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ringcentral-softphone - npm Package Compare versions

Comparing version 1.0.7 to 1.1.0-beta.2

dist/cjs/call-session/inbound.cjs

40

package.json
{
"name": "ringcentral-softphone",
"version": "1.0.7",
"version": "1.1.0-beta.2",
"homepage": "https://github.com/ringcentral/ringcentral-softphone-ts",
"license": "MIT",
"main": "dist/src/softphone.js",
"types": "dist/src/softphone.d.ts",
"main": "dist/cjs/index.js",
"types": "dist/cjs/index.d.ts",
"module": "dist/esm/index.js",
"exports": {
".": {
"import": "./dist/esm/index.js",
"require": "./dist/cjs/index.js",
"types": "./dist/esm/index.d.ts"
},
"./*": {
"import": "./dist/esm/*.js",
"require": "./dist/cjs/*.js",
"types": "./dist/esm/*.d.ts"
}
},
"scripts": {
"in": "rm -rf *.wav && tsx -r dotenv-override-true/config demos/inbound-call.ts",
"lint": "eslint --fix . && prettier --write . && sort-package-json",
"out": "rm -rf *.wav && tsx -r dotenv-override-true/config demos/outbound-call.ts",
"prepublishOnly": "rm -rf *.wav && rm -rf dist && yarn tsc --skipLibCheck"
"build": "tsc -p tsconfig.cjs.json && tsc -p tsconfig.esm.json",
"prepublishOnly": "yarn build",
"postpublish": "rm -rf dist"
},
"dependencies": {
"@evan/opus": "^1.0.3",
"wait-for-async": "^0.7.5",
"werift-rtp": "^0.8.3"
"wait-for-async": "^0.7.9",
"werift-rtp": "^0.8.4"
},
"files": [
"dist"
],
"devDependencies": {
"@rc-ex/core": "^1.4.2",
"@types/node": "^22.10.5",
"@rc-ex/core": "^1.5.0",
"@types/node": "^22.10.10",
"dotenv-override-true": "^6.2.2",
"eslint-config-tyler": "^0.2.2",
"sort-package-json": "^2.12.0",
"sort-package-json": "^2.14.0",
"tsx": "^4.19.2",
"ttpt": "^0.13.0",
"typescript": "^5.7.2",
"typescript": "^5.7.3",
"yarn-upgrade-all": "^0.7.4"
}
}
# 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](https://github.com/ringcentral/ringcentral-softphone-js)
This is a TypeScript SDK for RingCentral Softphone. It is a complete rewrite of
the
[RingCentral Softphone SDK for JavaScript](https://github.com/ringcentral/ringcentral-softphone-js)

@@ -24,14 +26,19 @@ Users are recommended to use this SDK instead of the JavaScript SDK.

7. Click the link "Set up manually using SIP"
8. You will find "SIP Domain", "Outbound Proxy", "User Name", "Password" and "Authorization ID"
8. You will find "SIP Domain", "Outbound Proxy", "User Name", "Password" and
"Authorization ID"
Please note that, "SIP Domain" name should come without port number. I don't know why it shows a port number on the page.
This SDK requires a "domain" which is "SIP Domain" but without the port number.
Please note that, "SIP Domain" name should come without port number. I don't
know why it shows a port number on the page. This SDK requires a "domain" which
is "SIP Domain" but without the port number.
### Programmatically
Invoke this RESTful API: https://developers.ringcentral.com/api-reference/Devices/readDeviceSipInfo
Invoke this RESTful API:
https://developers.ringcentral.com/api-reference/Devices/readDeviceSipInfo
Please note that, in order to invoke this API, you need to be familiar with RingCentral RESTful programmming.
Please note that, in order to invoke this API, you need to be familiar with
RingCentral RESTful programmming.
Here is a demo: https://github.com/tylerlong/rc-get-device-info-demo/blob/main/src/demo.ts
Here is a demo:
https://github.com/tylerlong/rc-get-device-info-demo/blob/main/src/demo.ts

@@ -96,5 +103,5 @@ The credentials data returned by that API is like this:

You will need to choose a outboundProxy value based on your location.
And please choose the `proxyTLS` value because this SDK uses TLS.
For example if you live in north America, choose `sip10.ringcentral.com:5096`.
You will need to choose a outboundProxy value based on your location. And please
choose the `proxyTLS` value because this SDK uses TLS. For example if you live
in north America, choose `sip10.ringcentral.com:5096`.

@@ -104,3 +111,3 @@ ## Usage

```ts
import Softphone from 'ringcentral-softphone';
import Softphone from "ringcentral-softphone";

@@ -135,8 +142,7 @@ const softphone = new Softphone({

The codec used between server and client is "OPUS/16000".
This SDK will auto decode/encode the codec to/from "uncompressed PCM".
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".
Bit rate is 16, which means 16 bits per sample. Sample rate is 16000, which
means 16000 samples per second. Encoding is "signed-integer".

@@ -149,3 +155,4 @@ You may play saved audio by the following command:

To stream an audio file to remote peer, you need to make sure that the audio file is playable by the command above.
To stream an audio file to remote peer, you need to make sure that the audio
file is playable by the command above.

@@ -168,13 +175,18 @@ #### ffmpeg

For Linux and Windows, please do some investigation yourself. Audio file generation is out of scope of this SDK.
For Linux and Windows, please do some investigation yourself. Audio file
generation is out of scope of this SDK.
### Multiple instances with same credentials
You can run multiple softphone instances with the same credentials without encountering any errors. However, only the most recent instance will receive inbound calls.
You can run multiple softphone instances with the same credentials without
encountering any errors. However, only the most recent instance will receive
inbound calls.
In the future, we may consider supporting multiple active instances using the same credentials. For now, we believe there is no demand for this functionality.
In the future, we may consider supporting multiple active instances using the
same credentials. For now, we believe there is no demand for this functionality.
### Invalid callee number
If you call an invalid number. The sip server will return "SIP/2.0 486 Busy Here".
If you call an invalid number. The sip server will return "SIP/2.0 486 Busy
Here".

@@ -186,4 +198,4 @@ This SDK will emit a "busy" event for the call session and dispose it.

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

@@ -194,6 +206,7 @@ ```

When you get audio from a call session, you may forward it to another call session:
When you get audio from a call session, you may forward it to another call
session:
```ts
callSession1.on('rtpPacket', (rtpPacket: RtpPacket) => {
callSession1.on("rtpPacket", (rtpPacket: RtpPacket) => {
if (rtpPacket.header.payloadType === 109) {

@@ -214,6 +227,14 @@ // 109 is for opus audio packet

Content below is for the maintainer of this SDK.
Content below is for the maintainer/contributor 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.
- 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.
- Caller Id feature is not supported. `P-Asserted-Identity` doesn't work. I
think it is by design, since hardphone cannot support it.
#### Code style
We use `deno fmt` to format all code.
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