[Asterisk-Users] Polycom 600 problem

[Adam Goryachev]

On Thu, 2004-12-23 at 11:08, Andrei (MPI) wrote:
> Hi there,
> 
> We are using 10+ Polycom SP IP 600 phones with Asterisk and TMD400P with 
> 4 FXO lines.
> 
> So far we have 3 phones with following problem: more or less frequently, 
> for every call or ever other call, user of the phone would hear brief 
> interruptions on the line when dialing out via PSTN, like if someone 
> would be disconnecting and reconnecting the external line a few times 
> per second. At the same time Asterisk Voicemail, Music on hold, SIP to 
> SIP calls would work just perfect. Loud and clear.

I had a similar problem, which was solved by forcing the phones to use
ALAW instead of ULAW. (I presume this was because the calls arrived in
ALAW from the PRI, and so needed to be transcoded to ULAW, so removing
this transcoding solved the problem, in my case).

Of course, I never figured transcoding from ULAW to ALAW would be that
much effort, but I guess it was....

Here is my 'show translation' output:
aster0x*CLI> show translation
         Translation times between formats (in milliseconds)
          Source Format (Rows) Destination Format(Columns)

        g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
   g723    -   -    -    -    -     -    -     -    -     -    -
    gsm    -   -    2    2    3     2    1     5    -    31   18
   ulaw    -   3    -    1    3     2    1     5    -    31   18
   alaw    -   3    1    -    3     2    1     5    -    31   18
   g726    -   4    3    3    -     3    2     6    -    32   19
  adpcm    -   3    2    2    3     -    1     5    -    31   18
   slin    -   2    1    1    2     1    -     4    -    30   17
  lpc10    -   4    3    3    4     3    2     -    -    32   19
   g729    -   -    -    -    -     -    -     -    -     -    -
  speex    -   3    2    2    3     2    1     5    -     -   18
   ilbc    -   4    3    3    4     3    2     6    -    32    -

If I read that right, it should take 1 ms to transcode 1 second worth of
audio data? Or is it 1 ms to transcode 20ms (one packet) of audio??
Either way, it should not have caused any interruptions/etc...

Hope this helps.

Regards,
Adam