- Google Chrome (system requirements for Chrome are listed here)
- Minimum 4Gb of RAM (8Gb recommended)
You'll need to make sure that you have enough bandwidth to support the number of simultaneous calls you expect your agents to make. The bandwidth used is 64 kbps per phone call (upstream and downstream). It is important to note, this number is only for the audio traffic.
Other actions in TalkPro will send/receive data, so more headroom is required. Also, other tabs the agent may have open, such as email or CRM will be consuming bandwidth as well. It's not possible for us to say exactly what you might need, but as a rough rule of thumb, 1Mbps per person sharing the connection is a good start. For example, if there are 100 people in your office, we recommend 100Mbps symmetrical connection. (It is crucial to have adequate download and upload bandwidth for the number of users).
As additional traffic on the same network can impact audio quality, here are other suggestions you should follow as well:
- Don't run any network-intensive applications on the computers, such as internet radio or streaming video. Also, do not run significant uploads or downloads that might compete with your audio. Close unused desktop apps that might also hoard CPU %
- Check with your IT dept to see if higher Quality of Service is possible for your audio connection.
- Open network ports in your router / firewall / antivirus software (advanced info in table below)
- WebRTC (Chrome/Firefox browser)
- TCP: port 80, 8080 and 443
- UDP: Server port: 10,000 – 30,000. Client will select any available port from the ephemeral range: 1,024 to 65,535.
- If your router supports QoS, prioritize the ports mentioned above, or the IP address of the computer(s) making calls.
- If your router includes SIP Application Level Gateway (ALG) function or Stateful Packet Inspection (SPI), disable both these functions.
Further Networking Info:
WebRTC is supported natively in most modern browsers, however, TalkPro only officially supports Google Chrome. (Edge and Firefox are unsupported).
WebRTC usually works without a problem using inbuilt networking technologies (STUN and TURN). However, environments with very restrictive firewalls may require some setup, the details below have further information for your IT-networking department.
WebRTC client connects using the following details:
|Component||Address||Client-side port||Server side port||Protocol|
|Signaling||janus-dt.convoso.com||Any†||443, 80, 8080||TCP|
|Signaling||janus-lax.convoso.com||Any†||443, 80, 8080||TCP|
|Media (SRTP)||Any†||10,000 - 30,000||UDP|
† The client will select any available port from the ephemeral range. On most machines, these means the port range 1,024 to 65,535.