C2901-VSEC-CUBE/K9

Product Code

C2901-VSEC-CUBE/K9

Bundle

C2901 VSEC CUBE Bundle, PVDM3-16, UC SEC Lic, FL-CUBEE-25

Rack Units

1U

Interfaces

2 integrated 10/100/1000 Ethernet ports

Expansion Slot(s)

2 onboard digital signal processor (DSP) slots

4 enhanced high-speed WAN interface card slots

1 onboard Internal Service Module for application services

RAM

512 MB (installed) / 2 GB (max)

Flash Memory

256 MB (installed) / 8 GB (max)

Dimensions

43.9 cm x 43.8 cm x 4.5 cm

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 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

Codes

· 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 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 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.