Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5

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• Read Me First
• New and Changed Information
• Supported Platforms
• CUBE Fundamentals and Basic Setup
  • Overview of Cisco Unified Border Element
  • Virtual CUBE
  • Dial-Peer Matching
  • DTMF Relay
  • Introduction to Codecs
  • Call Admission Control
  • Basic SIP Configuration
  • SIP Binding
  • Media Path
  • SIP Profiles
  • SIP Out-of-Dialog OPTIONS Ping Group
  • Configure TCL IVR Applications
  • VoIP for IPv6
  • Monitoring of Phantom Packets
  • Configurable SIP Parameters via DHCP
  • Matching Inbound Dial Peers by URI
  • URI-Based Dialing Enhancements
  • Multiple Pattern Support on a Voice Dial Peer
  • Outbound Dial-Peer Group as an Inbound Dial-Peer Destination
  • Inbound Leg Headers for Outbound Dial-Peer Matching
  • Server Groups in Outbound Dial Peers
  • Domain-Based Routing Support on the Cisco UBE
  • ENUM Enhancement per Kaplan Draft RFC
  • Support for Multi-VRF
  • Configuring Multi-Tenants on SIP Trunks
  • Codec Support and Restrictions
  • Codec Preference Lists
  • Transcoding
  • Transrating
  • Call Progress Analysis Over IP-to-IP Media Session
  • Voice Packetization
  • Fax Detection for SIP Call and Transfer
  • Video Suppression
  • Configuring RTCP Report Generation
  • Network-Based Recording
  • SIPREC (SIP Recording)
  • Video Recording - Additional Configurations
  • Third-Party GUID Capture for Correlation Between Calls and SIP-based Recording
  • Cisco Unified Communications Gateway Services--Extended Media Forking
  • CUBE Media Proxy
  • Passing Headers Unsupported by CUBE
  • Copying SIP Headers
  • Manipulate SIP Status-Line Header of SIP Responses
  • Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls
  • Delayed-Offer to Early-Offer
  • H.323-to-SIP Interworking on CUBE
  • H.323-to-H.323 Interworking on CUBE
  • SIP RFC 2782 Compliance with DNS SRV Queries
  • SRTP-SRTP Interworking
  • SRTP-RTP Interworking
  • SRTP-SRTP Pass-Through
  • High Availability on Cisco 4000 Series ISR and Cisco Catalyst 8000 Series Edge Platforms
  • High Availability on Cisco ASR 1000 Series Aggregation Services Routers
  • High Availability on Cisco CSR 1000V or C8000V Cloud Services Routers
  • High Availability on Cisco Integrated Services Routers (ISR-G2)
  • DSP High Availability Support
  • Stateful Switchover Between Redundancy Paired Intra- or Inter-box Devices
  • CVP Survivability TCL support with High Availability
  • ICE-Lite Support on CUBE
  • Mid-call Signaling Consumption
  • Early Dialog UPDATE Block
  • Consumption of Forked 18x Responses with SDP During Early Dialog
  • Support for Pass-Through of Unsupported Content Types in SIP INFO Messages
  • Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
  • Dynamic Refer Handling
  • Cause Code Mapping
  • Hosted and Cloud Services Delivery with CUBE
  • CUBE SIP Registration Proxy
  • Survivability for Hosted and Cloud Services
  • SUBSCRIBE-NOTIFY Passthrough
  • Cisco Unified Communications Manager Line-Side Support
  • SIP TLS Support on CUBE
  • CUBE Call Quality Statistics Enhancement
  • Voice Quality Monitoring
  • CUBE Smart Licensing
  • VoIP Trace for CUBE
  • Support for Session Identifier
  • Common Criteria (CC) and The Federal Information Processing Standards (FIPS) Compliance
  • Additional References
  • Glossary
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  Book Title

  Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5

  Overview of Cisco Unified Border Element

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  Updated: August 15, 2022

  Chapter: Overview of Cisco Unified Border Element

  Chapter Contents
  • Overview of Cisco Unified Border Element
  • Information About Cisco Unified Border Element
    • SIP/H.323 Trunking
    • Typical Deployment Scenarios for CUBE
    • Enabling the CUBE Application on a Device
      • Verifying the CUBE Application on the Device

      Overview of Cisco Unified Border Element

      Cisco Unified Border Element (CUBE) bridges voice and video connectivity between two separate VoIP networks. It is similar to a traditional voice gateway, except for the replacement of physical voice trunks with an IP connection. Traditional gateways connect VoIP networks to telephone companies using a circuit-switched connection, such as PRI. The CUBE connects VoIP networks to other VoIP networks and is often used to connect enterprise networks to Internet telephony service providers (ITSPs).

      Information About Cisco Unified Border Element

      Cisco Unified Border Element (CUBE) can terminate and originate signaling (H.323 and Session Initiation Protocol [SIP]) and media streams (Real-Time Transport Protocol [RTP] and RTP Control Protocol [RTCP]).

