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Cisco 3900 Series Integrated Services Routers

Cisco 2900, 3900, and 4400 Series Integrated Services Router Interoperability with Cisco Unified Communications Manager Data Sheet

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Cisco® 2900, 3900, and 4400 Series Integrated Services Routers can be deployed as unified communications routers as part of the Cisco Unified Communications and Collaboration Solution. New and existing deployments can benefit by using any of these routers as unified communications gateways with Cisco Unified Communications Manager.

Cisco 2900, 3900, and 4400 Series unified communications routers communicate directly with Cisco Unified Communications Manager, allowing for the deployment of unified communications solutions that are ideal for small and medium-sized businesses, large enterprises, and service providers that offer managed network services.
These platforms provide a highly flexible and scalable solution for small and medium-sized branch and regional offices. These platforms support a wide range of packet telephony-based voice interfaces and signaling protocols within the industry, providing connectivity support for more than 90 percent of the world's private branch exchanges (PBXs) and public-switched-telephone-network (PSTN) connection points. Signaling support includes T1/E1 Primary Rate Interface (PRI), T1 channel associated signaling (CAS), E1-R2, T1/E1 QSIG protocol, T1 Feature Group D (FGD), Basic Rate Interface (BRI), foreign exchange office (FXO), ear and mouth (E&M), and foreign exchange station (FXS). You can configure these unified communications routers to support from 2 to 720 voice channels. Additionally, you can use these unified communications routers to terminate Session Initiation Protocol (SIP) trunking into the enterprise or branch office by enabling the Cisco Unified Border Element features. Additional details are available in the Cisco Unified Border Element data sheet.
As your enterprise seeks to deploy an expanding list of unified communications applications and services, Cisco unified communications routers - interoperating with Cisco Unified Communications Manager - can provide a solution that will grow with your changing needs.

Interoperability Using SIP, H.323, or MGCP

The unified communications routers can communicate with the Cisco Unified Communications Manager using Session Initiation Protocol (SIP), H.323, or Media Gateway Control Protocol (MGCP):

• In SIP and H.323 mode, the unified communications routers communicate with Cisco Unified Communications Manager as intelligent gateway devices.

• In MGCP mode, these routers operate in "slave" mode where Cisco Unified Communications Manager takes the "master" role. The gateway configuration and dial-plan configuration are centrally managed from Cisco Unified Communications Manager, which generates an XML config file that is downloaded by the gateway to autoconfigure.

IP Telephony Phased Migration

The Cisco 2900, 3900, and 4400 Series unified communications routers can help you immediately deploy an end-to-end unified communications network architecture or gradually shift voice traffic from traditional circuit-switched networks to a single infrastructure carrying data, voice, and video over packet networks.
Initially, you can use these unified communications routers to interconnect older PBXs over the packet infrastructure and still maintain PSTN (off-net) connectivity through your circuit-switched PBXs. Later, you can migrate PSTN (off-net) connectivity to the unified communications routers and start to incorporate IP phones at larger sites (Figure 1). After all sites are running IP telephony, you can begin deploying IP-based applications such as IP unified messaging, personal assistants, and extension mobility.
The unified communications routers are an ideal solution for circuit-switched PBX and PSTN access within a Cisco Unified Communications Manager-based IP telephony architecture.

Figure 1. IP Telephony Phased Migration: Migrate Circuit-Switched PSTN and PBX Connectivity to Unified Communications

As companies seek to deploy unified communications solutions across the entire enterprise - converging voice, video, and data across potentially thousands of sites - they require a solution that offers simple administration, virtually unlimited scalability, and high availability. The unified communications routers work in concert with the Cisco Unified Communications Manager, deployed in either a distributed or centralized call-processing model, to provide the unified communications solutions that enterprises require.

Centralized Call Processing

Demand for technology to help increase employee productivity and reduce costs is at an all-time high. At the same time, many organizations are struggling to deploy new applications and services because of unavailable capital budgets. The centralized call-processing model can provide technology to users who require it, while simultaneously providing ease of centralized management and maintenance of applications to network administrators.
Instead of deploying and managing key systems or PBXs in small offices, applications are centrally located at a corporate headquarters or data center, and accessed through the IP LAN and WAN. This deployment model allows branch-office users to access the full enterprise suite of communications and productivity applications for the first time, while lowering total cost of ownership (TCO). There is no need to "touch" each branch office each time a software upgrade or new application is deployed, accelerating the speed in which organizations can adopt and deploy new technology solutions.
The ability to quickly roll out new applications to remote users can provide a sustainable competitive advantage versus having to visit each of many branch-office sites to take advantage of new applications. An architecture in which a Cisco Unified Communications Manager and other Cisco IP Communications applications are located at the central site offers the following benefits:

