Re: [Freeswitch-users] XML config file parsing

2009-11-23 Thread Eliot Gable
Or, you can use something like Smarty to cache your generated XML on
your web server and only invalidate those cached results when you
change something that will impact them.

On Mon, Nov 23, 2009 at 11:38 AM, Anthony Minessale
 wrote:
> There is a formula to implement caching but it's very complicated and nobody
> has had time to work on it.
> You have to take every single input variable into account when caching
> because who is calling the extension, why they are calling it when they are
> calling it all make a difference.
>
> Web servers are designed to get thousands of hits per second so typically
> they can handle delivering custom xml instruction quite well.
>
> If you do not require such a dynamic setup, you could generate static files
> instead.
>
>
> On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun  wrote:
>>
>> On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman  wrote:
>> > Hi Sam,
>> > Take a look at mod_xml_curl.  Pretty sure it'll do everything you're
>> > looking
>> > for.
>>
>>
>> Looking at that diagram it seems like mod_xml_curl makes a call for
>> every SIP connection. That seems like overkill.  Is there a way to set
>> it up so that it caches the XML it got for a period of time?
>>
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>
>
> --
> Anthony Minessale II
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] suggestions for hardware.

2009-11-13 Thread Eliot Gable
I have built some low-wattage systems before, and from what I have
seen, you want to stay away from the low-wattage processors. On a
price per performance per watt scale, the lowest wattage Core 2 Duo
processors are the best bet. I have a logic-supply ITX system here at
home that has a VIA processor in it and it is dirt slow. It cost about
$350 - $400. It sucks about 5W max, according to my Watt meter but it
takes forever to do anything. Compiling FreeSWITCH on it is an
absolute nightmare (it takes hours). I have an alternate MicroATX
system with a Core 2 Duo and it pulls about 15W all the time. If I
compile FreeSWITCH on that system, it takes a few minutes. The Core 2
Duo system was bought about 6 months after the VIA system. The Core 2
Duo cost about $270 for everything, including 2 GB of RAM. The amount
of money saved by running the VIA system at 5W is nowhere close to the
inconvenience of waiting for it to do anything. Also, I had to go
through about 5-6 different Linux distributions before I found one
that would actually install on the VIA processor. I think Suse Linux
was the one that finally worked on it. Tried CentOS, Slackware,
Ubuntu, Debian, and a couple of others.

Now, I have not tried the Atom processor, so it could be very
different. However, I have read the Tom's Hardware review that also
showed that the Core 2 Duo was several times better on the price per
performance per watt scale than the Atom processor, and again, it was
mainly because the Atom processor took so much longer to do anything.


On Fri, Nov 6, 2009 at 6:21 PM, Orien Love  wrote:
> First of all, Thanks to the help I received on my pfSense installation,
> especially to Michael.  I have a basic test system up and running. I am
> still waiting on some hardware but the base system is working
>
> I am looking on advice on how to set up a simple office PBX, 20 phones
> and 4 outside lines.with 2 or 3 "operator" phones and the rest will be
> extensions.
>
> Here is my plan, please let me know if it does not make sense, or if I
> am going about it
>
> System Hardware
>  4 spa3000's to handle the outside lines.
>  2-3 polycom 601 phones with expansion modules (Operator phones)
>  18 polycom 330 or other phones for desks.
>  2-24 port cisco POE switches
>  1 pfSense server.
>
> System Design.
>
>  Extension Numbers 2xx
>  Outside line access 1xx
>  groups 3xx
>  auto-attendent ???
>
> here are my questions
>    #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of
> memory handle this proposed system? (Here is the MB I am thing of using
> MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010)
>    #2 how do I pool my spa 3000 FXO lines so that the outgoing calls
> use the first available line? also how do insure that metro (non long
> distance) calls go to a specific line if available?
>
> I have learned a lot on how to set up Polycom 601 phones, I am planning
> on writing a how to document, is there any specific format?
>
> Thanks Orien
>
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] Large number of destinations

2009-11-13 Thread Eliot Gable
Performance is not an issue. I clocked 300 calls per second on such a
setup using a Dell R710 with two XEON X5570s and 32 GB RAM as the
FreeSWITCH server and a Dell 2950 4-core system with 8 GB RAM as the
app server. The app server was at 15% - 20% idle at that rate and the
Dell R710 was 65% - 70% idle. The main bottleneck I ran into was using
the limit application with ODBC. A mutex lock around the ODBC calls
meant that I could only pull 160 calls per second, even though the app
server was 55% - 60% idle at that rate, because the ODBC call took
1/160th of a second to complete and all the requests were serialized.

