C2911-VSEC-CUBE/K9

Feature

Suppt Details

Protocols

· H.323 and SIP

Protocol and signal interwking

· 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 suppt

· RTP, RTCP, and Binary Flo Control Protocol (BFCP)

· Sub-RTCP f media statistics

Media interwking

· SIP delayed-offer to SIP early-offer interwking f audio video calls

· H.323 Slow Start to H.323 Fast Start f audio calls

Media modes

· Media flow-through

· Media flow-around

Signaling transpt mode

· TCP

· User Datagram Protocol (UDP)

· TCP-to-UDP interwking

Fax suppt

· T.38 fax relay

· Fax pass-through

· Fax over G711

Modem suppt

· 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)

· Interwking capabilities include:

◦   H.323 to SIP

◦   RFC 2833 to G.711 in-band DTMF *

◦   Various SIP-to-H.323 DTMF interwking options

◦   RFC 2833 to KPML

Supplementary services

· Call hold, call transfer, and call fwarding f H.323 netwks using H.450 and transparent passing of Empty Capability Set (ECS)

· SIP-to-SIP supplementary services (holds and transfers) suppt using REFER

· SIP-to-SIP supplementary services (holds and transfers) suppt using REINVITE

· H.323-to-SIP supplementary services f Cisco Unified Communications Manager with media termination point (MTP) on the H.323 trunk

Internetwking

· Configurable SIP profiles to manipulate SIP message content, including header fields andSession Descript Protocol (SDP) attributes

· P-Asserted-Identity (PAI), P-Preferred-Identity (PPI), and Remote-Part-ID (RPID) internetwking**

· Unsuppted Multipurpose Internet Mail Extensions (MIME)-type attachment pass-through**

· Unsuppted SIP header pass-through**

· Dial-peer bind (allows Cisco Unified Bder Element to connect to multiple different service providers)

· Incoming dial-peer match based on remote IP address

· Assisted RTCP f Microsoft Lync Interoperability

Call routing and dialing options

· E164-based dialing

· Unifm Resource Identifier (URI)-based dialing

· Routing based on nonsequential lists (either E164 URI both)

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

· Server Groups to define der of selection of alternative backup routing paths f 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, memy use threshold

· CAC based on bandwidth availability and call-spike detection

· Resource Reservation Protocol (RSVP)

OPTIONS SIP message suppt

· Suppt f response to OPTIONS-PING messages with OPTION- PING groups based on session target

· Suppt f generation of in-dialog OPTIONS-PING messages

· Suppt f generation of out-of-dialog OPTIONS-PING messages to control dial-peer status**

Media recding

· Media fking features f both voice and video to integrate with Cisco TelePresence Media Recding Servers

· Active (SIP-based) and passive (application programming interface [API]-based) mechanisms f invoking media fking

IP Routing feature

· Suppt f Cisco IOS Software-based routing features, including Bder Gateway Protocol (BGP), Enhanced IGRP (EIGRP), and Multiprotocol Label Switching (MPLS)

· Suppt f Cisco IOS Software-based policy routing features

· Suppt f 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 infmation

· Sub-RTCP statistics collection

QoS

· IP Precedence and differentiated-services-code-point (DSCP) marking

· Per-call QoS packet marking

Netwk Address Translation (NAT) traversal

· NAT traversal suppt f SIP phones deployed behind non-Application Line Gateway (ALG) data routers

· Stateful NAT traversal

· IPv4-to-IPv6 translation

Netwk hiding

· IP netwk privacy and topology hiding

· IP netwk security boundary

· Intelligent IP address translation f call media and signaling

· Back-to-back user agent, replacing all SIP-embedded IP addressing

· Histy infmation-based topology hiding and call routing

Number translation

· Number translation rules f 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 f rogue packet activity

· IP Security (IPsec)

· Secure RTP (SRTP)

· Transpt Layer Security (TLS)

· SRTP-to-RTP interwking

Authentication, authization, and accounting (AAA)

· AAA with RADIUS

Voice media applications

· Tool Command Language (Tcl) scripts suppt f application customization

· VoiceXML 2.0 script suppt f application customization

· Web-based API to monit and control signaling and media traffic

API

· Web-based API compatible with Web Service Description Language (WSDL) development tools to suppt call moniting and control, call-detail recds (CDRs), and serviceability attribute interaction with external application; specifically designed f voice-policy applications

Billing

· Standard CDRs f accurate billing available through:

◦   AAA recds

◦   Syslog

◦   Simple Netwk 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 Pt Adapter (SPA) and other third-party SIP endpoints with registration pass‑through and survivability f use with third-party hosted call-control service provider services

Inter-Cluster Lookup Service (ILS) routing

· Suppt f ILS routing to complement ILS dial-plan exchange between Cisco Unified Communications Manager clusters to simplify call-routing complexity between multiple clusters

Video

Protocols

· H.323 and SIP

Cisco endpoints suppted

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

Rich media

· Simultaneous suppt f data, audio, and video

Signaling interwking

· SIP delayed-offer to SIP early-offer calls

Media

· Suppt f multiplex RTP calls (f Cisco TelePresence solution)

· Simple Traversal of UDP through NAT (STUN)/Datagram TLS (DTLS) pass-through f telepresence

H.323-enhanced features

· H.235 pass-through f secure calls

· H.239 pass-through f picture-in-picture feature

QoS

· DSCP markings to priitize video streams as they traverse the netwk

Data suppt

· T.120 data collabation flow-around only

Camera control

· Far-end camera control (FECC)

Video codecs

· H.261

· H.263

· H.264

Netwk Management

Manageability and serviceability

· Resource usage moniting 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 f active calls at time of failover in each redundancy configuration listed.