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Q: What is "packet loss (%)?"
A:
Packet Loss measures the reliability of a connection. A known chunk of data is sent to the router
and then the router is supposed to send the same data back unaltered (echo).
In the case of something like ping, several packets are sent out over the course of a couple seconds.
So, if 10 packets were sent out, but only 8 made it back, then that would be 20% packet loss;
so the more packets that are sent, the more accurate the picture of what the actual packet loss is.
In a perfect world 0% packet loss is what we all want - every packet we send out makes it
to where it's supposed to go. In reality, some packet loss is probably going to happen,
but as long as it is under 5% or so you shouldn't even notice. So just remember that the
higher the packet loss percentage, the slower the connection will work because in most instances
it has to send the same piece of information several times.
What is roundtrip?
A:
This is a measure of the latency of your network connection . Roundtrip times of 150ms to 200ms will mean excellent call quality
and means that theres very little need for you and the other caller to leave pause in your conversation just to know the other has stopped talking.
When roundtrip times stretch to 350ms and above you will find yourself having to leave longer and longer pauses in your conversation
just to know without ambiguity that the other party has stopped talking. Anything over 500ms and conversations will become stuttered and very strained,
as both calling parties will run the risk of talking over one another. If you are a Skype user that uses a satellite link to connect to the internet,
you can expect this number to be quite high, but there is little you can do about it as it is a consequence of the finite nature of the speed of light.
Similar to jitter, roundtrip data packet transfer time is primarily a function of the speed and quality of your internet connection.
Also, malware and adware, which are common forms of spyware (unwanted software installed on your computeroften without your knowledgethat
consumes processing power and internet bandwidth, see http://arstechnica.com/articles/paedia/malware.ars/ for a good overview of the topic)
can have a detrimental effect on roundtrip time. An excellent tool for removal of spyware is Spybot Search & Destroy, freely available
from http://www.safer-networking.org/en/spybotsd/index.html. Lastly, a firmware upgrade to your router can improve roundtrip time,
and (for real techies) you can tinker with the router port settings for those ports used by Skype (Skypes port settings can be found
by going to Tools--> Option--> Connection).
What is jitter?
A:
JITTER is defined here http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci213534,00.html
as "the variation in the time between packets arriving, caused by network congestion, timing drift,
or route changes" and JITTER BUFFER is defined here http://searchenterprisevoice.techtarget.com/sDefinition/0,,sid66_gci906844,00.html.
In the case of Skype, the jitter buffer would be managed by the Skype client software on the receiving end of a voice call.
Since there are multiple causes of jitter, and voice quality is very much time-dependent,
it would be difficult to draw conclusions about cause and effect by simply referring to the jitter number itself
that is reported on the CallInfo display. After all, we don't even know what the units of measure are
as nothing on the CallInfo display is documented.
However, to state that the jitter stat is 'meaningless' would be a mistake. One might argue that this merely a matter of semantics,
but the fact that jitter is occurring indicates that Skype is busy trying to sort it out on the fly.
That also makes it useful to the user who is trying to make sense out of the QOS issues that he or she may be witnessing at the time.
On the other hand, voice quality may be acceptable even in the presence of certain levels of jitter, and at other times not.
It all depends on how much jitter is occurring, what combination of factors is causing it, over what period of time
it is being measured, and how efficiently Skype is in dealing with it. There are tradeoffs involved.
You can't wait forever for packets that are late arriving and/or out of order, or QOS will suffer.
The goal would be to minimize delays which interfere with 2-way conversations and to mitigate the effects
that missing packets may have on voice fidelity.
Having said all this, there is nothing the user can do about jitter since it is a network issue. However,
it is important to realize its potential to interfere with QOS, and to view it in combination with the rest of the CallInfo display data.
I think that makes it both meaningful and useful.
Jitter
Jitter is a measure of the variations between consecutive data packets arriving at the Skype client.
The lower this number is the better your voice connection. The jitter buffer in your network router holds
incoming data packets until a bunch of them have been received and sequenced, then it starts to release the packets
as a even stream so that the sound you hear is continuous. Meanwhile, new data packets are being received. The idea
behind the jitter buffer is that buffering increases the chances that, even if some subsequent packets are late in their arrival,
there will still be enough packets available in the buffer that the sound you hear continues uninterrupted.
Clever algorithms implemented in firmware within the router detect and try to compensate for the effects of jitter.
In addition, Skype does its own jitter correction internally in software so, presumably, Skypes jitter number
is some composite measure of the residual jitter arriving from the router and its own efforts at correcting jitter.
High levels of jitter cause large numbers of data packets to be discarded by the jitter buffer. This may result in degraded call quality.
Skypes jitter measurement is difficult to interpret in any precise way (outside of Skype that is); however, as already mentioned, the lower this number is the better your voice connection. The causes of jitter are many, and the time spacing between arriving data packets and other characteristics of the packets themselves can vary for a multitude of reasons, but common causes are network congestion and packet route changes.
Skypes jitter number can peak in the thousands at the beginning of a call, but should then drop back to 300 or less. Since the jitter buffer and its corrective algorithms are built into your router, then the level of jitter you experience is in part a function of the quality and sophistication of your router. Upgrading to a new and improved router is one way to improve jitter. However, before doing this it is worth checking your current router manufacturers website for firmware updates that will improve the performance of your existing router. Roundtrip (see below) has some advice on router setup that may also help with jitter.
What are Relays
A:
When a Skype call is a relayed transfer, it means that a direct connection between caller and called could not be established, and so data packets must be routed via intermediate Skype nodes (peers, just like you, in the Skype network). The fewer the number of relay nodes, the smaller the number of hops data packets have to make from node-to-node, and so the better will be your call quality. This is another call attribute for which the lower it is the better it is for call quality. The ideal is to have zero relays, in which case the call is said to be in direct transfer. The most common cause for relayed transfer is because you, or the other party, are using a machine that lives behind a restrictive firewall or use Network Address Translation (NAT). These are the areas to look at if you are suffering badly from relayed transfer. Note, relayed transfer not only affects voice call quality, it also severely impacts file transfers made through Skype, as file transfer traffic is throttled back by Skype to a measly 1 kiliobyte per second (so a 1 MB file would take in excess of 15 minutes to transfer, and larger files correspondingly longer). You can improve file transfer speed by not talking at the same time as a transfer is taking place, because Skype gives voice packets priority over data packets. A high number of relays can also be a symptom of a malfunctioning or old router. In the case of an old router, it can sometimes be given a new breath of lifeincluding reduced relaysby upgrading its firmware.
Gerhard
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