asterisk users - Oct 2005 - Voice Quality bad on one side of Frame Relay

Hi ,

Does anyone encounter this problem ? We have installed Asterisk at Site 
A and have 128k Frame Relay over to Site B.
We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A. 
We are using Ulaw at Site A and G729 at Site B.

When the calls are originated from Site A to Site B, party at Site A can 
hear Site B voice clearly and no breaking up voice. But Site B user 
hears Site A voice is breaking up sometimes. The Total bandwidth usage 
is about 30k.

We have deployed the same setup to another Site , Site C with 64k Frame 
Relay. Same things happen.

Any Comments / Ideas ?

Regards,
Stephen

On Tue, 04 Oct 2005 14:28:47 +0800, Stephen wrote:
>Hi ,
>
>Does anyone encounter this problem ? We have installed Asterisk at Site 
>A and have 128k Frame Relay over to Site B.
>We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A. 
>We are using Ulaw at Site A and G729 at Site B.
>
>When the calls are originated from Site A to Site B, party at Site A can 
>hear Site B voice clearly and no breaking up voice. But Site B user 
>hears Site A voice is breaking up sometimes. The Total bandwidth usage 
>is about 30k.
>
>We have deployed the same setup to another Site , Site C with 64k Frame 
>Relay. Same things happen.
It sounds like you're very bandwidth constrained. A ULAW leg is going
to be 64k, or actually around 80k with IP overhead. Not sure abour FR
overhead. I can't see how you'll get anythin at site C without using a
compressed codec.

G729 is a good choice for codec to get over the bandwidth issue. Why
not use it in all cases? You can always experiment with GSM for no
cost.

Also, is there any other network activity beyond the voip streams? In
such a bandwidth limited instalation QoS management is going to be
critical.

Michael

--
Michael Graves                           mgraves@pixelpower.com
Sr. Product Specialist                          www.pixelpower.com
Pixel Power Inc.                                 mgraves@mstvp.com

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245

I have configured my asterisk to connect to an H323 gateway in order to 
place calls to the PSTN. The calls go through with no problem, but what I 
experience is a loss of received sound after about 5 mins in the call (the 
sound comes in very intermittent), while the other party continues to 
receive the call with no problem (it's a one-way loss). When this happens, I
can see that the pc (using top) CPU utilization goes upto 75% and the 
computer becomes sluggish. I have tried this on another server using a P4 
3GHz with 2GB of RAM, but the problem exists even on that server. Has anyone 
experienced this, or knows where the problem is?

SIP to SIP calls or SIP to IAX does not give such a problem