To ensure a smooth experience with Talkdesk, we provide a set of requirements for the use of our web application.
- 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 Talkdesk 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 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:
- When possible, use a wired network connection, rather than a Wi-Fi connection. This will generally provide a more consistent and better quality network connection.
- Don't run any network-intensive applications on the computers, such as internet radio or streaming video, or 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 the table below)
- WebRTC (Chrome/Firefox browser)
- TCP: port 80 and 443
- UDP: Server port: 10,000 – 20,000.
- The client will select any available port from the ephemeral* range: 1,024 to 65,535.
- * An ephemeral port is typically used by the Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or the Stream Control Transmission Protocol (SCTP) as the port assignment for the client end of a client–server communication to a well-known port on a server.
- 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.
- Do not use a VPN as it will likely affect audio quality. If you do need to use a VPN, you can exclude the voice traffic using the IP addresses listed here. It is important that the voice traffic does not go over a VPN.
To test network latency and quality you can refer to this article.
Further Networking Info:
webRTC is supported natively in most modern browsers however Talkdesk only officially supports Google Chrome. (Edge and Firefox will also work but 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|
|Media (SRTP)||Click here to view IP addresses||Any†||10,000 - 20,000||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.
Talkdesk Specific Domains:
- td-p-talkforce.herokuapp.com (Specific to widget usage)
These domains should be excluded from inspection/policing on ports 80 and 443 wherever possible.
Where * is all subdomains.
These Talkdesk domains are hosted on Amazon Web Services. You can find Amazon’s current public ranges by checking here. Amazon changes these ranges often, and services could originate from any address in these ranges.
If you utilize SIP devices with Talkdesk, make sure you follow the instructions present on the SIP Phone Networking Details article.