      CUBE extends the functionality provided by conventional session border controllers (SBCs) in terms of protocol interworking, especially on the enterprise side. As shown in the chart below, the CUBE provides the following additional features:

      

      The CUBE provides a network-to-network interface point for:

      • Signaling interworking—H.323 and SIP.
      • Media interworking—dual-tone multifrequency (DTMF), fax, modem, and codec transcoding.
      • Address and port translations—privacy and topology hiding.
      • Billing and call detail record (CDR) normalization.
      • Quality-of-service (QoS) and bandwidth management—QoS marking using differentiated services code point (DSCP) or type of service (ToS), bandwidth enforcement using Resource Reservation Protocol (RSVP), and codec filtering.

      CUBE functionality is implemented on devices using a special IOS feature set, which allows CUBE to route a call from one VoIP dial peer to another.

      Protocol interworking is possible for the following combinations:

      • H.323-to-SIP interworking
      • H.323-to-H.323 interworking
      • SIP-to-SIP interworking

      The CUBE provides a network-to-network demarcation interface for signaling interworking, media interworking, address and port translations, billing, security, quality of service, call admission control, and bandwidth management.

      The CUBE is used by enterprise and small and medium-sized organizations to interconnect SIP PSTN access with SIP and H.323 enterprise unified communications networks.

      A CUBE interoperates with several different network elements including voice gateways, IP phones, and call-control servers in many different application environments, from advanced enterprise voice and/or video services with Cisco Unified Communications Manager or Cisco Unified Communications Manager Express, as well as simpler toll bypass and voice over IP (VoIP) transport applications. The CUBE provides organizations with all the border controller functions integrated into the network layer to interconnect unified communications voice and video enterprise-to-service-provider architectures.

      

      If an enterprise subscribes to VoIP services offered by an ITSP, connecting the enterprise CUCM through a CUBE provides network demarcation capabilities, such as security, topology hiding, transcoding, call admission control, protocol normalization and SIP registration, none of which is possible if CUCM connects directly to the ITSP. Another use case involves mergers or acquisitions in an enterprise and the need to integrate voice equipment, such as CUCMs, IP PBXs, VM servers, and so on. If the networks in the two organizations have overlapping IP addresses, CUBE can be used to connect the two distinct networks until the acquired organization can be migrated into the enterprise addressing plan.

      SIP/H.323 Trunking

      H.323 protocol is no longer supported from Cisco IOS XE Bengaluru 17.6.1a onwards. Consider using SIP for multimedia applications.

      The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over IP networks. SIP (or H.323) trunking is the use of VoIP to facilitate the connection of PBX to other VoIP endpoints across the Internet. To use SIP trunking, an enterprise must have a PBX (internal VoIP system) that connects to all internal end users, an Internet telephony service provider (ITSP), and a gateway that serves as the interface between the PBX and the ITSP. One of the most significant advantages of SIP and H.323 trunking is the ability to combine data, voice, and video in a single line, eliminating the need for separate physical media for each mode.

      

      SIP trunking overcomes TDM barriers, in that it:

      • Improves efficiency of interconnection between networks
      • Simplifies PSTN interconnection with IP end-to-end
      • Enables rich media services to employees, customers, and partners
      • Carries converged voice, video, and data traffic
      

      For Cisco IOS XE Gibraltar 16.11.1a and later releases, the SIP processes are initiated only when either of the following CLIs is configured:

      • Voice dial-peer with session protocol as SIP.
      • voice register global
      • sip-ua

      In the releases before Cisco IOS XE Gibraltar 16.11.1a, the following commands initiated the SIP processes:

      • dial-peer voice (any)
      • ephone-dn
      • max-dn under call-manager-fallback
      • ds0-group 0 timeslots 1 type e&m-wink-start

      Typical Deployment Scenarios for CUBE

      CUBE in an enterprise environment serves two main purposes:

      

      • External Connections—CUBE is the demarcation point within a unified communications network and provides interconnectivity with external networks. This includes H.323 and SIP voice and video connections.
      • Internal Connections—When used within a VoIP network, CUBE increases flexibility and interoperability between devices.

      How to Configure Basic CUBE Features

      Consider a scenario where XYZ corporation uses a VoIP network to provide phone services and uses a PRI connection for telecommunications services, and the PRI trunk is controlled by MGCP. Migration from MGCP PRI to SIP trunk is provided by ITSP telecommunications. CUCM sends the telephone number, as 10 digits, to CUBE. CUCM may send only the extension (4 digits) to the CUBE. When the call is diverted (using call-forward), the requirement of the ITSP is that they need the full 10-digit number in the SIP Diversion field.

      

      The following sections describe the basic setup of CUBE through the steps involved in migrating the XYZ corporation to CUBE using a SIP trunk.

      Enabling the CUBE Application on a Device

      SUMMARY STEPS

      1. enable
      2. configure terminal
      3. voice service voip
      4. mode border-element license [ capacity sessions | periodicity < mins value | hours value | days value >]
      5. allow-connections from-type to to-type
      6. end