Hello Pradyumna
I don’t believe you will have to remember the specific conventions for particular QoS codes such as EF for voice and AF31 for call signalling. Such information is not usually asked for on exams. But you should know that EF and AF31 are definitions for Expedited Forwarding and Assured Forwarding.
I’m still not clear on the specific question but I’ll do my best. Remember that on an interface, only one packet/frame can be sent at a time. Also, remember that queuing will take place only if there is congestion. If there is no congestion, that means that there is free bandwidth that can be used immediately, so packets that arrive on the interface are served immediately. If there is congestion, then over any specific period of time, a certain number or percentage of packets will be served from each queue.
Let’s take 100ms as a duration of time and let’s look at the following setup on a GigabitEthernet port:
Remember from this post, that a port can either transmit at its full speed or at 0 Mbps. What we see as throughput is really the average speed over time. If these queues are all full, within those 100ms, a total of 50 ms will be used to send the packets found in output queue1, 20ms for queue2, 20 for queue3, and 10 for queue 4 (based on the percentage). If it is based on bandwidth, then you calculate what percentage of the rated speed of the interface that bandwidth takes up. Now, these time periods are not continuous but are interspersed throughout each time interval. Does that make sense? If I didn’t answer your question sufficiently, feel free to let me know…
By specifying the values found within the DSCP/ToS field of the ping, you can see what kind of behaviour your QoS mechanisms will have on your network. You can send a series of pings and see the response times of each, and see if any are dropped. But this makes sense only if you ping during times of high congestion on your network, otherwise, any values you put in will not have any effect.
I hope this has been helpful!
Laz