Troubleshooting Codec Mismatches in Complex Multi-Carrier BYOC Environments

What This Guide Covers

You are diagnosing audio quality failures - one-way audio, garbled speech, silence after answer - in a Genesys Cloud or NICE CXone BYOC (Bring Your Own Carrier) deployment where multiple SIP trunks from different carriers are active simultaneously. When this is resolved, every call leg negotiates a common codec, RTP flows bidirectionally without transcoding gaps, and your SBC logs confirm clean INVITE/200OK/ACK exchanges with matching codec lists in both directions.

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Prerequisites, Roles & Licensing

• Licensing: Genesys Cloud CX 1/2/3 with BYOC Cloud or BYOC Premises; NICE CXone with Carrier Integration entitlement
• Permissions required:
  • Telephony > Trunk > View/Edit
  • Telephony > SIPProxy > View (Genesys Cloud Edge diagnostics)
  • Analytics > Conversation Detail > View (for media stats)
• Tools required: SBC vendor access (AudioCodes, Ribbon, Oracle SBC, or Cisco CUBE), Wireshark or equivalent packet capture capability at the SBC, sngrep on Linux SBC hosts
• External dependencies: At least two carrier SIP trunk configurations, PCAP files from affected call legs

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The Implementation Deep-Dive

1. Understanding How Codec Negotiation Works in a Multi-Carrier SBC Topology

In a BYOC deployment, the SBC sits between the PSTN carrier trunks and the Genesys Cloud/CXone media plane. Every call leg has two SDP negotiation phases:

1. Carrier-leg SDP: The INVITE from the carrier’s SBC arrives with its offered codec list. Your SBC responds with the subset it accepts.
2. Platform-leg SDP: Your SBC re-INVITEs toward Genesys Cloud (or CXone’s media servers), offering the codec selected in phase 1.

The mismatch problem occurs when:

• Your SBC accepts a codec from Carrier A that Genesys Cloud doesn’t support
• Your SBC offers a codec to Genesys Cloud that doesn’t match what was negotiated with the carrier (the SBC isn’t transcoding - it’s pass-through configured)
• Two carriers offer the same codec name but different parameters (e.g., G.729 with and without Annex B, or PCMA vs. PCMU in a=rtpmap ordering)
• DTMF payload mismatch: carrier sends telephone-event on PT 101, platform expects PT 96

Genesys Cloud supported codecs (BYOC Cloud trunks):

The Trap - assuming G.711 is always available: Some carriers, particularly in East Asia and parts of the Middle East, operate SIP trunks that de-prioritize or omit PCMU from their SDP offers in favor of G.729 or G.722. If your SBC is in pass-through mode and passes the G.729-only SDP to Genesys Cloud, and your Genesys trunk configuration doesn’t have a codec override, the call either fails with 488 Not Acceptable Here or answers with silence if the platform attempts to force PCMU on a non-transcoding RTP path.

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2. Capturing the Right Evidence: SIP Traces and Media Stats

Before touching any configuration, capture evidence. Changing codec lists blindly will introduce new mismatches.

Step 1: Identify the affected call via Genesys Cloud Media Stats

GET /api/v2/conversations/{conversationId}/participants/{participantId}/communications/{communicationId}/wrapup
Authorization: Bearer {access_token}

More useful - the media summary endpoint:

GET /api/v2/conversations/{conversationId}/communications
Authorization: Bearer {access_token}

Look at the response for mediaStatsMinConversationMos - values below 3.5 indicate codec or packet loss issues. For a full media quality breakdown, use the Analytics API:

POST /api/v2/analytics/conversations/details/query
Authorization: Bearer {access_token}
Content-Type: application/json

{
  "conversationFilters": [
    {
      "type": "and",
      "predicates": [
        {
          "dimension": "conversationId",
          "operator": "matches",
          "value": "conv-uuid-affected"
        }
      ]
    }
  ]
}

In the response, inspect mediaEndpointStats[].codecs and mediaEndpointStats[].rxSummary.packetLoss.

