Genesys Softphone

Welcome to the Genesys Softphone Deployment Guide. This document describes how to deploy and configure the Genesys Softphone in your environment.

Features and functionality

DTMF

The Genesys Softphone supports Dual-Tone Multi-Frequency (DTMF) signalling according to the RFC 2833 standard for third-party call control.

After receiving a NOTIFY with DTMF event, the Softphone Endpoint generates DTMF signals.

DTMF can be sent by using one of the three possible methods:

• InbandRTP
• RFC 2833
• SIP INFO message

Third-party call control

When the Genesys Softphone Endpoint has registered on the Genesys SIP Server, it will support the following third-party call control scenarios:

• Make a call
• Answer a call
• Hold and retrieve a call
• Single-step and two-step transfers
• Participate in a conference that is provided by the GVP
• Play DTMF signals.

SIP Voice

The Genesys Softphone supports the following codecs for SIP signaling:

• PCMU/8000 (G.711/mu-law)
• PCMA/8000 (G.711/A-law)
• G722/16000
• iLBC/8000 (iLBC — internet Low Bitrate Codec)
• iSAC/32000 ((iSAC/32kHz) — internet Speech Audio Codec)
• iSAC/16000
• G729/8000
• OPUS/48000/2

WebRTC Voice

The Genesys Softphone supports the following codecs for WebRTC signaling:

• OPUS
• G711

WebRTC and OAuth Support

WebRTC with OAuth is supported in WWE/Softphone connector mode only. If you use the standalone Softphone mode, you must migrate to connector mode. To enable this feature in WWE you must configure the value of the sipendpoint.enable_webrtc_auth option to true.