Difference between revisions of "RN/WebRTCMedia Service/100.0.067.0000"

Revision as of 10:01, March 21, 2023

Component	WebRTC Media Service
Deployment Type	Genesys CX on AWS, Genesys CX on Azure
Release Number	100.0.067.0000 (Change release number)
Release Type
Highlight	Support for Ambassador cloud design pattern on Azure. New bandwidth-related metrics. Other improvements and resolved issues.
Boilerplate(s) Used
Release Date	2023-03-23
Private Edition Release Date
Mixed Mode Release ReleaseDate
Private Edition Containers List
JQL	labels=webrtcrn067
Links	Links to customer-facing pages in use: Draft:ReleaseNotes/Current/GenesysEngage-cloud/WebRTC ReleaseNotes/Current/GenesysEngage-cloud/WebRTC
Test Links	Links to test pages (for RN Admins only) None yet!

Issue	Issue Category	Description	SupportingDocumentation
Issue	Issue Category	Description	SupportingDocumentation
WRTCMS-1186	New	WebRTC now retries sending a SIP REGISTER request in case of an error or timeout when SIP Registrati...
WRTCMS-1185	New	WebRTC Gateway now supports RTP inactivity timeout to detect dropped or no-audio calls. This is turn...
WRTCMS-1145	New	The following new bandwidth-related metrics are added for the WebRTC Gateway: `wrtc_current_a...`
WRTCMS-1077	New	Helm-charts are updated to support the Ambassador design pattern in Azure.
WRTCMS-1237	Resolved Issue	The KEDA autoscaler is now created with `ignoreNullValues` set to false to enable K...
WRTCMS-981	Resolved Issue	WebRTC now uses log streaming to Grafana Cloud Loki for Docker based services in Engage Cloud on AWS...
WRTCMS-915	Resolved Issue	Now, session cookies are cleared on receiving a `sign-out` API request.
WRTCMS-861	Resolved Issue	WebRTC clients can now automatically re-login at a new active WebRTC deployment after cutover.

@@ Line 21: / Line 21: @@
 |IssueCategoryId=720446c8-10b6-42b8-af36-34a298aa1c72
 |Content=WebRTC now retry SIP REGISTER in case of error/timeout - In cases when SIP Registration fails due to the network disruption, SIP Proxy instance unavailability, webrtc gateway will attempt to retry/recover SIP registration.
-|LocalContent=WebRTC now retries sending a SIP REGISTER request in case of an error or timeout when SIP Registration fails due to a network disruption or the SIP Proxy instance is unavailable. Previously, when a SIP REGISTER transaction failed for some reason, WebRTC Gateway immediately terminated the corresponding session, that may have resulted in unexpected call terminations. This functionality is turned off by default.
+|LocalContent=WebRTC now retries sending a SIP REGISTER request in case of an error or timeout when SIP Registration fails due to a network disruption or the SIP Proxy instance is unavailable. Previously, when a SIP REGISTER transaction failed, WebRTC Gateway immediately terminated the corresponding session and that may have resulted in unexpected call terminations. This functionality is turned off by default.
 }}
 {{Issue