@genesys/web-sdk initialization hangs on the RTCPeerConnection handshake. The signaling proxy routes createOffer() correctly, which means the fault lies upstream in the MediaStream binding. Running version 5.18.2 inside hardened Chrome. Logs dump ICE candidate gathering failed immediately after the promise resolves. We’ve already patched the STUN config twice. Softphone UI renders fine, but calls drop after three seconds with SIP 408 Request Timeout. Mic stays hot, audio does jack all. Checking the signaling_url override next.
const sdkConfig = {
region: 'us-east-1',
iceServers: [
{ urls: 'stun:relay.genesyscloud.com:3478' },
{ urls: 'turn:relay.genesyscloud.com:3478', username: 'svc_account', credential: 'turn_secret' }
]
};
genesys.init(sdkConfig);
Hardened Chrome blocks default STUN traversal unless you explicitly pass iceServers through the genesys.init() payload, otherwise the RTCPeerConnection constructor drops the MediaStream binding before the createOffer() promise settles. You’ll need to inject the TURN credentials directly into the genesys.config object before calling platformClient.init(), since the SDK’s internal signalingProxy won’t fallback when candidateGatheringTimeout fires and your app token lacks the voice:call:write scope for /api/v2/interactions/callbacks. Usually just a stale token tripping the handshake.
That iceServers override shared in the thread earlier is exactly what you need. Makes sense. Hardened browsers just drop the default Genesys relay if the work policy blocks unencrypted STUN. When you’re routing media through custom AWS infrastructure, the security group rules often get overlooked. You’ll need UDP ports 3478 and 3479 open inbound on whatever NAT Gateway or Elastic IP is handling the TURN traffic.
Here is a quick CloudFormation snippet for the security group attachment.
TurnRelaySecurityGroup:
Type: AWS::EC2::SecurityGroup
Properties:
GroupDescription: Allow UDP for WebRTC TURN
SecurityGroupIngress:
- IpProtocol: udp
FromPort: 3478
ToPort: 3479
CidrIp: 0.0.0.0/0
You mentioned patching the STUN config twice. Are you dynamically pulling the TURN credentials via a Lambda Data Action, or hardcoding them in the frontend? The handshake usually times out if the credential token expires before createAnswer completes. It’s a common timeout scenario. Check the EventBridge logs for throttling on that auth endpoint.
Does the custom TURN endpoint support TCP fallback, or is it strictly UDP? The genesys-cloud-webrtc-sdk library forces UDP-only negotiation when iceServers are injected via the config payload. Testing showed that forcing iceTransportPolicy: 'all' inside the RTCPeerConnection constructor doesn’t resolve the hang. The handshake stalls at checking state regardless of the STUN response time. Logs indicate candidate gathering failed immediately after the offer is created. It’s apparent the hardened browser blocks the TCP fallback packets before they reach the relay. The security group rules might be rejecting the fallback traffic.
The genesys-cloud-webrtc-sdk library doesn’t retry with TCP if the initial UDP probe fails silently. This configuration introduces a silent failure risk when perimeter firewalls block UDP. Testing confirmed that static turn_secret values trigger 403 Forbidden errors after 24 hours. The TURN server rejects the binding request without logging the rejection. Dynamic credential generation is required. The generateTurnCredentials endpoint must be called before every createOffer(). Using a static secret leads to dropped calls during long sessions. The credential expiration aligns with the session timeout. Calls drop immediately after the window closes.
How many agents are currently hitting this handshake hang, and are they concentrated in a single management unit or spread across the entire desk?
When the softphone fails to establish that media connection, the WEM module typically defaults the agent to a generic unavailable state. That creates a cascade of adherence variances. It’s pretty messy when the system stops tracking skill availability. The dashboard just goes blank for that hour. Management units get penalized for unforecasted idle time anyway, and shift trade approvals get flagged because the agent status never transitions to ready.
The workaround usually sits in the status mapping configuration. You can adjust the timeout threshold for the softphone initialization state. Instead of letting the agent sit in a limbo status that breaks compliance reporting, the WFM admin console allows you to route that specific handshake failure to a predefined maintenance status. That keeps the adherence calculation intact. Shift planners can then view the actual available capacity without the noise from failed media bindings.
The screenshot from the configuration panel shows the status transition rules pretty clearly. The timeout value sits right next to the skill group assignment. Setting it to ninety seconds gives the browser enough time to negotiate the ICE candidates without triggering a hard adherence penalty. If the call still drops after that window, the agent gets pushed to a training or break status automatically. That preserves the schedule integrity.
You’ll want to verify how the current management unit handles custom status codes. Some setups force a manual logout when the softphone times out. That breaks the shift trade workflow completely. Adjusting the status routing rules in the WEM settings usually clears it up. The adherence reports will look much cleaner once the initialization state maps to a trackable status.
Check the compliance settings for your primary desk. The status timeout usually defaults to thirty seconds. Bump it up if the variance stays high.