Difference between revisions of "RN/WebRTCMedia Service/100.0.067.0000"
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|ComponentId=0371606b-df5e-41e2-869d-2009e76b916e | |ComponentId=0371606b-df5e-41e2-869d-2009e76b916e | ||
|JQL=labels=webrtcrn067 | |JQL=labels=webrtcrn067 | ||
− | |DeploymentTypeId=ec194bf2-b79a-436d-8ff6-eaff94d9f43a, 5439f1be-1868-4091-b058-1667389b6ce1 | + | |DeploymentTypeId=ec194bf2-b79a-436d-8ff6-eaff94d9f43a, 5439f1be-1868-4091-b058-1667389b6ce1, 8b480b3c-2733-433a-9166-eab2c2d0663a |
|ReleaseDate=2023-03-23 | |ReleaseDate=2023-03-23 | ||
+ | |PrivateEditionReleaseDate=2023-04-04 | ||
|Highlight=Support for Ambassador cloud design pattern on Azure. | |Highlight=Support for Ambassador cloud design pattern on Azure. | ||
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|IssueCategoryId=720446c8-10b6-42b8-af36-34a298aa1c72 | |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. | |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 | + | |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 | {{Issue |
Latest revision as of 09:31, April 5, 2023
Component RN Definition[edit source]
Component | WebRTC Media Service |
---|---|
Deployment Type | Genesys CX on AWS, Genesys CX on Azure, Private Edition |
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 | 2023-04-04 |
Mixed Mode Release ReleaseDate | |
Private Edition Containers List | |
JQL | labels=webrtcrn067 |
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-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:
| |
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. |
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