Designing Interoperable SIP Trunking between Genesys Cloud and Cisco CUBE Gateways

How-To-Articles
1

Designing Interoperable SIP Trunking between Genesys Cloud and Cisco CUBE Gateways

What This Guide Covers

This guide details the architectural requirements and exact configuration steps to establish a robust, secure, and highly available SIP trunk between Genesys Cloud CX (BYOC Premises or Cloud) and Cisco Unified Border Element (CUBE) session border controllers. You will configure SIP profiles, dial peers, codec negotiation, and TLS/SRTP encryption to ensure seamless interoperability for enterprise voice traffic bridging legacy Cisco infrastructure and the Genesys Cloud Edge.

Prerequisites & Infrastructure Sizing

  • Genesys Cloud BYOC: Bring Your Own Carrier (Premises or Cloud) enabled on your Genesys Cloud organization.
  • Cisco CUBE: IOS-XE version 16.12.x or later recommended. Must have appropriate CUBE licensing (app-sec, hsec if high throughput encrypted media is required).
  • Network Reachability: Firewall rules permitting SIP TLS (TCP 5061) and RTP/SRTP (UDP 16384-32767) between the CUBE public/private interfaces and the Genesys Cloud SIP proxy subnets (or local Edges).
  • Certificates: A valid X.509 certificate signed by a globally trusted Certificate Authority (CA), or a Private CA if using BYOC Premises.

The Implementation Deep-Dive

1. Architecting the CUBE Configuration

The Cisco CUBE acts as the demarcation point between the enterprise network (Cisco UCM) and Genesys Cloud. The integration relies on SIP Normalization (SIP Profiles) to translate proprietary Cisco SIP headers into standard RFC compliance that Genesys Cloud expects, and Dial Peers to route the calls.

Step 1.1: Global SIP Service Configuration
Ensure the basic SIP service is bound and strictly using TLS.

voice service voip
 ip address trusted list
  ipv4 192.168.10.0 255.255.255.0  ! CUCM Subnet
  ipv4 52.x.x.x 255.255.255.0      ! Genesys Cloud AWS Subnet (Region specific)
 allow-connections sip to sip
 sip
  bind control source-interface GigabitEthernet0/0/0
  bind media source-interface GigabitEthernet0/0/0
  listen-port tcp 5061

Step 1.2: Configuring Trustpoints and TLS
Genesys Cloud BYOC Cloud requires mutual TLS (mTLS) or strictly enforced inbound TLS.

crypto pki trustpoint GENESYS_CLOUD_ROOT
 enrollment terminal
 revocation-check none
!
crypto pki authenticate GENESYS_CLOUD_ROOT
! Paste the Genesys Cloud CA chain here

voice class tenant 1
 session transport tcp tls
 srtp-crypto 1

2. SIP Profiles for Header Normalization

Genesys Cloud strictly adheres to RFC 3261. Cisco Call Manager often sends proprietary headers (like X-Cisco-CallInfo) or formats the From and To fields in ways that can cause call drops or missing Caller ID.

Apply a SIP Profile to normalize outbound traffic to Genesys Cloud.

voice class sip-profiles 100
 ! Strip proprietary Cisco headers
 request ANY sip-header X-Cisco-CallInfo remove
 ! Normalize the P-Asserted-Identity for accurate Caller ID
 request INVITE sip-header P-Asserted-Identity modify "<sip:(.*)@.*>" "<sip:\1@genesys-cloud.internal>"
 ! Force Early Offer (required by Genesys Cloud for accurate media negotiation)
 request INVITE sdp-header Audio-Attribute add "a=sendrecv"

3. Dial Peer Configuration

Dial peers define the routing logic. You need inbound dial peers from CUCM and outbound dial peers pointing to the Genesys Cloud SIP Proxies. High Availability (HA) is achieved by configuring multiple outbound dial peers with preference mechanisms.

! Outbound to Genesys Cloud (Primary Proxy)
dial-peer voice 2000 voip
 description OUTBOUND_TO_GENESYS_PRIMARY
 destination-pattern +1..........
 session protocol sipv2
 session target ipv4:52.x.x.100:5061
 voice-class codec 1
 voice-class sip profiles 100
 voice-class sip tenant 1
 dtmf-relay rtp-nte
 srtp
 preference 1

! Outbound to Genesys Cloud (Secondary Proxy)
dial-peer voice 2001 voip
 description OUTBOUND_TO_GENESYS_SECONDARY
 destination-pattern +1..........
 session protocol sipv2
 session target ipv4:52.y.y.200:5061
 voice-class codec 1
 voice-class sip profiles 100
 voice-class sip tenant 1
 dtmf-relay rtp-nte
 srtp
 preference 2

4. Genesys Cloud BYOC Configuration

Now switch to the Genesys Cloud Admin UI.

  1. Navigate to Telephony > Trunks.
  2. Click Create Trunk and select BYOC Cloud (or BYOC Premises).
  3. Inbound SIP Routing: Define the public IP address of your Cisco CUBE. Genesys Cloud will only accept SIP requests from these exact IPs.
  4. Outbound SIP Routing: Input the CUBE IP address and port (e.g., 203.0.113.50:5061).
  5. SIP Access Control: Verify that the Access Control List (ACL) matches your CUBE subnet.
  6. Identity: Set the Asserted Identity behavior. If the CUBE is sending P-Asserted-Identity (PAI) for Caller ID, ensure Genesys Cloud is configured to trust the PAI header over the From header.

Validation & Troubleshooting

Scenario 1: One-Way Audio

The Problem: The call connects, but the Genesys Agent cannot hear the caller.
The Root Cause: NAT traversal issues, asymmetric routing, or a mismatch in codec negotiation (usually missing Early Offer).
The Fix:

  1. Run debug ccsip messages on the CUBE. Check the 200 OK SDP body.
  2. Ensure early-offer forced is configured on the outbound dial peer if the SIP profile normalization isn’t functioning.
  3. Verify UDP ports 16384-32767 are open bidirectionally.

Scenario 2: TLS Handshake Failures

The Problem: SIP OPTIONS pings are timing out; calls fail instantly with 503 Service Unavailable.
The Root Cause: Certificate chain mismatch or strict SNI (Server Name Indication) requirements not being met.
The Fix:

  1. Run debug ssl openssl errors.
  2. Ensure the Genesys Cloud Root and Intermediate CAs are successfully installed in the CUBE truststore.
  3. If Genesys Cloud requires SNI, ensure the CUBE iOS version supports SIP TLS SNI injection and configure voice class sip host-domain genesys.cloud.

Scenario 3: DTMF Recognition Failures in Architect IVR

The Problem: Callers enter account numbers, but Architect fails to recognize the digits.
The Root Cause: Mismatched DTMF payload types. Cisco uses payload 101 by default, but intermediate carriers might negotiate 96 or 100.
The Fix: Force asymmetric payload support on the CUBE:

voice service voip
 sip
  asymmetric payload full

Ensure dtmf-relay rtp-nte is explicitly declared on the dial peers interfacing with Genesys Cloud.

Official References