How to fix SIP echo?

Asterisk Asterisk General
1

On my installation I have problems with SIP echo, but not all the times.

When I phone to a specific destination from an ISDN line I have no echo, but when they transfer me to another ext on their PBX echo comes. When they switch me back to previous ext, there is no echo again. Their PBX is a Panasonic.

Is this a SIP problem on my PBX?
Can anyone has a HowTo about tuning SIP?

my SIP Phones=Planet VIP-152
SIP codec=uLaw
my PBX = * 1.2.13

2

[color=darkred]Is there anybody out there? (Pink Floyd)[/color]
Please give me a hint to search the SIP echo.
Should I upgrade to v1.2.16?

SIP echo doen’t always happen.

When I do > sip show peers
My 7 sip channels are from 157 to 360 ms away
Is that normal?

What is to blame?

My codecs are gsm|ulaw|alaw|speex|h263 and order:alaw,gsm,ulaw

For ISDN connection I use AVM C4 CAPI

3

I’ve read these days about echo cancellation and learned that this is a GREAT problem, remaining unsolved(?).
I was trying to eliminate it (checked network, changed codecs), but nothing happened.
Is the only way buying another ISDN BRI card than AVMC4?
Should I buy hardware only from Digium?
Please help.
Filippos

4

Have you read what you have posted ?

If you dont have a problem unless you call this user, Then the problem is on THEIR pbx not yours. and as you say its only when they transfer you to a particular extension, Then the Problem is with likely to be with that extension.

Ian

5

Thanks Ian.
I’ve read that the problem of EC is not applicable to PSTN PBXs.
So, if the problem is from their PBX, it does reflect ours.
Filippos

6

Ok let’s go to the source. In pure VoIP or ISDN environment ecko is possible only as a result of the poor design of UA (endpoint, terminal, soft/hardphone) with a feedback from a speaker to a mic. In this case YOU are causing the problem. Other story when your voice is getting converted to 2-wire line inside PSTN. It uses special transformers which separate analog signals going in oposite directions. This is the major source of the problem. Besides that if you have a problem with one specific extension of one specific PBX should it indicate the problem with that extension. Could you give me a link to the article which claims that ecko doesn’t exist in analog PBXs? I’d be interesting in contacting the author and discuss some aspects of the theory of telecommunications.

I totaly understand that it’s much easyer to make you feel guilty becuse you use “noname” brand PBX (Asterisk in eyes of management) and they installed and use for years some kind of Avaya and just don’t wont to think that it could have a problem too.

Back in days I was working on experimental implementation of drivers for one of ISDN cards and had exactly the same problem, it was extremely sensentive to ecko from the PSTN and was working just fine in pure ISND environment. The only fix was to implement EC and autogain algorithms in the driver.

7

As far as I know, the other end of line is ISDN and this end too, so the problem exists between them?
The same problem occurs and in another installation, where our end is ISDN and the other is ISDN with wireless analog phone or an overseas connection.
Filippos

8

Probably you didn’t read it carefuly. It describes exactly the same as I said.
Please read carefuly the part which describes the source of echo.

Here are the most commen issues.

  1. Nobody knows what’s between your two ISDN connections.
  2. Does remote party use PBX with analog ports?
  3. Do they use different model of the phone set connected to the extension with the problem
  4. Did you test your phone sets? Try to call from different lines and listen from the remote side.
  5. Did you try to max up the gain of your ISDN/IP gateway? You should keep it on the default level except some special cases.
  6. International connections cross multiple codecs even in digital form and add even mode delay.
9

So, what’s the solution to this? EC cards?

10
  1. Use G.711 whenever it’s possible
  2. Search for latest driver.
  3. Check if the driver will allow you to control gain and EC time.
  4. EC inside the asterisk codec processing.
11

what’s this?
OK about the other 3 but how to do this?

12

I thought that the problem was caused by the AVM-C4, but yesterday I’ve replaced it with a Digium B410P which has the same problem even if it has Echo Correction.