Cisco Network Modules

AVBO provides a way to busy out a voice port or DS0 group (timeslot) based on monitored network congestion to remote IP destinations. Service Assurance Agent (SAA) probes are sent periodically to IP destinations, and if the returned values exceed the configured thresholds, the voice interface can be busied out.

Interoperates with Cisco IP phones, analog phones, fax machine connections, and PBX connections to and from any other Cisco voice enabled product.

Supports debit and credit card (prepaid and post-paid calling card) applications.

Interoperable within the Cisco AVVID architecture.

Uses Resource ReserVation Protocol (RSVP) to determine end-to-end. Resource availability before establishing a call across an IP infrastructure.

Businesses incur significant recurring monthly costs for leased lines purely for the interconnection of telecom PBXs and switches. This product gives these enterprises the ability to remove these costly rigid-bandwidth leased lines and replace them with flexible bandwidth lines, which will be used to carry data, voice, and video. The ability to support proprietary PBX signaling types exists by using connection trunking and transparent CCS.

While using VAD, the DSP at the destination emulates background noise from the source side, preventing the perception that a call is disconnected.

Creates a tie-line replacement structure while only consuming bandwidth during a call (digital-to-digital or digital-to-analog).

Carries DTMF tones/information out-of-band for clearer transmission and detection.

E1 version supports both balanced and unbalanced modes.

E1 G.703 unframed for utilizing the full 2.048 Mbps is also supported.

Transmit Group III fax over any voice channel without sacrificing voice processing resources regardless of compression type.

Enables a connection for incoming and outgoing calls to and from the PSTN or classical PBXs using Cisco IP phones.

Enables a connection for incoming and outgoing calls to and from the PSTN originating from and destined for legacy PBXs, phones, fax machines, and key communication systems connected to a data, voice, and video infrastructure.

Uses industry-standard signaling protocols for call setup between gateways, gatekeepers, and H.323 end points (such as Intel Internet Phone or Microsoft Netmeeting).

Connect one or two T1 or E1 interfaces on this network module as a WAN interface either as one or multiple groups of DS0s or as an entire T1 or E1 frame while still providing packet voice support for connections to PBXs and the PSTN.

Enterprises and Service Providers can use a single box for ATM-based multi-service switching and IP routing applications.

Utilizes IVR to provide automated-attendant, voice-mail support, and call routing based on desired service.

Extends the capability of legacy PBX to off-premise phones.

Provides the ability to settle account billing between service providers who are sharing resources to expand geographical coverage using third-party tools and standards-based OSP.

Provides a direct connection to another digital or analog voice port by lifting a telephone handset on one end.

Up to 30 URM modules are supported in one Cisco IGX.

Users can choose to either transmit voice across their networks as uncompressed PCM (G.711, u-law and A-law) or compressed from 5.3 kbps to 32 kbps using standards-based compression algorithms (G.729, G.729a/b, G.723.1, G.726, G.728).

Standards-based ITU-T G.711 PCM encoding provides 64 kbps analog to digital conversion using u-law or A-law.

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.

Consume bandwidth during a call only when there is voice traffic to send (silence suppression).

Transport voice directly over ATM networks using AAL 5 encapsulation. VoIP is also transported over ATM.

Transmits data, voice, and video end-to-end with IP, and interworks with other Cisco IOS-based voice-enabled platforms.

225 MHz RISC QED RM5271

32 MB

128 MB SDRAM (upgradable to 256 with MEM3660-128D=)

Up to 85 Kpps fast switching and up to 8.5 Kpps process switching (at 80% processor utilization)

60 Calls per URM

1 slot

Two autosensing 10/100 Mbps Ethernet ports

Located on URM Back Card

2 x T1 (with BC-URI-2FE2V-T1 Back Card)

2 x E1 (with BC-URI-2FE2V-E1 Back Card)

Transmit Bit Rate

1.544 Mbps +/- 50 bps/32 PPM

Receive Bit Rate

1.544 Mbps +/- 50 bps/32 PPM

Line Code

AMI, B8ZS

AMI Ones Density

Enforced for N x 56 Kbps channels

Framing Format

D4 (SF) and ESF

Output level (LBO)

0, -7.5, or -15 dB

Input Level

+1dB0 down to -24 dB0

DTE Interface (data mode)

Fractional Service

DTE Interface (voice mode)

G.704/structured

DCE Interface

G.704/structured

Transmit Bit Rate

2.048 Mbps +/- 100 bps/50 PPM

Receive Bit Rate

2.048 Mbps +/- 100 bps/50 PPM

Data Rate

1.984 Mbps (framed mode) Per E1 Port

Clocking

Internal and Loop (recovered from network)

E1 National Bits

Software-configurable

Encoding

HDB3

DTE Interface (data mode)

Fractional Service

DTE Interface (voice mode)

G.704/structured

DCE Interface

G.704/structured

RJ48 Connector

T1 ANSI, ATT T1.1, ANSI T1.403

IETF: RSVP, WFQ, WRED, CRTP, PQ, and other IOS capabilities

ITU-T: H.323 v1 and v2

Pull-in range +/-64 PPM, Pass-through +/-32 PPM

AT&T Accunet (62411)

ATT 54016

ANSI T1.403 Annex B/V.54 loopup/down code recognition, network loopback, and user initiated loopbacks, network payload loopback, local DTE loopback, remote line (codes: V.541, loop up, and loop down)

Alarm indication signal (AIS), remote alarm, far-end block error (FEBE), out of frame (OOF), cyclic redundancy check (CRC) multiframe OOF, signaling multiframe OOF, frame errors, CRC errors, Loss of network signal (red alarm), loss of network frame, receive (blue alarm) (AIS) from network, receive (yellow) from network Performance Reports / Error Counters CRC, errored seconds, burst errored seconds, severely errored seconds, Ft and Fs framing errors for SF framing, FPS framing errors for ESF framing, 24-hour history stored in 15-minute increments

Data carrier detect (CD)

Loopback (LP)

Alarm (AL)

Voice DSP processing status

Selectable DSX-1 cable length in increments from 0 to 655 feet in DSU mode