WebRTC Release Notes

From Genesys Documentation
Jump to: navigation, search

Not all releases or changes listed below may pertain to your deployment. Check the table below to see which releases apply to you.

Important
The Release table lists the initial availability date for each release and the deployment environments for which a release is made available. Except when otherwise stated in the description for a specific release, each release includes all of the features and resolved issues that were introduced on earlier dates, regardless of the deployment environment. The features and resolved issues that apply only to a specific deployment environment are noted as such.
First availabilityReleased forHighlightsRelease number
January 05, 2022
AWS.png
WebRTC Media Service now starts the session cleanup timer when it receives the /wait request with an incorrect CSeq number. 100.0.019.0000
October 21, 2021
PrivateEdition.png
Support for deploying all private edition services in a single namespace.

WebRTC now supports redirection of logs to stdout.
CPU limits for WebRTC gateway pods can be configured now.

Early Adopter Program support for Genesys Multicloud CX private edition deployments on GKE.
100.0.016.0000
October 20, 2021
AWS.png Azure.png
v100.0.015.0000
Codec configuration support for WebRTC Media Service.
100.0.015.0000
September 30, 2021
AWS.png Azure.png PrivateEdition.png
v 100.0.011.0000
Support for arbitrary UIDs in private edition deployments on OpenShift.
Call hold operation handled properly now.
100.0.011.0000
September 17, 2021
AWS.png Azure.png
v 100.0.009.0000. DN@SwitchName format handled properly now. 100.0.009.0000
August 31, 2021
Azure.png
v 100.0.008.0000. Fixed WebRTC Media Service processing of ICE/DTLS completion and ICE errors. 100.0.008.0000
July 14, 2021
AWS.png PrivateEdition.png
v 100.0.003.0000. Tested for Hunt Group feature in AWS. 100.0.003.0000
July 09, 2021
AWS.png Azure.png PrivateEdition.png
v 9.0.000.89

Early Adopter Program support for Genesys Multicloud CX private edition deployments on OpenShift.

Hunt Group feature successfully tested in SIP Cluster deployments with WebRTC Media Service (AWS only).

200 OK status code sent for a pending/wait request when there is an authorization time out expiration.
9.0.000.89

January 05, 2022 AWS.png

Resolved Issues

  • WebRTC Media Service now starts the session cleanup timer when it receives the /wait request with an incorrect CSeq number. Previously while receiving the /wait request with an incorrect CSeq number, WebRTC Media Service sent a 400 Bad Request to the client but did not restart the cleanup timer. After this, if the browser client sent a new /sign_in request with a new tab ID without signing out from the previous session, the session stopped responding in the WebRTC Media Service side. (WRTCMS-883)

October 21, 2021 PrivateEdition.png

PrivateEdition.png

Private Edition: 100.0.016.0000 available October 21, 2021

For private edition

  • WebRTC now supports configurable namespaces. For more information, see Configure WebRTC. (WRTCMS-836)
  • CPU limits for WebRTC gateway pods can be configured now. (WRTCMS-828)
  • WebRTC now supports redirection of logs to stdout . (WRTCMS-823)
  • As of December 23, 2021, WebRTC supports deployments on Google Kubernetes Engine (GKE) in Genesys Multicloud CX private edition, as part of the Early Adopter Program. (CPE-1963)

October 20, 2021 AWS.png Azure.png

What's New

  • WebRTC Media Service allows configuring the list of codecs used for browser-based WebRTC endpoints.
    Configure the new option, webrtc.codecs on the Cloud Cluster application/Agent group/Person level to enable WWE 9 to provide the list of codecs that can be used for the WebRTC endpoint. The option webrtc.codecs defines the comma-separated list that specifies codecs that can be used for the WebRTC endpoint. The default values for this option are opus, pcmu, pcma. The changes take effect after the next platform configuration refresh interval.
    The browser-based WebRTC endpoint provides the list of codecs to WebRTC Media Service in the /sign_in request body. These codecs are used by WebRTC Media Service for OFFER/ANSWER during SDP negotiation.
    Important
    For information on webrtc.codecs support in WWE, see the WWE RN page.
    For Genesys Softphone in WebRTC mode, you can use the existing options codecs.enabled.audio and codecs.enabled.video. (WRTCMS-758)

