Datasheet

Data Sheet
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Table 6. IP Communications High-Density Digital Voice/Fax Network Module Feature Summary
Feature Description and Benefits
IP Telephony Voice Gateway
Integrates all Cisco IP Communications solutions by providing flexible and reliable connectivity to public
or private switched telephone networks around the world.
Provides high-density gateway for Cisco IP phones to PSTN or legacy PBX/PABXs. (Private Automatic
Branch Exchange).
Provides high-density gateway to PSTN for traditional PBX systems, phones, fax machines, and key
communication systems connected to a voice, data, and video infrastructure.
Interoperable within a Cisco AVVID architecture and Cisco IP Communications solution.
Toll Bypass
Reduce or eliminate toll charges assessed by long distance and local carriers by transporting voice and fax
traffic across the enterprise intranet, LAN, metropolitan-area network (MAN), or WAN.
Works with existing phones, fax machines, PBX systems, and key systems.
Connection trunk: Creates a permanent tie-line replacement structure (digital-to-digital, digital-to-analog, or
analog-to-analog capabilities).
Interoperates end-to-end with Cisco IP phones, analog phones, fax machine connections, and PBX/PABX
connections to and from other Cisco voice-enabled products.
Voice over Packet Transport
Voice/Fax over IP: VoIP traffic at Layer 3 can travel over any Layer 1 or Layer 2 media, including ISDN, leased
lines, serial connections, Frame Relay, Ethernet, Token Ring, and Asynchronous Transfer Mode (ATM).
Voice/Fax over Frame Relay: VoFR is supported using FRF.11 and FRF.12 standards. This solution also uses
features found only in Cisco IOS Software for maintaining voice quality.
Voice over ATM: VoATM is supported using AAL2 or AAL5 encapsulation. Leverages existing ATM networks
as a direct transport method for voice. VoATM requires ATM interfaces such as T1/E1 ATM, IMA, DS3/E3 or
OC-3, or DSL WICs.
Compressed Real-Time Protocol (cRTP) offers RTP header compression and packet fragmentation
techniques that allow toll-quality voice and fax transmissions over any WAN connection.
Call admission control and PSTN fallback: Uses Service Assurance Agent (SAA) to determine latency, delay
and jitter and provide real-time ICPIF calculations before establishing a call across an IP infrastructure. SAA
packets emulate voice packets receiving the same priority as voice throughout the entire network.
Advanced quality of service (QoS) mechanisms: These configurable Cisco IOS Software features reserve
appropriate bandwidth and prioritize voice and fax traffic to ensure transparent delivery of toll-quality voice and
fax. They include Resource Reservation Protocol (RSVP), Queuing Techniques (such as Low Latency
Queuing), IP Precedence, and DiffServ Code Points.
Call Control Signaling
Supports H.323 V1/V2/V3/V4, MGCP 0.1/1.0, and SIP call control protocols. Also supports CiscoUCM3.3 using
MGCP or H.323, and Cisco UCME 3.1.
ITU Standard Voice Codecs
G.711, G.729, G.729a/b, G.723.1, G.726, G.728, iLBC, G.722—These are standards-based compression
technologies allowing transmission of voice across IP, Frame Relay, and ATM. The G.711 standard employs 64
kbps PCM modulation using either u-law or A-law. Other codecs employ lower bit rates.
Telephony Interface Signaling
Support
Supports the following signaling protocols:
FXO/FXS loop-start and ground-start signaling
E&M (wink, immediate, delay)
Inbound signaling (such as DTMF, MF support)
T1 and E1 CAS
T1 and E1 PRI Q.931 user side and network side
T1 and E1 PRI QSIG
E1 MelCAS
E1 R2 CAS
T1 and E1 Transparent CCS (with Multi-D channel)
Country-specific signaling
Voice Features
Conferencing/Transcoding
1
3
: Supports up to 50 eight-party G.711 ad-hoc conferencing sessions, or up to 32
eight-party multi-codec (G.711/G.729) ad-hoc conferencing sessions. A maximum of 128 transcoding sessions
between G.729a and G.711 codecs, and 96 transcoding sessions between G.729 and G.711 codecs.
Echo cancellation: Industry-leading G.168-compliant software echo cancellation provides echo cancellation on
tail circuits up to 64 Mbps.
Silence suppression, voice activity detection (VAD): Bandwidth is used only when someone is speaking.
During silent periods of a phone call, bandwidth is available for data traffic.
Comfort noise generation: This feature reassures the phone user that the connection is being maintained,
even when no voice packets are being transmitted.
Private line automatic ring-down (PLAR): Provides a direct connection to another digital or analog voice port
by lifting a telephone handset on one end. Includes "Trader Turret" PLAR.
Local/Advanced Voice Bus
y
-
Out (LVBO/AVBO)
: Automatically busies out any desired voice trunk line to a
3
Conferencing/transcoding feature available on the 2800 Series starting with 12.3(8)T4, 3800 Series starting with 12.3(11)T, 2900
and 3900 Series starting with 15.0(1)M.