Network Requirements and Quality of Service (QoS)

Your firewall must allow outgoing traffic from Talkdesk Workspace (Chrome web browser or desktop application) to the Talkdesk Global Communications Network WebRTC Gateways via the public internet, and allow return traffic in response to the Agent Workspace egress connectivity. The required ports and protocols that must be opened to assure resilient and redundant communications are detailed below. Additionally, please review these guidelines:

  • If your router supports QoS, prioritize the media IP ranges listed here below.
  • If your router includes SIP Application Level Gateway (ALG) or Stateful Packet Inspection (SPI), disable both these functions.

Avoid the use of a VPN, as encapsulating VoIP traffic within an IPSec tunnel could affect audio quality. If you do need to use a VPN, you can exclude the media traffic from the IPSec tunnel by applying split-tunneling. It is important that this traffic from the Talkdesk application goes directly to the internet at the agent's physical location where ever possible to assure the lowest latency to our WebRTC Gateways and the highest call quality.


Media IP Ranges

The following IP ranges and protocols are used to establish Web Real Time Communication (WebRTC) between the Talkdesk Workspace (Chrome web browser or desktop application), Talkdesk’s Global Communication Network WebRTC Gateway, and Regional Cloud Platform and, therefore, must be reachable. If you want to prioritize voice traffic, and your router supports QoS, you can set up rules using the following media IP ranges:


WebRTC Gateway


Media IP Range









*For global geographical redundancy, Talkdesk utilizes multiple WebRTC Gateways and Locations, therefore access to all WebRTC Gateways is required by the Talkdesk Workspace to assure resilient and redundant operation of Talkdesk through the Global Communications Network.

**These Global Media IP ranges are required for the purpose of running the Talkdesk Network Test via ports 80 and 443.

  • TCP: ports 3478, 80, and 443. Note: All port 80 traffic is redirected to HTTPS port 443.
  • UDP: port 3478 and [10000 – 20000], used for media:
    • The client will select any available port from the ephemeral range: 1024 - 65535, typically used as an assignment for the client end of a client-server communication to a well-known port on a server. 


🌍  Global low-latency

We highly recommend setting up all the above ranges, regardless of your location. Our service uses Global Low-Latency (GLL) routing to assign the closest WebRTC Gateway Location to the Agent with the lowest latency. GLL region selection reduces audio latency in call scenarios where two or more parties are connected in a region.

Thanks to GLL, conference audio latency will be reduced in cases where two or more parties are physically close to one another, but far from the United States. For example, a call from Sydney to Sydney will see the greatest benefit from Global Low-Latency as the difference between a locally routed media path and a media path that routes through the United States is the greatest. A conference call where all participants are dialing in from European countries, which is mixed in Ireland, will have lower audio latency for all parties compared to the same conference mixed in the United States.


🚦 Quality of Service: QoS Traffic Shaping

Voice traffic on an organization’s local area network is similar to data traffic, in the sense that it is transmitted as packets over different devices. The main difference between data and voice traffic is that data traffic has the ability to resend information if it initially gets lost in transit.


Voice traffic, on the other hand, cannot resend information because the packets must be received in order, as a continuous stream, for the information to make sense. As such, the way voice packets are treated in your network will have a significant effect on your call quality.


We recommend configuring your network in a way that voice traffic has higher priority than data traffic, and setting up QoS rules based on the Media IP addresses listed above.

This will ensure that your calls have optimal audio quality, without having a noticeable effect on your data traffic.

  • Please reach out to your organization’s network / IT team to determine the best way to set up traffic prioritization.
  • If you do not have an IT team available, we suggest reaching out to your internet provider to check if prioritization can be configured on your network.

Additionally, Talkdesk enables DSCP (Differentiated Services Code Point) by default, with Google Chrome and Callbar, tagging WebRTC media packets, which enables differentiated handling on a LAN, so that real-time media can be prioritized above other network traffic. These will be tagged as EF (101110): Expedited Forwarding (46).


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