Most cases of poor audio quality are due to jitter in the call audio caused by bandwidth saturation on-site. Without a QoS system in place, your Internet Service Provider (ISP) dumps packets into large queues when your traffic hits the bandwidth limit. The VOIP packets fill into this queue and are often competing against other data traffic, so they are delayed and "jittered". This can cause sound artifacts such as echoes, long audio delays (more than a second), pops, clicks, and intermittent garbled/robotic voices.

Using QoS to prioritize your VOIP traffic ensures your call audio prompt delivery on your network. That is, your call audio will be delivered at a higher priority than other network traffic, such as checking your email. Users will typically never notice a few millisecond delays in downloading data when checking email – whereas a 400-millisecond delay in call audio packets (RTP) can be significant enough to cause choppy or jittery audio. Using QoS priorities allows you to prioritize the traffic that you would notice a small delay in – i.e. call audio (RTP) – above the traffic you wouldn't notice a small delay in – checking your email, browsing the internet, etc. 

Additionally, your packets flagged for high QoS priority may be honored and expedited on your ISP over the public internet (or they may offer CoS – "Class of Service"). Contact your ISP for information about the honoring of QoS and/or CoS on their services. 

We recommend that you configure QoS on your network to prioritize all packets delivered via Flowroute Direct Media. 

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