• Centralized configuration and management

• Access at every site to all Cisco Unified Communications Manager features, next-generation contact centers, unified messaging services, personal productivity tools, mobility solutions, and software-based phones all the time

• IT staff not required at each remote site

• Ability to rapidly deploy applications to remote users

• Easy upgrades and maintenance

• Lower TCO

Survivable Remote Site Telephony

As enterprises extend their IP telephony deployments from central sites to remote offices, an important consideration is the ability to cost-effectively provide failover capability at remote branch offices. However, the size and number of these small-office sites preclude most enterprises from deploying dedicated call-processing servers, unified messaging servers, or multiple WAN links to each site to achieve the required high availability.
Cisco Unified Communications Manager with Survivable Remote Site Telephony (SRST) allows companies to extend high-availability IP telephony to their remote branch offices with a cost-effective solution that is easy to deploy, administer, and maintain. The SRST capability is embedded in the Cisco IOS ® Software that runs on the Cisco 2900, 3900, and 4400 Series unified communications routers. Cisco Unified Communications offers two types of call-processing survivability: Survivable Remote Site Telephony (SRST), which helps ensure survivability of basic and most critical IP telephony services, and Enhanced Survivable Remote Site Telephony, which survives basic IP telephony services as well as advanced Unified Communications supplementary services. For more information about SRST, go to http://www.cisco.com/go/srst.
SRST software automatically detects a connectivity failure between Cisco Unified Communications Manager and IP phones at the branch office. Using the Cisco Simple Network Automated Provisioning capability, SRST initiates a process to automatically configure the unified communications routers to provide call-processing backup redundancy for the IP phones and PSTN access in the affected office. The router provides essential call-processing services for the duration of the failure, helping ensure that critical phone capabilities are operational.
Upon restoration of the connectivity to the Cisco Unified Communications Manager, the system automatically shifts call-processing functions back to the primary Cisco Unified Communications Manager cluster. Configuration for this capability is performed only once in the Cisco Unified Communications Manager at the central site (Figure 2).

Figure 2. Centralized Cisco Unified Communications Manager Deployment with SRST

Cisco Unified Communications Router Features and Benefits

Simple Administration

• Provides centralized administration and management

• Enables administration of large dial plans

• Provides a single point of configuration for a Cisco IP Telephony network

Availability

• Provides for Cisco Unified Communications Manager redundancy; if a primary host Cisco Unified Communications Manager fails, call control fails over to the next available Cisco Unified Communications Manager server

• Offers branch-office survivability using SRST when connection to the Cisco Unified Communications Manager cluster is lost

Scalability

• Meets enterprise office requirements of small offices to large corporations

• Scales up to 40,000 users per cluster with Cisco Unified Communications Manager clustering

Investment Protection

• Provides a modular platform design with a growing list of more than 90 interface combinations

• Allows you to increase voice capacity while taking advantage of your existing investments in Cisco Unified Communications routers

Unified Communications Router with Cisco Unified Communications Manager Feature Summary

Table 1 summarizes the features of the unified communications routers with Cisco Unified Communications Manager.

Table 1. Cisco Unified Communications Routers with Cisco Unified Communications Manager Feature Summary

Cisco ISR 2900 and 3900

Cisco ISR 4400

   

SIP

MGCP

H.323

SIP

MGCP

H.323

Feature

Benefits

Y

Y1

Y

N

N

N

Analog FXS interfaces loop-start and ground-start signaling

This signaling facilitates direct connection to phones, fax machines, and key systems.

Y

N

Y

N

N

N

Analog E&M (wink, immediate, and delay) interfaces

These interfaces make direct connection to a PBX possible.

Y

Y

Y

N

N

N

Analog FXO interfaces loop-start and ground-start signaling

This feature facilitates connection to a PBX or key system and provides off-premises connections to or from the PSTN. Calling line ID (CLID) is available in MGCP mode.2

Y

N

Y

N

N

N

Analog direct inward dialing (DID)

Analog DID enables connection to the PSTN with DID operation.

Y

N

Y

N

N

N

Analog Centralized Automated Message Accounting (CAMA)

Analog CAMA facilitates analog PSTN connection for E-911 support.

Y

Y

Y

N

N

N

BRI Q.931 user side (NET3)

This feature enables connection to the PSTN.

Y

N

Y

N

N

N

BRI Q.931 network side (NET3)

This feature enables connection to a PBX.

Y

Y

Y

N

N

N

BRI Q.SIG-basic call (including calling number)

This feature facilitates connection to a PBX or key system.

Y

N

N3

N

N

N

BRI Q.SIG forward, transfer, and conference

These services enable connection to a PBX or key system.

