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Audio Issues

What's the easiest way to test and adjust my audio?

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 problem?

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 FAQ.

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 default]". 

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.

If you are running Windows 98 or Windows 95, be sure there is no other program on the computer that might be using the sound card for recording.  For example, if you have speech-recognition or dictation software installed, be sure it is not running when you use EchoLink.

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 arrive completely.

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.  If cable modem or DSL service is available in your area, you might consider upgrading to it, as it usually makes all Internet use more enjoyable, including EchoLink.

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.

Why do some stations sound smooth, but other stations sound choppy?

The kinds of problems described above are more likely to occur over narrowband connections, such as dial-up lines, than broadband connections, such as cable-modem or DSL lines.  They are also more likely if there is a long Internet path between stations, as is common when the stations are very distant geographically.

Is there anything I can adjust to solve this problem?

If you notice choppy-sounding audio from most stations, and you are using a narrowband connection such as a dial-up modem, you might get better results if you increase the Network Buffering setting.  Choose Setup from the Tools menu, and click on the Audio tab.  Move the slider marked "Network Buffering" all the way to the right, and click OK.

The only disadvantage of increasing Network Buffering is that there will be an additional delay after you "turn it over" to the other station, and vice versa.

Most stations sound OK, but many of them tell me my audio is broken up.  What's the problem?

Many Internet connections work better in one direction than the other.  For example, if you are using a so-called "56k" modem, the best possible connection you can achieve is 53k downstream (from the Internet), and 33.6k upstream (to the Internet).  When you connect, you may see a message reporting your downstream speed, but your upstream speed is usually not displayed.  You may find that hanging up and re-connecting will give better results.

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 normal.  

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, particularly on Windows 95, 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.

Sometimes while I'm transmitting, I see a yellow triangle flashing in the bottom right-corner of the screen.  What does this mean?

This indicates that your Internet connection is not fast enough to support EchoLink.  If you are on a dial-up connection, be sure you are using at least a 33.6K modem (a 56k modem is preferred); a 14.4k modem will not work.  Also, be sure to disable the Conferencing feature (on the Preferences tab).

Even if you are using a 56k modem, your connection to the Internet may not be fast enough to fully support EchoLink.  There are many other factors, such as the quality of the telephone connection between your computer and your Internet provider (ISP).  Check with your ISP for suggestions on how to improve your connection.

I have no trouble connecting to other stations, but everyone I talk to says my audio is too low, or distorted.  Any suggestions?

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 under Tools->Setup).

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 performance?

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.

Why doesn't the Adjust Volume menu item (on the Tools menu) work correctly?

These functions do not work on Windows Vista or Windows 7/8/10. If you are running Windows Vista or Windows 7, 8, or 10, please open the Playback and Recording volume controls using the Windows Control Panel.



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