Cisco 26/3600 2 PORT DATA INTERFACE Guide De Spécification

Cisco Systems, Inc.

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Connection Trunk

Creates a permanent tie-line replacement structure (digital-to-digital, digital-to-analog,
or analog-to-analog capabilities).

LVBO (Local Voice
Busy-Out)

Automatically busy out any desired voice trunk line to a PBX or PSTN when a direct
WAN or LAN connection to the router is down. Also, busy out a far end trunk
connection when configured for Connection Trunk.

Caller ID Support

Per-port configurable caller ID to phones connected to analog FXS voice ports using
per call un-blocking if desired. Also provide caller ID over analog FXO and DID voice
interfaces. Interoperates with analog phones, PSTNs, PBXs, H.323 terminals (i.e.
Microsoft Netmeeting), Cisco Call Manager and IP phones.

Call Admission Control
using RTR

Uses Response Time Reporter (RTR) to determine latency, delay and jitter and provide
real-time ICPIF calculations before establishing a call across an IP infrastructure. RTR
packets emulate voice packets receiving the same priority as voice throughout the
entire network. A superior method to data and ping packets for determining
congestion levels.

Voice and Fax over Same
Port

Ports can be used for both voice and fax traffic--no dedicated ports are required.

Works with Existing
Phones, Faxes, PBXs,
and Key Systems

No user retraining is required.

H.323 v3/v4
Compatibility

The Cisco voice/fax modules are interoperable with numerous emerging voice and
videoconferencing applications, such as Microsoft NetMeeting, Intel Internet Phone,
LAN-based IP telephony equipment, and Cisco Call Manager.

High-Performance DSP
Architecture

The Cisco voice/fax modules offer extremely low latency, which is essential for
high-quality voice and fax traffic; the DSP architecture also enables all critical functions
to be handled in software, which allows for simple code updates, scalability, and new
features.

ITU Standards G.729,
G.729a/b, G.711, G.723.1,
G.726 and G.728

These are standards-based compression technologies allowing transmission of voice
across IP, Frame Relay and ATM. G.711 is standard 64 kbps PCM modulation using
either u-law or A-law.

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

To better simulate phone calls over voice networks, this feature reassures the phone
user that the connection is being maintained, even when no voice packets are being
transmitted.

Dial Plan Mapping

Automatic mapping of dialed phone numbers to IP addresses simplifies configuration
and management.

Dual Tone
Multifrequency (DTMF)
Tone Processing

This feature enables access to voice-mail and Interactive Voice Response (IVR)
systems.

Fax and Modem
Passthrough

Allows fax and modem traffic to pass through a voice port.