Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-09 Thread John Timms
Thanks for suggestions, everyone- I should have thought about jitter and
latency as I began to use up more  more bandwidth. I was concerned that it
was a problem with my configuration of Asterisk, but it looks like is really
is a bandwidth issue. By the way, Joe- I've been in another situation with
my cableco  Asterisk/VoIP (on a business connection!) and would frequently
have trouble getting *one* call that sounded good, even though we had
several megabits up  down, with no other traffic on the network. Charter's
service is horrible- there were several times pinging Google took over 1
second.

John Timms


On Sat, Nov 7, 2009 at 2:45 PM, John Timms johngti...@gmail.com wrote:

 Hi. I'm having trouble figuring out why I'm not able to make many
 concurrent VoIP calls on my system. I'm not aiming for a huge number,
 because I have purposely bought a low powered system, but I would
 think that I could get more. Here are the details:

 I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
 (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu
 Server 9.04 with the default Debian package manager installation of
 Asterisk. (version 1.4)

 Here is what is going on: I'm making outgoing calls (with .call files)
 via SIP (using Vitelity's service, if anyone wants to know) with about
 55.0 ms latency between my Bellsouth DSL connection  their servers.
 I'm using GSM-format prompts with GSM encoding (disallow=all,
 allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls.
 I have a very fast internet connection, so there is still plenty of
 bandwidth, and the top command shows that Asterisk is only at about
 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will
 skip occcasionally, but cell phones have perfect quality.

 I don't think that 7 calls is very many, I'll be happy if I can get 10
 good-sounding calls. Can anyone give suggestions? (If this has been
 hashed out elsewhere, I'm happy with a link to more information!)

 Thanks.

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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-08 Thread Todd Reese

Hi Joe,

What is the app that generates your bandwidth table shown below?

Joe Greco wrote:

By fast I mean the best Business DSL Bellsouth has to offer: Up to
6.0 Mbps downstream - Up to 512 Kbps upstream



That almost sounds like an invitation to check out what business service
your cableco offers.

One thing to be aware of with DSL and cable modems is that there can be
various ill effects as your line gets closer to its rated capacity; do
not expect that you'll get a reliable 512Kbps upstream.  VoIP is sensitive
to loss, latency, and jitter.  You may be able, for example, to only get
384Kbps reliably out of the link (before packet loss/jitter/etc wreck its
suitability for VoIP).  That's a good time to look seriously at a gateway
package like pfSense that can prioritize certain classes of traffic while
also limiting overall bandwidth.

As an example, we noticed on the local business cable offering (2Mbps up)

Shaped  PL  min avg max stddev
2.2M3   6.4 251 557 176
2.1M1   7.8 350 584 134
2.0M3   6.4 271 535 132
1.9M1   7   254 527 131
1.8M0   6   79  339 90
1.75M   0   5.9 14  92  11
1.7M0   5.4 13  77  10
1.65M   0   4.9 11  69  7
1.6M0   5.4 13  55  9
1.5M0   5.3 11  59  7
1.4M0   5   11  57  7
1.3M0   4.9 11  54  6
1.2M0   4.9 11  52  7
1.1M0   4.8 14  53  11

The max starts trending up after 1.6M (helps to graph it) and pretty much
everything goes to hell after 1.75M.

... JG
  


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[asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread John Timms
Hi. I'm having trouble figuring out why I'm not able to make many
concurrent VoIP calls on my system. I'm not aiming for a huge number,
because I have purposely bought a low powered system, but I would
think that I could get more. Here are the details:

I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
(1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu
Server 9.04 with the default Debian package manager installation of
Asterisk. (version 1.4)

Here is what is going on: I'm making outgoing calls (with .call files)
via SIP (using Vitelity's service, if anyone wants to know) with about
55.0 ms latency between my Bellsouth DSL connection  their servers.
I'm using GSM-format prompts with GSM encoding (disallow=all,
allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls.
I have a very fast internet connection, so there is still plenty of
bandwidth, and the top command shows that Asterisk is only at about
5% CPU and 10% RAM. Even with only 7 calls, a landline phone will
skip occcasionally, but cell phones have perfect quality.

I don't think that 7 calls is very many, I'll be happy if I can get 10
good-sounding calls. Can anyone give suggestions? (If this has been
hashed out elsewhere, I'm happy with a link to more information!)

