Anyone free to help troubleshoot this discrepancy between WebRTC softphone recording ingestion and our bulk export jobs. We are running a compliance workflow that triggers exports via the Recording API for all channels, but WhatsApp and Voice (PSTN) behave correctly while WebRTC sessions fail the metadata validation step. The environment is Genesys Cloud version 24.3.1, and we are using the standard S3 integration for storage. The issue appears specifically when agents use the built-in softphone within the Agent Desktop.
The bulk export job completes with a status of ‘SUCCESS’, but the resulting JSON manifest files lack the participant_id and interaction_id fields required for our chain of custody audit trails. These fields are present in the PSTN recording payloads. We verified that the Architect flow correctly sets the custom attributes cust_legal_ref and cust_case_id before routing to the recording action. The error occurs downstream during the export process, not during the live interaction.
We checked the POST /api/v2/recordings/bulkexport response, and the initial job creation returns a 202 Accepted. However, when polling the job status at GET /api/v2/recordings/bulkexport/{id}, the final payload shows metadata_status: incomplete for WebRTC items. The S3 objects are uploaded, but the associated metadata files are empty or null. This breaks our legal hold process because we cannot map the audio files to the specific case references without those identifiers.
Has anyone seen this specific metadata drop for WebRTC channels? We suspect it might be related to how the softphone session terminates or how the recording service tags the stream before archiving. We need the metadata to align with the WhatsApp export format to maintain a single legal discovery pipeline. Any insights on debugging the metadata injection step for softphone recordings would be appreciated.
It depends, but generally…
“The environment is Genesys Cloud version 24.3.1, and we are using the standard S3 integration for storage. The issue appears specifically when agents use the built-in softphone”
The discrepancy usually stems from how the recording_type field is populated during the ingestion phase for WebRTC versus PSTN. The bulk export job likely filters or validates metadata based on legacy PSTN schemas, causing the WebRTC objects to fail validation if they lack specific telecom-level attributes like call_leg_id in the expected format.
Check the raw JSON of a failed WebRTC recording in the Genesys Cloud developer tools. You will often find that the media_url is present, but auxiliary fields like direction or participant_id might be null or structured differently compared to PSTN.
To fix this, update your Data Action or webhook payload mapping to explicitly handle the webRTC channel type. Ensure the ServiceNow ticket creation script does not enforce strict schema validation on fields that are optional for digital channels. A common workaround is to add a conditional check in the export logic: if channel == "webRTC", bypass the PSTN-specific metadata validation step.
I normally fix this by switching the export filter to use media_type: webrtc instead of relying on the generic recording type, which bypasses the legacy pstn schema validation entirely.
check your recording ingestion settings in the admin console, specifically under the voice tab. coming from zendesk, i’m used to the ticket metadata being rigid, but genesys cloud is more flexible, which can be confusing. the issue here is likely that the webrtc softphone recordings are tagged with a different media_type than the legacy pstn calls.
in zendesk, we’d just map the channel type to a tag, but in gc, the bulk export api expects specific fields. try adjusting your export filter to explicitly include media_type: webrtc alongside media_type: voice. if you’re using the api directly, your payload should look something like this:
{
"filter": {
"type": "and",
"children": [
{ "type": "equals", "field": "media_type", "value": "webrtc" },
{ "type": "equals", "field": "status", "value": "completed" }
]
}
}
this bypasses the default pstn schema validation that’s failing for you. also, make sure your s3 bucket policy allows the specific metadata headers that webrtc generates, as they differ slightly from sip trunks. it’s a small tweak, but it mirrors how we had to adjust our zendesk export rules for different channel types. the documentation mentions this in the recording api section, but it’s easy to miss if you’re used to the simpler zendesk workflow. hope this helps you get those legal holds sorted!