Release Number | AIX | HP-UX PA | HP-UX IPF | Linux | Solaris | Windows |
---|---|---|---|---|---|---|
8.5.210.35 [01/28/2020] – Hot Fix | X | X | ||||
8.5.210.34 [12/20/2017] – Hot Fix | X | X | ||||
8.5.210.31 [05/12/2017] – General | X | X | ||||
8.5.210.27 [01/31/2017] – Hot Fix | X | X | ||||
8.5.210.25 [10/07/2016] – Hot Fix | X | X | ||||
8.5.210.23 [04/29/2016] – Hot Fix | X | X | ||||
8.5.210.16 [10/23/2015] – Hot Fix | X | X | ||||
8.5.210.08 [03/24/2015] – General | X | X | ||||
8.5.210.07 [12/12/2014] – Hot Fix | X | X | ||||
8.5.210.03 [10/06/2014] – General | X | X | ||||
8.5.200.10 [07/18/2014] – General | X | X | ||||
8.5.200.07 [06/02/2014] – General | X | X |
This release note applies to all 8.5 releases of Genesys WebRTC JavaScript API.
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
Corrections and Modifications
This is a hot fix for this product. This release contains no new features or functionality.
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This section describes new features that were introduced in this release of Genesys WebRTC JavaScript API.
This release does not include any corrections or modifications.
Supported Operating Systems
New in This Release
Corrections and Modifications
There are no restrictions for this release. This release contains no new features or functionality.
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This section describes new features that were introduced in this release of Genesys WebRTC JavaScript API.
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)onIceDisconnected
, was added to Grtc.Client
. This event can be used to reestablish a disconnected ICE connection. (MWA-561)
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release contains no new features or functionality.
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This section describes new features that were introduced in this release of Genesys WebRTC JavaScript API.
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)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)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release contains no new features or functionality.
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
There are no restrictions for this release. This release contains no new features or functionality.
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release contains no new features or functionality.
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
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.
talk
and hold
on the SIP-side, which are part of the Broadsoft
SIP Extensions.
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)
Supported Operating Systems
New in This Release
Corrections and Modifications
There are no restrictions for this release. This release contains no new features or functionality.
This release also includes the following correction and modification:
The JSAPI now correctly handles browser-generated DeviceNotFound
errors. (MWA-423)
Supported Operating Systems
New in This Release
Corrections and Modifications
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
This first release of this product does not include any corrections or modifications.
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:
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)
Found In: 8.5.200.10 | Fixed In: |
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.
Found In: 8.5.200.07 | Fixed In: 8.5.210.31 |
Mozilla Firefox does not support SDES-SRTP. Although SDES-SRTP is supported
by Chrome, it has been made obsolete, and should not be used.
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.
Information in this section is included for international customers.
There are no internationalization issues for this product.
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.