Session Initiation Protocol (SIP) is an ASCII-character-based signaling protocol designed for real-time transmission using Voice over IP (VoIP). In Speech Server, SIP establishes sessions with requests and responses. When Speech Server is connected to a client, session initiation and call control are handled over SIP.
SIP is hosted through a SIP peer, which connects Speech Server to the caller's endpoint. The SIP peer can be a VoIP client, SIP client, or traditional telephony client.
|A SIP client, such as a SIP phone or softphone, is a SIP peer and an endpoint because it connects the caller directly to Speech Server through a SIP line without requiring a gateway.|
Examples of SIP peers include:
IP PBX (telephony)
SIP phones and softphones
Telephony Interface Manager Connector (TIMC)
In previous Speech Server versions, Speech Server connected to the PSTN exclusively through a traditional circuit-switched interface by means of a COM-based non-standard interface known as Telephony Interface Manager (TIM).
Speech Server now includes Telephony Interface Manager Connector (TIMC), a compatibility layer between a TIM and the Speech Server SIP interface. TIMC provides a Speech Server upgrade path for TIM users.
Telephony Interface Manager
TIM software is a separate component that is tightly coupled to the installed telephony board. All telephony audio information, either from a caller (using a telephony client) or from Speech Server, passes through the TIM software. TIM software accepts incoming telephone calls and the Telephony Application Proxy routes them to the appropriate server. The audio portion of the call is sent to Speech Server.
The telephony board provides the physical connection between the telephone network and the TIM. The telephony board includes the drivers necessary to interface with the host computer's operating system. TIM software supports certain manufacturer's telephony boards.