Cisco Router C3945E-VSEC-CUBEK9

3 Enhanced High-Speed WAN Interface Card slots

3 onboard digital signal processor (DSP) slots

①

②

③

On/Off switch

Power connector

LEDs area

①

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

⑦

SFP 1 and SFP 2 (1, Top)

②

USB serial port

⑧

USB 0 and USB 1 (1, Left)

③

AUX port

⑨

Ground

④

RJ-45 serial console port

⑩

Service module slots

⑤

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

Compact flash 0 and 1 (0, right)

⑥

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

• 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

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

• IP phone to Cisco Unified Survivable Remote Site Telephony router signaling and encryption uses TLS
(Transport Layer Security)

• 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

• H.323 and SIP

• 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)

• RTP, RTCP, and Binary Floor Control Protocol (BFCP)
• Sub-RTCP for media statistics

• 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 flow-through
• Media flow-around

• TCP
• User Datagram Protocol (UDP)
• TCP-to-UDP interworking

• T.38 fax relay
• Fax pass-through
• Fax over G711

• Modem pass-through
• Modem over G711

• H.245 alphanumeric
• H.245 signal
• RFC 2833
• SIP notify
• Key Press Markup Language (KPML)
• Interworking capabilities include:H.323 to SIPRFC 2833 to G.711 in-band DTMF *Various SIP-to-H.323 DTMF interworking options

  RFC 2833 to KPML

• 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

• 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

• 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

• 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)

• 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 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

• 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

• 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

• IP Precedence and differentiated-services-code-point (DSCP) marking
• Per-call QoS packet marking

• NAT traversal support for SIP phones deployed behind non-Application Line Gateway (ALG) data routers
• Stateful NAT traversal
• IPv4-to-IPv6 translation

• 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 rules for voice-over-IP (VoIP) numbers
• URI-based dialing translations

• 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 between any two different families of codecs from the following list:G.711 a-law and mu-lawG.729, G.729A, G.729B, and G.729ABiLBC

  G.722

   

• Midcall transcoder insert and drop

• 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

• AAA with RADIUS

• 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

• 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

• Standard CDRs for accurate billing available through:AAA recordsSyslogSimple Network Management Protocol (SNMP)

• Provision of replicated packets to third-party mediation device

• 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)

• 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

• 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

• H.323 and SIP

• Cisco Unified Video Advantage (UVA) and Cisco TelePresence endpoints

• Simultaneous support for data, audio, and video

• SIP delayed-offer to SIP early-offer calls

• Support for multiplex RTP calls (for Cisco TelePresence solution)
• Simple Traversal of UDP through NAT (STUN)/Datagram TLS (DTLS) pass-through for telepresence

• H.235 pass-through for secure calls
• H.239 pass-through for picture-in-picture feature

• DSCP markings to prioritize video streams as they traverse the network

• T.120 data collaboration flow-around only

• Far-end camera control (FECC)

• H.261
• H.263
• H.264

Network Management

• Resource usage monitoring over SIP trunk
• SNMP per-call quality traps
• SNMP and syslog SIP trunk status messages

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.

CISCO3945E/k9

C3945E-VSEC-CUBEK9

N/A

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

w/SPE 250

w/SPE 250

4GE

4GE

3

3

4

4

0

0

3

3

3U

3U