WebRTC Release Notes

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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.
Service Available Genesys CX on Private edition Highlights Release
AWS Azure GCP
WebRTC Media ServiceJune 21, 2022 AWS.png

Azure.png

This release includes important fixes.100.0.050.0000
WebRTC Media ServiceJune 16, 2022 AWS.png

Azure.png

WebRTC Media Service now implements golden signals-based observability that enables a global view of the overall service health.100.0.046.0000
WebphoneMay 30, 2022 PrivateEdition.png This release includes important fixes.100.0.031.0000
WebRTC Media ServiceMay 27, 2022 PrivateEdition.png This release includes important improvements and fixes.100.0.044.0000
WebRTC Media ServiceMay 13, 2022 AWS.png

Azure.png

This release includes important improvements and fixes.
100.0.038.0000
WebRTC Media ServiceMarch 21, 2022 AWS.png WebRTC Media Service now supports TLS 1.2. It no longer supports TLS 1.0.100.0.032.0000
WebRTC Media ServiceFebruary 24, 2022 AWS.png

Azure.png

PrivateEdition.png The Content-length HTTP header is now handled as a case-insensitive field.100.0.025.0000
WebRTC Media ServiceJanuary 5, 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
WebRTC Media ServiceOctober 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
WebRTC Media ServiceOctober 20, 2021 AWS.png

Azure.png

v100.0.015.0000
Codec configuration support for WebRTC Media Service.
100.0.015.0000
WebRTC Media ServiceSeptember 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
WebRTC Media ServiceSeptember 17, 2021 AWS.png

Azure.png

v 100.0.009.0000. DN@SwitchName format handled properly now.100.0.009.0000
WebRTC Media ServiceAugust 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
WebRTC Media ServiceJuly 14, 2021 AWS.png PrivateEdition.png v 100.0.003.0000. Tested for Hunt Group feature in AWS.100.0.003.0000
WebRTC Media ServiceJuly 9, 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

WebRTC Media Service: June 21, 2022AWS.png Azure.png

Resolved Issues

  • WebRTC Media Service now avoids the creation of a stale session while authenticating a sign_in request from a client. Previously, when WebRTC Media Service handled the scenario in the following way, it led to a situation where the WebRTC Gateway created a stale session without a HTTP client, affecting the active agent’s sessions.

    1. The WebRTC client sends a sign_in request to the WebRTC Gateway.
    2. The WebRTC Gateway starts the authentication procedure on GAuth service.
    3. The client is disconnected from the WebRTC Gateway and the authentication process is completed. (WRTCMS-1066)

  • WebRTC JS library now uses a generic API path for all "/sign_in" requests and sends the requests to the appropriate endpoint. (WRTCMS-1062)

WebRTC Media Service: June 16, 2022AWS.png Azure.png

What's New

  • WebRTC Media Service now implements golden signals-based observability that enables a global view of the overall service health. The four golden signals include metrics for traffic, latency, errors, and saturation. For more information, see Google SRE book. (WRTCMS-976)

Webphone: May 30, 2022PrivateEdition.png

Resolved Issues

  • Webphone now correctly handles the pre-released version (v1.22.8-gke.200) of Kubernetes. Previously, the ingress object was not deployed properly for the pre-released version. (WRTCMS-1035)

  • Webphone now decreases the number of active agent sessions in the active session metric now decrements when an agent logs out of the session. Previously, the active session metric kept increasing even when the agent logged out of the session. (WRTCMS-956)

WebRTC Media Service: May 27, 2022PrivateEdition.png

What's New

  • Network policies can now be enabled and configured independently for WebRTC Gateway and Coturn service. (WRTCMS-985)

  • Starting with version 100.0.025+0001, WebRTC Media Service now includes the enableServiceLinks Helm chart option in the deployment section of the values.yaml file. This option controls whether service information is added to the environment variables or not. Valid values are true and false. The default value is false. (WRTCMS-942)

Resolved Issues

  • WebRTC Media Service now deploys the Coturn service even if the deployment.coturnService section includes no values. (WRTCMS-1034)

  • WebRTC Media Service now correctly handles the pre-released version (v1.22.8-gke.200) of Kubernetes. Previously, the ingress object was not deployed properly for the pre-released version. (WRTCMS-1010)

WebRTC Media Service: May 13, 2022AWS.png Azure.png

What's New

  • WebRTC Gateway now logs a message into Elasticsearch/stdout even if there are no relay candidates received from the client side, where the wrtc_system_error {type="norelay", domain="%DOMAIN NAME%"} metric is incremented.
     (WRTCMS-966)

  • The Elasticsearch\Grafana log messages now include the following fields for troubleshooting purposes:

    • loginName - Agent name for the given WebRTC session.
    • callUUID - Genesys Call UUID that is tracked through voice and WebRTC services. (WRTCMS-965)

Resolved Issues

  • WebRTC Gateway now clears the existing DTLS context when a new SIP re-invite is received. Previously, when the WebRTC Gateway did not complete processing of the existing DTLS context and a new SIP re-invite was received, the existing DTLS context was used for re-establishing media connection which might have resulted in no media for the call. (WRTCMS-1014)

  • WebRTC Media Service now correctly handles all the audio calls in SIP re-invite state.

    Previously, when WebRTC Media Service handled the scenario in the following way (below), it led to a situation where the audio call got stuck in the SIP re-invite state, declining all subsequent SIP re-invite attempts with the 486 Busy Here error:

    1. The WebRTC Call is established.
    2. The agent performs a consult call.
    3. The agent puts the consult call on hold and SIP NOTIFY is sent to the WebRTC Gateway.
    4. The WebRTC Gateway processes the SIP NOTIFY and sends an OK message so that the SIP Server sends an EVENT-TALK message to the client.
    5. While the hold operation is in progress, the agent releases the consult call and SIP Server sends a SIP re-invite request to the WebRTC Gateway to recover the main call.
    6. The WebRTC Gateway processes the SIP re-invite and sends OFFER SDP to the client.
    7. Simultaneously, the client finishes processing the EVENT-TALK message and sends OFFER SDP to the WebRTC Gateway. (WRTCMS-1005)

WebRTC Media Service: March 21, 2022AWS.png

What's New

  • WebRTC Media Service now supports TLS 1.2. It no longer supports TLS 1.0. (WRTCMS-959)

WebRTC Media Service: February 24, 2022AWS.png Azure.png PrivateEdition.png

Resolved Issues

  • The Content-length HTTP header is now handled as a case-insensitive field. Previously, when the Content-length HTTP header was handled as a case-sensitive field, the body of the HTTP request was parsed incorrectly if the Content-length header field had lowercase characters. (WRTCMS-923)

WebRTC Media Service: January 05, 2022AWS.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)

WebRTC Media Service: October 21, 2021PrivateEdition.png

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)

WebRTC Media Service: October 20, 2021AWS.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)

WebRTC Media Service: September 30, 2021AWS.png Azure.png PrivateEdition.png

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

WebRTC Media Service: September 17, 2021AWS.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)

WebRTC Media Service: August 31, 2021Azure.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)

WebRTC Media Service: July 14, 2021AWS.png PrivateEdition.png

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)

WebRTC Media Service: July 09, 2021AWS.png Azure.png PrivateEdition.png

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