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Feature
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Support Details
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Protocols
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● H.323 and SIP
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Protocol and signal interworking
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● 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)
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Media support
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● RTP, RTCP, and Binary Floor Control Protocol (BFCP)
● Sub-RTCP for media statistics
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Media interworking
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● 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
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Media modes
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● Media flow-through
● Media flow-around
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Signaling transport mode
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● TCP
● User Datagram Protocol (UDP)
● TCP-to-UDP interworking
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Fax support
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● T.38 fax relay
● Fax pass-through
● Fax over G711
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Modem support
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● Modem pass-through
● Modem over G711
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Dual-tone multifrequency (DTMF)
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● 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
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Supplementary services
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● 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
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Internetworking
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● 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
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Call routing and dialing options
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● 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
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Cisco Call Admission Control (CAC)
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● 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)
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OPTIONS SIP message support
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● 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
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Media recording
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● 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
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IP Routing feature
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● 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
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Voice-quality statistics
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● 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
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QoS
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● IP Precedence and differentiated-services-code-point (DSCP) marking
● Per-call QoS packet marking
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Network Address Translation (NAT) traversal
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● NAT traversal support for SIP phones deployed behind non-Application Line Gateway (ALG) data routers
● Stateful NAT traversal
● IPv4-to-IPv6 translation
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Network hiding
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● 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
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Number translation
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● Number translation rules for voice-over-IP (VoIP) numbers
● URI-based dialing translations
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Codecs
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● 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
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Transcoding
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● 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
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Security
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● 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
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Authentication, authorization, and accounting (AAA)
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● AAA with RADIUS
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Voice media applications
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● 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
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API
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● 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
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Billing
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● Standard CDRs for accurate billing available through:
AAA records
Syslog
Simple Network Management Protocol (SNMP)
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Lawful intercept
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● Provision of replicated packets to third-party mediation device
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Remote phone proxy sessions
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● 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)
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Line-side back-to-back user agent NANOCUBE sessions
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● 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
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Inter-Cluster Lookup Service (ILS) routing
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● 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
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Video
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Protocols
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● H.323 and SIP
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Cisco endpoints supported
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● Cisco Unified Video Advantage (UVA) and Cisco TelePresence endpoints
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Rich media
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● Simultaneous support for data, audio, and video
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Signaling interworking
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● SIP delayed-offer to SIP early-offer calls
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Media
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● Support for multiplex RTP calls (for Cisco TelePresence solution)
● Simple Traversal of UDP through NAT (STUN)/Datagram TLS (DTLS) pass-through for telepresence
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H.323-enhanced features
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● H.235 pass-through for secure calls
● H.239 pass-through for picture-in-picture feature
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QoS
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● DSCP markings to prioritize video streams as they traverse the network
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Data support
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● T.120 data collaboration flow-around only
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Camera control
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● Far-end camera control (FECC)
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Video codecs
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● H.261
● H.263
● H.264
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Network Management
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Manageability and serviceability
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● Resource usage monitoring over SIP trunk
● SNMP per-call quality traps
● SNMP and syslog SIP trunk status messages
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High Availability
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High availability
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● 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.
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