[asterisk-users] Two Connected Servers Sound Quailty

[Steve Totaro]

Try SIP if at all possible.  I have had mixed results with IAX that SIP
made go away.  If you try SIP, you can at least rule out IAX as the
cause.

 

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
  

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yossi Ben
Hagai
Sent: Saturday, April 28, 2007 4:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Two Connected Servers Sound Quailty

 

Hi Matt,

 

you didn't mention what type/bw of each site Internet connection, i
suggest that you try to split the scenario into smaller pieces:

- run long term pings between the server while you make a call and check
for packet loss.

- make internal calls between extensions on the same branch and verify
that both servers work okay (eliminating Internet connectivity)

- register a UA from one site to a server on the other site, make a call
and viceversa (eliminating a problem on one of the servers).

- check for speed/duplex setting on NIC and switch port.

- check if the sound quality issues are symmetric (does both sides
experience the sound cut or it only happens on a specific site).

- make sure you don't use G.711 as it consumes bw and from the codec
list you've mentioned has the lowest tolerance to packet loss.

 

Since the problems are intermittent my bet is that someone in the office
is have the p2p client work overtime or sending lots emails with funny
attachments 

 

On 4/28/07, Matt Gardner <garddawg at gmail.com> wrote: 

Ok this is my first post and I will try to keep it short.

 

I have searched everywhere and haven't found an answer to my question

 

I have two Trixbox servers that are connected over the Internet via an
IAX2 connection.  We are experiencing very poor sound quality.  I have
tried many different codecs gsm, ilbc, g729, g711 and all seem to have
the same problem. (All though g729 seems to work the best but still
isn't reliable)  The problems are intermittent sometimes the sound will
cut out for 3-4 seconds and other times the sound will just be loosing
every other word, and other times it sounds just fine. 

 

Also, we have been using Skype over this same Internet connection and
have very good sound quality with very few lost words.

 

So here are my questions.

 

First, is it a correct assumption to say that because Skype works well
over this connection then I should be able to get asterisk to work over
this connect.  I am hoping that Skype isn't "better" then asterisk in
this area. 

 

If I should be able to get the same sound quality could you point me in
the right direction on how to achieve this.  (I have tried messing with
the jitterbuffer but haven't been able to find very good docs on how to
utilize this functionality so about all I have done is set
jitterbuffer=yes) 

 

Thanks in advance.


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