Get WebRTC SIP credentials

Provision (or fetch) SIP credentials for a WebRTC softphone tied to a contact-center user.

Endpoint: POST https://<base>/api/v2/GetWebRTC.php

Returns the SIP user / secret / realm needed for a WebRTC softphone (e.g., the Velip browser client). When the user does not have a SIP entry yet, the endpoint creates one in cd_sip_webrtc with a freshly generated random secret.

The returned credentials are tied to the contact-center user (userid) and the SIP realm picked from the customer’s WebSocket server config (cdcsu_cc_ws_server).

Authentication

Token authentication required. See Authentication.

Request

tsid — type: string — required

Token for the account.

userid — type: string — required

Contact-center user id (numeric). Used to derive the SIP username VW{cdcs_id}U{userid}.

Request example

```bash curl curl -X POST ‘https:///api/v2/GetWebRTC.php’
-d ‘tsid=YOUR_TSID&userid=42’

## Response
```json 200 OK
{
  "return": {
    "status": "OK",
    "status_code": "0",
    "sipname": "VW1234U42",
    "sipsecret": "x9d2KqLm",
    "sipserver": "wss.velip.com.br"
  }
}

• return.sipname (string) — SIP username to register from the softphone.

• return.sipsecret (string) — SIP password. Generated once and persisted; subsequent calls return the same secret.

• return.sipserver (string) — Realm portion of the WebSocket server (host without port). Use it as the SIP domain.

Error codes

Code	status	Cause
200	No userid	Missing userid parameter.

The endpoint inherits the global authentication codes.

Warning The SIP secret is generated by generatePassword(12) and stored in plain text in cd_sip_webrtc.secret. Treat the returned credentials like any other password — deliver them over HTTPS and keep them out of client-side logs.