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Call Center - Echo Problem


Hello everyone,


We have recently migrated our call center to our XA/XD 1912 farm due to the recent pandemic. Agents are working from home using the internet infrastructure their ISP provides. The sound is very clear on agent's side and they can hear customers with no interruptions in the audio. However, the performance is fluctuating on customers' side. The sound is sometimes very echoey and parts of agent's sentences intermittently repeat on the call as long as the problem remains and it causes a very unpleasant customer experience.


The virtual desktops are located behind a virtual gateway of Citrix ADC 12.1. DTLS is enabled on the virtual server. We expect users to stay connected to their virtual machines via UDP. However, we sometimes observe fallback to TCP during which the problem resurfaces for the most of the time. There's a soft phone application running on the virtual desktop which navigates voice packets accordingly between both ends. We believe the issue occurs due to high retransmission rate on the TCP connection, even with little to no delay (15-20 ms). The connection becomes extremely lossy and it build up to an increasing retransmission rate, which is reflected as an echo in a phone call.


We have set the "Audio Quality" policy to "Medium - optimized for speech". However, the required ports are not open on Firewall UDP Real-time Transport to kick in over the port range UDP 16500-16509. Here's the tricky part why we have it not enabled thus far; when it is enabled if there's no audio traffic between end user and client for the amount of time greater than the value set for UDP Timeout on the firewall, the firewall thinks that returning UDP packets to the user are deemed as a result of a new connection attempt and blocks them. We believe this is because UDP is a stateless protocol and the only setting that governs this behaviour is UDP Timeout on the firewall. I can't guarantee if there'll be an audio traffic between the user and the virtual desktop and tell the network team to set it to a reasonable value as it could be even an hour until the first audio traffic. Interestingly running a small audio sample after startup enables UDP transfer of audio. When this sample is not run after startup, agents can't hear the customer as the firewall starts to think it's a new connection back to the user (after a long time of no packets) and starts dropping packets as explained thoroughly above.


I'm looking for your suggestions as to how we can improve the audio quality for our call Canter infrastructure. I'd really appreciate if you guys could shed some light on UDP Real-time Transport settings.

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For the issue where the audio stops working after a time this is pointing to a UDP time out for audio. You can fix it by applying KeepAliveTimer registry key & this work only in 1912CU2 (1912.0.2000.2345).


This registry fix provides a timer to send a small datagram over a UDP connection to keep the connection alive between the host and the client due to this audio continue to stay active.


Name: KeepAliveTimer

The recommended value is 15.



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