Step 2: Capture SDP from both call legs at the SBC

Using sngrep on a Linux-based SBC:

sngrep -I /path/to/capture.pcap host <carrier_sbc_ip>

Export the specific call using Call-ID as filter. Look at:

• The INVITE from the carrier: m=audio ... RTP/AVP 18 0 8 101 (G.729, PCMU, PCMA, telephone-event)
• Your SBC’s 200 OK toward the carrier: m=audio ... RTP/AVP 0 101 (you accepted PCMU and telephone-event)
• Your SBC’s INVITE toward Genesys Cloud: does it match m=audio ... RTP/AVP 0 101?

If the Genesys-ward INVITE shows m=audio ... RTP/AVP 18 101 (G.729), your SBC is passing the carrier’s preference without normalization. That’s the mismatch.

Step 3: Pull the Genesys Edge SIP log (BYOC Premises)

In Genesys Cloud Admin > Edge Groups > [Your Edge] > Logs, download the SIP log for the time window. Search for the Call-ID from your PCAP to find the corresponding leg. The Edge log will show exactly what codec it received in the SDP offer and what it accepted in the 200 OK.

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3. Normalizing Codec Preference at the SBC Layer

The SBC is the correct enforcement point. Do not fight codec mismatches in Genesys Cloud trunk configuration if you can normalize at the SBC - trunk-level codec restrictions in Genesys Cloud apply globally and may break calls from carriers that legitimately require a restricted codec.

AudioCodes Mediant - SIP Profile codec configuration:

configure voip
  sip-definition profile 1
    codec-name PCMU
    codec-name PCMA
    codec-name G729
    codec-name telephone-event
    codec-priority-list PCMU PCMA G729
  exit
exit

Set codec-priority-list to force PCMU first. The SBC will re-order the SDP offer to the platform-ward leg to present PCMU as the preferred codec regardless of what the carrier offered.

Oracle SBC (ACME Packet) - codec policy:

codec-policy Brand-A-Policy
  add-codecs-on-egress enabled
  codec-priority-list pcmu pcma g729 telephone-event/8000/1
  audio-transcoding disabled

Bind this policy to the realm facing the Genesys Cloud platform (not the carrier realm). The carrier can still offer G.729 as preferred on the ingress - the SBC will negotiate it, then re-offer PCMU on egress.

The Trap - enabling transcoding without CPU profiling: Setting audio-transcoding enabled solves the codec mismatch immediately but introduces RTP transcoding for every call. On a high-volume SBC (1,000+ concurrent calls), transcoding CPU usage can spike 60-80%. Transcode only the specific codec pairs that are actually mismatching (e.g., G.729 ↔ PCMU only for Carrier B), and validate under load before production cutover. Use sngrep --stats to confirm transcoding sessions are bounded.

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4. Diagnosing telephone-event (RFC 2833 DTMF) Payload Type Collisions

Payload Type (PT) collisions are the most overlooked cause of DTMF failures in multi-carrier environments. RFC 2833 defines telephone-event as a dynamic payload type - the value is negotiated per call in SDP, but different carriers use different defaults.

Common carrier PT assignments:

Genesys Cloud expects PT 101 by default. If Carrier B negotiates PT 96 and your SBC passes it through, Genesys Cloud may silently drop the DTMF packets (interpreting PT 96 as an unknown dynamic codec rather than telephone-event), causing IVR input failures.

SDP normalization for telephone-event at the SBC:

In your egress SDP manipulation rule (platform-ward), force the PT remapping:

# AudioCodes CLI example
sip-definition manipulations
  message-condition 1
    condition sdp-line "m=audio" "telephone-event"
    action replace-sdp-attr "a=rtpmap:96 telephone-event/8000" "a=rtpmap:101 telephone-event/8000"
    action replace-sdp-attr "a=fmtp:96 0-16" "a=fmtp:101 0-16"

Validate by capturing RTP packets during a test DTMF press. The RTP header’s PT field in the DTMF packet must be 101. Use Wireshark: filter rtp && rtp.payload_type == 101 - if you see DTMF packets on a different PT, the normalization is not applied.

