Datasheet

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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 Cisco CallManager
3.3 using MGCP or H.323, and Cisco CallManager Express 3.1.
ITU Standard Voice
Codecs
• G.711, G.729, G.729a/b, G.723.1, G.726, G.728, GSM, GSM-EFR, GSM-ER: 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