N

Y4

N

N

Y

N

T1 E&M hookflash

This feature is used to transfer a call from time-division multiplexing (TDM) interactive voice response (IVR) to a PSTN or IP phone destination.

Y

Y

Y

Y

Y

Y

T1-CAS E&M (wink-start and immediate-start) interfaces

These interfaces facilitate connection to a PBX, key system, or PSTN.

Y

N

Y

Y

N

Y

T1-CAS E&M (delay dial) interfaces

These interfaces facilitate connection to a PBX, key system, or PSTN.

Y

N

Y

N

N

N

T1-CAS feature group D5

This feature is used to connect to a PBX or PSTN.

Y

N

Y

Y

N

Y

T1-CAS FXO (ground-start and loop-start) interfaces

These interfaces are used to connect to a PBX or key system and to provide off-premises connections.

Y

N

Y

Y

N

Y

T1-CAS FXS (ground-start and loop-start) interfaces

These interfaces are used to connect to a PBX or key system.

Y

N

Y

Y

N

Y

E1 CAS

E1 CAS enables connection to a PBX or PSTN.

Y

N

Y

Y

N

Y

E1 MelCAS

E1 MelCAS facilitates connection to a PBX or PSTN.

Y

N

Y

N

N

N

E1 R2 (more than 30 country variants)

E1 R2 enables connection to a PBX or PSTN.

Y

Y

Y

Y

N

Y

T1/E1 ISDN PRI Q.931 interfaces

These interfaces are used to connect to a PBX or key system and to provide off-premises connections to or from the PSTN or post, telephone, and telegraph (PTT).

Y

Y

Y

Y

N

Y

T1/E1 Q.SIG basic call (including calling number)

This feature is used to connect to a PBX.

Y6

Y

N3

Y

N

N

T1/E1 Q.SIG, including call diversion and forward, transfer, calling and connected ID services, and message-waiting indicator

This feature is used to connect to a PBX.

N

Y

Y

N

N

N

T1/E1 External Signaling (ext-sig)

This feature is used to enable a connection trunk for common channel signaling (TCCS) application.

Y

Y

Y

Y

N

Y

Out-of-band dual-tone multifrequency (DTMF)

This feature carries DTMF tones and information out of band for clearer transmission and detection.

N

Y

N

N

N

N

Single point of gateway configuration for a Cisco IP Telephony network

This feature centralizes and automates the configuration process for MGCP unified communications routers by making them configurable on the Cisco Unified Communications Manager. Configuration information is automatically downloaded at startup and after any configuration change.

Y

Y

Y

Y

Y

Y

Cisco Unified Communications Manager failover redundancy

When the unified communications router loses contact with the primary Cisco Unified Communications Manager, the gateway uses the next available Cisco Unified Communications Manager.

Y

Y

Y7

Y

N

Y

Cisco Unified Communications Manager call preservation during failover

Existing calls are preserved during a failover to the next available Cisco Unified Communications Manager. Calls are also preserved upon restoration of the primary host Cisco Unified Communications Manager.

Y

Y

Y

Y

Y

Y

SRST and gateway fallback

When contact with the Cisco Unified Communications Manager cluster is lost, SRST provides basic call handling for the IP phones. Gateway fallback provides support for PSTN telephony interfaces on the branch-office router for the duration of the loss.

Y

N

Y7

Y

N

Y

Call preservation for existing BRI and PRI calls during gateway fallback and recovery

Existing calls are preserved during a loss of connection to the Cisco Unified Communications Manager cluster and gateway fallback. Calls are also preserved upon restoration of the Cisco Unified Communications Manager connection.

Y

Y

Y7

Y

N

Y

Call preservation for existing T1/E1 (CAS) and analog calls during gateway fallback and recovery

Existing calls are preserved during a loss of connection to the Cisco Unified Communications Manager cluster and gateway fallback. Calls are also preserved upon restoration of the Cisco Unified Communications Manager connection.

Y

Y

Y

N

N

N

Multicast music on hold (MoH) - centralized

This feature helps the unified communications router deliver music streams from a MoH server to users on on- and off-net calls.

N

Y

N

N

N

N

Multicast MoH - distributed

This feature helps the unified communications router deliver music streams to users through the router-embedded MoH server to on- and off-net calls.

N

Y

Y

N

N

Y

Tone on hold

Tone indicates when a user is placed on hold.

N

Y

N

N

N

N

Tone-on-hold timer tuning

Tone on hold is generated locally in the gateway for play to the PSTN. Tone-on-hold timer tuning allows the use of service parameter settings in Cisco Unified Communications Manager for specification of the time between beeps.

Y

Y

Y

Y

N

Y

Caller ID support8

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.

N

Y

Y

N

N

Y

Malicious caller ID (MCID) over PRI

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.

