asterisk users - Sep 2004 - Maximum tollerable lag/jitter for IAX2 w/o jitterbuffer enabled?

I'm having a problem with intersite calls over IAX2 being abruptly
terminated. Nothing odd shows in any of the logs for Asterisk or the host.
The only think I can think it might be is a lag-spike on the site to site
connection. 

 How sensitive is IAX2 to lost frames, lag spikes or large variations in
jitter with the GSM codec and:

 bandwidth=low
 jitterbuffer=no
 trunkfreq=100		; Raised from 20
 tos=lowdelay

 notransfer=no
 trunk=no

All calls are running as GSM, even though g.729 is also an 'allowed'
codec
(w/5 licenses installed). During an average call 'iax2 show channels'
provides:

Peer             Username    ID (Lo/Rem)  Seq (Tx/Rx)  Lag      Jitter
JitBuf  Format
10.0.40.140      astpbx-woo  00002/00002  00005/00006  00040ms  0036ms
0000ms  GSM

Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District

----- Original Message -----
From: "Kris Boutilier" <Kris.Boutilier@scrd.bc.ca>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Tuesday, September 07, 2004 4:26 PM
Subject: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2
w/ojitterbuffer enabled?

> I'm having a problem with intersite calls over IAX2 being abruptly
> terminated. Nothing odd shows in any of the logs for Asterisk or the host.
> The only think I can think it might be is a lag-spike on the site to site
> connection.
>
>  How sensitive is IAX2 to lost frames, lag spikes or large variations in
> jitter with the GSM codec and:
>
>  bandwidth=low
>  jitterbuffer=no
>  trunkfreq=100 ; Raised from 20
>  tos=lowdelay
>
>  notransfer=no
>  trunk=no
>
> All calls are running as GSM, even though g.729 is also an
'allowed' codec
> (w/5 licenses installed). During an average call 'iax2 show
channels'
> provides:
>
> Peer             Username    ID (Lo/Rem)  Seq (Tx/Rx)  Lag      Jitter
> JitBuf  Format
> 10.0.40.140      astpbx-woo  00002/00002  00005/00006  00040ms  0036ms
> 0000ms  GSM
>
If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP
doesn't just disconnect sockets unless it recieves a RESET or a FINISHED or
there's a timeout (usually like 5 minutes or more depending on your TCP/IP
stack). Needless to say that to disconnect a TCP connection, that would have
to be one hell of a lag spike... * must be actively disconnecting the
connection....

I've heard the jitter buffer is a bit buggy, have you tried turning it off
completely?

Hell even SIP won't just disconnect you unless your UA tells it to do so...

    -Chris

Kris Boutilier wrote:
>  I'm having a problem with intersite calls over IAX2 being abruptly
> terminated. Nothing odd shows in any of the logs for Asterisk or the host.
> The only think I can think it might be is a lag-spike on the site to site
> connection. 
> 
Has this happened before when making calls over IAX2, or did this just 
crop up? I once had a problem where internal calls over IAX2 was fine, 
but when calling the outside world to another * box it would cut me off 
after ten seconds everytime.

Fixed it by setting "notransfer=yes" in iax.conf. But as the Wiki says
this directly affects CDR and billing information.

It sounds too probable a solution, so most likely may not be the same 
problem you're facing. My 2 cents.

Flynn

On Tue, 7 Sep 2004 16:26:24 -0700, Kris Boutilier
<kris.boutilier@scrd.bc.ca> wrote:
>  I'm having a problem with intersite calls over IAX2 being abruptly
> terminated. Nothing odd shows in any of the logs for Asterisk or the host.
> The only think I can think it might be is a lag-spike on the site to site
> connection.
When does the cut off occurr? Is it always after about 8-10 seconds?
If so, you may have a problem with IAX transfer. You can verify this
by using notransfer=yes.

>  How sensitive is IAX2 to lost frames, lag spikes or large variations in
> jitter with the GSM codec <snip>
> 
> During an average call 'iax2 show channels'
> provides:
> 
> Peer             Username    ID (Lo/Rem)  Seq (Tx/Rx)  Lag      Jitter
> JitBuf  Format
> 10.0.40.140      astpbx-woo  00002/00002  00005/00006  00040ms  0036ms
> 0000ms  GSM
Those values are certainly no problem for IAX at all. I have made hour
long IAX calls with both lag and jitter often going well above 1
second and the calls never terminated. All you get is a heavy delay on
the audio and occasional drop outs, but you shouldn't get cut off.

Even if the lag goes above 2 seconds and you have qualify=yes, the
calls will not normally be cut off when Asterisk reports "peer now to
lagged". As long as the lag will go back below 2 seconds within a
reasonable time frame the connection will recover. IAX is extremely
robust, it is rare to have a connection terminate due to network
problems.

rgds
benjk
-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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