Found on https://skype.forum.com / by Gerhard
There are 4 possibilities for this ECHO/DELAY issue
1. Echo hardware
This echo is only caused by your contact, not by you
realtek or other manufacturers of AC97 sound chip
The solution is as follows:
1. Download the latest version of the realtek or other manufacturer driver (important)
2. Double click on the windows speaker icon
3. Select playback properties
4. MUTE the microphone in the playback section
5. Select recording properties
6. Select the microphone in the recording section
Empty Windows Temp folder(s)
Empty prefetch folder in Windows\prefetch
Delete internet cache
Select soundcard/device for *audio-in* and *audio-out* and not the *windows default device*.
2. Echo/delay via internet line
There is echo known from 0,5s to a couple of minutes.
This echo is NOT caused by any kind of headset/speaker etc.
This is only caused by packet loss on the data highway by one or more ISP in the chain.
You can NOT influence this.
Accept trying to call different times.
Pingplotter will show you this trouble
There is only a small chance to reduce this by downloading a new nodes list called shared.xml.
In same cases your call will routed now differently with better result.
Just delete this file.
It will be downloaded newly with your next start of Skype.
You need to select "show hidden files" in folder options
You can find this file in
Documents and settings
--> delete shared.xml
3. heavy downloads or other heavy network load at the same time
Think about your neigbors in an router surrounding (if you have)
Check this for more informations
4. Check entries in config.xml
*Under the <General> section:
* Toggle its value between 1 and 0 (if there is <AEC>0</AEC> change it to <AEC>1</AEC> or vice versa)
*In the same config.xml file insert (and/or try toggling)
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.