Skip to content

Working with audio using the Cloud Phone SDK

Two of the Cloud Phone SDK's most popular uses is to gain direct access to the audio streams of a phone call. To see code samples for these two use cases, consult the links below.

  • Receive audio stream from peer

    Tap into the audio stream of a phone call and pipe the stream to a file, or translator, or transcription engine.

    See example code

  • Stream local audio to remote peer

    Tap into the audio stream of a phone call and inject your own custom audio. Play audio from a file, or stream audio from a speech synthesizer.

    See example code

Selecting the right right audio codec

The Cloud Phone SDK supports the following audio codecs:

Codec Sample rate Channels Description
OPUS/16000 (default) 16 kHz Mono Best for telephony. High quality, but low bitrates and low latency
OPUS/48000/2 48 kHz Stereo Best for music and full-band audio, but requires more bandwidth and CPU power
PCMU/8000 8 kHz Mono Widely supported in PSTN and SIP networks, but poorest overall audio quality

Here is a quick selection guide for common use cases:

Scenario Recommended codec
Voice chat with good quality OPUS/16000
High-fidelity stereo audio OPUS/48000/2
Interoperability with legacy phones PCMU/8000
Mixed voice and music OPUS/48000/2
Low-bandwidth VoIP OPUS/16000

Setting the audio codec

One is able to select their preferred audio codec when instantiating a Cloud Phone instance.

import Softphone from "ringcentral-softphone";

const softphone = new Softphone({
    codec: "PCMU/8000",
    // ...
});

Working with audio codecs

OPUS/16000

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

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

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

ffplay -autoexit -f s16le -ar 16000 test.wav

PCMU/8000

If you choose this codec, make sure audio is playable using the following command:

play -b 8 -r 8000 -e mu-law test.raw

If the file's extension is .wav, then play will complain

play FAIL formats: can't open input file `6fdbbf2f-74fe-437a-b5a7-80c0c546baf0.wav': WAVE: RIFF header not found
To fix, change the file extension to .raw or use ffplay instead.

ffplay -autoexit -f mulaw -ar 8000 test.wav

OPUS/48000/2

If you choose this codec, make sure audio is playable using the following command:

=== "play command" play -t raw -b 16 -r 48000 -e signed-integer -c 2 test.wav

Generating an audio file for testing

say "Hello world" -o test.wav --data-format=LEI16@16000

The audio file generated can be used for codec OPUS/16000

Please consider contributing an example command to help Windows users

Please consider contributing an example command to help Linux users