[Asterisk-Users] GXP-2000 Audio Quality

Wed Jun 14 11:47:40 MST 2006

That may not be such a bad idea. I've read people trying to put  
Asterisk on a WRTG54 or something like that. Would that be good? I  
guess I could do SIP in the office and trunk via IAX2 and save on  
bandwidth plus internal calls would be local.

I tried to upgrade them to 512K but because they're borderline to the  
18K feet, the best BellSouth can offer them is 256K. I'm talking to  
Comcast to see if they can get their broadband service which can go  
up to 768K.

Thanks,
Daniel

On Jun 14, 2006, at 12:45 PM, Tim Panton wrote:

> Well, with 16 phones, it might be worth putting a
> 'satellite' asterisk in their office, have it handle local
> transfers, and act as a protocol converter, talking sip to the
> phones and (trunked) IAX2 to the outside world.
>
> An embedded low power system would do fine.
>
> You might even get away with an nslu2, but I'm not sure
> it has the RAM for 16 calls.
>
> A better alternative is to get them to upgrade the DSL to 512 uplink.
>
> Tim.
>
> On 14 Jun 2006, at 17:11, 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
>>>> _______________________________________________
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation provided by Easynews.com --
>>>
>>> Asterisk-Users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Tim Panton
> tim at mexuar.com
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users