[Asterisk-Users] GXP-2000 Audio Quality

[Daniel Salama]

Wow! 22Kbps of overhead? Are you sure? That sounds like way too much  
overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any  
other suggestion?

Thanks,
Daniel

On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote:

> G729 uses 8kbps but with the IP overhead it actually uses 30kbps so  
> for
> 256k upstream you should be able to handle 8 calls but this is in  
> ideal
> conditions.
>
> If you were to use IAX and enable trunking then you would use  
> 30kbps for
> the 1st call and 10kbps for each additional call.
> See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth 
> +iax2
>
> On Wed, 2006-06-14 at 04:17, Daniel Salama wrote:
>> 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
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