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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

we can configure this by command allow-connections h323 to h323allow-connections h323 to sipallow-connections sip to h323allow-connections sip to sip under voip service configuration.

SIP/H.323 Trunking

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. SIP (or H.323) trunking is the use of VoIP to facilitate the connection of a private branch exchange (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.

 

Configuration Guide

So first setup is to activate the CUBE funtion:

Router#conf t
Router(config)#hostname CUBE
CUBE(config)#voice service voip
CUBE(confi-voi-serv)# allow-connections h323 to h323
CUBE(confi-voi-serv)# allow-connections h323 to sip
CUBE(confi-voi-serv)# allow-connections sip to h323
CUBE(confi-voi-serv)# allow-connections sip to sip
CUBE(confi-voi-serv)# supplementary-service h450.12
CUBE(confi-voi-serv)# no supplementary-service sip moved-temporarily
CUBE(confi-voi-serv)# no supplementary-service sip refer
CUBE(confi-voi-serv)# fax protocol pass-through g711ulaw
CUBE(confi-voi-serv)# h323
CUBE(confi-voi-serv)# sip
CUBE(confi-voi-serv)# bind control source-interface Vlan1
 bind media source-interface Vlan1
 registrar server expires max 3600 min 3600

Outgoing dial-peer.

dial-peer voice 1000 voip

translation-profile outgoing DID
destination-pattern 0[2348]…….. //the number can be 02, 03, 04, or 08 followed by a 8 digit number, which is a Australian landline.
session protocol sipv2
session target dns:sip20.mynetfone.com.au
voice-class codec 1
dtmf-relay rtp-nte
fax-relay ecm disable
no vad

Reference

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-overview.html

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-multi-trunks.html

https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

https://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html

Config example: https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/99863-cube-config.html

Voice command reference: https://www.cisco.com/c/en/us/td/docs/ios/voice/command/reference/vr_book/vr_a1.html

Step by step configure:

  • http://what-when-how.com/cisco-voice-over-ip-cvoice/implementing-sip-gateways-examining-voip-gateways-and-gateway-control-protocols-part-1/
  • http://what-when-how.com/cisco-voice-over-ip-cvoice/implementing-sip-gateways-examining-voip-gateways-and-gateway-control-protocols-part-2/
  • http://what-when-how.com/cisco-voice-over-ip-cvoice/implementing-sip-gateways-examining-voip-gateways-and-gateway-control-protocols-part-3/