Cisco AS5300-96VOIP-A Manual de usuario descargar pdf (Pagina 8)

Page 8 of 12

High Performance Co-Processor Design This is a highly integrated single-device solution that minimizes packet latency, essential for high-quality voice. PC-based

solutions using loosely coupled components cannot achieve the same performance characteristics.

Modular Architecture The system provides ﬂexibility and investment protection. The Cisco AS5300/Voice Gateway modular design allows for scaling

from 24 to 120 voice connections per device. In addition, modem and voice modules can both be used in the same AS5300, giving

customers added ﬂexibility.

Compatible with Existing Phones, Faxes,

PBXs, and Key Systems

This feature provides a standard interface to your existing telephony equipment. Users continue to use familiar equipment, with

no special adaptation or retraining.

Real-Time CODEC Selection This sophisticated DSP architecture supports simultaneous H.323 capability negotiation on all channels. The Cisco AS5300/Voice

Gateway voice feature card loads appropriate voice or fax CODEC on the ﬂy for all 120 channels, simultaneously if necessary.

E.165 Echo Cancellation This feature provides echo cancellation into the circuit-switched network with a tail of up to 32 msec, more than adequate to

support carrier quality.

Adaptive Jitter Buffering Adaptive jitter buffer intelligently balances delay and packet loss through the gateway for maximum call clarity and quality.

Voice Activity Detection (silence

suppression)

Bandwidth on the packet network is used only when someone is speaking. During silent periods ofa phone call (up to 50 percent

of the time), bandwidth is available for data trafﬁc.

Front-End Clipping (Time Before

Speech Activity is Detected After

a Period of Silence)

0 msec

Hang-Over Time (Maximum Time Before

Silence is Recognized After a Period of

Speech)

200 msec

Comfort Noise Regeneration To better simulate phone calls over voice networks, this feature reassures the phone user that the connection is being

maintained, even when no voice conversation is in progress.

Voice Quality Statistics Call parameters used for the ITU-T G.113 recommendation for voice quality impairment calculations are supplied. These include

CODEC type, bandwidth used, end-to-end delay, circuit noise, loudness, echo, packet loss, and other statistics.

ITU StandardCODECsG.711, G.729,G.729a,

and G.723.1

These standards-based compression technologies, allow for high-quality voice and compression as low as 53kbps to minimize

bandwidth required to transmit packet telephony.

Compressed Real-Time Protocol (CRTP)

and Multilink PPP Fragmentation and

Interleave

These are header compression and packet fragmentation techniques that allow toll-quality voice and fax transmissions over low

bandwidth WAN connections.

Dial Plan Mapping This feature entails mapping of dialed phone numbers to IP addresses. Mapping can be programmed directly in the Cisco

AS5300/Voice Gateway or alternately maintained in H.323 gatekeepers that communicate to multiple gateways via H.323 RAS

messages.

Number Expansion This feature enables speed dialing and simpliﬁes dial plan conﬁguration. It expands all numbers matching a deﬁned pattern, so

you need to conﬁgure and dial only the last few signiﬁcant digits of the number.

Direct Inward Dial Direct Inward dial allows direct dialing of each user sitting behind a PBX; there is no need to dial the main number and then dial

an extension. This feature is also useful in service provider applications with a VCO/4K used as a service node.

Secondary Dial Tone This feature allows explicit access to a VoIP network for two-stage calling implementation. The ﬁrst dial tone is generated by the

local phone company and the second dial tone, or prompt, is generated by the VoIP carrier.

Call Progress and Tone Generation This feature generates call progress tones, including dial, busy, ring-back, and congestion tones, with local country variants.

Dual Tone Multifrequency (DTMF)

Transport

This feature enables the use of touch tones for voice-mail applications and IVRsystems; it also supports out-of-band DTMF relay

when high-compression CODECs such as G.723.1 are used, which may corrupt inband DTMF tones.

Fax Autodetect Any port can accept a fax relay call; the port is automatically reconﬁgured when an incoming fax call is detected. A scalable

design gives the CiscoAS5300/VoiceGateway the unique abilityto reload all fax algorithmssimultaneously,meeting the stringent

timing requirements of legacy fax machines.

-law and A-law Encoding on Any Channel This featurefacilitates international callingby transparentlytranscoding between -lawencoding(used in T1countries) anda-law

(used in E1 countries).

Music on Hold Threshold This feature offers intelligent music on hold handling.

Table 2 Features and Beneﬁts Summary (Continued)

Feature Beneﬁt

1 2 3 4 5 6 7 8 9 10 11 12

Comentarios a estos manuales

Sin comentarios

Cisco AS5300-96VOIP-A Manual de usuario Pagina 8

Comentarios a estos manuales

Relacionado con productos y manuales para Servidores Cisco AS5300-96VOIP-A