Revision as of 12:18, June 18, 2018 by Umamaheswari.85 (talk | contribs)
WebRTC
Note: Not all changes listed below may pertain to your deployment.
June 29, 2018 (9.0.000.xx)
What's New
(Release notes including review comments from Softphone team)
- Genesys Softphone now supports the WebRTC protocol specification, including Opus and G.711 audio codecs.
- Genesys WebRTC supports the traditional audio codecs through transcoding.
- Genesys WebRTC supports all the call flows supported by Genesys Softphone on the Pure Engage Cloud platform.
- Genesys WebRTC enables agents to communicate with the PEC platform over the internet (public or private networks).
- Genesys WebRTC also supports call recording functionalities through Genesys Softphone.
Old release notes
- Provides the ability to Agents to connect from private enterprise networks, remote contact centers, etc.
- Provides the ability to Agents to connect from their home network.
- Supports integration with SIP Cluster.
- Supports Opus and G.711 codecs for voice.
- Supports call flows like inbound, outbound, single step transfer, two step transfers, conference, and call supervisions.
- Supports transcoding from one codec to the other.
- Provides call recording capabilities through Genesys Interaction Recording (GIR).
- Supports single party and third party call control procedures.
- Supports integration with Media Control Platform (MCP) to offer core Genesys features.
Known Issues
There are currently no known issues.
Comments or questions about this documentation? Contact us for support!
