[asterisk-users] TCP port, VPN and resolving the cutting voice problem

[Steve Totaro]

I wouldn't bother with their hardware.  You can run it on most servers
providing the drivers for the hardware are supported.

Just install it on a box with two NICs and put it between the router and
your LAN, both static IPs, simple

If I were you, I would find out  what kind of DSL modem you have, but if it
is doing NAT, DHCP, and all of that,  you may be able to turn off everything
except for the modem and use Vyatta for everything from NAT, DHCP, QoS,
Squid, Firewall.

In this case, one NIC would have your public IP, I suspect you would get it
via DHCP or worst case, from your ISP, the second NIC is for the LAN, you
can add more NICs for various purposes as well.

I run Asterisk on Vyatta systems and it works great.  No NAT issues with
remote phones, QoS, and whatever else your imagination can come up with.

I also install Webmin and NTOP.

Just be aware that as soon as you activate the firewall, everything is
blocked, so if you are going to use it as a firewall, get as many rules in
place as you can think of.

Thanks,
Steve T

On Thu, Dec 2, 2010 at 3:14 PM, bilal ghayyad <bilmar_gh at yahoo.com> wrote:

> Dear;
>
> I understood that Vyatta is the solution for the QoS, but I am not able to
> know if I can use a Vyatta hardware router to be DSL router and I set my QoS
> in it to resolve the voice problem. Is it possible?
>
> Thanks for the help.
> Regards
> Bilal
>
> ------------
> > > Thanks all for ur participation and kindly advise.
> > >
> > > As I noticed that jitterbuffer could help if the ping
> > does not have request time out but the voice is also cutting
> > .. but in that case, I have to set the jitterbuffer at the
> > IP Phones and Asterisk boxes.
> > >
> > > I have a polycom phone for example, and to set the
> > jitterbuffer there are the following paramters:
> > >
> > > Payload Size
> > > Jitter Buffer Minimum
> > > Jitter Buffer Shrink
> > > Jitter Buffer Maximum
> > >
> > > When it use the minimum, and when it use the Shrink
> > and when it use the maximum?
> > >
> > > If to look at the asterisk (in the SIP or IAX files)
> > then there are a paramters for the jitterbuffer also, but
> > really I am not able to know when to use this and when to
> > use this:
> > >
> > > jenable, jbforce, jbmaxsize, jbresyncthreashold,
> > jbimpl, jblog
> > >
> > > How to use the jbresyncthreashold? In which case?
> > >
> > > Regarding to the QoS, which will be need in case
> > having a packet loose, correct?
> > >
> > > I just need to ask about something:
> > > What I will be able to do if my ISP did not setup the
> > QoS at his side? What kind of settings I can do in my DSL
> > router (in case of Cisco, or in case of Linksys that running
> > linux firmware)?
> > >
> > > From the other side, if I used linux server to set the
> > QoS, so do I have to let all the network elements to pass
> > this linux server (so it will be the default gateway for
> > other elements)?
> > >
> > > Appreciate the kindly help.
> > > Regards
> > > Bilal
> > >
> > >
> >
> > If getting a second circuit is out of the question.
> >
> > 1.  Switch to SIP
> > 2.  Install and Learn Vyatta for QoS (Squid may help
> > you quite a bit
> > as well) as your router (or whatever you prefer)  I
> > use the paid
> > versions of Vyatta but the free edition should be
> > sufficient.
> >
> > I did the same setup over OpenVPN VSAT links in Iraq, 700ms
> > ping
> > times.  I used GSM and some tricks on the Vyatta box.
> >
> > Originally, before I deployed the above, it was a wild west
> > situation
> > like what you have now.  Going from G729 to GSM made a
> > big improvement
> > in conjunction with QoS.
> >
> > My theory on that is that G729 is already a very lossy
> > codec, so any
> > more loss, garbled audio.  GSM is less lossy.
> >
> > Switch from IAX to SIP was another huge improvement, and
> > then finally
> > putting Vyatta and QoS as my router made calls almost
> > crystal clear.
> >
> > There was the obvious lag time but users get used to that
> > and wait a
> > second or two before speaking so they don't talk over each
> > other and
> > the quality was five by five, except for solar flares,
> > sandstorms,
> > rain.  Things beyond my control.
> >
> > Thanks,
> > Steve T
>
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101205/51caa285/attachment.htm