[asterisk-users] Voice "broken" during calls

Antony Stone Antony.Stone at asterisk.open.source.it
Mon Jun 15 14:24:31 CDT 2020

On Monday 15 June 2020 at 18:55:23, Luca Bertoncello wrote:

> Absolutly *no changes* on the behaviour compared with my Thomsons...

Okay, I'm glad we can rule out the specific make / model of phone - that would 
have been bizarre.

> I try to summarize:
> 1) Phones are not the problem, since 3 phones of 2 different
> companies/model have the same issue.

Good (if you see what I mean).

> 2) Asterisk seems not to be the problem, too, since I have the same
> behaviour if I connect to phone directly to the server of Deutsche Telekom.

Is that also via the Banana, or with the phone directly on a DSL modem?

> 3) Traffic shaping seems not to be the problem, too, since I tried to
> deactivate it.

Good test / check.

> 4) The problem happens *only* on active call, not by voicemail.

So, only when there are two SIP clients active on each side of the Asterisk 

> 4a) To test it I read a text and my partner just listen it, and then he
> read a text and I listen it. *No* simulaneously speak!

But, what were the results - each of you could hear the other perfectly well?

This sounds interesting - more ideas below.

> 5) A *single call* (since I couldn't reproduce it anymore), made using
> my Android phone as SIP-client connected to my Asterisk, had not the
> problem. Any other try to call someone using my mobile phone via SIP had
> the problem.

You seem to have the problem in general, so a single (or small number of) 
instances of no problem doesn't mean there isn't something to be resolved.

> I could *not* test connecting to the server of Deutsche Telekom using
> the Internet connection of someone other, since Telekom bounds my
> VoIP-login to my IP.


> I really think, the problem should be by Deutsche Telekom...

Especially since you say you do not get the problem when you have calls in via 
Messagenet for your Italian calls.

> What is your opinion? Do you see some other tests I should try?


I'm intrigued by the "only one party speaking at a time" test you did.

What happens if:

a) you call someone external, speak for about 30 seconds without them making 
any sound, then they start speaking *at the same time as you*, then you stop 
talking and they carry on.

b) exactly the same, except this time they call you, so it's an inbound call.

Do you get good quality while only one person speaks, and bad while both do?  
Does the quality return to good when one person stops speaking?



f u cn rd ths, u cn gt a gd jb n nx prgrmmng

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