7.3 Cisco Unified Border Element (CUBE) Configuration & Redundancy
Key Takeaways
Cisco Unified Border Element (CUBE) is an enterprise Session Border Controller (SBC) terminating external SIP trunks from ITSPs and connecting internal call control systems (CUCM, Webex Calling).
In media flow-through mode, CUBE terminates both signaling and RTP media, regenerating media packets to provide topology hiding, NAT traversal, transcoding, and security; in media flow-around mode, RTP flows directly between endpoints without CUBE media handling.
Voice class SIP profiles allow regular-expression-based inspection, addition, modification, and removal of SIP headers (such as Remote-Party-ID, P-Asserted-Identity, Contact, and Diversion) to resolve ITSP interoperability requirements.
Voice class server groups provide outbound SIP trunk redundancy with round-robin load distribution or preference-based failover, monitored dynamically by SIP OPTIONS ping keepalives.
On IOS XE, CUBE box-to-box High Availability uses the Redundancy Group (RG) infrastructure: two routers share virtual IP addresses and checkpoint signaling and media state over a dedicated link, so established calls survive a failover.
7.3 Cisco Unified Border Element (CUBE) Configuration & Redundancy
As enterprise voice communications transition entirely from legacy TDM circuits to IP-based Session Initiation Protocol (SIP) trunking, the Cisco Unified Border Element (CUBE) serves as the indispensable enterprise Session Border Controller (SBC). Operating natively on Cisco IOS XE platforms, CUBE acts as an intelligent, protocol-aware voice firewall and demarcation point between trusted enterprise internal networks (such as Cisco Unified Communications Manager or Webex Calling) and untrusted public networks (such as Internet Telephony Service Providers [ITSPs] or cloud communications clouds).
+-----------------------------------------------------------------------------------------+
| CISCO UNIFIED BORDER ELEMENT (CUBE) |
| |
| +--------------------+ +------------------+ |
| | INTERNAL TRUSTED | | UNTRUSTED PUBLIC | |
| | NETWORK | | NETWORK | |
| | | LAN Leg WAN Leg | | |
| | +--------------+ | SIP Signaling SIP Signaling | +------------+ | |
| | | CUCM | |================> [ CUBE ] ================> | ITSP SIP | | |
| | +--------------+ | / \ | | SBC / | | |
| | | RTP Media / \ RTP Media | | CARRIER | | |
| | +--------------+ | (Call Leg 1) / \ (Call Leg 2) | +------------+ | |
| | | IP Endpoint | |<-----------------+ +------------>| | |
| | +--------------+ | [ Media Flow-Through ] | | |
| +--------------------+ +------------------+ |
+-----------------------------------------------------------------------------------------+
1. CUBE Architectural Role & Core Capabilities
CUBE terminates and re-originates both signaling and media sessions at the enterprise perimeter. Its primary responsibilities include:
- Demarcation & Security: Shields internal network topologies by hiding internal IP addresses, directory structures, and device types from the carrier (Topology Hiding). Prevents toll fraud via the
ip address trusted list. - Protocol & Header Normalization: Resolves interoperability disparities between carrier SIP implementations and internal CUCM formatting using SIP profiles.
- Media Interworking: Bridges disparate media streams requiring codec transcoding (e.g., G.711 to G.729), dual-tone multi-frequency (DTMF) conversion (e.g., RFC 2833 to SIP KPML), or media encryption conversion (SRTP to RTP).
- Call Admission Control (CAC): Enforces bandwidth and call volume thresholds to prevent WAN link saturation.
- High Availability: Delivers uninterrupted call continuity via active/standby stateful failover.
2. Global Media Flow Modes: Flow-Through vs. Flow-Around
CUBE can be configured in one of two fundamental media modes under voice service voip:
voice service voip
mode border-element
media flow-through ! (Or 'media flow-around')
Media Flow-Through (Standard Enterprise Default)
In media flow-through mode, CUBE acts as a true back-to-back user agent (B2BUA) for both signaling and media:
- CUBE terminates the incoming RTP media stream on one interface (LAN), inspects and buffers the media packets, and re-originates a completely independent outbound RTP media stream on the other interface (WAN).
- Topology Hiding: Internal IP addresses of IP phones are never exposed to the public carrier; the carrier only sees the public WAN IP of CUBE in the Session Description Protocol (SDP)
c=connection line. - Full Media Services: Enables hardware transcoding, DTMF interworking, audio conferencing, media recording, and SRTP-to-RTP interworking.
- Resource Consumption: Consumes router CPU, packet memory, and DSP resources for all active concurrent calls.