Thanks.

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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread Fred Posner
On Sat, Nov 7, 2009 at 2:45 PM, John Timms johngti...@gmail.com wrote:
 I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
 (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu
 Server 9.04 with the default Debian package manager installation of
 Asterisk. (version 1.4)

What kind of NIC are you using and what's the network config? ie
Bellsouth - router - switch - you

Are you NAT'd?

Where are your endpoints connected? (locally, outside?)

 I have a very fast internet connection, so there is still plenty of
 bandwidth

what is the specs for fast?

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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread Darrick Hartman
John Timms wrote:
 Hi. I'm having trouble figuring out why I'm not able to make many
 concurrent VoIP calls on my system. I'm not aiming for a huge number,
 because I have purposely bought a low powered system, but I would
 think that I could get more. Here are the details:
 
 I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
 (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu
 Server 9.04 with the default Debian package manager installation of
 Asterisk. (version 1.4)

Most of my installations are Soekris net5501's with 512MB ram and a 
500mhz Geode LX processor.  Unless Ubuntu is running a ton of extra junk 
in the background, that processor should be more than adequate.

 Here is what is going on: I'm making outgoing calls (with .call files)
 via SIP (using Vitelity's service, if anyone wants to know) with about
 55.0 ms latency between my Bellsouth DSL connection  their servers.
 I'm using GSM-format prompts with GSM encoding (disallow=all,
 allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls.
 I have a very fast internet connection, so there is still plenty of
 bandwidth, and the top command shows that Asterisk is only at about
 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will
 skip occcasionally, but cell phones have perfect quality.

Your only connection to the PSTN is via SIP right?  Then this is likely 
coincidental that 'landline' calls are different than 'cell phone' 
calls.  The ONLY possibility is that the problem is with your SIP 
termination provider, but even that is unlikely.  As Fred pointed out 
your DSL connection is likely the cause.  Do you have any traffic 
shaping on the network?  If not, you really should have a firewall 
that's capable of prioritizing voice traffic over bulk data traffic. 
What is the actual down and up speed of your DSL connection?

 I don't think that 7 calls is very many, I'll be happy if I can get 10
 good-sounding calls. Can anyone give suggestions? (If this has been
 hashed out elsewhere, I'm happy with a link to more information!)

Use this calculator to see how much bandwidth 10 concurrent calls will take.

http://www.asteriskguru.com/tools/bandwidth_calculator.php

Darrick

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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread Tom Moore
If you've got a bellsouth dsl connection because of the way their system
works even with doing qos on the link you can really only do about 8 calls
before you start to run into problems with their setup.

Tom
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms
Sent: Saturday, November 07, 2009 2:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Help with concurrent VoIP calls

Hi. I'm having trouble figuring out why I'm not able to make many concurrent
VoIP calls on my system. I'm not aiming for a huge number, because I have
purposely bought a low powered system, but I would think that I could get
more. Here are the details:

I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
(1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04
with the default Debian package manager installation of Asterisk. (version
1.4)

Here is what is going on: I'm making outgoing calls (with .call files) via
SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms
latency between my Bellsouth DSL connection  their servers.
I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in
sip.conf) and I'm able to make about 7 concurrent calls.
I have a very fast internet connection, so there is still plenty of
bandwidth, and the top command shows that Asterisk is only at about 5% CPU
and 10% RAM. Even with only 7 calls, a landline phone will skip
occcasionally, but cell phones have perfect quality.

I don't think that 7 calls is very many, I'll be happy if I can get 10
good-sounding calls. Can anyone give suggestions? (If this has been hashed
out elsewhere, I'm happy with a link to more information!)

Thanks.

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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread John Timms
Hi Fred.

The NIC chip is a Realtek RTL8101E, on the motherboard. Network is
Bellsouth = modem/router = Asterisk
Yes, I am using NAT (assuming you mean that the Asterisk server does
not have its own public IP address)
Endpoints are outside the network, just standard POTS phones. Vitelity
is my SIP provider.
By fast I mean the best Business DSL Bellsouth has to offer: Up to
6.0 Mbps downstream - Up to 512 Kbps upstream
I've used iftop on my server while running calls, and I'm under 200
Kbps while my calls are running.

