C3945 VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25

Model:

C3945-VSEC-CUBE/K9 Cisco 3925 Secure Voice & Unified Border Element

Brand:

Cisco

C3945-VSEC-CUBE/K9 Overview

C3945-VSEC-CUBE/K9 is the Cisco 3945 router with VSEC CUBE Bundle, including PVDM3-64, UC SEC License and FL-CUBEE-25.

Quick Specs

Table 1 shows the Quick Specs.

Product Code

C3945-VSEC-CUBE/K9

Bundle

PVDM3-64, UC SEC License and FL-CUBEE-25

Services Performance Engine module

w/SPE 150

Rack Units

3U

Interfaces

3 integrated 10/100/1000 Ethernet ports with 2 ports capable of RJ-45 or SFP connectivity

Expansion Slot(s)

4 ( 4 ) x HWIC

3 ( 2 ) x PVDM

2 ( 1 ) x CompactFlash Card

1 ( 1 ) x expansion slot

RAM

1 GB (installed) / 2 GB (max)

Flash Memory

256 MB (installed) / 8 GB (max)

Dimensions

18.7 in x 17.2 in x 5.2 in

Net Weight

39 lbs

Product Details

Note:

①

②

③

On/Off switch

Power connector

LEDs area

Note:

①

Compact flash 0 and 1 (0, right)

⑦

10/100/1000 Ethernet port GE 0/1 and GE 0/2 (GE 0/1, down)

②

EHWIC 0,1,2 and 3 (0, far right)

⑧

10/100/1000 Ethernet port GE 0/0

③

USB serial port

⑨

USB 0 and USB 1 (1,top)

④

RJ-45 serial console port

⑩

Ground

⑤

AUX port

Service module slots

⑥

SFP 1 and SFP 2 (2, top)

Table 2. Voice Security Bundle Features Table

Authentication and Encryption Features

• Media encryption of voice RTP streams using SRTP

• Exchange of RTP Control Protocol (RTCP) information using secure RTCP

• SRTP to RTP fallback for calls between secure and insecure endpoints

• Secure calls supported in Cisco Unified Survivable Remote Site Telephony (SRST) mode during WAN failover

• Compressed RTP (CRTP) supported with media encrypted calls using SRTP

Authentication and Encryption Algorithm

• Supports AES-128 encryption algorithm

• Supports the HMAC secure hash authentication algorithm (SHA 1)

Signaling Authentication and Encryption Features

• Gateway to Cisco Unified Communications Manager signaling and encryption uses IPSec for Media Gateway Control Protocol (MGCP), H.323 and SIP gateways

• IP phone to Cisco Unified Survivable Remote Site Telephony router signaling and encryption uses TLS

(Transport Layer Security)

Protocol Support

• MGCP 0.1 (supports MGCP gateways with Cisco Unified Communications Manager)

• H.323 (supported on H.323 gateways and CUBE; Cisco Unified Communications Manager interoperability is optional)

• Session Initiation Protocol (SIP)

• SCCP (Cisco Unified IP Phone) in SRST mode

Module Support

• Any module that has PVDM2, PVDM3 and/or built-in DSP

Codec Support

• G.711, G.729A, and G.729

Table 3 shows the Cisco Unified Border Element Features (CUBE Versions Include 9.5.1 or Later).

Feature

Support Details

Protocols

H.323 and SIP

Protocol and signal interworking

H.323 to H.323 (including Cisco Unified Communications Manager)

H.323 to SIP (including Cisco Unified Communications Manager)

SIP to SIP (including Cisco Unified Communications Manager)

SIP to SIP (including Cisco TelePresence calls)

Media support

RTP, RTCP, and Binary Floor Control Protocol (BFCP)

Sub-RTCP for media statistics

Media interworking

SIP delayed-offer to SIP early-offer interworking for audio or video calls

H.323 Slow Start to H.323 Fast Start for audio calls

Media modes

Media flow-through

Media flow-around

Signaling transport mode

TCP

User Datagram Protocol (UDP)

TCP-to-UDP interworking

Fax support

T.38 fax relay

Fax pass-through

Fax over G711

Modem support

Modem pass-through

Modem over G711

Dual-tone multifrequency (DTMF)

H.245 alphanumeric

H.245 signal

RFC 2833

SIP notify

Key Press Markup Language (KPML)

Interworking capabilities include:H.323 to SIP

RFC 2833 to G.711 in-band DTMF *

Various SIP-to-H.323 DTMF interworking options

RFC 2833 to KPML

Supplementary services

Call hold, call transfer, and call forwarding for H.323 networks using H.450 and transparent passing of Empty Capability Set (ECS)

SIP-to-SIP supplementary services (holds and transfers) support using REFER

SIP-to-SIP supplementary services (holds and transfers) support using REINVITE

H.323-to-SIP supplementary services for Cisco Unified Communications Manager with media termination point (MTP) on the H.323 trunk

