| License Details |
|---|
| Model | FLASR1-SBC-RTU |
| Description | DBE Base Right-To-Use Lic for Release 2.3.x and prior |
| Type | Perpetual Right-To-Use (RTU) |
| Availability | 10 Units In Stock - Immediate Dispatch |
| Pricing | Special Offer: $1893 (Save 92%) |
| Documentation | PDF Datasheet Available |
| Protocol and Signal Interworking |
|---|
| Support | SIP to SIP (including Cisco Unified Communications Manager and Cisco TelePresence) |
| Media Support |
|---|
| Protocols | RTP and RTCP, Binary Flow Control Protocol (BFCP) passthrough |
| Media Interworking |
|---|
| Support | SIP delayed-offer to SIP early-offer interworking for audio or video calls |
| Media Modes |
|---|
| Modes | Media flow-through, Media flow-around |
| Signaling Transport Mode |
|---|
| Protocols | Transport Control Protocol (TCP), Transport Layer Security (TLS), User Datagram Protocol (UDP), TCP TLS and UDP interworking |
| Fax Support |
|---|
| Types | T.38 fax relay, Fax pass-through, Fax over G.711 |
| Modem Support |
|---|
| Types | Modem pass-through, Modem over G.711 |
| Dual-Tone Multifrequency (DTMF) |
|---|
| Methods | RFC 2833 /RFC 4733, SIP notify, Key Press Markup Language (KPML) |
| Interworking | RFC 2833/4733 to G.711 in-band DTMF, sip-info to rtp-nte interworking, RFC 2833/4733 to KPML |
| Supplementary Services |
|---|
| Support | SIP supplementary services (holds and transfers) support using REFER or REINVITE, Multicast Music on Hold (MMoH) to Unicast MoH conversion, Call Progress Analysis (CPA) to analyze far-end media (live versus recorded media) for outbound call centers |
| Internetworking |
|---|
| Features | Configurable SIP profiles to manipulate SIP message content including header fields and Session Descriptor Protocol (SDP) attributes, Conditional SIP profiles performing header modification dependent on header content, P-Asserted-Identity (PAI) P-Preferred-Identity (PPI) and Remote-Party-ID (RPID) internetworking, Unsupported Multipurpose Internet Mail Extensions (MIME)-type attachment pass-through, Unsupported SIP header pass-through, SDP attribute pass-through, Dial-peer bind (allows CUBE to connect to multiple service providers), Incoming dial-peer match based on remote IP address, Assisted RTCP for Microsoft Lync/Skype for Business interoperability, Mid-call signaling block or pass-through when media changes, Early dialog UPDATE /183 consumption, Block incoming 180 and 183 signaling messages, Restrict video call to audio only, Media Anti-trombone, IPv4 to IPv6 interworking, Configurable SIP error codes, SIP error code pass-through |
| Call Routing and Dialing Options |
|---|
| Options | E164-based dialing, Uniform Resource Identifier (URI)-based dialing, Routing based on nonsequential E164 and/or URI lists, Destination-based or source-based routing, 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, Routing based on duple header variables (both AND OR logic), Refer and call redirect consumption and pass-through, Outbound call load distribution with random or round robin schemes, Call re-routing based on network errors or error responses, P-called-party-ID support |
| Multitenancy, Multi-VRF, and Trunk Realms |
|---|
| Support | Support for dial plan scenarios requiring either or both inter- and intra- IP VRF routing tables, Per-VRF-domain SIP user agent for multi-tenancy support (up to 100 VRFs), Realm commonality of multiple trunks even with different user agent definitions per trunk |
| Cisco Call Admission Control (CAC) |
|---|
| Types | CAC based on 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 |
| OPTIONS SIP Message Support |
|---|
| Features | 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 | Media forking features for voice and video to integrate with media recording or analysis servers, API-based mechanisms for invoking media forking, Support for standard SIPREC media forking, Raw media forking using secure WebSockets for Cisco Contact Center solutions (Requires Enhanced License), Media Proxy mode for forking calls to up five different destinations, Secure forking of a non-secure call |
| IP Routing Features |
|---|
