Really bad echo through sip trunk to handsets

Nigel Rennie · Post by **Nigel Rennie** » 02 Feb 2010 01:07

Hi Guys,

I have the following setup:

ISDN <----> OXE <---> Asterisks

Connection between OXE and Asterisks is a SIP trunk.

I am able to make calls from a client attached to Asterisk through the OXE, through the ISDN to external and sound is fine.

However when I call from Asterisks client to a handset attached to the OXE the sound is terrible.

I get a lot of ambient noise coming through the mic of the OXE handset.

The handsets on the OXE work perfectly from internal and external calls. This problem is only apparent when trying to communicate with the Asterisks clients.

Thanks in advance for any responses.
Nige

Nigel Rennie · Post by **Nigel Rennie** » 02 Feb 2010 22:23

I have done further testing and the problem is isolated to calls through the sip trunk from a handset attached to the OXE (4018 / 4028).

The problem is isolated to the mic / input on the phones (which work perfectly in other situations). The sound coming through the ear peace on the handsets is perfect.

Are there settings to adjust the voice gain / volume of the handsets when dialing through different trunks?

alex · Post by **alex** » 04 Feb 2010 02:52

First you wrote

Nigel Rennie wrote: The handsets on the OXE work perfectly from internal and external calls. This problem is only apparent when trying to communicate with the Asterisks clients.

Then

Nigel Rennie wrote: The problem is isolated to the mic / input on the phones (which work perfectly in other situations).

From my point of view that means that the mic works properly if it works fine in other cases. What makes you think that mic is to blame?

Nigel Rennie wrote: Are there settings to adjust the voice gain / volume of the handsets when dialing through different trunks?

Actually you have allway-through digital connection. There is no need to play with gain/attenuation. All levels of transmission at all points of transmission are normalyzed exactly for this purpose - do not have audio loops.

Also do you mean that only 4018/4018 sets have this problem? What about 4038/4068?

Nigel Rennie · Post by **Nigel Rennie** » 04 Feb 2010 21:34

I have only tested this on 4018 and 4028 so expect that it will be the same for all IP sets.

I have done further testing that suggests that the problem is to do with compression.

I am working on it now. I expect that I am going to have to force g711 somehow.

alex · Post by **alex** » 05 Feb 2010 05:55

Nigel Rennie wrote:I have only tested this on 4018 and 4028 so expect that it will be the same for all IP sets.

If you have a possibility to check all 40x8 range do check it.

Nigel Rennie wrote: I have done further testing that suggests that the problem is to do with compression.

I am working on it now. I expect that I am going to have to force g711 somehow.

Hope you will post your solution here

Nigel Rennie · Post by **Nigel Rennie** » 07 Feb 2010 16:52

I have got the phones working however I am not entirely happy with how I achieved it.

I turned compression on by default for calls within the default (0) domain, which is where the phones and asterisks box are.

This was the opposite of what I though would work (given that asterisks does not know how to deal with the compressed protocol) however I guess now I am forcing the traffic over a compressor which is doing the thinking for me.

If anyone knows the internals it would be good to know exactly why this worked out for me.