Release Note

Contents

Introduction

This release note applies to all 8.5 releases of Genesys WebRTC JavaScript API.

• Links in the Contents section enable you to access information regarding a specific release. Releases are listed by version number rather than in date order. For this reason, a recent release may be listed after earlier releases, if the version number is lower. Except when otherwise noted in the information for a specific release, each release includes all of the features and corrections that were introduced for the applicable operating system at earlier dates, regardless of the version numbering.
• The Release Number table lists the supported operating systems for each release at a high level. For more detailed information, see the Genesys Supported Operating Environment Reference Guide.
• The Known Issues and Recommendations section is a cumulative list. It includes information on when individual items were found and, if applicable, corrected.
• The Corrections and Modifications section for each release may list additional issues that were corrected without first being documented as Known Issues.
• The Discontinued Support section is cumulative. It lists functionality that is no longer supported and the release number of the version in which support was discontinued.

Use of Third-Party Software

Genesys follows applicable third-party redistribution policies to the extent that Genesys solutions utilize third-party functionality. For specific information on any third-party software used in this product, see the Legal Notices for WebRTC. Please contact your Genesys Customer Care representative if you have any questions.

Supported Operating Systems

New in This Release

This is a hot fix for this product. This release contains no new features or functionality.

Corrections and Modifications

Genesys WebRTC JavaScript API now passes the session ID in the Cookie header of the HTTP request and does not include it in the request URL parameters. Previously, Genesys WebRTC JavaScript API passed the session ID to the Genesys WebRTC Gateway in the request URL parameters.

Note: This version of Genesys WebRTC JavaScript API is compatible with Genesys WebRTC Gateway, from version 8.5.210.93.

(MWA-707)

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Supported Operating Systems

New in This Release

This is a hot fix for this product. This section describes new features that were introduced in this release of Genesys WebRTC JavaScript API.

• Genesys WebRTC JavaScript API has been rebranded to reflect the new Genesys logo in the applicable places. (MWA-601)

Corrections and Modifications

This release does not include any corrections or modifications.

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Supported Operating Systems

New in This Release

There are no restrictions for this release. This release contains no new features or functionality.

Corrections and Modifications

This release also includes the following correction and modification:

Video calls to Chrome 57 now work correctly. Previously, there was no video on the Chrome (callee) side when Chrome received the call from a peer, due to Chrome Issue 7027. (MWA-589)

Sending DTMF tones from a WebRTC client now works in Firefox version 53 and higher. (MWA-581)

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Supported Operating Systems

New in This Release

This is a hot fix for this product. This section describes new features that were introduced in this release of Genesys WebRTC JavaScript API.

• A new event, onSessionHold, was added to Grtc.MediaSession. This event is triggered when a call on-hold is detected. Note: Due to a bug in Mozilla Firefox, this event does not work well with that browser. (MWA-569)
• A new event, onIceDisconnected, was added to Grtc.Client. This event can be used to reestablish a disconnected ICE connection. (MWA-561)

Corrections and Modifications

This release also includes the following corrections and modifications:

You can now fully control the logging in WebRTC JavaScript API (JSAPI) via the Grtc Client log level, except for some messages during initialization. The log levels are named NONE, ERROR, NOTICE, INFO, and DEBUG. These correspond to numeric levels 0–4 respectively where no logging is done during a call at level NONE (0). New log functions have also been added in JSAPI that can be used by the application to log messages at different levels. (MWA-570)

When a connection error is detected with an HTTP request to the WebRTC Gateway, the request is now retried by the JavaScript API after three (3) seconds. Previously, failure of a hanging GET request that triggered an onConnectionError event stopped the hanging GET requests from being made, making it impossible for the application to recover. (MWA-561)

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Supported Operating Systems

New in This Release

This is a hot fix for this product. This release contains no new features or functionality.

Corrections and Modifications

This release also includes the following correction and modification:

WebRTC JavaScript API now completes the DTLS (Datagram Transport Layer Security) negotiation and sends/receives media with Chrome 52 and higher. Previously, due to the default use of ECDSA (Elliptic Curve Digital Signature Algorithm) for certificate generation, introduced in Chrome 52, DTLS negotiation failed, resulting in no media connection. (MWA-556)

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Supported Operating Systems

New in This Release

This is a hot fix for this product. This section describes new features that were introduced in this release of Genesys WebRTC JavaScript API.

