FAQ - General
FAQ - Installation
FAQ - Callsign Validation
FAQ - Nodes
FAQ - Servers
FAQ - Connecting
FAQ - Operating
FAQ - Advanced Operating
FAQ - Audio
FAQ - iPhone
FAQ - Android
What's the easiest way to test and adjust my
There is a special "echo" conference server, called *ECHOTEST*, to
which you can connect to test your audio. Once connected, the server
simply records anything you transmit and plays it back. This is a
convenient way to verify that your transmitted audio is clean, and to adjust
record and playback sound levels. You can connect by choosing "Connect to
Test Server" from the Station menu. If you are connecting to EchoLink via
an RF link, the node number for *ECHOTEST* is 9999.
I seem to be able to connect to other
stations, but I can't hear them, or they can't hear me. What's the
If you're not hearing any audio, first check to
see if a text message appears on the right-hand side of the screen when you
establish the connection. (Typically, this message says a few words about
the station you have connected to.) If not, you might have a "firewall"
problem that needs to be resolved -- see the Connecting
Next, double-check the settings under
Tools->Setup, on the Audio tab. The two audio devices listed should
match the name of the sound card you're using, or should be set to "[system
Finally, be sure your computer's
microphone and speakers are plugged in properly, and the audio levels
are set right. Go to Tools->Adjust Volume->Recording and
be sure the Microphone is turned up, selected, and not muted. Then,
go to Tools->Adjust Volume->Playback and be sure that Volume Control
and Wave are both turned up, selected, and not muted. Also, be sure
the volume control on your speakers is turned up.
Why do I sometimes hear gaps, or break-ups,
in the transmissions from other stations?
Despite its amazing capabilities, the Internet does not form a
perfect connection between your computer and the other station's
computer. Information on the Internet moves along many diverse paths,
most of which are shared with others. As a result, information can be
delayed, or sometimes even get lost.
Both of these problems, "late" and "lost", can cause trouble
for computer software such as EchoLink which converts voice into data, sends it
over the Internet, and converts it back into voice at the other end. The
reason is that timing is everything. Spoken words must be processed in
real time as a continuous stream, otherwise they will be unintelligible at the
other end. This is not much of a problem for other types of data, such as
e-mail messages and Web pages, which can arrive "late" as long as they can
Often, the problem is in the last mile of the path --
the part between your PC and your internet service provider (ISP). Often,
this is the narrowest path of the whole connection, especially if you are using
a dial-up modem to access the Internet.
Avoid using other Internet programs while you are in QSO with someone
on EchoLink. Downloading Web pages and e-mail, or running other messaging
software, tends to clog your "pipe" to the Internet temporarily, making it more
difficult for audio information to get through on time.
Most stations sound OK, but many of them
tell me my audio is broken up. What's the problem? Can any of these audio problems
be blamed on the sound card itself?
Perhaps. EchoLink operates the computer's sound card at 8,000
samples per second, but a few sound cards are not able to run their "clock" at
exactly this frequency, due to limitations in their internal design.
Tests have shown that some sound cards are off-frequency by as much as
2%. This can cause break-ups in audio at either end of the connection, or
other problems, such as voices sounding higher-pitched or lower-pitched than
One way to test this is to connect to the *ECHOTEST* server. If
the test server seems to work smoothly in both directions, but you get
consistent reports of broken-up audio from other stations, this could be the
problem. If possible, try using a different sound card.
I'm running in Sysop mode, and
have no trouble connecting to other stations, but the audio only seems to be
working in one direction. What's the problem?
Check to be sure your sound card (and sound drivers) are capable of
full-duplex operation. Some older sound cards, cannot play and record at the same time. You will need a full-duplex
sound card in order to use Sysop mode with VOX and/or Internal DTMF decoding.
