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VoIP Audio Service API

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Created: ltomuta (22 May 2008)
Last edited: hamishwillee (07 Feb 2012)

Note.pngNote: This API is not part of the public SDK. It can be found in the SDK API Plug-in.

VoIP Audio Services API is a secure single utility interface to audio services on the S60 platform, which allows control of audio by 3rd party VoIP client applications. It provides high level access to audio device making possible to play and record audio, play DTMF and ring tones, configure supported codecs, adjust microphone gain, volume level, and switch audio routing between private and public output device.

It removes the need for use of Audio Proxy Server (APS) on S60 3rd Edition, Feature Pack 1 and Feature Pack 2 and largely extends its feature set.

Use cases

The API provides interface for clients to perform the following audio tasks:

  • Initialization of downlink stream for audio playback on the local device.
  • Initialization of uplink stream for audio recording on the local device.
  • Playback of DTMF tones through downlink stream.
  • Sending DTMF tones to the uplink stream.
  • Playback of Ring Tones from a specified source or from general Profile.
  • Dynamic configuration of supported encoders and decoders.
  • Dynamic configuration of input and output audio devices (for example, volume or gain).
  • Dynamic configuration of audio output routing.
  • Verifying capability and secure allocation of resources for audio activity.

Example code

Provided with the API package.

Example project

Provided with the API package.

Known issues

KIS001634 - CRingTonePlayer instance stops with KErrUnderflow error while playing a file


The latest version of the VoIP Audio Services API device binaries (sis file) can be found at File:VoIPAudioSrvc v2.03.1 SA S6031S6032S6050

The latest version of the VoIP Audio Services API device binaries (sis file) can be found at File:VoIPAudioServicesAPI

VoIP Example Application

This page was last modified on 7 February 2012, at 06:21.
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