For UDP applications such as an IP phone call, any lost packet cannot be retransmitted because the voice packets come in as an ordered stream; re-transmitting packets is useless.
As a result, any lost or delayed packets for applications running the UDP protocol are a real problem. In the voice call example, losing even a few packets will result in the voice quality becoming choppy and unintelligible. Additionally, the packets are sensitive to jitter, the variation in delay of a streaming application. If the network has plenty of bandwidth and no traffic that bursts above what it can handle, there won't be a problem with packet loss, delay or jitter. But in many enterprise networks, there will be times where links become overly congested to the point where routers and switches start dropping packets because they are coming in/out faster that what can be processed. If that's the case, streaming applications are going to suffer. This is where QoS comes in.

QoS helps manage packet loss, delay and jitter on the network infrastructure. Since there is a finite amount of bandwidth, the first objective is to identify which applications would benefit from managing these three things.

Once network and application administrators identify the applications that need to have priority over bandwidth on a network, the next step is to identify that traffic.
There are several ways to identify or mark the traffic. Class of Service (CoS) and Differentiated Services Code Point (DSCP) are two examples. CoS will mark a data stream in the layer 2 frame header while DSCP will mark a data stream in the layer 3 packet header. Various applications can be marked differently, which allows the network equipment to be able to categorize data into different groups.

Now that data steams can be categorised into different groups, that information can be used to place policy on those groups in order to provide preferential treatment of some data streams over others. This is known as queuing.
For example, if voice traffic is tagged and policy is created to give it access to the majority of network bandwidth on a link, the routing or switching device will move these packets/frames to the front of the queue and transmit them immediately.
But if a standard TCP data transfer stream is marked with a lower priority, it will wait (be queued) until there is sufficient bandwidth to transmit. If the queues fill up too much, these lower-priority packets/frames are the first to get dropped.