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C3945 VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25 - C3945-VSEC-CUBE/K9

Name: C3945 VSEC CUBE Bundle, PVDM3-64, UC SEC Lic, FL-CUBEE-25 - C3945-VSEC-CUBE/K9
Brand: Cisco
SKU: G5N-C3945-VSEC-CUBE/K9
Price: 15795.0 GBP
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Description

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

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	B serial port	⑨	B 0 and B 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)

Authentication and Encryption Features	• Media encryption of voice RTP streams ing SRTP • Exchange of RTP Control Protocol (RTCP) information ing 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 ing 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 es 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 es 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 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 ● er 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 * Vario SIP-to-H.323 DTMF interworking options RFC 2833 to KPML
Supplementary services	● Call hold, call transfer, and call forwarding for H.323 networks ing H.450 and transparent passing of Empty Capability Set (ECS) ● SIP-to-SIP supplementary services (holds and transfers) support ing REFER ● SIP-to-SIP supplementary services (holds and transfers) support ing 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 e, or memory e 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 stat
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 er 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 RADI
Voice media applications	● Tool Command Language (Tcl) scripts support for application ctomization ● VoiceXML 2.0 script support for application ctomization ● 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 Biness Edition, or Cisco HCS (not included with NANOCUBE license)
Line-side back-to-back er agent NANOCUBE sessions	● Termination of Cisco Shared Port Adapter (SPA) and other third-party SIP endpoints with registration passthrough and survivability for e with third-party hosted call-control service provider services
Inter-Clter Lookup Service (ILS) routing	● Support for ILS routing to complement ILS dial-plan exchange between Cisco Unified Communications Manager clters or to simplify call-routing complexity between multiple clters
Video
Protocols	● H.323 and SIP
Cisco endpoints supported	● Cisco Unified Video Advantage (UVA) and Cisco TelePresence endpoints
Rich media	● Simultaneo 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 age monitoring over SIP trunk ● SNMP per-call quality traps ● SNMP and syslog SIP trunk stat 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.

Communications
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 Stat Lights Indicators	Link activity, power
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-B Type B Serial : 1 x auxiliary - RJ-45 B : 2 x 4 pin B 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 / Miscellaneo
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)

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

Figure 1 shows the appearance of the C3945-VSEC-CUBE/K9.

C3945-VSEC-CUBE/K9

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

Figure 2 shows the front panel of the C3945-VSEC-CUBE/K9.

C3945-VSEC-CUBE/K9 Front Panel

Note:

①	②	③
On/Off switch	Power connector	LEDs area

Figure 3 shows the back panel of the C3945-VSEC-CUBE/K9 with slots and ports.

C3945-VSEC-CUBE/K9 Back Panel

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	B serial port	⑨	B 0 and B 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 ing SRTP • Exchange of RTP Control Protocol (RTCP) information ing 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 ing 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 es 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 es 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 ● er 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 * Vario SIP-to-H.323 DTMF interworking options RFC 2833 to KPML
Supplementary services	● Call hold, call transfer, and call forwarding for H.323 networks ing H.450 and transparent passing of Empty Capability Set (ECS) ● SIP-to-SIP supplementary services (holds and transfers) support ing REFER ● SIP-to-SIP supplementary services (holds and transfers) support ing 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 e, or memory e 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 stat
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 er 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 RADI
Voice media applications	● Tool Command Language (Tcl) scripts support for application ctomization ● VoiceXML 2.0 script support for application ctomization ● 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 Biness Edition, or Cisco HCS (not included with NANOCUBE license)
Line-side back-to-back er agent NANOCUBE sessions	● Termination of Cisco Shared Port Adapter (SPA) and other third-party SIP endpoints with registration passthrough and survivability for e with third-party hosted call-control service provider services
Inter-Clter Lookup Service (ILS) routing	● Support for ILS routing to complement ILS dial-plan exchange between Cisco Unified Communications Manager clters or to simplify call-routing complexity between multiple clters
Video
Protocols	● H.323 and SIP
Cisco endpoints supported	● Cisco Unified Video Advantage (UVA) and Cisco TelePresence endpoints
Rich media	● Simultaneo 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 age monitoring over SIP trunk ● SNMP per-call quality traps ● SNMP and syslog SIP trunk stat 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

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