| Support | Support for Cisco IOS XE Software-based routing features including Border Gateway Protocol (BGP) Enhanced IGRP (EIGRP) and Multiprotocol Label Switching (MPLS), Support for Cisco IOS XE Software-based policy routing features, Support for Cisco IOS XE Software-based Access-Control-List (ACL) features |
| Voice-Quality Statistics |
|---|
| Metrics | RTCP data from incoming and outgoing call legs used to provide: Packet loss jitter and Round-Trip Time (RTT), Per-call leg call-quality statistics |
| QoS |
|---|
| Features | IP precedence and Differentiated-Services-Code-Point (DSCP) marking, Per-call QoS packet marking |
| Network Address Translation (NAT) Traversal |
|---|
| Support | NAT traversal support for SIP phones deployed behind non-Application Line Gateway (ALG) data routers, Stateful NAT traversal, ICE-Lite |
| Network Hiding |
|---|
| Features | 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 |
|---|
| Support | Number translation rules for Voice-over-IP (VoIP) numbers, URI-based dialing translations |
| Codecs |
|---|
| Supported Codecs | OPUS low bitrate 6 kbps to very high-quality 510 kbps, G.711 mu-law and a-law, G.722, 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) 13330 or 15200 bps, Internet Speech Audio Code (iSAC) 10 to 32 kbps, AAC-LD MP4A-LATM, Mid-call codec renegotiation and preservation, Narrowband Adaptive Multi-rate (AMR-NB) 4750-12200 bps |
| Transcoding |
|---|
| Support | 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, OPUS (PVDM4 modules only), Mid-call transcoder insert and drop |
| Transrating |
|---|
| Support | Transrating of packetization rates for the following codecs: G.711 a-law and mu-law, G.723 5.3/6/3 kbps, G.729 G.729A G.729B and G.729AB, G.722 |
| Security |
|---|
| Features | Rogue SIP invite and rogue RTP packet detection with alerting, Configurable RTP port range, IP security (IPsec), SRTP flow-through, Transport Layer Security (TLS) version 1.2 with exclusivity, SRTP-to-RTP and STRP-to-SRTP interworking with Next-Generation Encryption (NGE) cipher suites, Configurable SIP listening port per trunk, Disable unused transport mechanisms, SIP registration and digest authentication support, Various mechanisms for control of RTP and UDP packet flooding, Voice security policy application integration (via HTTP API), Peer whitelisting /IP Trusted List, Silent discard of SIP messages from untrusted peers, Compatible with IOS Zone Based Firewall |
| Authentication, Authorization, and Accounting (AAA) |
|---|
| Support | AAA with RADIUS |
| Voice Media Applications |
|---|
| Support | Tool Command Language (TCL) scripts support for application customization, Web-based API to monitor and control signaling and media traffic (for external policy control) |
| API |
|---|
| Support | 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 |
|---|
| Support | Standard CDRs for accurate billing available through: AAA records, Syslog, Simple Network Management Protocol (SNMP) |
| Line-side Registration Proxy |
|---|
| Features | Proxy registration of endpoints using the standard SIP registration process (including third-party SIP endpoints) for connecting with third-party hosted call-control services (e.g. Cisco BroadSoft), Local and PSTN survivability in the event of loss of WAN connectivity to a hosted call control, Proxy endpoint registration with 10 endpoints per SIP registration event |
| Inter-Cluster Lookup Service (ILS) Routing |
|---|
| Support | 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 - Rich Media |
|---|
| Support | Simultaneous support for data audio and video |
| Video - Signaling Interworking |
|---|
| Support | SIP delayed-offer to SIP early-offer calls |
| Video - Media |
|---|
| Support | Support for multiplex RTP calls (for Cisco TelePresence solution), Simple Traversal of UDP through NAT (STUN) /Datagram TLS (DTLS) pass-through for telepresence |
| Video - QoS |
|---|
| Support | DSCP markings to prioritize video streams as they traverse the network |
| Video - Data Support |
|---|
| Support | T.120 data collaboration (flow-around only) |
| Video - Camera Control |
|---|
| Support | Far-End Camera Control (FECC) |
| Video - Video Suppression |
|---|
| Support | Terminate video media session for connection to audio-only sessions |
| Video Codecs |
|---|