Internetworking

Configurable SIP profiles to manipulate SIP message content, including header fields andSession Descriptor Protocol (SDP) attributes

P-Asserted-Identity (PAI), P-Preferred-Identity (PPI), and Remote-Part-ID (RPID) internetworking

Unsupported Multipurpose Internet Mail Extensions (MIME)-type attachment pass-through

Unsupported SIP header pass-through

Dial-peer bind (allows Cisco Unified Border Element to connect to multiple different service providers)

Incoming dial-peer match based on remote IP address

Assisted RTCP for Microsoft Lync Interoperability

Call routing and dialing options

E164-based dialing

Uniform Resource Identifier (URI)-based dialing

Routing based on nonsequential lists (either E164 or URI or both)

Dial Peer Groups (Trunk Groups) (outbound routing determined by inbound dial pattern)

Server Groups to define order of selection of alternative or backup routing paths for outbound routing

Cisco Call Admission Control (CAC)

Maximum number of calls per trunk (maximum number of calls)

CAC based on IP circuits

CAC based on total calls, CPU use, or memory use threshold

CAC based on bandwidth availability and call-spike detection

Resource Reservation Protocol (RSVP)

OPTIONS SIP message support

Support for response to OPTIONS-PING messages with OPTION- PING groups based on session target

Support for generation of in-dialog OPTIONS-PING messages

Support for generation of out-of-dialog OPTIONS-PING messages to control dial-peer status

Media recording

Media forking features for both voice and video to integrate with Cisco TelePresence Media Recording Servers

Active (SIP-based) and passive (application programming interface [API]-based) mechanisms for invoking media forking

IP Routing feature

Support for Cisco IOS Software-based routing features, including Border Gateway Protocol (BGP), Enhanced IGRP (EIGRP), and Multiprotocol Label Switching (MPLS)

Support for Cisco IOS Software-based policy routing features

Support for Cisco IOS Software-based access-control-list (ACL) features

Voice-quality statistics

Packet loss, jitter, and round-trip time (RTT)

Per-call leg call-quality statistics

Flexible NetFlow call-quality statistics and information

Sub-RTCP statistics collection

QoS

IP Precedence and differentiated-services-code-point (DSCP) marking

Per-call QoS packet marking

Network Address Translation (NAT) traversal

NAT traversal support for SIP phones deployed behind non-Application Line Gateway (ALG) data routers

Stateful NAT traversal

IPv4-to-IPv6 translation

Network hiding

IP network privacy and topology hiding

IP network security boundary

Intelligent IP address translation for call media and signaling

Back-to-back user agent, replacing all SIP-embedded IP addressing

History information-based topology hiding and call routing

Number translation

Number translation rules for voice-over-IP (VoIP) numbers

URI-based dialing translations

Codecs

G.711 mu-law and a-law

G.722 and G.722.2

G.723ar53, G.723ar63, G.723r53, and G.723r63

G.726r16, G.726r24, and G.726r32

G.728

G.729, G.729A, G.729B, and G.729AB

Internet Low Bitrate Codec (iLBC)

Midcall codec renegotiation

Adaptive Multirate (AMR) wideband

AAC-LD

Transcoding

Transcoding between any two different families of codecs from the following list:G.711 a-law and mu-law

G.729, G.729A, G.729B, and G.729AB

iLBC

G.722

Midcall transcoder insert and drop

Security

Rogue SIP invite and rogue RTP packet detection

Alerts for rogue packet activity

IP Security (IPsec)

Secure RTP (SRTP)

Transport Layer Security (TLS)

SRTP-to-RTP interworking

Authentication, authorization, and accounting (AAA)

AAA with RADIUS

Voice media applications

Tool Command Language (Tcl) scripts support for application customization

VoiceXML 2.0 script support for application customization

Web-based API to monitor and control signaling and media traffic

API

Web-based API compatible with Web Service Description Language (WSDL) development tools to support call monitoring and control, call-detail records (CDRs), and serviceability attribute interaction with external application; specifically designed for voice-policy applications

Billing

Standard CDRs for accurate billing available through:AAA records

Syslog

Simple Network Management Protocol (SNMP)

Lawful intercept

Provision of replicated packets to third-party mediation device

Remote phone proxy sessions

Termination of SIP-TLS and SRTP with registration pass-through to allow SIP-based endpoints, including Cisco Unified IP Phone 7900, 8900, and 9900 models and Jabber® Voice Client, to connect from remote sites through the Internet without requiring IPsec VPN to Cisco Unified Communications Manager, Cisco Business Edition, or Cisco HCS (not included with NANOCUBE license)

Line-side back-to-back user agent NANOCUBE sessions

Termination of Cisco Shared Port Adapter (SPA) and other third-party SIP endpoints with registration passthrough and survivability for use with third-party hosted call-control service provider services

Inter-Cluster Lookup Service (ILS) routing

Support for ILS routing to complement ILS dial-plan exchange between Cisco Unified Communications Manager clusters or to simplify call-routing complexity between multiple clusters