September 30, 2021 AWS.png Azure.png PrivateEdition.png

PrivateEdition.png

Private Edition: 100.0.011.0000 available September 22, 2021

Resolved Issues

  • The WebRTC Media Service JavaScript library now correctly processes call hold operations. Previously, when an agent tried to put a consultation call on hold, the library silently ignored the request from SIP Server. As a result, the call hold and its subsequent operations were delayed for 30 seconds due to the hold operation timeout. (WRTCMS-798)

For private edition

  • WebRTC now supports the use of arbitrary, or random, user IDs (UIDs) in OpenShift.
    • The Dockerfile has been modified to specify container and file ownership as user=500 (genesys) and group=0 (root).
    • The securityContext settings exposed in the default values.yaml file specify the user and group IDs for the genesys user (500:500:500). You must override these Helm chart values if you want OpenShift to use arbitrary UIDs. For more information, see Configure WebRTC.
    • WebRTC is deployed using ServiceAccounts that use the restricted Security Context Constraint (SCC). In an earlier implementation, Genesys required you to deploy all private edition services using a ServiceAccount associated with the custom genesys-restricted SCC, to control permissions for the genesys user (500). Genesys now expects OpenShift to use arbitrary UIDs in your deployment, and the genesys-restricted SCC has been deprecated. If you previously deployed WebRTC using the genesys-restricted SCC, Genesys recommends that you redeploy WebRTC so that you use arbitrary UIDs.
      (WRTCMS-764)
    More info: OpenShift security settings

September 17, 2021 AWS.png Azure.png

Resolved Issues

  • WebRTC Media Service now handles the DN@SwitchName format properly in sign-in request. Previously, WebRTC Media Service incorrectly handled the DN@SwitcName format for JSON-based signaling protocol, resulting in an invalid SIP DN being registered. (WRTCMS-792)

August 31, 2021 Azure.png

Resolved Issues

  • WebRTC Media Service no longer processes SIP re-invite until the current ICE/DTLS establishment process completes. Previously, because SIP re-invite did not require that the WebRTC connection be reestablished, WebRTC Media Service did not wait for the completion of the current ICE/DTLS establishment process. This resulted in the agent missing some part of the call (for example, the call did not include the whisper). (WRTCMS-773)
  • The WebRTC Media Service Java Script API library now handles Interactive Connectivity Establishment (ICE) errors and reconnects the media if possible. Previously, the library did not handle the ICE errors caused by temporarily network disconnection during active calls. This resulted in the media connection being lost. (WRTCMS-750)

July 14, 2021 AWS.png PrivateEdition.png

PrivateEdition.png

Private Edition: 100.0.003.0000 available July 14, 2021

What's New

  • AWS only. The Hunt Group feature (parallel call distribution strategy) is successfully tested in SIP Cluster deployments with WebRTC Media Service. Supported in WebRTC Media Service v9.0.and v100.0.x. Calls are initiated/received by the browser-based Workspace Web Edition (WWE) 9.0.x. (WRTCMS-725)

July 09, 2021 AWS.png Azure.png PrivateEdition.png

PrivateEdition.png

Private Edition: 9.0.000.89 available June 30, 2021

What's New

  • AWS only. The Hunt Group feature (parallel call distribution strategy) is successfully tested in SIP Cluster deployments with WebRTC Media Service. Supported in WebRTC Media Service v9.0.and v100.0.x. Calls are initiated/received by the browser-based Workspace Web Edition (WWE) 9.0.x. (WRTCMS-725)

Resolved Issues

  • WebRTC Media Service now sends a 200 OK status code for a pending/wait request when there is an authorization time out expiration (<60 seconds). The subsequent wait request will be rejected with 403/401 messages. (WRTCMS-698)

For private edition

  • Starting with this release, WebRTC is available for select customers in Genesys Multicloud CX private edition, as part of the Early Adopter Program. Deployments on OpenShift Container Platform (OpenShift) are supported. (WRTCMS-661)
    More info: WebRTC Private Edition Guide



Prior Releases

For information about prior releases of WebRTC, click here: WebRTC Release Notes