[Asterisk-Users] GXP-2000 Audio Quality

[Steve Underwood]

Welcome to the wonderful world of VoIP, where people are eager to move 
from 8kbps G.729 to 6.3kbps G.723.1, and accept a substantial drop in 
voice quality, and then throw over 20kbps of RTP, IP and related 
overhead on top of them. Isn't IP wonderful? :-)

Regards,
Steve

Daniel Salama wrote:

> 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
>>