[asterisk-users] SIP devices with packet loss tolerance

[Eric "ManxPower" Wieling]

Michael Graves wrote:

> Ah, of course you are completely correct. My use of the term QoS was in error and out of context.
> 
> That said, at the remote user end they will most certainly suffer poor voip performance if there is no form of traffic prioritisation. In my home office I rely upon the traffic shaping feature found 
> in m0n0wall to ensure that oubound traffic from my Asterisk server gets priority over general outbound internet activity. Strictly speaking this is not QoS at all since it has nothing to do with 
> packet tagging.

A good jitter buffer on both ends of the call can do amazing things to 
audio quality.  Add in  LAN->WAN outbound traffic priority for VoIP 
traffic and WAN->LAN traffic shaping and you can do a pretty good job of 
  keeping call quality acceptable.  One thing to be concerned about is 
UDP based services like Bittorrent.

On the networks I manage we use Cisco routers with QoS and 
point-to-point T-1s or Fractional T-1s for locations where we do 
VoIPoWAN.  I've not done any LAN QoS yet, as our LANs do not have all 
that much traffic on them and we've not had any VoIPoLAN call quality 
issues.