• The methods hasAudioEnabled() and hasVideoEnabled() were added to Grtc.MediaSession. These methods determine whether a call has audio/video enabled after an SDP negotiation with the peer. (MWA-516)

Corrections and Modifications

This release includes the following corrections and modifications:

WebRTC JavaScript API now logs a warning message when an unexpected message is received, possibly due to a race condition, and continues long-polling the WebRTC Gateway. Previously in this scenario, WebRTC JavaScript API would throw an exception and stop long-polling. (MWA-540)

WebRTC JavaScript API now detects failure of ICE liveness checks when the network connection is lost, and then throws the connection error event so the application can take the necessary corrective action. Previously, the error might not have been immediately detected and the application would lose media connection, even if the network connection was restored. Currently, the ICE liveness check works only with Chrome, and the connection error is thrown in 15 seconds. (MWA-539)

In WebRTC JavaScript API, the long-polling mechanism is now updated to have a configurable timeout (default 30 seconds) and to reconnect to the gateway after the timeout occurs. During a reconnection, if there is already a network issue between the client and the gateway, it is immediately detected because opening the new connection to the gateway will fail. As a result, the connection error event is thrown and the client can make an appropriate decision for an active call. (MWA-538)

WebRTC JavaScript API is now updated with the latest adapter.js and some additional changes to avoid issues with the latest versions of Chrome and Firefox browsers. (MWA-523)

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Supported Operating Systems

New in This Release

This is a hot fix for this product. This release contains no new features or functionality.

Corrections and Modifications

This release includes the following corrections and modifications:

New calls are no longer affected when a previous call is dropped that had been muted using the JavaScript API updateCall() method. Previously, a new call using the same client was still muted. (MWA-509)

WebRTC JavaScript API now throws a grtcClient.onConnectionError event when a hanging GET connection to the WebRTC Gateway fails. This enables the application to handle such a failure. Previously, the failure was silently ignored. (MWA-510)

WebRTC JavaScript API no longer waits for ICE candidate gathering when it is not needed; for example, during a session renegotiation, given a new media type is not being added or media bundle is being used. Previously, a Chrome issue occurred where ICE candidate gathering did not complete in some scenarios. (MWA-511)

Downgrading a video call to an audio-only call now works correctly with Chrome. Previously, it could have resulted in one-way audio due to Chrome still having video in the SDP offer. (MWA-514)

The RTP proto value UDP/TLS/RTP/SAVPF in an SDP offer, created by Chrome version 46 and higher, is now handled correctly. Without this, a call from Chrome 46 failed with a NOMATCH error from the gateway. (MWA-515)

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Supported Operating Systems

New in This Release

There are no restrictions for this release. This release contains no new features or functionality.

Corrections and Modifications

This release includes the following correction or modification:

Outbound calls using Broadsoft extensions and 3PCC now work properly. Previously, they could fail due to a race condition in the WebRTC Gateway. (MWA-477)

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Supported Operating Systems

New in This Release

This is a hot fix for this product. This release contains no new features or functionality.

Corrections and Modifications

This release includes the following corrections or modifications:

The ICE gathering timer is now cleared in all scenarios. (MWA-469)

A workaround for Chrome Issue 3962 is now included, which will avoid renegotiation failures when the VP8 payload type in the original offer is not 100.

A workaround for Firefox is now included, which will avoid media issues by setting the video port to 0 in the answer SDP, when there is no local video stream and the application has set the OfferToReceiveVideo constraint for creating answer SDP to false.

(MWA-470)

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Supported Operating Systems

New in This Release

There are no restrictions for this release. This section describes new features that were introduced in this release of Genesys WebRTC JavaScript API.

• The Genesys WebRTC JavaScript API now supports adding video to an audio-only call.

• The Genesys WebRTC JavaScript API now supports automatically answering a call, as well as placing a call on hold and resuming it, without direct user interaction with the Web application. This feature works using remote CTI control events talk and hold on the SIP-side, which are part of the Broadsoft SIP Extensions.

Corrections and Modifications

This release includes the following correction or modification:

The WebRTC JavaScript API now provides a configuration for the ICE candidate gathering timer with a default timeout of 3 seconds. Without this timer, Chrome may take up to 10 seconds for ICE candidate gathering, if the client machine has many virtual network interfaces. (MWA-452)

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Supported Operating Systems

New in This Release

There are no restrictions for this release. This release contains no new features or functionality.

Corrections and Modifications

This release also includes the following correction and modification:

The JSAPI now correctly handles browser-generated DeviceNotFound errors. (MWA-423)

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Supported Operating Systems

New in This Release

There are no restrictions for this release. This section describes new features that were introduced in this initial 8.5 release of Genesys WebRTC JavaScript API.

