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SIP

MGCP

Requires Cisco IOS Software Release 12.4(20)T or later

Requires Cisco Unified Communications Manager 4.2(3)

G.722 is not supported with MGCP. G.722 requires Cisco IOS Software Release 12.4(20)T or later with Cisco Unified

Communications Manager 5.0 or later. iLBC requires Cisco IOS Software Release 12.4(15)T or later with Cisco Unified

Communications Manager 6.0 or later

H.323

Feature

Caller ID support

Malicious caller ID (MCID) over PRI

Multilevel precedence and preemption (MLPP) for T1-PRI

(backhaul) and T1-CAS (wink start only)

Group III fax support

T.38 standards-based fax support

Private-line automatic ringdown (PLAR)

Standards-based codecs

Voice activity detection (VAD)

Comfort-noise generation

Busy out

H.323 ITU Version 1, 2, 3, and 4 support

SIP IETF RFC 3261 support

Authentication, authorization, and accounting (AAA)

IVR support

Automated Attendant

VoiceXML

Overlap sending over voice over IP (VoIP)

Benefits

This feature helps the unified communications router

send the caller ID of a caller for display:

In MGCP mode, to and from IP phone, FXS, T1/E1 PRI;

and FXO to IP phone, not conversely (caller ID currently

not supported on T1-CAS).

In SIP and H.323 mode, to and from IP phone, FXS, BRI,

T1/E1 PRI; and from FXO to IP phone, FXS, BRI, and

T1/E1 PRI, not conversely.

MCID over PRI facilitates malicious call notification to on-

net personnel, flags the on-net call detail record (CDR),

and notifies the off-net (PSTN) system (through the

network interface) of the malicious nature of the call.

This feature helps assure high-ranking personnel

communication to critical organizations and personnel

during network stress situations. It allows priority calls for

validated users to preempt lower-priority calls.

Group III fax support facilitates transmit of Group III faxes

between the PSTN and IP using either fax relay or fax

pass-through methods.

This feature enables transmit T.38 fax between the PSTN

and IP.

PLAR provides a dedicated connection to another

extension or an attendant.

You can choose to transmit voice across your network as

either uncompressed pulse code modulation (PCM) or

compressed from 5.3 to 64 kbps using standards-based

compression algorithms (G.711, G.729, G.729a/b, G.722,

Internet Low Bitrate Codec [iLBC], G.723.1, G.726, or

G.728).

VAD conserves bandwidth during a call when there is no

active voice traffic to send.

While using VAD, the digital signal processor (DSP) at

the destination end emulates background noise from the

source side, preventing the perception that a call is

disconnected.

When the WAN or LAN connection to the router is down

or network conditions are such that a call cannot be

admitted, this feature will "busy out" the trunk to the PBX

or PSTN.

These versions of H.323 use industry-standard signaling

protocols for setting up calls between gateways,

gatekeepers, and H.323 endpoints.

This feature uses industry-standard signaling protocols

for setting up calls between gateways and SIP proxies or

SIP Back-to-Back User Agents.

AAA supports debit card and credit card (prepaid and

postpaid calling card) applications.

IVR offers Automated-Attendant support, voicemail

support, or call routing based on service desired.

This feature uses IVR to provide automated call-

answering and -forwarding services.

VoiceXML controls calls "in queue" at the gateway for

call-center applications. Calls are redirected only when an

agent becomes available.

This feature speeds variable-length dial strings dialing.

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