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

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

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