Introducing the SIP audio provider
This section introduces the Nuance SIP audio provider and describes the features it supports. This document assumes that you are familiar with SIP technology.
The SIP audio provider is a software-based audio provider supporting VoIP connectivity through Session Initiation Protocol (SIP).
SIP is an application-layer signaling protocol for creating, modifying, and terminating sessions with one or more participants. SIP defines mechanisms for call routing, call signaling, capabilities exchange, media control, and supplementary services. Because SIP integrates with web, email, and other IP applications, it provides a powerful and extensive URL definition that allows SIP URLs to be embedded in web browsers and email tools.
The SIP audio provider is designed for use in a managed network environment that guarantees Quality of Service (QoS), such as a local area network (LAN) or a properly provisioned wide area network (WAN).
The SIP audio provider, based on the DynamicSoft User Agent stack, provides recognition and verification accuracy equal to that provided by other audio services based on the PSTN with the following constraints:
- You must disable voice activity detection (VAD) in the gateway or VoIP client.
- Nuance highly recommends that you use the UDP protocol as the underlying SIP transport protocol.
Note: Real-Time Transport (RTP) audio is handled by the Nuance Speech Server. The SIP audio provider sets up the RTP connection between the Speech Server and the telephony gateway.