N

Y

N

N

N

N

Multilevel precedence and preemption (MLPP) for T1-PRI (backhaul) and T1-CAS (wink start only)

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.

Y

Y

Y

Y

N

Y

Group III fax support

Group III fax support facilitates transmit of Group III faxes between the PSTN and IP using either fax relay or fax pass-through methods.

Y

Y9

Y

Y

N

Y

T.38 standards-based fax support

This feature enables transmit T.38 fax between the PSTN and IP.

Y

N

Y

Y

N

Y

Private-line automatic ringdown (PLAR)

PLAR provides a dedicated connection to another extension or an attendant.

Y

Y

Y

Y

N

Y

Standards-based codecs10

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).

Y

Y

Y

Y

N

Y

Voice activity detection (VAD)

VAD conserves bandwidth during a call when there is no active voice traffic to send.

Y

Y

Y

Y

N

Y

Comfort-noise generation

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.

Y

N

Y

Y

N

Y

Busy out

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.

-

-

Y

-

-

Y

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

These versions of H.323 use industry-standard signaling protocols for setting up calls between gateways, gatekeepers, and H.323 endpoints.

Y

-

-

Y

-

-

SIP IETF RFC 3261 support

This feature uses industry-standard signaling protocols for setting up calls between gateways and SIP proxies or SIP Back-to-Back User Agents.

Y

Y

Y

Y

N

Y

Authentication, authorization, and accounting (AAA)

AAA supports debit card and credit card (prepaid and postpaid calling card) applications.

Y

N

Y

Y

N

Y

IVR support

IVR offers Automated-Attendant support, voicemail support, or call routing based on service desired.

Y

N

Y

Y

N

Y

Automated Attendant

This feature uses IVR to provide automated call-answering and -forwarding services.

Y

N

Y

N

N

N

VoiceXML

VoiceXML controls calls "in queue" at the gateway for call-center applications. Calls are redirected only when an agent becomes available.

Y

Y

Y7

Y

N

Y

Overlap sending over voice over IP (VoIP)

This feature speeds variable-length dial strings dialing.

Y

N

Y

N

N

N

Voice + Data integrated access

This feature makes the voice and serial data interfaces available on the same T1/E1.

Y

N

Y

Y

N

Y

Fractional PRI

This feature allows for use of fewer than 23/30 channels on a T1/E1. Other channels are either unused or used for data.

Y

N

Y

N

N

N

FXO tone answer supervision

This feature facilitates the use of tones to signal answering a call and the start of a CDR.

Y

Y

Y

N

N

N

FXO disconnect supervision

This feature makes battery reversal or tones available for use to disconnect FXO calls.

Y

N

Y

N

N

N

ISDN video switching on gateway (drop DSPs)

This feature allows ISDN-based videoconferencing systems to connect and be switched back out the ISDN.

Y

N

Y

Y

N

Y

Set numbering plan type of outgoing calls

You can change the numbering plan on the gateway before your call goes out over the PSTN.

Y

Y

Y

Y

N

Y

Name display on PRI using FACILITY IE (caller name [CNAM])

This feature provides caller name display on IP phones for PSTN calls.

N

Y11

N

N

N

N

Secure Telephone Unit (STU) and Secure Terminal Equipment (STE) phone support

STU and STE support the U.S. Department of Defense analog and BRI secure phones.

N

Y12

N

N

N

N

Connection to Defense Switched Network (DSN)

This feature supports the U.S. Department of Defense private TDM network.

Y13

Y14

Y15

Y

N

Y

Secure Real-Time Transport Protocol (SRTP): Media authentication and encryption on unified communications routers

This feature enables secure gateway-to-gateway calls and secure IP phone-to-gateway calls.

Y

N

N

Y

N

N

SRTP-Real-Time Transport Protocol (RTP) fallback operations

This feature enables the fallback from SRTP to RTP during capabilities negotiation at the time of call setup.

Y16

Y17

Y18

Y

N

Y

Signaling encryption SIP: Transport Layer Security (TLS), MGCP/H,323: IP Security IPsec)

This feature encrypts signaling communication between unified communications and Cisco Unified Communications Manager.

Y

N

Y

N

N

N

H.320 video gateway support

This feature integrates ISDN trunks with both voice and video traffic.

Y

N

Y

Y

N

Y

Virtualization (Virtual Route Forwarding [VRF])

This feature supports virtual segmentation of the network using VRF.

Y

N

N

Y

N

N

IPv6

IPv6 support enables interworking with IPv6-capable networks.

Y

N

N

Y

N

N

Dynamic Host Configuration Protocol (DHCP)

DHCP enables acquisition of gateway configuration parameters from the DHCP server.

Y

Y

Y

Y

N

Y

Resource Reservation Protocol (RSVP) support

This feature helps assure high-quality voice by enabling resource reservation for call admission control.

