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

[Steve Totaro]

On Thu, Dec 2, 2010 at 4:15 AM, bilal ghayyad <bilmar_gh at yahoo.com> wrote:
> 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