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

[bilal ghayyad]

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)?