We’ve got SIP 408s on the secondary BYOC trunk right when the failover triggers during peak volume. The dialer drops about 15% of the attempts before the invite leaves the Genesys edge, doing jack all to help the backup carrier.
The SIP registration stays solid on the backup trunk, but the outbound invites still fail at the carrier gateway. Throttling the outbound concurrency on the failover path keeps the invite rate under 50 per second and stops the drops for now.
Platform SDK for JS handles the dynamic failover routing a lot better than static throttling. You can spin up a webhook subscription on api/v2/webhooks that watches telephony.sessions.events, then hit the trunk API to adjust the concurrency or failover flags on the fly. Hardcoding a cap at 50 calls just kills your ACD efficiency when the primary trunk comes back online.
Here is how you patch the secondary trunk directly through the Node client when the 408 rate spikes:
const platformClient = require('purecloud-platform-client-v2');
const client = new platformClient.PureCloudPlatformClientV2();
async function adjustTrunkConcurrency(edgeId, trunkId, newMax) {
await client.loginOAuthClientCredentials({
clientId: process.env.GC_CLIENT_ID,
clientSecret: process.env.GC_CLIENT_SECRET,
grantType: 'client_credentials',
scope: ['telephony:edge:write', 'telephony:edge:read']
});
const edgesApi = client.TelephonyProvidersEdgesApi();
const body = {
maxConcurrentCalls: newMax,
failover: true,
status: 'active'
};
return await edgesApi.updateEdgeTrunk(edgeId, trunkId, body);
}
You’ll want to wrap that in a retry loop with exponential backoff since the edge sometimes locks the trunk config during active failover. The webhook payload usually contains the callId and leg details, so you can track the 408 density per minute and only trigger the patch when it crosses your threshold. Running this in a lightweight Express server keeps the latency down. The SDK handles the token refresh automatically, so you don’t have to juggle scopes manually. Just make sure your OAuth client has telephony:edge:write attached, otherwise the PATCH call returns a 403 before it even hits the edge.
The SDK patching approach introduces latency. The POST /api/v2/telephony/trunks call runs async against the gateway state machine and creates a race condition when the dialer pushes INVITE requests faster than the config sync completes. You can bypass the external webhook loop by routing the failover logic through the routing:queue expression layer. The platform caches trunk health inside telephony.sessions.events, so binding a trunk selection expression to the outboundContact payload handles the switch locally. When the primary trunk returns a 408 or 503, the expression checks trunk.status and maps to the secondary trunkId without waiting for an API response. This keeps the dialer queue aligned with the actual SIP registration state.
Try updating the flow configuration to use a routing:script that reads session.trunk.lastFailure and applies a trunk:override field on the outbound request. Set the expression to evaluate trunk.available == false before the INVITE leaves the edge node. You’ll need to lock the maxConcurrency inside the outbound:campaign settings to match the secondary carrier buffer, but the expression handles the routing decision in memory. The gateway stops dropping invites once the failover path matches the queue capacity rules.
The suggestion above hits the mark on the expression layer. Patching the trunk config via PUT /api/v2/telephony/trunks/sip/{trunkId} creates a hard sync gap that the dialer engine absolutely does not wait for. The platform already exposes trunk health through the routing:queue context, so you can bind a dynamic routing rule that checks system.telephony.trunks[trunkId].status before queuing the INVITE.
When you’re validating this in the custom desktop, the client_app_sdk actually caches the trunk state inside the telephony.sessions.events stream. You can assert the failover path directly without hitting the REST API. Here’s how the Playwright suite verifies the routing decision before the SIP stack fires:
const trunkStatus = await page.evaluate(() => {
return window.genesys?.clientAppSdk?.getTelephonyState()?.trunks?.secondary?.health;
});
expect(trunkStatus).toBe('degraded');
The expression evaluator runs locally in the desktop runtime. Routing happens before the outbound dialer even constructs the SIP packet. You’ll want to map the routing:queue expression to if(system.telephony.trunks[trunkId].status == 'healthy', trunkId, fallbackTrunkId). The dialer respects that evaluation and drops the INVITE entirely if the target isn’t green. Kills the 408 loop at the source. Just make sure your OAuth scope includes telephony:trunks:read when spinning up the test harness. Otherwise the SDK returns a masked state object. Tests just time out.