Stuck on a persistent 504 Gateway Timeout when attempting to initiate a WebRTC media stream via the /api/v2/flegs/interactions endpoint within our custom softphone integration. The issue occurs specifically when the Architect flow attempts to bridge a digital channel interaction to a voice queue using the v2.1 SDK, resulting in a failed handshake before the media session establishes. Environment details: Node.js 18, @genesyscloud/genesys-cloud-sdk v2.1.0, US-1 Region. Has anyone encountered similar latency issues with multi-org tenant configurations?
Make sure you check the interaction type mapping. In Zendesk, ticket types were static, but Genesys Cloud requires explicit voice capability flags on digital interactions. The 504 often indicates a handshake mismatch in the flow configuration. Review the interaction object here: https://developer.genesys.cloud/api/v2/interactions
have you tried checking the websocket keep-alive intervals in your jmeter script? the interaction mapping mentioned above is solid, but i often see 504s when the load tester drops the connection before the flow completes the bridge. in my recent tests with eu-west-1, the default timeout was too aggressive for the media handshake. try increasing the connection timeout in your http sampler to 60 seconds. also, verify that your custom softphone isn’t sending duplicate ice candidates too quickly. the api rate limits can throttle the signaling if you’re hammering it during peak load. i usually add a small pause between the create interaction and the media setup calls. here is a snippet of how i handle the timing in my java controller:
Thread.sleep(2000); // wait for flow to initialize
sendMediaSetupRequest();
this helped stabilize the handshake in my concurrent call simulations.
This is a standard case of misaligned media handling policies rather than a simple network timeout. When bridging WebRTC to a voice queue, especially within a BYOC environment, the system must resolve the correct SIP trunk and carrier settings before the media handshake completes. If the outbound routing rules do not explicitly account for the digital-to-voice transition, the gateway can hang waiting for a carrier response that never arrives, resulting in the 504 error.
You need to verify that your BYOC trunks have the correct ‘media handling’ configuration enabled for WebRTC sources. Specifically, check if the ‘Force Codec’ settings are compatible with the softphone’s offered codecs. Many carriers require G.711, but WebRTC often defaults to Opus or PCMU. If the codec negotiation fails silently, the flow times out.
Here is a snippet to check your trunk configuration via API:
GET /api/v2/architect/trunks
# Look for "mediaHandling": { "forceCodec": "PCMU" }
Additionally, ensure that your failover logic does not inadvertently route this traffic to a carrier that does not support WebRTC bridging. We often see this issue when the primary trunk is busy or registered incorrectly, forcing traffic to a secondary carrier with stricter firewall rules. The documentation on BYOC media policies provides a detailed breakdown of these interactions. See this support article for specific configuration steps: https://support.genesys.com/articles/BYOC_WebRTC_Media_Handling_Gotchas.
A common fix is to explicitly set the ‘allowWebRtc’ flag to true in your outbound routing rules for the specific queue. This ensures the platform attempts to use the appropriate media path before falling back to traditional SIP, which often causes these handshake failures. Double-check your carrier’s firewall settings as well, as some block the high ports used by WebRTC data channels.
The easiest fix here is this is to align the outbound routing configuration with the specific media requirements of the digital-to-voice transition. While the interaction mapping is correct, the 504 Gateway Timeout often stems from the gateway waiting for a carrier response that is never initiated due to missing routing rules. Ensure the Outbound Contact Policy explicitly includes the digital channel origin as a valid source for voice bridging. Without this, the system may drop the media handshake before the SIP trunk is engaged.
From an operational perspective, this divergence between stable queue metrics and failing media streams is a common bottleneck in BYOC environments. The Performance dashboard will show the interaction as active in the queue because the signaling layer accepts the request, but the media layer fails during the handshake. This creates a false sense of stability in agent performance views while digital channel exports or recordings fail. Review the Queue Activity metrics specifically for “Failed Handshakes” or “Media Errors” rather than just total interactions. This metric often spikes before the API errors manifest. Additionally, verify that the WebRTC recording settings in the Organization Settings do not conflict with the custom softphone’s codec preferences. If the system attempts to record a codec that the softphone does not support during the bridge, the timeout occurs. Adjusting the recording policy to use a fallback codec, such as Opus or G.711, can often resolve the handshake hang without requiring code changes. This approach prioritizes operational visibility and configuration alignment over complex API debugging.