Network Requirements: Quality of Service (QoS)


  • Updated Upgrade Schedule: On January 23, 2024 Talkdesk will be migrating 100% of voice traffic using new media IPs and port ranges for all calls in all regions to and expanding the UDP port range to 10000-60000. Old IP and port ranges will no longer accept or send voice traffic after this date. Before this migration date, we will perform two days of testing with limited traffic. For additional information, please refer to the FAQ.

To assure resilient and redundant voice communications, your firewall must allow outgoing traffic from the Conversations App to the Talkdesk Global Communications Network and allow return traffic in response to the Conversations App egress connectivity. Connections are always outbound from Talkdesk voice applications. Once a connection is established, data will be bi-directional. Note: Talkdesk will never initiate a connection from the Talkdesk Cloud or Talkdesk Global Communications Network to a Talkdesk client. 

1 - When a call is initiated, either an inbound call or an outbound call, the Talkdesk client sends a connection request to the Talkdesk Global Communications Network to establish a WebRTC connection.

2 - When the connection is established, bi-directional communications are allowed between the two voice endpoints, the Talkdesk user and the caller.

IP ranges and protocols are used to establish Web Real Time Communication (WebRTC) between the Conversations App, 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 Quality of Service (QoS), you can set up rules using the media IP ranges below.


Please review these guidelines:

  • 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 the traffic from the Conversations App goes directly to the internet at the agent's physical location wherever possible to assure the lowest latency to our WebRTC Gateways and the highest call quality.


Media IP Ranges

Global Low-Latency

QoS Traffic Shaping

Additional Resources


Media IP Ranges

WebRTC Gateway Location & Service*

Media IP Range


Global Media Range (Conversations voice)

New as of September 26, 2023. To replace region specific IP ranges.

Global (Testing via Talkdesk Network Test Tool)**

No changes.


Scheduled for deprecation on Jan 23, 2024.


Scheduled for deprecation on Jan 23, 2024.


Scheduled for deprecation on Jan 23, 2024.


Scheduled for deprecation on Jan 23, 2024.


Scheduled for deprecation on Jan 23, 2024.


Scheduled for deprecation on Jan 23, 2024.


Scheduled for deprecation on Jan 23, 2024.

*For global geographical redundancy, Talkdesk utilizes multiple WebRTC Gateways and Locations. Starting December 5, 2023, the new global range will encompass all Gateways and Locations. Note: this change will complete on January 23, 2024, when voice traffic will be 100% migrated to the new Global IP range and region specific IP ranges will be deprecated.

**These IP ranges are required for the purpose of running the Talkdesk Network Test Tool via ports 80, 443, and 3478.

  • TCP: ports 3478, 80, and 443. Note: All port 80 traffic is redirected to HTTPS port 443.
  • UDP: ports 3478 and 10000 – 60000, 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. Note: We will be using UDP ports 10000 - 60000 beginning December 5th.
  • TCP/UDP: port 3478 is used for Talkdesk Network Test tool and Screen recording.


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, Microsoft Edge, and Conversations App, 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).


Additional Resources:


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