Video

Protocols

H.323 and SIP

Cisco endpoints supported

Cisco Unified Video Advantage (UVA) and Cisco TelePresence endpoints

Rich media

Simultaneous support for data, audio, and video

Signaling interworking

SIP delayed-offer to SIP early-offer calls

Media

Support for multiplex RTP calls (for Cisco TelePresence solution)

Simple Traversal of UDP through NAT (STUN)/Datagram TLS (DTLS) pass-through for telepresence

H.323-enhanced features

H.235 pass-through for secure calls

H.239 pass-through for picture-in-picture feature

QoS

DSCP markings to prioritize video streams as they traverse the network

Data support

T.120 data collaboration flow-around only

Camera control

Far-end camera control (FECC)

Video codecs

H.261

H.263

H.264

Network Management

Manageability and serviceability

Resource usage monitoring over SIP trunk

SNMP per-call quality traps

SNMP and syslog SIP trunk status messages

High Availability

High availability

Inbox redundancy on Cisco ASR 1006

Box-to-box redundancy on Cisco ASR 1000 (based on RG Infrastructure)

Box-to-box redundancy on Cisco ISRs (Hot Standby Router Protocol [HSRP]-based)Note: Media is preserved for active calls at time of failover in each redundancy configuration listed.

Compare to Similar Items

Table 4 shows the comparison.

Model

C3945-VSEC-CUBE/K9

CISCO3945/k9

SPE

w/SPE 150

w/SPE 150

Interface

3GE

3GE

Rack Unit

3U

3U

Bundle

C3945 VSEC CUBE Bundle, PVDM3-64,

UC SEC Lic, FL-CUBEE-25

None

Get more information

C3945-VSEC-CUBE/K9 Specification

C3945-VSEC-CUBE/K9 Specification

Manufacturer

Cisco Systems, Inc

Manufacturer Part Number

C3945-VSEC-CUBE/K9

Product Type

Router

Form Factor

Desktop – modular – 3U

Connectivity Technology

Wired

Data Link Protocol

Ethernet, Fast Ethernet, Gigabit Ethernet

Network / Transport Protocol

TCP/IP, UDP/IP, RSVP, IPSec, PPPoE, L2TPv3

Routing Protocol

OSPF, IS-IS, BGP, EIGRP, DVMRP, PIM-SM, IGMPv3, GRE, PIM-SSM, static IPv4 routing, static IPv6 routing, policy-based routing (PBR)

Remote Management Protocol

SNMP, RMON

Encryption Algorithm

SSL

Features

Firewall protection, NAT support, VPN support, MPLS support, Syslog support, content filtering, IPv6 support, Class-Based Weighted Fair Queuing (CBWFQ), Weighted Random Early Detection (WRED), Quality of Service (QoS), Dynamic Multipoint VPN (DMVPN)

Compliant Standards

IEEE 802.3, IEEE 802.1Q, IEEE 802.1ah, IEEE 802.1ag

DRAM Memory

1 GB (installed) / 2 GB (max)

Flash Memory

256 MB (installed) / 8 GB (max)

LED Status Lights Indicators

Link activity, power

Communications

Type

Voice / fax module

Digital Ports Qty

64

Protocols & Specifications

ITU T.38

IP Telephony

VoIP Protocols

H.323, SIP

Voice Codecs

G.711, G.722, G.723, G.728, G.729, G.729a, G.729ab, G.711u, G.711a, G.726, iLBC

IP Telephony Features

Call forwarding, automatic fax/modem detection and pass-through, call hold, call transfer, DTMF generation/detection

Connectivity Slots

Router Interfaces

3 x 10Base-T/100Base-TX/1000Base-T – RJ-45

Management : 1 x console – RJ-45

Management : 1 x console – mini-USB Type B

Serial : 1 x auxiliary – RJ-45

USB : 2 x 4 pin USB Type A

2 x SFP (mini-GBIC)

Expansion Slots Total (Free)

4 ( 4 ) x HWIC

3 ( 2 ) x PVDM

2 ( 1 ) x CompactFlash Card

1 ( 1 ) x expansion slot

Power Supply

Power Device

Power supply – internal

Voltage Required

AC 120/230 V ( 50/60 Hz )

Dimensions / Weight / Miscellaneous

Approximate Width

18.7 in

Depth

17.2 in

Height

5.2 in

Weight

39 lbs

Compliant Standards

CISPR 22 Class A, CISPR 24, EN55024, EN55022 Class A, EN50082-1, CAN/CSA-E60065-00, ICES-003 Class A, CS-03, AS/NZS 3548, FCC CFR47 Part 15, EN300-386, UL 60950-1, IEC 60950-1, EN 60950-1, CSA C22.2 No. 60065, BSMI CNS 13438

System software

Software Included

Cisco IOS Unified Communications, Cisco IOS Security, Cisco Unified Border Element Enterprise Edition (25 sessions)