WebRTC softphone blocking KB article suggestions during chatbot-to-voice handoff

Voice & Telephony
1

The chatbot handoff flow works fine until the call actually bridges to the agent desktop. Once the WebRTC session establishes, the agent assist panel just hangs. No knowledge base articles show up. The auto-answer suggestions that worked perfectly in the chat window completely vanish. Console throws a media negotiation error right when the transfer happens.

Environment details and steps tried:

  • Genesys Cloud 24.8.1
  • Bold360 DX integration v2.3.4
  • Chrome 124.0 on Windows 11
  • WebRTC softphone enabled with G.711 codec
  • Architect flow: Chatbot → IVR transfer → Voice queue → Agent desktop
  • Tested this early morning in CET
  • Disabled third-party extensions
  • Cleared the cache
  • Rebooted the VM

The article suggestions still don’t surface. Does the WebRTC stream interfere with the DX knowledge base sync? Not sure if the bearer token expires during the media handshake.

Logs look like this:

[INFO] WebRTC connection established
[WARN] ICE candidate gathering delayed
[ERROR] Failed to fetch article suggestions: 401 Unauthorized
at KnowledgeSyncHandler.fetchArticles (dx-integration.js:142)
at async AgentAssistPanel.render (panel.js:89)

The rest of the log cuts off. Browser console just shows net::ERR_CONNECTION_TIMED_OUT on the /api/v2/integrations/genesysdx/knowledge/articles endpoint.

Any idea why the softphone media stream would block the KB API calls? The documentation mentions something about concurrent WebSocket connections but it’s pretty vague. Maybe the WebRTC config needs a specific header for the DX sync to work?

Checking the network tab shows the POST request hanging at 200ms before dropping. The chatbot context payload definitely makes it to the interaction. Just the article suggestions fail. Mic stays hot but the KB panel is doing jack all.

{
 "status": "pending",
 "kb_sync": null,
 "error_code": "WEBSOCKET_TIMEOUT_401"
}
2

Console throws that error when the handshake drops. Usually just a timing issue. You’ll need to patch the TRANSFER CONFIG with the correct ASSIST METADATA before the WEBRTC SESSION initializes. Hit the routing endpoint directly:

curl -X PATCH "https://api.mypurecloud.com/api/v2/conversations/$CONV_ID/transfer" \
-H "Authorization: Bearer $TOKEN" \
-H "Content-Type: application/json" \
-d '{"routing":{"assist":{"enabled":true,"kbSync":true}}}'
3

The suggestion above about patching transfer config before WebRTC session starts is correct direction. The real issue usually sits in SDP offer-answer exchange. When Bold360 es assist payload, Genesys Edge sometimes mangles the a=mid and a=ssrc lines. This breaks DTLS handshake per RFC 5763. You’ll see Chrome console throw media negotiation error because ICE candidates never finish gathering.

Check SIP trace on Edge side. Look for initial INVITE and 183 Session Progress response.

INVITE sip:agent@genesyscloud.com SIP/2.0
v=0
o=- 123456 1 IN IP4 10.0.0.5
s=-
c=IN IP4 10.0.0.5
t=0 0
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9
a=ice-ufrag:xxxx
a=ice-pwd:yyyy

If a=setup:actpass line is missing or codec order shifts, SBC will drop stream. Assist metadata needs to ride on separate data channel. It shouldn’t interfere with media path.

Do not KB sync flag in routing payload if SBC is doing strict RFC 3261 compliance checks. It will cause 488 Not Acceptable Here. Instead, inject assist_metadata into custom headers before INVITE leaves Architect flow. Set X-BOLD360-SYNC: true in SIP header mapping. This keeps SDP clean. Agent desktop will pick up articles once RTP stream stabilizes.

Watch ICE candidate gathering timeout in Edge config. Sometimes it needs extra 2 seconds. Trace usually shows STUN binding failing right at handoff.