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Cisco Router C3925E-VSEC-CUBEK9

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About Product

C3925E VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25

Model:

C3925E-VSEC-CUBEK9 Cisco 3925E Secure Voice & Unified Border Element

Brand:

Cisco

C3925E-VSEC-CUBEK9 Overview

C3925E-VSEC-CUBEK9 is the Cisco 3925E router with Voice Sec and CUBE Bundle, including PVDM3-64, UC and SEC License PAK, and FL-CUBEE-25.

Quick Specs

Table 1 shows the Quick Specs of the C3925E-VSEC-CUBEK9.

Product Code

C3925E-VSEC-CUBEK9

Bundle

C3925E VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25

Services Performance Engine module

w/SPE 200

Rack Units

3U

Interfaces

4 integrated 10/100/1000 Ethernet ports with 2 SFP ports

Expansion Slot(s)

2 service module slots

3 onboard digital signal processor (DSP) slots

3 Enhanced High-Speed WAN Interface Card (EHWIC) slots

RAM

1 GB (installed) / 2 GB (max)

Flash Memory

256 MB (installed) / 8 GB (max)

Dimensions

47.6 cm x 43.8 cm x 13.3 cm

Product Details

C3925E-VSEC-CUBE/K9 provides Voice Sec and CUBE bundle.

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 Routers

Table 4 shows the comparison between CISCO3925E/k9 and C3925E-VSEC-CUBEK9.

Model

CISCO3925E/k9

C3925E-VSEC-CUBEK9

Bundle

N/A

C3925E VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25

SPE

w/SPE 200

w/SPE 200

Interface

4GE

4GE

EHWIC slots

3

3

SM slots

2

2

ISM slots

0

0

DSP

3

3

SFP

2

2

Rack Unit

3U

3U

Get more information

C3925E-VSEC-CUBEK9 Specification

C3925E-VSEC-CUBEK9 Specifications

Manufacturer

Cisco Systems, Inc

Manufacturer Part Number

C3925E-VSEC-CUBEK9

Bundle

C3925E VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25

Product Type

Router

Form Factor

Desktop – modular – 3U

Connectivity Technology

Wired

Data Link Protocol

Ethernet, Fast Ethernet, Gigabit Ethernet

Capacity

Cisco Unified Communications Manager Express sessions : 400

Cisco Unified SRST sessions : 1350

Network / Transport Protocol

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

Features

Firewall protection, VPN support, MPLS support, Syslog support, content filtering, IPv6 support, Class-Based Weighted Fair Queuing (CBWFQ), Weighted Random Early Detection (WRED), Access Control List (ACL) support, 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 G.168

IP Telephony

Voice Codecs

G.711, G.722, G.723.1, G.728, G.729, G.729a, G.729ab, G.726

Connectivity Slots

Router Interfaces

WAN : 4 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)

3 ( 3 ) x EHWIC

3 ( 2 ) x PVDM

2 ( 1 ) x CompactFlash Card

2 ( 2 ) x service module (SM)

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, AS/NZS 60950-1, AS/NZ 3548 Class A, CAN/CSA-E60065-00, ICES-003 Class A, FCC CFR47 Part 15, EN300-386, UL 60950-1, IEC 60950-1, EN 60950-1, CSA C22.2 No. 60065, BSMI CNS 13438, NEBS

System software

Software Included

Cisco Unified Border Element (25 sessions)

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