Y

N

N

Y

N

N

History Info support

This feature enables support for the History Info header to transport the history information of a call.

Y

N

N

Y

N

N

SIP privacy and identity

This feature enables transport of identity, both preferred (P-Preferred Identity [PPI]) and asserted (P-Asserted Identity [PAI]).

Y

N

N

Y

N

N

Signaling health monitoring

This feature enables monitoring of the signaling connection across the signaling trunk.

Y

Y

Y

Y

N

Y

Q.SIG and Q.931 Tunneling

This feature enables transparent tunneling of ISDN signaling over VoIP signaling.

Y19

N

Y19

N

N

N

Ad hoc videoconference service and unified video transcoding service on Cisco Integrated Services Routers Generation 2 (ISR G2)

This feature enables ad hoc video-conferencing and unified video transcoding on the Cisco 2900 and 3900 Series Integrated Services Routers (ISRs)

N

Y19

N

N

N

N

Cisco V.150.1 Minimum Essential Requirements

This feature delivers enhancements to the voice gateways to satisfy requirements outlined in the UCR2008 specification. Specifically, support is added for the V.150.1 Minimum Essential Requirements (modem relay) and Modem over IP (MoIP) and Fax over IP (FoIP).

1Supports loop-start signaling only
2Requires Cisco IOS Software Release 12.4(20)T or later and Cisco Unified Communications Manager 8.0 or later
3Supported between gateways in the absence of Cisco Unified Communications Manager
4Requires Cisco IOS Software Release 12.4(4)T or later and Cisco Unified Communications Manager 4.2 or later
5Not supported on the Cisco 1700 Series unified communications routers
6Support is for forward, transfer, and conference; message-waiting indicator is from SIP to QSIG (not the reverse) and requires Cisco IOS Software Release 12.4(11)T; calling and connected ID are not supported
7Requires Cisco Unified Communications Manager 4.1(3)SR2 or later and Cisco IOS Software Release 12.4(9)T or later; no gatekeeper support
8Requires Cisco IOS Software Release 12.4(20)T or later
9Requires Cisco Unified Communications Manager 4.2(3)
10G.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
11Requires Cisco IOS Software Release 12.3(14)T or later; BRI operations limited: single B-channel voice only; testing limited to three phones; no data call support
12Requires Cisco IOS Software Release 12.4(2)T or later
13Requires Cisco IOS Software Release 12.4(15)T or later and Cisco Unified Communications Manager 5.0 (line-side) or later; Cisco Unified Communications Manager trunk-side support currently not available
14Requires Cisco IOS Software Release 12.4(3) or later and Cisco Unified Communications Manager 4.1 or later
15Requires Cisco IOS Software Release 12.4(6)T2 or later and Cisco Unified Communications Manager 5.0 or later
16Requires Cisco IOS Software Release 12.4(6)T or later and Cisco Unified Communications Manager 5.0 or later
17Requires Cisco IOS Software Release 12.4(3) or later and Cisco Unified Communications Manager 4.1 or later
18Requires Cisco IOS Software Release 12.4(6)T1 and Cisco Unified Communications Manager 5.0 or later
19Requires Cisco IOS Software Release 15.1(4)M or later and Cisco Unified Communications Manager 8.6 or later

Unified Communications Router with Cisco Unified Communications Manager Minimum System Requirements

Tables 2 through 5 give system requirements for the unified communications routers.

Table 2. Cisco Unified Communications Routers with Cisco Unified Communications Manager Minimum System Requirements Using SIP

Platform

TDM Protocol or Feature

Minimum Cisco IOS or Cisco IOSXE Software Release*

Minimum Cisco Unified Communications Manager Release

2900 and 3900 ISRs

Analog (FXS and FXO)

12.4(6)T

5.0

2900 and 3900 ISRs

BRI

12.4(6)T

5.0

2900 and 3900 ISRs

T1 CAS and T1/E1 PRI

12.4(6)T

5.0

4400 ISRs

Analog (FXS and FXO)

-

-

4400 ISRs

BRI

-

-

4400 ISRs

T1 CAS and T1/E1 PRI

XE 3.10

8.6

* This table shows when a Cisco IOS Software particular interface type was first tested with Cisco Unified Communications Manager. It does not document when individual network modules (NMs), advanced integration modules (AIMs), service modules (SMs), integrated service modules (ISMs), and platforms are first supported in Cisco IOS Software. For this information refer to the data sheet for the relevant interface. Note that when using SIP, Cisco Unified Communications Manager does not need to know which NM, SM, AIM, ISM, or platform is used. Hence, when Cisco Unified Communications Manager supports a particular protocol or feature, this support is sufficient for operation.