The Trap - normalizing SDP but not RTP: SDP manipulation on the SBC changes the signaling (what both sides agreed to). But if the carrier continues sending RTP with PT 96 in the actual media packets (ignoring the renegotiated PT), the SBC must also remap the RTP payload type in the media stream - not just in SDP. This requires RTP manipulation mode, which is distinct from SDP manipulation. Verify the SBC vendor’s capability and enable rtp-pt-remap or equivalent if SDP-only normalization doesn’t resolve the DTMF failure.

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5. BYOC Cloud vs. BYOC Premises: Different Debugging Surfaces

BYOC Cloud (SBC connects directly to Genesys Cloud SIP endpoints):

• You have no Edge to inspect. Your SBC is the termination point.
• Codec enforcement must happen entirely at your SBC before the INVITE reaches audiocodes.us-east-1.genesys.cloud:5061.
• Genesys Cloud trunk configuration provides a secondary codec filter: Admin > Telephony > Trunks > [Trunk] > SIP > Preferred Codec Override. Use this only as a last resort - it rejects calls from carriers offering non-matching codecs.

BYOC Premises (SBC connects to Genesys Edge):

• The Edge is an additional SDP negotiation point. You have two SBC hops: your carrier SBC → Edge → Genesys Cloud media.
• Capture logs from both the carrier SBC and the Edge.
• Edge codec configuration is managed via Edge > Media > Codec Profiles in Genesys Cloud Admin.

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Validation, Edge Cases & Troubleshooting

Edge Case 1: One-Way Audio on Transferred Calls Only

If audio is fine on direct inbound calls but breaks on transfers, the issue is likely a re-INVITE codec renegotiation during the transfer. The transferee SBC sends a re-INVITE with a different codec list (or with a=sendonly/a=recvonly flags reversed). Capture re-INVITEs specifically - search for CSeq: 2 INVITE in your SIP trace. The codec list in the re-INVITE must be a strict subset of the original negotiated codec list.

Edge Case 2: Calls Answering Then Going Silent After 30 Seconds

This is almost always an RTP timeout or firewall state table issue, not a codec mismatch. The codec negotiated correctly, but the carrier’s RTP stream stopped mid-call and the Genesys media server timed out. Look for the carrier’s RTP RTCP sender reports - if they stop after 30 seconds, the carrier’s session timer (Session-Expires header) is expiring without a re-INVITE refreshing the session. Enable SIP session timer on the SBC carrier-ward leg.

Edge Case 3: G.729 Negotiated but Voice Sounds Robotic

Robotic or digital-sounding audio on G.729 is almost always caused by G.729 Annex B (VAD/silence suppression) mismatch. One side is sending comfort noise packets, the other is not expecting them. In SDP, the difference is: a=fmtp:18 annexb=yes vs. a=fmtp:18 annexb=no. Standardize on annexb=no across all trunks unless your carrier explicitly requires Annex B for cost reasons.

Edge Case 4: NICE CXone Virtual Agent Calls Losing Audio on Bot Handoff

In CXone, virtual agent (voice bot) calls use a separate media path during bot hold. When the call is handed back to ACD for agent delivery, a codec renegotiation occurs. If the bot platform (Nuance, Google CCAI) negotiated Opus during the bot leg and the PSTN carrier only supports G.711, the renegotiation after handoff may fail. Ensure your CXone carrier trunk is configured with a codec exclusion list that removes Opus from any legs that touch PSTN carriers.

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Official References

• Genesys Cloud BYOC Cloud SIP Trunk Configuration - Resource Center
• Genesys Cloud Edge Audio Settings - Resource Center
• AudioCodes Mediant SBC Codec Configuration Guide
• RFC 3264 - An Offer/Answer Model with Session Description Protocol (IETF)