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

Mon Dec 6 20:21:50 UTC 2010

What you probably have is a DSL MODEM that can act as a ROUTER but most
likely doesn't have to.

Your device probably has the same capabilities as most modems, the added
features of NAT, DHCP, and whatever else.  Normally you can disable that
additional functionality.  Now you just have a DSL modem.

If you can turn off the "ROUTER" functions on the MODEM then you can use a
Vyatta server to be a "ROUTER" that just so happens to be connected to DSL,
but could just as easily be connected to a gigabit connection.

Have you tried dumping IAX and using SIP?

Have you verified that your bandwidth is saturated?  Have you run NTOP or a
similar tool to see what is eating all the bandwidth?

I would start with the above because you have no idea what the problem is at
this point.

You need to come to a consensus of how many simultaneous calls are going to
be allowed.  You can QoS your VoIP all day long, but if one too many people
get on the phone, everyone suffers.

Once you get that number, you have to do the math as far as bandwidth to
reserve and limit the calls on the Asterisk side.  If this leaves you with
less than enough bandwidth for business activities, you have to get more
bandwidth, it is that simple.

1.  No, I don't think so.  Why do you?  You want voice to be #1 correct?  I
presume your LAN connection is faster than your DSL.  Any modern server can
handle these chores.  You are talking DSL, so I cannot imagine you have much
call volume, setups and tear downs.  Any G729 or codec conversion should be
very light.  If you are using G729 then set the phones to use it as well.
You could probably run World Community Grid and consume all of your cycles
without a hitch (not recommended, I use it for burn in on new machines)

2.  Yes, you could setup a failover but I have servers with years of uptime
and over a year of Asterisk not being restarted 1.0 and 1.2.  Besides
internal communication, would you not lose phone service now if your DSL
"ROUTER" had to be rebooted?  You don't need to activate the firewall if you
feel NAT is adequate protection.  QoS is your goal, the rest is just icing
on the cake.

3.  You are not tagging the packets for the ISP, you are controlling the
rate at which protocols can consume on outbound traffic.  You assign a port
a piece of the pie, you have to let Vyatta know how big the pie is and how
much of a slice each protocol gets.

Inbound is a little trickier, what kind of DSL do you have, inbound may not
be the problem.  If it is, last I knew Vyatta used "Rate-limiting" which
would essentially drop packets from the sender causing them to slow down,
the protocols that you do not limit will not drop packets.
http://en.wikipedia.org/wiki/Rate_limiting

It has been a while since I looked at the latest and greatest or talked to
the dev guys at Vyatta but they were discussing another method on the
inbound side.  Nevertheless, rate-limiting works for VoIP when correctly
applied.

Use google for God's sake.  There are very well done videos and diagrams
that are specific to Asterisk, Vyatta, and all of your questions.

http://www.google.com/search?q=vyatta+asterisk+qos

Thanks,
Steve T

On Sun, Dec 5, 2010 at 1:36 PM, bilal ghayyad <bilmar_gh at yahoo.com> wrote:

> Dear Steve;
>
> I am fully thanks for your advise and kindly help.
>
> I am asking about the ability to use vyatte hardware DSL router because of
> the following reasons:
>
> 1) I am afraid to make Asterisk the gateway for the whole network and this
> might effect on the performance and might cause a big load, u do not think
> so?
>
> 2) If any problem happened regarding to the QoS rules or regarding to the
> firewall or any other thing and they decided to do hardware restart for the
> server (or the PC machine), then the Asterisk will be restarted and that
> will effect on the telephony service at the site?
>
> 3) I am afraid if we applied the QoS and bandwidth divsion at Vyatte, and
> then we route the traffic to the DSL router (which will do the NAT to ISP),
> then all the QoS rules will be ignored (or become not effected)? What do u
> think?
>
> Again, special thanks for the guide and special help.
>
> Regards
> Bilal
> ---------------------
>
> > 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
>
>
>
>
>
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