Table 3. Cisco Unified Communications Routers with Cisco Unified Communications Manager Minimum System Requirements Using H.323

Platform

TDM Protocol or Feature

Minimum Cisco IOS or Cisco IOS XE Software Release*

Minimum Cisco Unified Communications Manager Release

2900 and 3900 ISRs

Analog (FXS and FXO)

12.2(1)M

3.0(5a)

2900 and 3900 ISRs

BRI

12.2(1)M

3.0(5a)

2900 and 3900 ISRs

T1 CAS and T1/E1 PRI

12.1(2)T

3.0(5a)

2900 and 3900 ISRs

T1/E1 QSIG

12.1(2)T

3.0(5a)

2900 and 3900 ISRs

MCID

12.3(11)T

4.0

4400 ISRs

Analog (FXS and FXO)

-

-

4400 ISRs

BRI

-

-

4400 ISRs

T1 CAS and T1/E1 PRI

3.10

8.6

4400 ISRs

T1/E1 QSIG

3.10

8.6

4400 ISRs

MCID

3.10

8.6

* This table shows when a particular interface type is first supported in Cisco IOS Software. It does not document when individual NMs, SMs, AIMs, ISMs, and platforms are first supported in Cisco IOS Software. For this information refer to the data sheet for the relevant interface. Note that in H.323 mode, Cisco Unified Communications Manager does not need to know which NM, SM, AIM, ISM, or platform is used. Hence, when Cisco Unified Communications Manager supports a particular protocol or feature, this support is sufficient for operation.

Table 4. Cisco Unified Communications Routers with Cisco Unified Communications Manager Minimum System Requirements Using MGCP

Active Platforms

Interface Part Number

TDM Protocol or Feature

Minimum Cisco IOS or Cisco IOS XE Software Release

Minimum Cisco Unified Communications Manager Release

Cisco 2901, 2911, 2921, and 2951

EVM-HD-8FXS/DID with EM3-HDA-8FXS, EM-HDA-6FXO, or EM-HDA-3FXS/4FXO (Cisco 2911, 2921, and 2951 only)

Analog FXS and FXO

15.0.1M

6.1.5, 7.1.3, or 8.0

VIC3-2FXS/DID, VIC3-2FXS-E/DID, VIC3-4FXS/DID, NM-HD-1V/2V/2VE, and NM-HDV2

Analog FXS and FXO

15.0.1M

6.1.5, 7.1.3, or 8.0

EVM-HD-8FXS/DID with EM-4BRI-NT/TE, VIC2-2BRI-NT/TE, NM-HD-1V/2V/2VE, and NM-HDV2

BRI

15.0.1M

6.1.5, 7.1.3 or 8.0

NM-HDV2

T1 CAS E&M, T1/E1 PRI, and T1/E1 QSIG (basic)

15.0.1M

6.1.5, 7.1.3 or 8.0

VWIC2-1MFT-T1/E1, VWIC2-2MFT-T1/E1, VWIC2-1MFT-G703, and VWIC2-2MFT-G703

T1 CAS E&M, T1/E1 PRI, QSIG**, and MLPP***

15.0.1M

6.1.5, 7.1.3 or 8.0

VWIC3-1MFT-T1/E1, VWIC3-2MFT-T1/E1, VWIC3-1MFT-G703, and VWIC3-2MFT-G703

T1 CAS, T1/E1 PRI, QSIG**, MLPP*** and channel-groups

15.0.1M3

7.1.5 or 8.0.2

VWIC3-4MFT-T1/E1

Not supported on Cisco 2901

T1 CAS, T1/E1 PRI, QSIG**, MLPP*** and channel groups

15.1.3T

7.1.5 or 8.0.2

Cisco 3925 and 3945

EVM-HD-8FXS/DID with EM3-HDA-8FXS, EM-HDA-6FXO, or EM-HDA-3FXS/4FXO

Analog FXS and FXO

15.0.1M

6.1.5, 7.1.3, or 8.0

VIC3-2FXS/DID, VIC3-2FXS-E/DID, VIC3-4FXS/DID, NM-HD-1V/2V/2VE, and NM-HDV2

Analog FXS and FXO

15.0.1M

6.1.5, 7.1.3, or 8.0

EVM-HD-8FXS/DID with EM-4BRI-NT/TE, VIC2-2BRI-NT/TE, NM-HD-1V/2V/2VE, and NM-HDV2

BRI

15.0.1M

6.1.5, 7.1.3, or 8.0

NM-HDV2

T1 CAS E&M, T1/E1 PRI, and T1/E1 QSIG (basic)

