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

bilal ghayyad bilmar_gh at yahoo.com
Thu Dec 2 14:14:17 CST 2010


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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