[Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Denis Galvão - iSolve]

Hi Dan, Steve, Michael, Bruno and others.

I will try to describe my VoIP environment below:

SERVER:
- FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17
- iax.conf
[general]
bindport = 4569
bindaddr = 0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
tos=lowdelay
jitterbuffer=no
dropcount=2
maxjitterbuffer=100
maxexccessbuffer=100
mailboxdetail=yes

[1001]
callerid="Ramal 1001" <1001>
context=from-internal
host=dynamic
mailbox=1001
notransfer=yes
port=4569
secret=****
type=friend
username=1001

[1002]
callerid="Ramal 1002" <1002>
context=from-internal
host=dynamic
mailbox=1002
notransfer=yes
port=4569
secret=****
type=friend
username=1002

CLIENT 1001:
- Windows XP
- DIAX 0.9.9g
- Firefly 1.9.6 Build 3944
- USB Phone NTP200E - Compatible with ATCOM USB Phone
- AMD 1.8Ghz with 256Mb

CLIENT 1002:
- Windows XP
- DIAX 0.9.9g
- Firefly 1.9.6 Build 3944
- USB Phone NTP200E - Compatible with ATCOM USB Phone
- AMD 1.66Ghz with 256Mb


ADDITIONAL INFORMATION
- All machines are in the same network(192.168.*.*) no firewall in the 
middle;
- With Firefly I have a VERY GOOD conversation, without any delay;
- With DIAX I have a one way delay of 10 sec. Only the person who recieve 
the call get the delay, the person who make the call listen without 
problems;
- Firefly in one side and DIAX in the other side, same delay problem;
- No problems with SIP;
- No problems(delay) with Linux clients runnig IaxComm 0.99pre11;
- Same problem with DIAX oldest DLL;
- Ping from clients to server: 0% packet loss and < 1ms;
- No problems calling PSTN, Voicemail, etc, just between DIAX clients;

If you need something else, let me know!

Thanks for your help!

Denis Galvão.



Em Dom 16 Jan 2005 19:52, Steve Kann escreveu:
> On Jan 16, 2005, at 2:53 PM, Dan wrote:
> > Hi Steve,
> >
> > ----- Original Message ----- From: "Steve Kann" <stevek at stevek.com>
> >
> >> On Jan 14, 2005, at 2:03 PM, Dan wrote:
> >>> Hi,
> >>>
> >>> \> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> >>>>> > I dont have problems when calling PSTN extensions, and calling
> >>>>> > VoceMail, EchoTest, etc. The problem is related with the
> >>>>>
> >>>>> conversation
> >>>>>
> >>>>> > between two DIAX Softphones.
> >>>>>
> >>>>> Between 2 DIAX phone and the delay is in one direction only??
> >>>>
> >>>> Yes. One direction only... Just who make the call get the delay.
> >>>
> >>> Then try
> >>> jitterbuffer=no
> >>> in iax.conf
> >>> to see if it solves this issue.
> >>
> >> Dan et. al,
> >> I think this might be a problem with native transfers, and needing to
> >> reset the jitterbuffer history when this happens, or something like
> >> this..
> >> -SteveK
> >
> > But I have tried and I do don't have this problem here...
> > What can I do to make this happen here?
>
> I don't know...
>
> Maybe if we could get a packet trace of the situation that causes the
> problem?
>
> Maybe try notransfer or whatever the iax.conf parameter is, and see if
> that changes things.  If it does, it points towards this being the
> problem.
>
> If the delay goes down after a couple of minutes after the transfer,
> this could be the problem.  If it doesn't, there's something else
> really wrong..
>
> (I'm assuming you're using the new JB code here..).  Also, if you're
> using the new JB code, you should implement the stuff to get the
> network stats, so we can see if calculated jitter is substantially
> higher..)
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
D e n i s   G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 252-2977
http://www.isolve.com.br