15.0.1M

6.1.5, 7.1.3, or 8.0

VWIC2-1MFT-T1/E1, VWIC2-2MFT-T1/E1, VWIC2-1MFT-G703, and VWIC2-2MFT-G703

T1 CAS E&M, T1/E1 PRI, QSIG**, and MLPP***

15.0.1M

6.1.5, 7.1.3, or 8.0

VWIC3-1MFT-T1/E1, VWIC3-2MFT-T1/E1, VWIC3-1MFT-G703, and VWIC3-2MFT-G703

T1 CAS, T1/E1 PRI, QSIG**, MLPP***, and channel groups

15.0.1M3

7.1.5 or 8.0.2

VWIC3-4MFT-T1/E1

T1 CAS, T1/E1 PRI, QSIG**, MLPP***, and channel groups

15.1.3T

7.1.5 or 8.0.2

Cisco 3925E and 3945E

EVM-HD-8FXS/DID with EM3-HDA-8FXS, EM-HDA-6FXO, or EM-HDA-3FXS/4FXO

Analog FXS and FXO

15.1.1T

7.1.5 or 8.0.2

VIC3-2FXS/DID, VIC3-2FXS-E/DID, VIC3-4FXS/DID, NM-HD-1V/2V/2VE, and NM-HDV2

Analog FXS and FXO

15.1.1T

7.1.5 or 8.0.2

EVM-HD-8FXS/DID with EM-4BRI-NT/TE, VIC2-2BRI-NT/TE, NM-HD-1V/2V/2VE, and NM-HDV2

BRI

15.1.1T

7.1.5 or 8.0.2

NM-HDV2

T1 CAS E&M, T1/E1 PRI, and T1/E1 QSIG (basic)

15.1.1T

7.1.5 or 8.0.2

VWIC2-1MFT-T1/E1, VWIC2-2MFT-T1/E1, VWIC2-1MFT-G703, and VWIC2-2MFT-G703

T1 CAS E&M, T1/E1 PRI, QSIG**, and MLPP***

15.1.1T

7.1.5 or 8.0.2

VWIC3-1MFT-T1/E1, VWIC3-2MFT-T1/E1, VWIC3-1MFT-G703, and VWIC3-2MFT-G703

T1 CAS, T1/E1 PRI, QSIG**, MLPP***, and channel groups

15.0.1M3

7.1.5 or 8.0.2

VWIC3-4MFT-T1/E1

T1 CAS, T1/E1 PRI, QSIG**, MLPP***, and channel groups

15.1.3T

7.1.5 or 8.0.2

Cisco 4451

NIM-1MFT-T1/E1,

NIM-2MFT-T1/E1,

NIM-4MFT-T1/E1,

NIM-8MFT-T1/E1

T1 CAS, T1/E1 PRI, QSIG**, MLPP***, and 2 channel groups

XE 3.10

8.6

NIM-1CE1T1-PRI,

NIM-2CE1T1-PRI,

NIM-8CE1T1-PRI

T1 CAS, T1/E1 PRI, QSIG**, MLPP***, and 24 channel-groups

XE 3.10

8.6

** QSIG supplementary requires Cisco Unified Communications Manager 4.0 or later. QSIG basic services were first introduced with Cisco Communications Manager 3.3 and Cisco IOS Software Release 12.2.11T.
*** MLPP requires Cisco Unified Communications Manager 4.0.2 or later.

Table 5. Cisco Unified Communications Routers with Cisco Unified Communications Manager Minimum System Requirements for Conferencing, Transcoding, and Media Termination Point

