Requirements

The agent only needs a compatible web browser:

• Desktop PC – with Windows 10+
  • Microsoft Edge version 12+
  • Google Chrome version 28+
  • Mozilla Firefox version 22+
  • Safari version 11+
• Android
  • Google Chrome version 28+ (enabled by default since 29)
  • Mozilla Firefox version 24+
• Chrome OS
• Firefox OS
• iOS 11+
  • Mobile Safari/ WebKit

The device running the browser must provide access to microphone and speaker(s), either built in or via a headset

The device will require at least

• a 1.4GHz 64-bit quad-core processor
• 2 GBytes of RAM

  This amount of RAM is necessary in cases where the debug mode need to be enabled for troubleshooting. Please also consider all other applications running in the same computer or device.

By default, the Webphone uses the G.711 codec , requiring a minimum of 100 kilobytes/ second of bandwidth for the voice channel and the web application communication.

If a lower bandwidth is required, the Opus codec can also be used (15 to 30 kbps). For usage details, please contact Sytel.

The client device firewall should allow

• outgoing UDP to the public internet from the browsers that will be using the Webphone
• return traffic in response

If you are operating a restricted network that requires white-listing of media IPs, specify the WebRTC servers.

If your router includes the SIP Application Level Gateway (ALG) function or Stateful Packet Inspection (SPI), disable both functions.

Latency (delay) is the time (in milliseconds) taken for voice data to travel between its source and destination. It affects quality of experience for voice when remote agents (e.g. in India) are at some physical distance from the infrastructure (e.g. in the UK). In such cases, using public networks is not optimal and an International Private Leased Circuit (IPLC) should be used to manage latency and bandwidth.

Callers typically start to notice the effect of latency once it exceeds 200ms, so below 200ms is recommended. Above ~600ms, the experience becomes unusable.