• Communications:

  • Supports two-way audio-only calls

  • Supports two-way audio and video calls

  • Supports transitions between audio-only and video/audio sessions (within browser limitations)

  • Supports incoming calls from either the web or the SIP side

  • Supports sending context data from a web client to the SIP Server as attached data when a call is established

  • Supports sending mid-call user data either to SIP Server as mapped data, or to the remote peer

  • Supports call transfer with Genesys SIP Server

  • Supports sending DTMF tones as telephone-events

• Web browser support:

  • Google Chrome

  • Google Chrome for Android

  • Mozilla Firefox

  • Mozilla Firefox for Android

  • Opera (Desktop and Mobile)

• Supports anonymous access from the web (such as in click-to-dial scenarios)

• JavaScript libraries for integration with web applications, communications, and attached data transfer

• Security:

  • DTLS-SRTP Support

• The WebRTC JSAPI provides a configuration parameter to specify the time (in milliseconds) to wait for an answer from the peer side after making an offer. If that timeout expires and the offer is still not answered, then the JSAPI sends the onPeerNoanswer event to the client application.

  The minimum valid value for this timeout is 18000 (18 seconds) and the default value is 60000 (60 seconds).

• The WebRTC JSAPI provides a mechanism for mid-session data transfer for the following scenarios:

  • Between two peers

  • From a peer to the SIP Server as mapped user data

Corrections and Modifications

This first release of this product does not include any corrections or modifications.

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Known Issues and Recommendations

This section provides the latest information on known issues and recommendations associated with this product.

Mozilla Firefox has an outstanding issue that affects WebRTC JavaScript API:

• Bug 1268533 - ICE restart is falsely detected on some WebRTC renegotiation scenarios

The Chrome browser, since version 47, supports user media operations only from secure origin. If you are using Genesys WebRTC service with Chrome browser version 47 or higher, you must use HTTPS for the application server, as well as the WebRTC Gateway. To do this, configure the WebRTC Gateway and set up the application server with the appropriate security certificates as described in the Genesys WebRTC Service Deployment Guide.

The WebRTC JSAPI Installation Package for Windows can only be installed in the default directory. (MWA-422)

Firefox does not currently support renegotiation of an ongoing media session: once a media session has been set up, its parameters are fixed. For all practical purposes, this means that you cannot, for example, start an audio-only call and then add video to that same PeerConnection later in that session.

In order to add video mid-call, the recommended workaround is to destroy the audio-only PeerConnection and create a new PeerConnection that uses both audio and video. This is handled internally in the JSAPI. If you wish to track this issue, the current Firefox bug can be found at https://bugzilla.mozilla.org/show_bug.cgi?id=857115

Note: This issue has now been resolved by Firefox. However, the default behavior in JSAPI has not changed, so it will still create a new PeerConnection on every renegotiation. This behavior in JSAPI can be overridden to reuse the same PeerConnection by calling the Grtc.Client method setRenewSessionOnNeed(false) with the value false during the initialization part in the client application. Also, the WebRTC Gateway option rsmp.new-pc-support must be set to 0 for this to work.

Mozilla Firefox does not currently support sending DTMF tones. The JSAPI returns an error (-1) when its DTMF-sending API is used with Firefox.

Mozilla Firefox does not support SDES-SRTP. Although SDES-SRTP is supported by Chrome, it has been made obsolete, and should not be used.

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Discontinued Support

This section documents features that are no longer supported in this software. This cumulative list is in release-number order with the most recently discontinued features at the top of the list. For more information on discontinued support for operating environments and databases, see Discontinued Support in the Genesys Supported Operating Environment Reference Guide.

There are no discontinued items for this product.

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Internationalization

Information in this section is included for international customers.

There are no internationalization issues for this product.

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Additional Information

Additional information on Genesys Telecommunications Laboratories, Inc. is available on our Customer Care website. The following documentation also contains information about this software. Please consult the Deployment Guide first.

• The Genesys WebRTC Service Deployment Guide gives you the information that you need to get started with Genesys WebRTC Service. This document contains product overview information, as well as deployment details.

• The Genesys WebRTC Service Developer's Guide gives you the information that you need to customize Genesys WebRTC Service.

Product documentation is provided on the Customer Care website, the Genesys Documentation website, and the Documentation Library DVD (produced monthly).

Note: For the DVD, the New Documents on this DVD page indicates the production date for that disc. Due to disc production schedules, documentation on the Genesys Documentation website may be more up-to-date than what is available on disc immediately after a product is released or updated. To determine the version of a document, check the version number that is located on the second page in PDFs or on the About This File topic in Help files.

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