Media Flow-Around
In media flow-around mode, CUBE terminates and processes SIP signaling, but directs the RTP media streams to bypass CUBE entirely:
- CUBE passes the endpoints' own SDP connection addresses through unchanged, so the internal IP phone and the carrier's gateway exchange RTP packets directly end to end.
- Resource Efficiency: Drastically reduces router CPU and memory overhead; zero DSP resources are consumed on CUBE for media handling.
- Architectural Limitations: Completely eliminates topology hiding (carrier sees internal endpoint IPs). Transcoding, DTMF translation, and SRTP-to-RTP interworking are impossible because media never touches the router.
| Architectural Capability | Media Flow-Through | Media Flow-Around |
|---|---|---|
| Signaling Termination | Terminated and regenerated by CUBE | Terminated and regenerated by CUBE |
| RTP Media Path | Traverses CUBE (Two discrete legs) | End-to-end between phone and ITSP |
| Topology Hiding | Complete (Internal IPs hidden) | None (Internal IPs exposed in SDP) |
| Transcoding & DTMF Interworking | Fully supported | Unsupported |
| CUBE CPU / DSP Overhead | Moderate to High | Extremely Low |
| NAT Traversal Support | Fully supported across dual VRFs/interfaces | Requires direct IP routability or external SBC |
3. Voice Class SIP Profiles (SIP Header Manipulation)
Public telephony carriers strictly enforce SIP header formatting for caller billing verification, regulatory compliance, and call tracking. Often, CUCM sends internal SIP headers that an ITSP rejects (e.g., private domain names or unroutable extensions). Cisco IOS XE provides Voice Class SIP Profiles (voice class sip-profiles)—a powerful regular-expression-based engine that modifies, adds, or removes SIP and SDP headers across any SIP request or response method.
SIP Profile Syntax Structure
voice class sip-profiles <tag>
request <method> sip-header <header-name> {modify | add | remove} ...
response <status-code> sip-header <header-name> {modify | add | remove} ...
request <method> sdp-header <header-name> {modify | add | remove} ...
Key Practical Applications
- Modifying the Remote-Party-ID (RPID) or P-Asserted-Identity (PAI) Header: Telecom providers require a valid, billed E.164 number in the PAI or RPID header to authorize outbound calls.
- Removing Private Internal Domain Names: Replacing internal CUCM hostnames (e.g.,
cucm01.corp.internal) with the carrier's registered domain (e.g.,carrier-trunk.com). - Diversion Header Manipulation: Stripping or reformatting SIP Diversion headers on forwarded calls so the ITSP can identify the original forwarding subscriber for billing.
Production SIP Profile Configuration Example
voice class sip-profiles 100
description == ITSP Outbound SIP Header Normalization ==
! Modify From header to replace internal domain with ITSP domain
request INVITE sip-header From modify "@cucm01.enterprise.local" "@sip.itsp-carrier.com"
!
! Add P-Asserted-Identity header with corporate billing pilot number
request INVITE sip-header P-Asserted-Identity add "P-Asserted-Identity: <sip:+14155551000@sip.itsp-carrier.com:5060>"
!
! Remove internal Cisco proprietary headers
request INVITE sip-header Cisco-Guid remove
!
! Modify Contact header in 200 OK responses to present CUBE WAN IP
response 200 sip-header Contact modify "<sip:(.*)@10.1.10.1:5060>" "<sip:\1@198.51.100.2:5060>"
Applying SIP Profiles
SIP profiles can be applied globally under voice service voip or selectively per dial peer (recommended for targeted carrier trunk control):
dial-peer voice 200 voip
description Egress Trunk to ITSP Carrier
destination-pattern +1[2-9]..[2-9]......
session protocol sipv2
session target ipv4:198.51.100.1
voice-class sip profiles 100
4. Server Groups and Outbound Trunk Redundancy
Enterprise SIP trunking requires high availability across redundant carrier Session Border Controllers and multiple internal CUCM subscriber nodes. In Cisco IOS XE, Voice Class Server Groups aggregate multiple SIP destinations under a single logical entity assigned to a dial peer.
Dynamic Heartbeats: SIP OPTIONS Keepalives
Without active heartbeat monitoring, a voice gateway only discovers a dead carrier trunk when an active call setup attempt times out (typically after 32 seconds of SIP INVITE retransmissions), causing noticeable user delay.
Configuring voice class sip-options-keepalive commands the gateway to periodically send SIP OPTIONS request messages to target servers. If a server fails to respond within the configured timeout window, CUBE marks the target BUSY-OUT and automatically reroutes subsequent calls to alternate servers without delay.
! --- Define Outbound Server Group with Active Health Monitoring ---
voice class server-group 50
description == ITSP Redundant Carrier SBCs ==
ipv4 198.51.100.10 preference 1
ipv4 198.51.100.20 preference 2
hunt-scheme preference
!
