Basics of VoIP Troubleshooting Problems

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Basics of VoIP Troubleshooting Problems and its Solutions.
Basics_of_VoIP_Troubleshooting

Some Basics of VoIP Troubleshooting Problems and its Solutions.

The basics of voice over IP troubleshooting comes down to a couple of things. And those things are quality of service. Do we have a quality of service from both sides and dropped packets or lost packets. So using your analyzer probably the easiest way is to ensure no one that you’re looking at your voice over IP.

Quality of service by properly

Quality of service by properly there are two ways to decode that byte. One is the original RF C type of service and the second is differentiated services code point which is what I’m using in the IP header you’ll see it’s labeled the type of service but it’s telling me DCP so I’m looking at the Voice signaling and again ship in this case to invite. That’s the request that’s being made and we notice that there is the quality of service on one side.

But when I look at the response I notice that what’s coming back is no quality of service. So there’s our first step in troubleshooting is to ensure that we have q OS and the same Kuo s on both sides and in this case, you can see that we do not. The invite is set for an f 30 one and the response is zero. Basically, there is no Q OS coming back from the endpoint. Also if I look at the real-time protocol let’s just do the same. 

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We want to ensure that we have quality of service and we see that we have a higher quality of service here on the RTP of 0 3 and let’s see if we get anything on the return again a f 0 3 and on the stuff coming from the other side we see that it’s 0 so potential voice over IP problems here being one side has quality of service and one side does not.

So now that we know there is a queue OS issue as far as both sides are not playing with the same properties we can expect one side probably will have some packet loss with Observer. There’s some easy ways to see this by clicking on expert analysis going to the voice and video conferencing you can see that we’ve got some calls going on here and really there’s this main one and I’ve moved my columns to make it really easy for me to see and then I do see on this first call that there are four packets lost and not that that’s a big thing out of thirteen thousand five hundred and thirty-seven that’s a pretty low percent.

As you can see packets lost percentage point 0 2 percent so that’s not a very high percentage probably not going to affect the quality of this call but some of the pluses with Observer and being able to troubleshoot voice very quickly is I can quickly identify I have lost packets I know what is the percentage and the other thing that observer gives me is based on each of these sips the session initiation protocol here I can see in the signaling.

One side has queue OS and I can see the other side does not. So I did it. Have to look at packets and I can also see on the RTP side the same issue I see that I have quite OS value of zero and I also have on the other side of the queue OS value of forty-six so I can very quickly see that one side of the conversation does have quite OS and one side of the conversation does not have quality of service so we can expect that’s going to cause us problems. 

What’s the MOS and R factor.

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The second step that we can look at here is what’s the MOS and R factor. So MOS is a scale between 0 and 5 and we can see the MOS is actually pretty good mean opinion score is four point one and our factor is a scale between one and one hundred and we see that’s eighty-three again pretty good very good. The other things I look for jitter. What is our jitter total maximum? And I see our max jitter is about one point six milliseconds and bad jitter would be somewhere approaching one hundred and fifty milliseconds so we can see that we’ve got pretty low jitter for this particular call. 

So again should not affect the quality of the call bursts and the gaps the way that I remember how these process works are burst begins with B burst is bad gap begins with G. Gap is good so to define it burst is a period of packet loss and Gap is a period between bursts. So the simple rule of thumb in using the expert observer in troubleshooting voice is if I take my duration of the call which is two minutes twenty-six seconds so that will be a hundred and forty-six seconds minus my setup two seconds.

One hundred and forty-six two minutes would be 120 plus twenty-six one forty-six minus two one forty-four. So my gap should be close to one forty-four if there are no problems with this call and if I look at my gap it’s not quite it’s only one minute seven seconds so that tells me that there’s a kind of a problem and that problem is packet loss that there were some periods of packet loss so that’s a very easy way as well you know our gap duration average gap duration should be basically the call duration minus set up.

So if I look at my duration subtract my setup that should equal my gap if there are no problems if-if it doesn’t equal the gap then that is another indicator that we do have potential voice problems and then finally we can listen to the call if you want that’s one of the things for 4 jitter purposes to see if there are any problems you can click a call right-click the call and say I want to play the audio and the analyzer will say OK . Which streams do you want to play it knows which two streams sort of go together and I can say I’m going to I’m going to pick this first stream and say play and we can listen in the call here a little journey going on.

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Nice nice choice of music. So that’s one of the pluses of observer as well as to be able to listen to the call for jitter purposes and see if there is a problem that we can hear on the network or if we don’t then it’s probably something with the handset the headset you know a physical hardware type problem. 

So hopefully that’s giving you a little bit of insight into voice troubleshooting and some of the simple techniques to do it. Looking for Q OS mismatch type problems looking for jitter problems greater than one hundred and fifty milliseconds in time or approaching one hundred and fifty milliseconds in time packet loss.

Those are all problems that will affect the quality of the voice calling. People will start complaining. 

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