Base unit of content for WebRTC Media Service - 100.0.038.0000
From Genesys Documentation
Component RN Definition[edit source]
Component | WebRTC Media Service |
---|---|
Deployment Type | Genesys CX on AWS, Genesys CX on Azure |
Release Number | 100.0.038.0000 (Change release number) |
Release Type | |
Highlight | This release includes important improvements and fixes. |
Boilerplate(s) Used | |
Release Date | 2022-05-13 |
Private Edition Release Date | |
Mixed Mode Release ReleaseDate | |
Private Edition Containers List | |
JQL | project = "WebRTC Media Service" AND id = WRTCMS-1014 OR id = WRTCMS-1005 OR id = WRTCMS-965 OR id = WRTCMS-966 |
Links | Links to customer-facing pages in use: |
Test Links | Links to test pages (for RN Admins only)
None yet! |
Issue | Issue Category | Description | SupportingDocumentation |
---|---|---|---|
Issue | Issue Category | Description | SupportingDocumentation |
WRTCMS-966 | New | WebRTC Gateway now logs a message into Elasticsearch/stdout even if there are no relay candidates re... | |
WRTCMS-965 | New | The Elasticsearch\Grafana log messages now include the following fields for troubleshooting purposes... | |
WRTCMS-1014 | Resolved Issue | WebRTC Gateway now clears the existing DTLS context when a new SIP re-invite is received. Previously... | |
WRTCMS-1005 | Resolved Issue | WebRTC Media Service now correctly handles all the audio calls in SIP re-invite state. Previously,... |
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