! --- Define SIP Options Keepalive Profile ---
voice class sip-options-keepalive 1
down-interval 15
up-interval 60
retry 3
!
! --- Assign Server Group to VoIP Dial Peer ---
dial-peer voice 500 voip
description Egress Route to Monitored ITSP Trunks
destination-pattern +1[2-9]..[2-9]......
session protocol sipv2
session server-group 50
voice-class sip options-keepalive profile 1
voice-class codec 1
dtmf-relay rtp-nte
no vad
With hunt-scheme preference (the default), CUBE sends calls to the lowest-preference target first; hunt-scheme round-robin spreads calls across the targets instead. The keepalive profile defaults are up-interval 60, down-interval 30, and retry 5; the example tightens them.
5. CUBE High Availability: Box-to-Box (B2B) Redundancy
For mission-critical enterprise environments, a single CUBE platform represents a single point of failure. Cisco IOS XE delivers Box-to-Box (B2B) High Availability, pairing two physical CUBE routers in an Active/Standby architecture.
+-----------------------------------------------------------------------------------------+
| CUBE BOX-TO-BOX (B2B) HIGH AVAILABILITY PAIR |
| |
| PRIMARY CUBE (Active) SECONDARY CUBE (Standby) |
| +--------------------------+ +--------------------------+ |
| | - Owns Virtual IPs | | - Monitors HSRP Health | |
| | - Processes SIP & Media | | - Hot Standby State | |
| +------------+-------------+ +------------+-------------+ |
| | | |
| +====================================================+ |
| 1. Dedicated Control / Checkpoint Link |
| 2. RG Control-Link Heartbeats |
| 3. SIP & Media State Checkpointing |
+-----------------------------------------------------------------------------------------+
Architectural Components of B2B Redundancy
- Redundancy Group (RG) infrastructure (older ISR G2 CUBE releases used HSRP instead):
- Both CUBE routers share a Virtual IP (VIP) address on both their inside (LAN) and outside (WAN) interfaces.
- CUCM and the ITSP configure SIP trunks pointing solely to the shared Virtual IP addresses, unaware that two physical routers exist.
- Inter-Chassis Communication & Checkpointing Protocol:
- A dedicated direct physical Ethernet link connects the two CUBE routers.
- The active CUBE continuously synchronizes (checkpoints) call control blocks, SIP dialog information, cryptographic keys, and active RTP socket bindings to the standby router in real time.
- Failover Execution:
- If the active router suffers a power failure, hardware crash, or interface link loss, the Redundancy Group moves the standby router to the Active state.
- The newly active CUBE assumes the Virtual IPs via gratuitous ARP.
- Because RTP media sockets and SIP call states were fully synchronized, active audio calls remain connected without dropping (zero dropped calls).
- In-flight call setups (calls currently ringing or negotiating SIP SDP) may need to re-originate, but established two-way voice conversations experience only an imperceptible audio gap (< 500 ms).
What is the primary architectural difference between media flow-through and media flow-around modes on a Cisco Unified Border Element (CUBE)?
Media flow-around provides full hardware transcoding and DTMF interworking, whereas media flow-through disables all media manipulation.
Media flow-around terminates SIP signaling and RTP media on CUBE, whereas flow-through bypasses signaling entirely.
Media flow-through terminates only signaling, whereas media flow-around requires dedicated DSP farm hardware on CUBE.
Flow-through terminates and re-originates RTP on CUBE, hiding internal addresses; flow-around lets RTP pass directly between the endpoints.
An enterprise requires all outbound SIP INVITE requests sent to an ITSP to contain a carrier-assigned billing number in the P-Asserted-Identity header. Which Cisco IOS XE tool is designed specifically to perform this SIP header modification?
Cisco Unified Communications Manager Service Parameters under Enterprise Phone Configuration
Translation Profiles bound to the dial peer using translation-profile outgoing
Voice Translation Rules configured with rule 1 /^.*$/ /P-Asserted-Identity/
Voice Class SIP Profiles using the command request INVITE sip-header P-Asserted-Identity add
A voice engineer configures an outbound VoIP dial peer with a voice class server group pointing to two redundant carrier SIP proxies. What mechanism should be added to ensure the gateway detects a dead carrier proxy dynamically before routing active calls to it?
Enable SIP OPTIONS keepalive on the dial peer or server group so a dead proxy is marked down before calls are sent to it.
Configure media flow-around so that carrier proxy failures do not affect active RTP media streams.
Deploy an analog FXO loop start port to serve as an out-of-band monitoring channel.
Configure huntstop on every outbound dial peer so that failed calls are dropped immediately instead of retrying another proxy.
Sections you finish are checked off in the contents.