[Asterisk-Users] Call quality degradation after time

[Adam Dobrin]

Thanks for the reply, Adam.

If this is the case, it would seem to me (because the degradation 
happens only after a period of time, and quite suddenly) that the issue 
lies with digium's implementation of g729.

As an interesting note, I had the same problems using ulaw -> ulaw over 
the local network (from internal phone to internal phone) with a much 
shorter period of 'good voice' before degradation--which was my reason 
for switching to g729, which seemed to solve the problem.

If there is no other current solution, it would seem to me the best 
thing to do would be to force g729 for non PSTN connections, and ulaw 
(im in the US) for calls going out the PRI.  Has anyone done this..?

I'm still hoping to be able to stick with g729; anyone else experience 
this kind of issue?

-a

Adam Goryachev wrote:

>On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote:
>  
>
>>I'm using Polycom 501's; with stable1.0.8, g729 and a very decent 
>>machine; we have a PRI interface to a T1. 
>>
>>Many users complain that after a given amount of time, say, 30 or 40 
>>minutes on a call, the outside party complains that their sound keeps 
>>'cutting in and out'.  I believe that the incoming sound quality remains 
>>fine.
>>
>>I had read that there may be some soft of memory issue with the 
>>Polycom's... what else could be causing this?  Will auto-rebooting the 
>>phones once every few days fix this problem?
>>    
>>
>
>We had this problem initially, except it happened from the beginning of
>the call. The fix was to tell the phones to prefer alaw and to tell
>asterisk to prefer alaw (and not allow anything else). This meant that
>we were using alaw from PSTN -> asterisk -> SIP phone ie, everywhere. It
>solved the above symptom.
>
>PS, this is in Australia and was with CVS stable 1.0.x from about 12
>months ago.
>
>Regards,
>Adam
>
>
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