[asterisk-users] Tired of dropouts and garbled phone, calls - where to go next?

[Dave Platt]

> In my case, I have good incoming quality and terrible quality going out.
> That is, I can hear people perfectly well but they complain that my 
> voice drops out and is garbled regardless of who places the call.

This suggests to me that you may have congestion problems in your
"upstream" traffic flow.

Setting QoS on the packets may not help, if whatever router you are
using for Internet connectivity isn't managing the traffic flow well.
In my experience, you need to do two things:

-  Make sure that the traffic with QoS for low latency, is placed
   in a separate transmission queue on the router from "bulk" traffic
   (e.g. web service, file transfer).

-  Make sure that your router uses a "traffic shaping" system, to
   ensure that data isn't being submitted to the network interface
   faster than it can actually be transmitted by the *slowest*
   link in the path to your Internet provider.

A lot of routers, switches, and network interface drivers these days
have a lot of buffering.  This is a mixed blessing.  Unless you are
careful, a burst of low-priority (bulk) traffic can be transmitted
into your switch/router, and fill up a bunch of the buffer... by the
time the system "knows" that you have some audio-QoS traffic to send,
there's a whole bunch of data ahead of it in some network router or
switch (or even the ring-buffer in a network interface card) and there's
no way for your audio data to "jump ahead" of the bulk data in order to
be delivered quickly.

Ideally, your system should send data upstream at a rate which never
"bursts" up to, or above your link's sustained data transmission rate.
You want the buffers in the "upstream" equipment to remain as empty
as possible.

For background reading on this, look up the "Bufferbloat" problem
and project, and the "Linux ultimate traffic shaper" scripts.
They may not be directly applicable to your problem but may
explain some of what you are seeing.