ringcentral-softphone

RingCentral Softphone SDK for TypeScript

This is a TypeScript SDK for RingCentral Softphone. It is a complete rewrite of the RingCentral Softphone SDK for JavaScript

Users are recommended to use this SDK instead of the JavaScript SDK.

Installation

yarn install ringcentral-softphone

Where to get credentials?

• Login to https://service.ringcentral.com
• Find the user/extension you want to use
• Check the user's "Devices & Numbers"
• Find a phone/device that you want to use
• if there is none, you need to create one. Check steps below for more details
• Click the "Set Up and Provision" button
• Click the link "Set up manually using SIP"
• At the bottom part of the page, you will find "Outbound Proxy", "User Name", "Password" and "Authorization ID"

Usage

import Softphone from 'ringcentral-softphone';

const softphone = new Softphone({
  outboundProxy: process.env.SIP_INFO_OUTBOUND_PROXY,
  username: process.env.SIP_INFO_USERNAME,
  password: process.env.SIP_INFO_PASSWORD,
  authorizationId: process.env.SIP_INFO_AUTHORIZATION_ID,
});

For complete examples, see demos/

domain

For UK accounts you need to explicitly specify the domain parameter:

{
  domain: 'sip.ringcentral.co.uk',
}

US accounts use sip.ringcentral.com by default if you don't specify it.

Please do not specify port number in domain.

Supported features

• inbound call
• outbound call
• inbound DTMF
• outbound DTMF
• reject inbound call
• cancel outbound call
• hang up ongoing call
• receive audio stream from peer
• stream local audio to remote peer
• call transfer

Notes

Audio formats

The codec used between server and client is "OPUS/16000". This SDK will auto decode/encode the codec to/from "uncompressed PCM".

Bit rate is 16, which means 16 bits per sample. Sample rate is 16000, which means 16000 samples per second. Encoding is "signed-integer".

You may play saved audio by the following command:

play -t raw -b 16 -r 16000 -e signed-integer test.wav

Invalid callee number

If you call an invalid number. The sip server will return "SIP/2.0 486 Busy Here".

This SDK will emit a "busy" event for the call session and dispose it.

You can detect such an event by:

callSession.once('busy', () => {
  console.log('cannot reach the callee number');
});

Dev Notes

Content below is for the maintainer of this SDK.

• We don't need to explicitly tell remote server our local UDP port (for audio streaming) via SIP SDP message. We send a RTP message to the remote server first, so the remote server knows our IP and port. So, the port number in SDP message could be fake.
• Ref: https://www.ietf.org/rfc/rfc3261.txt
• Caller Id feature is not supported. P-Asserted-Identity doesn't work. I think it is by design, since hardphone cannot support it.