Datasheet

Data Sheet
© 2009 Cisco Systems, Inc. All rights reserved. This document is Cisco Public Information. Page 4 of 14
Call Admission Control (CAC) helps ensure that voice quality of service (QoS) is maintained across constricted WAN
links, and it automatically diverts calls to alternate public-switched-telephone-network (PSTN) routes when WAN
bandwidth is not available. A web interface to the configuration database allows remote device and system
configuration. HTML-based online help is available for users and administrators.
Cisco Unified Communications Manager supports Resource Reservation Protocol (RSVP) agent capability. The
RSVP agent on a Cisco router extends CAC capability beyond a hub-and-spoke topology within a cluster. Now a call
can be routed directly between two locations without having to traverse the hub, allowing alternative network
topologies and more efficient use of networks.
The Cisco Unified IP Phone 7931G initially supported in Cisco Unified Communications Manager 6.0 with Skinny
Client Control Protocol (SCCP) is now optionally available with SIP. This phone provides functions that are
commonly needed in the commercial and retail environments. It provides 24 lighted line keys and four interactive
softkeys that guide you through call features and functions. In addition, it provides hard hold, redial, and transfer
keys to facilitate simple and rapid call handling.
SNMP is available to manage Cisco Unified Communications Manager, allowing managers to set and report traps on
conditions that could affect service and send them to remote monitoring systems.
System Capabilities Summary
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Alternate automatic routing (AAR)
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Attenuation and gain adjustment per device (phone and gateway)
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Audio message-waiting indicator (AMWI)
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Automated bandwidth selection
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Automatic route selection (ARS)
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AXL Simple Object Access Protocol (SOAP) application programming interface (API) with performance and
real-time information
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Basic Rate Interface (BRI) endpoint support: Registers BRI endpoints as SCCP devices
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CAC: Intercluster and intracluster
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Call coverage
Forwarding based on internal and external calls
Forwarding out of a coverage path
Timer for maximum time in coverage path
Time of day
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Call display restrictions
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Call preservation -- redundancy and automated failover -- on call-processing failure
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Call recording
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Codec support for automated bandwidth selection: G.711 (mu-law and a-law), G.722, G.722.1, G.723.1,
G.728, G.729A/B, Global System for Mobile-Enhanced Full Rate (GSM-EFR), Global System for Mobile-Full
Rate (GSM-FR) iLBC (internet Low Bitrate Codec), wideband audio (proprietary 16-bit resolution; 16-kHz
sampled audio), and Advanced Audio CODEC (AAC) for use with Cisco TelePresence devices
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Digit analysis and call treatment (digit string insertion, deletion, stripping, dial access codes, digit string
translation, and dial pattern transformation) [[Deleting the asterisk here because it doesn’t seem to refer to
anything.]]
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Database resiliency to increase feature availability for the following: