Loss and Discards Explained
Let's explore Lost Packets and Packet Discards. What are
they, and what do they mean?
First, to determine if you have a packet loss or discard problem, visit www.testyourvoip.com
and run a test. If you are on Vonage or interested in Vonage,
test using "Boston" as your location, because Vonage is located in New
Jersey so Boston will give you the most accurate and realistic results.
Now, for a little background on what VoIP is, and what lost or discarded packets will do to VoIP.
VoIP (Voice over Internet Protocol) works by encoding your voice as packets of data. These
packets are then numbered, packaged, and sent over the network to a
remote machine that receives the packets. Since the Internet
routes packets one at a time, some of these packets may be delayed for
various amounts of time or even lost in transmission.
The remote machine is tasked with the job of taking these packets,
sorting them, and re-creating the original voice. It does
this by assembling the packets as they come in, and waiting some
predetermined amount of time to play back the packets. Any
packets that come in AFTER they should have been worked into the voice
stream are discarded (thrown away), because they are useless - the need
for them has already passed by. Any packets that never make
it at all are called "Lost" packets.
So, discarded packets are packets that have been delayed long enough to
be useless. Lost packets are packets that never made it at
all. Both of them are problems to VoIP, as the missing data
can cause choppy audio ("Burst Loss"), or loss in voice quality
("Random Loss"), depending on whether large blocks of packets are lost
all at once (causing choppiness and silences), or the packet loss is
steady (causing a steady voice, but with occasional artifacts or a
"metallic" or distant/synthetic sound that some people call "toilet
How VoIP Works:
In the above diagram, packet 4 was lost entirely ("LOST") and packet 2
made it, but it was too late ("DISCARDED").
Packet discards can be fixed by either:
Packet LOSS can only be fixed by figuring out where the packets are
being lost, and fixing it. In some cases, if network
congestion gets severe, a packet will get thrown away (lost) by a
router en route to the destination.
- Making the remote machine wait longer before assembling the
voice. This means that more packets will make it, but it also
introduces an annoying delay between the time you say something and the
time the remote person hears it. Vonage has a fixed time
period of (I think) about 1/4 to 1/2 second. A rep MAY change it for you, if they are able to - I don't know.
- Making the packets make it to their destinations faster.
This is the ideal solution as it does not involve introducing
Solving Latency (packet
Packets can be delayed or lost anywhere
on your network. How do you find out where this is happening?
Simple. Using any good Route Tracing tool. When
your packets are sent to a server, they go through the Internet through
a series of routers (also commonly called "hops") to reach their
destination. Delays can occur anywhere along that path,
including your local network. To find out where delays may be
happening, you can either use the DOS-based tool "TRACERT", which is
included on every Windows operating system (and Linux, Unix, and other
operating systems as well). Or you can download a trial of a
visual tool called "PingPlotter" from http://www.pingplotter.com
and get your results in a more visual GUI display. If you use it and prefer it, please do the authors a favor and buy it.
One good site to trace to, for example, would be sip.vonage.net - this
is the server that handles the job of telling Vonage devices that they
have an incoming call, or routing out the calls that Vonage users make
(this is NOT the server the actual calls are handled by - that will
vary depending on where your call comes from or goes to, and Vonage is
not likely to give you that information).
Results from TRACERT will look like this:
1 <1 ms <1 ms <1 ms 192.168.1.1
2 50 ms 13 ms 9 ms host-1 [22.214.171.124]
3 10 ms 13 ms 62 ms host-2 [126.96.36.199]
4 15 ms 19 ms 74 ms host-3 [188.8.131.52]
14 * * * Request timed out.
When you start seeing "Request timed out", you have hit a firewall that
doesn't want you looking inside its network. Press CTRL-C to
cancel the test.
Each line has the following data in order:
- Hop Number
- Lowest Latency
- Average Latency
- Highest (worst) Latency
- Host Name (if available)
- IP Address (in brackets if host name was provided)
"Latency" is a measure of the amount of delay introduced by that
specific hop. The higher the number, the more delay. It is
measured in "ms" or milliseconds, which are 1/1000 of a second.
So a 200 ms delay means 2/10ths of a second. Route delays
are cumulative, meaning they "add up" as you go through the network.
5 hops with a 200 ms delay each means a delay of one second from
source to destination. They are also transient, meaning they will
vary wildly from moment to moment based on the amount of traffic they
are handling at that moment.
To see the TOTAL trip time to a given server, ping it ("PING
(servername)"). The "time=" portion of each response will give
you an idea of how long it took for your test packet to make it to the
server and back. Many servers are set up not to respond to ping
requests, unfortunately. sip.vonage.net is one of them.
The first hop will be your router (if you have one). Your
router should be easy to identify, since it does not have a host name,
and probably has an IP address you recognize if you know your router at
all. If you see high latency here, then you have a router
problem. Your router is on your Local Area Network (LAN), and
should generally have response times within a few ms, and most of the
time should be within 1 ms ("<1 ms" as shown above).
The second hop (or the first if you do not have a router) is your modem
or Internet connection. If you see high latency here, your
network is either saturated (try doing the test with only one computer
hooked up), or your modem has a problem. QoS might
fix your issue if your problem is caused by saturation (trying to
send too much data upstream on the router).
Any large latency numbers after your modem are the sole responsibility
of your ISP, or their upstream providers. Your ISP may or may not
be able to do anything about them, but you can call and ask.
If you are on an idle
network and your modem is still giving you high latency numbers, then
you should probably call your ISP and ask them to do a line test. There
may be something wrong with your cable modem, or your connection.
You can also try a new patch cable between your modem and router
or computer, and checking to make sure the wire coming on from the ISP
(RJ11 for DSL, Coax for Cable) is free of corrosion and plugged in
firmly. If everything looks good, and you want more information
about your cable modem, you can also read the next section on
"Interpreting your Cable Modem Data". But I'd spend that time
calling your ISP if I were you. Maybe you can read it while you
wait on hold. Then you won't be so bored listening to their
INTERPRETING CABLE MODEM DATA:
If you are experiencing a problem, and you've
gotten this far without success, I'm going to assume you are calling your ISP.
You can read this while you are on hold. Really. It's that non-riveting.
First, please understand that I am not
expert in Cable Modems. In fact, if you are, and you see gaps or
mistakes, I'd like to know about them. Really. See the
bottom of this page for feedback information.
However, if you really want to know, you can probably find out some of
the details about the quality of signal between your Cable Modem and
the Mother Ship. This can help you determine if there MAY be a
cabling problem in your coax, almost as easily as turning on the TV and
looking for clear picture.
This information may prove useful when calling Customer Service for
your ISP, though their technicians can probably retrive it faster than
you can read it off to them. But drop words like "low SNR" and
"insufficient downstream power", and they'll treat you with
newfound respect until you mangle up a term, and then they'll laugh at
you and start making up terms to test your knowledge (hint: a
cable modem does NOT come equipped with a flux capacitor). :)
Anyway, enough disclaimers. You're either curious or have a masochistic streak, so I'll do my best. Plus you've probably got some time left on hold.
Many modems will show their status on a local page, like
http://192.168.100.1 (which works on my Cable Modem - a Linksys
BEFCMU10). Addresses may vary by modem manufacturer and/or model,
and the DOCSIS standard says that your ISP can shut down this display
if they want to. If your modem was supplied by your ISP, contact
them for the address. Otherwise, a Google search on your modem
model may yield the address you need, if it exists.
Here are a few of the important bits of data you can get from your
Cable Modem. I've put the data from my Cable Modem here for
Standard Specification Compliant DOCSIS 1.0
DOCSIS is the industry standard for the way modems communicate.
Nowadays, it is VERY RARE to find a Cable ISP who does not use
DOCSYS. There are three levels of this specification.
DOCSYS 1.0 is hideously ancient and many people will tell you
that Vonage will simply not work on DOCSIS 1.0. Well, I get
crystal-clear audio on DOCSIS 1.0 but that does not mean it's not a
problem for anyone. Slightly newer is DOCSIS 1.1, and the latest
is DOCSIS 2.0.
Some modems can be upgraded to newer DOCSIS via a firmware upgrade.
As per the DOCSIS standard, ONLY your Cable ISP is allowed to
load new firmware releases on your modem. For example, my modem
is capable of 1.1, but my ISP uses Axxis equipment, which is not
capable of pushing firmware updates to anything but their proprietary
modems. Given that they want $50 to replace my modem with a
proprietary Axxis one, and given that they are scheduled to be bought
out by Comcast in three months, I'll wait and see what Comcast does.
Downstream (indicating data sent to you from the ISP):
Max Bit Rate 3000000 bps
Downstream Power -4.34 dBmV
SNR 36 dB
Max Bit Rate is the speed cap your Cable company has placed on your
modem. While you may be able to change this using techniques
commonly found on the Internet, your cable company WILL find out, and
they WILL be displeased with you. So just back away from that
concept and move on, OK. If this is on your modem, then this is
the fastest your modem will ever send packets to you.
Downstream Power indicates the amount of signal (or, technically,
signal loss) between you and your ISP. Cisco's web site claims
that this number should be -5 dBmV or greater, so as you can see, my
modem is supposedly in range.
SNR refers to "Signal to Noise Ratio", or how loud the actual data is
as compared to background noise and interference. My cable
signal is 36 times louder than any detected interference. Yay.
This is good. I have no clue what "Bad" would be.
Less than 1 would be "very bad", as in you would not be on this web
site looking at this tutorial.
Upstream (data sent to your ISP from your modem)
Max Bit Rate 256000 bps
Upstream Power 46.00 dBmV
Max Bit Rate is the same as the one above, except this is your upstream
limit (data you send to your ISP). For VoIP applications, this is
the amount of data you can send out, and determines the quality of
voice that others hear when you speak. This is the only area
where QoS is effective, and this number (or actually the real world
tested value) is critical to QoS.
Upstream Power is the level of signal your Cable Modem is pumping out
to your ISP. I could not find any specs on this, except that it
needs to be at least 25dB higher than any interference.
If you really can't sleep, here's a reference with more detail than you'll ever want on the subject: http://www.cisco.com/univercd/cc/td/doc/cisintwk/ito_doc/cable.htm
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