asterisk users - Feb 2006 - one way / irratic voice over iax and g729

Hi All,

We are experiencing a a problem when running calls over IAX with g.729. 
The call flow is as follows:

Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier

The first Asterisk system is running 1.2 and the second is running 1.0. 
When using g726 from the handset all the way thru to Asterisk2(then 729 
for the carrier leg) calls go thru fine, but when using g729, there is 
one way voice whereby the B party cannot hear the A party, however the A 
party can hear the B party  fine. Sometimes there is no audio for the B 
party, other times the B party can hear the A party but it is very 
broken up and stuttery, with only parts of the words coming through. The 
calls also work fine when using g711 from the A party.

Asterisk2 is running a couple of TDM04B's so there is a physical timing 
device on that side and Asterisk1 is running ztdummy on a 2.6 kernel - 
so there is timing on that side also (??)

Have done a fair bit of searching on this one, and as it only happens 
with g729 (both systems have the licensed codecs installed) it is a bit 
of a head scratcher - has anyone else experiencved this? Or does anyone 
have any feedback?

Cheers,

Ben

2006/2/17, Ben Dinnerville <bdinnerv@gmail.com>:
> Hi All,
>
> We are experiencing a a problem when running calls over IAX with g.729.
> The call flow is as follows:
>
> Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier
if you are calling asterisk-to-asterisk, you should try speex
compression. In my tests, speex had the better quality even with low
bandwidth or bandwidth very occuped by other applications (p2p).

--
Alejandro Vargas

The carrier does not support speex, only g729, 723 and 711, so to 
minimise codec coversions etc, and  due to the fact that licensing 723 
is so expensive and 711 is a bit fat on bandwidth (asterisk 1 is 
connecintg over 128k ISDN) we are kind of stuck with g729 (not that it 
has ever proved to be a problem anywhere else)


Alejandro Vargas wrote:
> if you are calling asterisk-to-asterisk, you should try speex
> compression. In my tests, speex had the better quality even with low
> bandwidth or bandwidth very occuped by other applications (p2p).
> 
> --
> Alejandro Vargas

So you have 2 asterisk systems connected, I am doing this for the first
time.  Any tips you can give me besides whats on the wiki?  I am not
sure the best way to set it up, I want to be able to have the 2
locations act as 1 over their internet connection to each other, I was
planning to use vpn...

Mike
 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ben
Dinnerville
Sent: Friday, February 17, 2006 4:03 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] one way / irratic voice over iax and g729

Hi All,

We are experiencing a a problem when running calls over IAX with g.729. 
The call flow is as follows:

Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier

The first Asterisk system is running 1.2 and the second is running 1.0. 
When using g726 from the handset all the way thru to Asterisk2(then 729
for the carrier leg) calls go thru fine, but when using g729, there is
one way voice whereby the B party cannot hear the A party, however the A
party can hear the B party  fine. Sometimes there is no audio for the B
party, other times the B party can hear the A party but it is very
broken up and stuttery, with only parts of the words coming through. The
calls also work fine when using g711 from the A party.

Asterisk2 is running a couple of TDM04B's so there is a physical timing
device on that side and Asterisk1 is running ztdummy on a 2.6 kernel -
so there is timing on that side also (??)

Have done a fair bit of searching on this one, and as it only happens
with g729 (both systems have the licensed codecs installed) it is a bit
of a head scratcher - has anyone else experiencved this? Or does anyone
have any feedback?

Cheers,

Ben

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Hi All,


An update for this problem. The problem seems to lie with the fact that 
1 Asterisk system was a 1.0 release (1.0.10) and the other was a 1.2 
release (1.2.4) - there seems to be some computability issues with the 
IAX implementations for certain codecs here. We setup another server 
with 1.2.4 on it and the A party was also running 1.2.4 and calls will 
now go through without a problem so all is now sorted.


Cheers,

Ben

Ben Dinnerville wrote:
> Hi All,
> 
> We are experiencing a a problem when running calls over IAX with g.729. 
> The call flow is as follows:
> 
> Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier
> 
> The first Asterisk system is running 1.2 and the second is running 1.0. 
> When using g726 from the handset all the way thru to Asterisk2(then 729 
> for the carrier leg) calls go thru fine, but when using g729, there is 
> one way voice whereby the B party cannot hear the A party, however the A 
> party can hear the B party  fine. Sometimes there is no audio for the B 
> party, other times the B party can hear the A party but it is very 
> broken up and stuttery, with only parts of the words coming through. The 
> calls also work fine when using g711 from the A party.
> 
> Asterisk2 is running a couple of TDM04B's so there is a physical timing
> device on that side and Asterisk1 is running ztdummy on a 2.6 kernel - 
> so there is timing on that side also (??)
> 
> Have done a fair bit of searching on this one, and as it only happens 
> with g729 (both systems have the licensed codecs installed) it is a bit 
> of a head scratcher - has anyone else experiencved this? Or does anyone 
> have any feedback?
> 
> Cheers,
> 
> Ben
> 
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