You should also check to see if your computer is automatically muting
the playback audio while it is recording. If so, this will prevent your sound
card from working correctly with EchoLink. Go into your Playback levels control
panel, be sure that Advanced controls are enabled, and look for an Advanced
button on the microphone slider. If you click this button and see a checkbox
labeled "Audio Monitor" (or something similar), be sure it is checked; if there is a checkbox labeled
"Recording Mute" (or something similar), be sure it is NOT checked.
I have no trouble connecting to other
stations, but everyone I talk to says my audio is too low, or distorted.
You may need to adjust your sound-card Recording Volume
controls. For details, see the section
Sound Card Adjustment in the Help files.
If your audio levels are correct, but your audio is sounding muffled
at the other end, try enabling the 300 Hz High-Pass Filter (on the Audio tab
I'm hearing (or getting reports of)
poor-quality audio. Is this a problem with the server?
No. Contrary to popular belief, the server only provides a list
of available stations -- it is not involved in connecting to another station,
or in exchanging audio or text during a QSO. All of your communication
with another station is transmitted directly to that station over the Internet,
without going through a server.
However, if you notice audio break-up while the station list is
refreshing (normally once every 5 minutes), it may be because the new
information coming down from the server is filling up the same Internet "pipe"
you use for your QSO, or because your computer is busy revising the station
list. To avoid this, un-check the option "Even while connected" on the
Stations tab of the Preferences window.
Is there any way to monitor the audio
Some interesting information is displayed in the Connection
Statistics window during a QSO. (To open this window, choose Connection
Statistics from the View menu). The numbers at the bottom right show the
total number of "packets" received from the other station, the number missed,
and the number received out of sequence.
Whenever possible, EchoLink will re-assemble out-of-sequence packets
in their proper order. However, if a single packet is missing or late to
arrive, it inserts a empty packet in its place, causing a slight "tick" in the
audio. If several packets are missing or arrive late, the audio stops
altogether, and then resumes after several more packets are "buffered up".
The bar graphs at the lower left of this window show the current
contents of the Network buffer and the PC buffer. The Network buffer
holds packets received from the Internet, and the PC buffer holds audio as it
is being sent to the sound card (for smoother playback). During normal
conditions, the Network bar graph will remain steady, near the center. If a lot
of packets are being missed, you'll see the Network bar graph shrink.
Do some sound cards work better than others?
Generally, no, but there is one interesting issue to be aware
of. Tests have revealed that PC sound cards run at slightly different
rates, from unit to unit. Normally, these differences are never noticed,
until a voice-over-IP (VoIP) program such as EchoLink is used.
During a QSO, the transmitting station sends out a stream of packets
to the receiving station, over the Internet. These packets are produced
by the transmitting station's sound card at a certain rate. The sound
card at the receiving station, which is running at the same rate, converts
these packets back into sound.
However, if the transmitting station's sound card is running at a
slightly higher rate, it will produce too many packets for the other station to
be able to accept. Conversely, if it is running slower, there will be
gaps at the receiving end because not enough packets are being received.
In most cases, this difference is too slight to be noticed.
However, if you notice a regular pattern whereby the Network bar graph
(described above) always moves gradually from the middle to one end or the
other, yet the "missed packets" counter is not increasing, it may be due to
this kind of mismatch. You may want to enable the Auto Sample Rate
Compensation option (on the Audio tab of Setup) if you notice this behavior.
Where can I get more information about
troubleshooting audio problems?
See VK2JTP's page, Audio
Set-Up for EchoLink.
I like the automatic Recording feature (on
the Audio tab) as a way to keep a permanent record of QSOs. However, it
seems to be creating hundreds of WAV files -- is there any way to avoid this?
Take a look at the Audio tab to see which Recording mode is
enabled. If you have selected "Record by Callsign", EchoLink will create
a separate WAV file for each transmission through your node, both sent and
received. If your node is fairly busy, or you tend to host conferences
frequently, this will create many, many files. Instead, you might prefer
the "Record by QSO" option, which creates one WAV file for each QSO, a file in
which all stations' transmissions can be heard. This is a more natural
way to record activity on your node if you wish to re-play it later; the Record
by Callsign is more appropriate as a logging function.