| Codecs | H.261, H.263/H.263+, H.264, MPEG4 |
| Network Management - Manageability, Serviceability, and Troubleshooting |
|---|
| Features | Resource usage monitoring over SIP trunk, Sortable dial peers, SIP session ID for end-to-end call tracing, SNMP per-call quality traps, SNMP and syslog SIP trunk status messages, DEBUG commands allowing user-selectable levels of debug information from critical to verbose, DEBUG commands allowing user-selectable information for specific call characteristics, VoIPTrace continuous diagnostic capture, Yang data model allowing configuration and management via RESTCONF and NETCONF |
| High Availability |
|---|
| Support | Inbox redundancy with Cisco ASR 1006 and ASR 1006-X, Box-to-box redundancy with Cisco 4000 Series ISRs Catalyst Edge 8000 Cisco ASR 1000 and CSR 1000V models (based on RG infrastructure), Use of port channels to allow a connection to redundant switches, Requires Enhanced Trunk or Media proxy session license |
| Router Platform Support |
|---|
| Cisco 1100 ISR | Control Plane Memory: Default, Maximum trunk sessions: 500, Maximum sustainable call setup rate: 5 Calls per second |
| Cisco 4321 ISR | Control Plane Memory: 4 GB, Maximum trunk sessions: 500, Maximum sustainable call setup rate: 4 Calls per second |
| Cisco 4331 ISR | Control Plane Memory: 4 GB, Maximum trunk sessions: 1000, Maximum sustainable call setup rate: 10 Calls per second |
| Cisco 4351 ISR | Control Plane Memory: 4 GB, Maximum trunk sessions: 2000, Maximum sustainable call setup rate: 13 Calls per second |
| Cisco 4431 ISR | Control Plane Memory: 8 GB, Maximum trunk sessions: 3000, Maximum sustainable call setup rate: 15 Calls per second |
| Cisco 4451-X ISR | Control Plane Memory: 8 GB, Maximum trunk sessions: 6000, Maximum sustainable call setup rate: 40 Calls per second |
| Cisco 4461 ISR | Control Plane Memory: 8 GB, Maximum trunk sessions: 10,000, Maximum sustainable call setup rate: 55 Calls per second |
| C8200L-1N-4T | Control Plane Memory: 4 GB, Maximum trunk sessions: 1500, Maximum sustainable call setup rate: 9 Calls per second |
| C8200-1N-4T | Control Plane Memory: 8 GB, Maximum trunk sessions: 2500, Maximum sustainable call setup rate: 14 Calls per second |
| C8300-1N1S-6T | Control Plane Memory: 8 GB, Maximum trunk sessions: 7000, Maximum sustainable call setup rate: 40 Calls per second |
| C8300-1N1S-4T2X | Control Plane Memory: 8 GB, Maximum trunk sessions: 8000, Maximum sustainable call setup rate: 45 Calls per second |
| C8300-2N2S-6T | Control Plane Memory: 8 GB, Maximum trunk sessions: 7500, Maximum sustainable call setup rate: 42 Calls per second |
| C8300-2N2S-4T2X | Control Plane Memory: 16 GB, Maximum trunk sessions: 10,000, Maximum sustainable call setup rate: 55 Calls per second |
| Cisco CSR 1000V /C8000V 1vCPU | Control Plane Memory: 4 GB, Maximum trunk sessions: 1000, Maximum sustainable call setup rate: 5 Calls per second |
| Cisco CSR 1000V /C8000V 2vCPU | Control Plane Memory: 4 GB, Maximum trunk sessions: 3000, Maximum sustainable call setup rate: 20 Calls per second |
| Cisco CSR 1000V /C8000V 4vCPU | Control Plane Memory: 8 GB, Maximum trunk sessions: 6000, Maximum sustainable call setup rate: 30 Calls per second |
| Cisco ASR 1001-X | Control Plane Memory: 16 GB, Maximum trunk sessions: 12,000, Maximum sustainable call setup rate: 50 Calls per second |
| Cisco ASR 1002-X | Control Plane Memory: 16 GB, Maximum trunk sessions: 14,000, Maximum sustainable call setup rate: 55 Calls per second |
| Cisco ASR 1006-X with RP3 and ESP100/ESP100X | Control Plane Memory: 16 GB, Maximum trunk sessions: 16,000, Maximum sustainable call setup rate: 65 Calls per second |
| Cisco ASR 1004 /ASR 1006 / ASR 1006-X with RP2 and ESP40 | Control Plane Memory: 16 GB, Maximum trunk sessions: 16,000, Maximum sustainable call setup rate: 70 Calls per second |
| CUBE Subscription Options (Related) |
|---|
| A-FLEX-ENH-CUBE | One CUBE trunk enhanced session subscription |
| A-FLEX-STD-CUBE | One CUBE trunk standard session subscription |
| A-FLEX-MP-CUBE | One Media Proxy stream subscription |
| Cisco Environmental Sustainability |
|---|
| Product Material Content | Materials |
| Electronic Waste Compliance | WEEE compliance |
