[Asterisk-Users] GXP-2000 Audio Quality

[Daniel Salama]

I have a client with about 16 GXP-2000. They complain that the audio  
quality is terrible after 2 or 3 simultaneous conversations. They are  
behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u  
codec, I know they upstream bandwidth is the limiting factor and they  
most likely won't be able to have more than 3 simultaneous  
conversations, and if they're surfing the net and/or checking email,  
things will only get worse.

So, I purchased some g729 codec licenses and forced their sip peer  
configuration to g729 codec. We made sample test calls and were able  
to make 8 simultaneous calls. On the eighth call, the audio started  
to sound choppy. Then we dropped the eighth call and tested with 7.  
We could hear just fine on the GXP-2000 but the remote end heard us a  
bit choppy and/or with a robot-like voice. So, we kept dropping calls  
until they were of acceptable quality.

My question is, if they were using g729 which, in theory uses 8kbps  
plus overhead, they should have been just fine handling eight calls.  
All the computers were turned off on the network, so there shouldn't  
have been any other traffic but VoIP. Does anyone have any ideas?

How can I improve their audio quality? I requested BellSouth to  
upgrade their capacity but because of where they are located, the  
best they can get is 900Kbps/256Kbps, so the upstream continues to be  
the limiting factor.

I purchased a Dlink-1226G switch to allow me to control QoS on the  
LAN. I also upgraded their Netopia DSL router to the latest firmware  
which allows me to configure VLANs and DiffServ. All the computers  
are connected to the PC port on the phone because there is no  
available second wiring. Can anyone suggest how to configure the QoS  
settings on the phones, the Dlink and the Netopia?

While there was "no traffic" on the wire, pinging from/to the  
Asterisk box gave me about 47ms latency. When we went passed the 4th  
call, the latency started increasing significantly and when we got to  
8 calls, the latency was up in the 2000ms. Obviously, if anything I  
did in the QoS configuration gave VoIP a priority, then ICMP packets  
would have the lowest priority and I could understand that to be the  
reason for such result. However, I'm not sure I configured QoS  
properly and that's why I'm asking for help.

Thanks,
Daniel