[asterisk-users] [UPDATE] bug(?) bandwidth problem

David fire ddfire at gmail.com
Mon Jan 12 11:01:38 CST 2009


2009/1/12 Jeff LaCoursiere <jeff at jeff.net>

>
> On Mon, 12 Jan 2009, David fire wrote:
>
> > hi again mybe this info is usefull to solve this problem
> >
> > *box1<--->*box2<---->*box3
> >
> > box2 originate 1 call to box1 and to box 3
> > using sip/box1/1
> >
> > extension 1
> > context default
> >
> > exten => 1,1,dial(sip/box3/1)
> >
> > box1 and box3 will exec musiconhold when they answer.
> >
> >
> >
> > 2009/1/12 David fire <ddfire at gmail.com>
> >
> >> hi
> >> i am using asterisk 1.4.22
> >> ubuntu 8.4
> >>
> >> i have two Ethernet one for ssh and other one only for voip calls
> >>
> >> when i start a call using originate in the manager or the cli
> >> in the voip Ethernet i get something like 4Mbits/sec of traffic only  1
> >> G711 call.
> >> if i start the call using a soft phone everything is normal.
> >>
> >> any idea?
> >> thanks
>
> Try using tcpdump or wireshark to look at the traffic you are claiming you
> are seeing.  What is it?  4Mbps is a lot of traffic.
>

i see a lot of checksum error a no more.
again this only happen when i do musiconhold connected to musiconhold()

ok before you ask i am doing this to see how many calls the server can
handle transcoding from gsm to g729
before the CPU go to 100%


>
> >>
> >>
> >> --
> >> (\__/)
> >> (='.'=)This is Bunny. Copy and paste bunny into your
> >> (")_(")signature to help him gain world domination.
> >>
>
> I think it may be bunny's fault.
>
> i too but hi is my master i cant blame him directly
>
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