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Ok, welcome back. In our next segment

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we will talk about analyzing poor voice quality.

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So what consitutes poor quality?

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Very simply, if you pick up the phone and you hear

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be very teeny or hollow or you hear static,

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or you hear anything other than a clear, crisp

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quality phone call, 
then that would constitute poor quality.

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As we've mentioned earlier, your data network 
needs to be pristine.

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And quite frankly, when you have issues with voice,

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a lot of times, the knee jerk reaction is 

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to start looking into possible issues with the 
voice configurations in the voice system.

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But in reality, the data network also plays very closely.

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So, I can give you an example with a port mismatch

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from a phone hanging off a PC, 
a PC hanging off a phone,

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an incorrect speed and duplex settings.

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Something as simple as that is going to cause 
you a tremendous issue with quality

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and you would have to correct that 
network data link layer issue

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instead of going and starting to mess 
with QoS and all that other stuff

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where really the underlying problem causing 
the poor quality was in fact the data network.

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So just remember that when 
you're troubleshooting this stuff

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that you're going to be looking not only 
at the voice but also at the data network.

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And how can Wireshark help?
Well, it can analyze the flow of calls

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which we will look at specifically.
We can look at the voice calls.

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We can reconstruct them in the player and
we can take a look and see what's going on.

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So, that being said, you will also be able 
to analyze the response time

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in the packets list or the flow graph
 and use the Expert to really

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get an understanding of what could be happening here.

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An example, or another example is
you may have hair-pinning on a network device.

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This may cause voice to only work 
in a, in a certain direction correctly.

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Incorrectly configured QoS causing queuing issues,

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that can also play a large role into quality.

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Not giving yourself enough of a buffer for voice 
or prioritizing it incorrectly.

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You can experience jitter which comes down to,
you can have latency in your network.

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That will cause extreme poor quality and you 
can also see issues with packets being dropped.

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And if packets are being dropped or 
re-transmitted, that can also cause poor quality.

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And what's really good is that Wireshark
once you place correctly and capture the data

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will start to give you some insight into 
what you can see.

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So up on the screen we have the call
and to get to that we go to the 

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telephony menu and we pull voice calls 
or I should say, VOIP calls.

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And if we want to take a look,
we can pull up the player

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and decode it and then we can see specifically

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what exactly was taking place during this call.

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Now, why is this helpful?

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For those of you that are familiar with 
audio concepts, this looks very much like as if

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you were recording something in pro-tools 
or a tool of that nature.

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It's essentially the same thing and when 
you look at it, it will tell you over time

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exactly what was taking place 
and, and the quality of the calls. 

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So obviously, if it's a smaller wave, 
it's more quiet or less,

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and if it's a bigger wave, it's louder or more.

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And then you could also suffer from similar things 

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like if you are clipping or you have distortion,
you have enough space or head room.

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But a lot of these things come into play 

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when you pull up the RTP player 
and you can view these problems.

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So, that being said,

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just be aware if you're using this player, 
it's also a security concern.

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You can, yes, replay calls that you've captured

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which is probably a concern of those 
who were using the network.

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So just be aware that you have to consider 
security here as well,

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because this will be a violation of security
if you open up someone's call and listen to it.

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So Wireshark, once you captured the data

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and then you start putting it into the tools
in the telephony menu, such as VOIP calls

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will allow you to see the quality of the call.

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You'll be able to see some issues with it.
You can open up the flow graph.

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And you could see if packets are being re-transmitted

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or if your underlying protocols are having an issue.

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And also with RTP, you can check for,

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you could do stream analysis and 
you could check for out-of-order packets,

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or any other issues that are happening with 
RTP while you're doing your analysis.

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And you could look at the RTP stream analysis,

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both in forward and reversed direction

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and see specifically how many packets,

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what the delta is in between the packets,

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if you're experiencing jitter, and so on and so forth.

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So this could be really helpful 
and it also provides the summary

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for you to give you the max amount of jitter
experienced during the call as an example,

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and the duration of the call, as well as the max delta.

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