John Timms



On Sat, Nov 7, 2009 at 4:25 PM, Fred Posner f...@teamforrest.com wrote:
 On Sat, Nov 7, 2009 at 2:45 PM, John Timms johngti...@gmail.com wrote:
 I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
 (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu
 Server 9.04 with the default Debian package manager installation of
 Asterisk. (version 1.4)

 What kind of NIC are you using and what's the network config? ie
 Bellsouth - router - switch - you

 Are you NAT'd?

 Where are your endpoints connected? (locally, outside?)

 I have a very fast internet connection, so there is still plenty of
 bandwidth

 what is the specs for fast?


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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread Fred Posner
On Sat, Nov 7, 2009 at 4:45 PM, John Timms johngti...@gmail.com wrote:
 Hi Fred.

 By fast I mean the best Business DSL Bellsouth has to offer: Up to
 6.0 Mbps downstream - Up to 512 Kbps upstream

If you're running the GSM codec,  7 calls will hit around 200 Kbps. If
you're running ulaw, 7 calls will hit over your max. The calculator
Darrick linked to is a great tool.

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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread Joe Greco
 By fast I mean the best Business DSL Bellsouth has to offer: Up to
 6.0 Mbps downstream - Up to 512 Kbps upstream

That almost sounds like an invitation to check out what business service
your cableco offers.

One thing to be aware of with DSL and cable modems is that there can be
various ill effects as your line gets closer to its rated capacity; do
not expect that you'll get a reliable 512Kbps upstream.  VoIP is sensitive
to loss, latency, and jitter.  You may be able, for example, to only get
384Kbps reliably out of the link (before packet loss/jitter/etc wreck its
suitability for VoIP).  That's a good time to look seriously at a gateway
package like pfSense that can prioritize certain classes of traffic while
also limiting overall bandwidth.

As an example, we noticed on the local business cable offering (2Mbps up)

Shaped  PL  min avg max stddev
2.2M3   6.4 251 557 176
2.1M1   7.8 350 584 134
2.0M3   6.4 271 535 132
1.9M1   7   254 527 131
1.8M0   6   79  339 90
1.75M   0   5.9 14  92  11
1.7M0   5.4 13  77  10
1.65M   0   4.9 11  69  7
1.6M0   5.4 13  55  9
1.5M0   5.3 11  59  7
1.4M0   5   11  57  7
1.3M0   4.9 11  54  6
1.2M0   4.9 11  52  7
1.1M0   4.8 14  53  11

The max starts trending up after 1.6M (helps to graph it) and pretty much
everything goes to hell after 1.75M.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.

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Re: [asterisk-users] Help with concurrent VoIP calls

2009-11-07 Thread covici
How are you connecting your land line phones since this is where you
have the problem?  Also, I would not expect very many calls at the same
time with that setup if each call takes 50K you can't get exactly the
maximum anyway, maybe 80% of maximum.

Hope this helps.

Tom Moore tommym2...@gmail.com wrote:

 If you've got a bellsouth dsl connection because of the way their system
 works even with doing qos on the link you can really only do about 8 calls
 before you start to run into problems with their setup.
 
 Tom
  
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms
 Sent: Saturday, November 07, 2009 2:45 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Help with concurrent VoIP calls
 
 Hi. I'm having trouble figuring out why I'm not able to make many concurrent
 VoIP calls on my system. I'm not aiming for a huge number, because I have
 purposely bought a low powered system, but I would think that I could get
 more. Here are the details:
 
 I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
 (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04
 with the default Debian package manager installation of Asterisk. (version
 1.4)
 
 Here is what is going on: I'm making outgoing calls (with .call files) via
 SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms
 latency between my Bellsouth DSL connection  their servers.
 I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in
 sip.conf) and I'm able to make about 7 concurrent calls.
 I have a very fast internet connection, so there is still plenty of
 bandwidth, and the top command shows that Asterisk is only at about 5% CPU
 and 10% RAM. Even with only 7 calls, a landline phone will skip
 occcasionally, but cell phones have perfect quality.
 
 I don't think that 7 calls is very many, I'll be happy if I can get 10
 good-sounding calls. Can anyone give suggestions? (If this has been hashed
 out elsewhere, I'm happy with a link to more information!)
 
 Thanks.
 
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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