Feature |
Support Details |
Protocols |
|
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
|
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) |
|
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 |
|
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 |
|
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.
|
