Summary:ASTERISK-01402: Horrible sound quality SIP -> IAX2
Reporter:jesses (jesses)Labels:
Date Opened:2004-04-13 12:44:34Date Closed:2011-06-07 14:04:47
Versions:Frequency of
Description:I have the latest CVS head, a SIP phone, and an IAX2 phone, both using ulaw. When the stations call eachother through asterisk (i.e. dialing outbound, then inbound with DID) the sound quality is perfect. However when either station calls the other, the sound going from IAX -> SIP is perfect, but the sound going in the other direction has a continual jitter/garble, almost like every other split second of voice is missing.
Comments:By: Brian West (bkw918) 2004-04-13 13:03:00

Seems to be an issue with FireFly's ulaw codec.

By: zoa (zoa) 2004-04-14 07:03:31

i'm not sure about this being a problem with the ulaw codec, it might be the same timestamp problems the cisco's don't like.
(and xlite doesnt like).

Jesses, can you tell us if you also have this problem with the -stable branch ?
If its also there with the stable  branch, its a problem with firefly, if its not there with the stable branch, its a timestamp problem in -head.

By: jesses (jesses) 2004-04-14 08:24:43

I haven't tried it with stable yet but changing the iax client codec to GSM fixes the problem for whatever reason, which is why we're assuming its a problem with the client codec...

By: zoa (zoa) 2004-04-14 08:42:25

aha !
damn, bkw will be right again :)
Well try with the stable anyway, firefly is supposed to be pretty compatible as the writers also use asterisk.

By: Brian West (bkw918) 2004-04-14 12:13:38

Never fear bkw is here!@!!!  NEXT!!!