WebRTC Media Service
- July 03, 2019 (9.0.000.37 UPDATE)
- April 11, 2019 (9.0.000.37)
- December 21, 2018 (9.0.000.27)
- June 29, 2018 (9.0.000.15)
- Known Issues
July 03, 2019 (9.0.000.37 UPDATE)
WebRTC Agent timeout
- WebRTC Agent will now be signed out of the session if the connection between the Endpoint and WebRTC gateway is broken for more than the configured time. For improved security reasons, this time period is set to 5 seconds by default. However, it is configurable up to 15 minutes.
April 11, 2019 (9.0.000.37)
- WebRTC Media Service now retrieves the SIP address from Genesys Web Services (GWS) version 9 automatically and users are not required to configure the SIP address while provisioning Agent Desktop. The Agent Desktop supported version is 9.0.000.21 and above.
December 21, 2018 (9.0.000.27)
- WebRTC Media Service now supports OAuth 2.0 authentication and authorization method to validate the user credentials passed from Agent Desktop. The Genesys Softphone compatible version to support OAuth 2.0 is 9.0.004.05 and above and the Agent Desktop version is 9.0.000.17 and above.
June 29, 2018 (9.0.000.15)
This is the initial release of WebRTC Media Service on the PureEngage Cloud (PEC) platform. Agents can handle both inbound and outbound voice calls through WebRTC-capable devices like Genesys Softphone by communicating with the PEC platform through the WebRTC Media Service. The WebRTC Media Service supports Genesys Softphone version 9.0.003.04+.
The key features of the WebRTC Media Service are:
- Supports G.711 and Opus codecs.
- Provides real-time media transcoding whenever required.
- Supports audio calls only.
- Signalling and media encryption capabilities of WebRTC Media Service ensures appropriate security for voice communications over the public network.
There are currently no known issues.