Active Platforms

Interface Part Numbers

TDM Protocol or Feature

Minimum Cisco IOS Software Release

Minimum Cisco Unified Communications Manager Release

Cisco 2901, 2911, 2921, and 2951

Onboard PVDM2 and PVDM3 DSPs

Conferencing and transcoding

15.0.1M

6.1.5, 7.1.3 or 8.0

Onboard PVDM2 and PVDM3 DSPs

MTP and RFC 2833

15.0.1M

6.1.5, 7.1.3 or 8.0

NM-HD-1V/2V/2VE and NM-HDV2

Conferencing and transcoding

15.0.1M

6.1.5, 7.1.3 or 8.0

NM-HD-1V/2V/2VE and NM-HDV2

MTP and RFC 2833

15.0.1M

6.1.5, 7.1.3 or 8.0

Cisco 3925 and 3945

Onboard PVDM2 and PVDM3 DSPs

Conferencing and transcoding

15.0.1M

6.1.5, 7.1.3, or 8.0

Onboard PVDM2 and PVDM3 DSPs

MTP and RFC 2833

15.0.1M

6.1.5, 7.1.3, or 8.0

NM-HD-1V/2V/2VE and NM-HDV2

Conferencing and transcoding

15.0.1M

6.1.5, 7.1.3, or 8.0

NM-HD-1V/2V/2VE and NM-HDV2

MTP and RFC 2833

15.0.1M

6.1.5, 7.1.3, or 8.0

Cisco 3925E and 3945E

Onboard PVDM2 and PVDM3 DSPs

Conferencing and transcoding

15.1.1T

7.1.5 or 8.0.2

Onboard PVDM2 and PVDM3 DSPs

MTP and RFC 2833

15.1.1T

7.1.5 or 8.0.2

NM-HD-1V/2V/2VE and NM-HDV2

Conferencing and transcoding

15.1.1T

7.1.5 or 8.0.2

NM-HD-1V/2V/2VE and NM-HDV2

MTP and RFC 2833

15.1.1T

7.1.5 or 8.0.2

Cisco 4451

Onboard PVDM4 DSPs

Conferencing,

Transcoding,

MTP, and RFC 2833

XE 3.10

8.6

Voice Performance

Tables 6 and 7 give information about connectivity and CPU performance, respectively, on the unified communications routers.

Table 6. Maximum Physical DS-0 Connectivity on the Cisco Unified Communications Routers*

 

Cisco 2901

Cisco 2911

Cisco 2921

Cisco 2951

Cisco 3925

Cisco 3945

Cisco 3925E

Cisco 3945E

Cisco 4451

FXS

16

40

40

64

64

112

60

108

-

FXO and CAMA

16

28

28

40

40

64

36

60

-

E&M

8

12

12

16

16

24

14

22

-

Analog DID

16

32

32

48

48

80

44

76

-

BRI ports

8

16

16

24

24

40

22

38

-

T1/E1 ports

8

20

20

24

24

32

20

28

24

T1 channels

192

480

480

576

576

768

480

672

576

E1 channels

240

600

600

720

720

960

600

840

720

* This table contains physical connectivity numbers. You should also use CPU performance as a guide to determine how many voice calls can actually be supported on each platform.

Table 7. CPU Performance on the Cisco Unified Communications Routers*

 

Cisco 2901

Cisco 2911

Cisco 2921

Cisco 2951

Cisco 3925

Cisco 3945

Cisco 3925E

Cisco 3945E

Cisco 4451

VoIP Performance: Maximum Number of Simultaneous Calls (not exceeding 75% platform CPU usage)

Cisco Unified Border Element

 

100

200

400

600

800

950

2100

2500

2500

Standalone Unified Communications Router1

 

No encryption

100

150

240

400

720

960

600

840

720

SIP TLS with SRTP

100

150

240

400

720

880

600

840

720

H.323 Signaling in IPsec with SRTP

100

150

240

400

720

780

600

840

720

H.323 Signaling and media in IPsec

100

150

195

325

360

385

600

840

720

WAN Edge Gateway2

 

No encryption

100

150

240

400

610

650

600

840

720

SIP TLS with SRTP

100

150

240

400

600

645

600

840

720

H.323 Signaling in IPsec with SRTP

100

150

240

400

530

565

600

840

720

H.323 Signaling and media in IPsec

100

125

145

235

265

285

600

840

720

WAN Edge Gateway with Compressed Real-Time Protocol (CRTP)3

 

No encryption

100

150

240

400

510

550

600

840

720

SIP TLS with SRTP

100

150

240

400

500

540

600

840

720

H.323 Signaling in IPsec with SRTP

100

150

240

400

445

475

600

840

720

H.323 Signaling and media in IPsec

95

105

120

200

220

240

600

840

720

VoIP Performance: Maximum Number of Calls per Second (not exceeding 75-percent CPU)

 
 

1

1.5

2

3

10

15

30

35

40

1Gigabit Ethernet or Fast Ethernet egress; no quality-of-service (QoS) features; voice traffic only
2T1/E1 or High-Speed Serial Interface (HSSI) serial egress; some QoS features; voice and small amount of data traffic
3T1/E1 or HSSI serial egress; some QoS features; CRTP; voice and small amount of data traffic

Notes:
1. All results represent G.729A or G.711 (20-ms packetization) switched H.323 calls with VAD turned off.
2. The call success rate (CSR) of all tests is 98 to 100 percent.
3. Call duration of tests is 180 seconds except for calls-per-second rate testing, where the duration is shorter.
The test release is Cisco IOS Software Release 15.0.1M for Cisco 2900 and 3900 ISRs. For the testing of the Cisco 4400, Cisco IOS XE Software Release 3.10 was used. This document contains general numbers as a guide to the approximate performance of the unified communications routers. The numbers are extrapolated from a large number of disparate tests, test conditions, and traffic patterns. Several nontesting factors have also been accounted for. Therefore, actual test results will vary, and we encourage you to do proof-of-concept testing for more specific performance numbers for a specific scenario, traffic pattern, or release.

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