss://pbx.example.com/ws",
  uri: `sip:${username}@pbx.example.com`,
  authorizationUsername: username,
  password,
  displayName: "Support",
  // Supply deployment-specific STUN/TURN servers if required.
  iceServers: [],
});

Option	Default / meaning
`websocketUrl`	Required; `wss://…` (`ws://` allowed only for localhost)
`uri`	Required SIP address of record
`authorizationUsername`	Optional digest identity, separate from the URI user
`password`	Required; keep in memory
`displayName`, `registrarServer`	Optional caller name / explicit registrar
`iceServers`	`[]`; supply your own STUN/TURN servers
`iceTransportPolicy`	`"all"`; set `"relay"` to require TURN
`registrationTimeoutMs`	`20000`
`maxReconnectAttempts`	`5`; retry delay managed by JsSIP (2–15 seconds)
`callTimeoutSeconds`	`45`; unanswered-call timeout
`dtmfTransport`	`"RFC2833"`; use `"INFO"` when required by your PBX
`audioConstraints`	Browser audio constraints; echo cancellation and noise suppression default on

Never put passwords into public build-time variables (VITE_*, PUBLIC_*), checked-in files, or URLs. The hosted playground accepts credentials at runtime and keeps them in memory. A browser SIP client necessarily has access to its own credentials; use a dedicated, least-privilege PBX device account.