WebRTC – Voice, Video, Screenshare

WebRTC – Voice, Video, Screenshare

Standalone  / Cisco Unified Contact Center Express (UCCX), Enterprise (UCCE or PCCE)

Expertflow’s WebRTC implementation allows agents and customers to securely share voice, video and screen content. It doesn’t require the customer or your agents to install any software – all that is required is a WebRTC-compatible browser.

A WebRTC session is typically initiated as part of a chat session . In a Cisco UCCX/ UCCE/ PCCE enviornment, it replaces Remote Expert Mobile that Cisco discontinued. Agents use a Finesse Gadget to initiate a session.

Customers can be identified automatically using a cookie/ token on the website.

WebRTC is only a secure end-to-end communication protocol between two endpoints. Therefore we make this technology available in to versions: WebRTC-to-WebRTC, and as a WebRTC-to-SIP gateway.

1/ WebRTC-WebRTC mode

In a WebRTC-to-WebRTC mode, both the customer and the agent are using a Webbrowser to host their voice, video, screenshare experience:

2/ WebRTC-SIP mode

In a WebRTC-to-SIP mode, the customer is using WebRTC as before. On the agent side, voice and video is transcoded to SIP and forwarded to the agent’s Voice/ Videophone. The screensharing element is kept in the agent’s browser. In this case, the solution includes a switch that performs transcoding multiplexing and encryption.

Available features:

  • Browser or SIP phone based voice and video
  • Browser based screen-sharing
  • Hybrid Chat (if installed), and  file sharing.

WebRTC doesn’t allow for remote screen, mouse or keyboard control or screen annotation.

The customer identifies himself to the browser to the client’s website. This broswer in turn is associated to a customer profile in CIM using a cookie that will be stored in the customer’s browser. The customer identity is transmitted through a tag manager data layer  on the website.

Technical and release notes

This solution includes a ICE (Interactive Connectivity Establishment) server that can be located on the client’s premises or using a public server for network address translation (ICE, STUN, TURN).

This solution is currently (July 2020) available as a standalone solution that can be started by a client and agent by sharing a common URL from an existing chat session. In the September 2020 release of Hybrid Chat/ CIM, WebRTC will be started as an additional media directly from the agent application, either from a chat or voice session.

Agents can use a normal Cisco SIP phone (such as Jabber), or Expertflow’s WebRTC client.

We provide a trade-in to current users of CafeX or REM that wish an alternative/ replacement of Cisco Remote Expert Mobile that went EOL (End-of-Life) and EOS (End-Of-Sale) on 11 Feb 2019 and End-of-Support on Feb 2022.