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
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