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.