C3945E-VSEC-CUBEK9

Product Code

C3945E-VSEC-CUBEK9

Bundle

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

Services Performance Engine module

w/SPE 250

Rack Units

3U

Interfaces

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

Expansion Slot(s)

4 service module slots

3 Enhanced High-Speed WAN Interface Card slots

3 onboard digital signal processor (DSP) slots

RAM

1 GB (installed) / 2 GB (max)

Flash Memory

256 MB (installed) / 8 GB (max)

Dimensions

18.7 x 17.2 x 5.7 in

?

?

?

On/Off switch

Power connector

LEDs area

?

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

?

SFP 1 and SFP 2 (1, Top)

?

USB serial port

?

USB 0 and USB 1 (1, Left)

?

AUX port

?

Ground

?

RJ-45 serial console port

?

Service module slots

?

10/100/1000 Ethernet ports (GE 0/3, Top; GE 0/2, Bottom)

?

Compact flash 0 and 1 (0, right)

?

10/100/1000 Ethernet ports (GE 0/1, Top; GE 0/0, Bottom)

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

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 audiovideo 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 E164URIboth)

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

?  Server Groups to define order of selection of alternativebackup 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,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,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 pass?through 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 clustersto 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.

Model

CISCO3945E/k9

C3945E-VSEC-CUBEK9

Bundle

N/A

C3945E VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25

SPE

w/SPE 250

w/SPE 250

Interface

4GE

4GE

EHWIC slots

3

3

SM slots

4

4

ISM slots

0

0

DSP

3

3

SFP

2

2

Rack Unit

3U

3U