In theory, you should get better performance using mod_xml_curl
because FreeSWITCH will not have to parse a large XML dial plan. One
of the drawbacks of the XML dial plan is that any time it tries to
locate a route element, it must perform an XML linear search until it
finds the correct child (as can be seen in the source code). Thus,
searching the XML dialplan is O(n) operation while mod_xml_curl is
typically constant time, or at worst, O(log n), depending on how you
are storing / querying your data from your database system. Actually,
I suppose you could just be a bad programmer and end up making it
exponential, but I'm assuming you know how to write code and design
your database in a way that avoids that.

I have been considering writing a hash cache for the XML dialplan so
that lookups can become constant time, but I have no idea when or if I
will find the time to do that. :)

On Fri, Nov 13, 2009 at 5:23 AM, Robin Vleij  wrote:
> On 11/13/09 2:49 AM, Eliot Gable wrote:
>
> Hi Eliot,
>
>> Or, of course, there is always mod_xml_curl. Basically, XML dialplan
>> on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a
>> web application server, web application server responds with XML
>> routing response, FreeSWITCH routes the call.
>
> Yeah, been looking at that one, really cool idea. Then I could build my
> routing database in any way I want. I'm just worried about performance
> and the extra delay it'll introduce. But technically with my complex
> routing demands this would be the right solution, instead of a mix of
> modules (which probably brings the same extra load on the machine).
>
> I'll fiddle a bit. :)
>
> /Robin
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] Large number of destinations

2009-11-12 Thread Eliot Gable
Or, of course, there is always mod_xml_curl. Basically, XML dialplan
on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a
web application server, web application server responds with XML
routing response, FreeSWITCH routes the call.


On Thu, Nov 12, 2009 at 5:53 PM, Rupa Schomaker  wrote:
> On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij  wrote:
>> On 11/12/09 9:59 PM, Rupa Schomaker wrote:
>> If I read it right, this is suited for "complete" nrs. So would I have a
>> system connected with lots of DIDs, I would put them in easyroute. Then
>> for systems with lots of number ranges, I would use mod_lcr.
>
> lcr is based on prefix, so the boundaries for which the range is
> assigned may not match a prefix.  You may be better off either:
>
>
> 1) denormalize your ranges and just insert all distinct #s
>
> 2) Modify mod_easyroute to support ranges
>
> 3) talk to SWK (he is on irc here and there) about his (non free)
> fancier routing options
>
>
> --
> -Rupa
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] SIP provider with extern rtp server

2009-10-30 Thread Eliot Gable
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold  wrote:
> I have already set debug to 9, on both profiles.
>
> Ivan
>
>
> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
>
>> See that 200 OK that keeps coming in over and over and over and over
>> again? That's because they never received your ACK. If you can turn on
>> sofia loglevel to 9 and then watch where you send the ACK, you will
>> probably have your answer to why the other system did not receive it.
>> If you're still not sure what's going on, post another pastebin with
>> sofia loglevel set to 9.
>>
>>
>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold 
>> wrote:
>>> Oh, what happened to it?
>>> Anyway, here is a new pb:
>>> http://pastebin.freeswitch.org/10867
>>> Ivan
>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>>>
>>>
>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold 
>>> wrote:
>>>>
>>>> Here is a debug log from a call from an internal phone out to an
>>>> external (my iPhone with nbr 91316356):
>>>> http://pastebin.freeswitch.org/108578
>>>>
>>>> Ivan
>>>>
>>> Uh... you wanna try that PB number again?
>>> -MC
>>>
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>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> ___
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>
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] pass arguments into javascript

2009-10-28 Thread Eliot Gable
Should work fine. I use this:

var calling_num = argv[0];
var called_num = argv[1];

Are you sure you actually had valid data in $1 and $2? Try to call it
from the CLI:

jsrun test.js testvar1 testvar2



On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis  wrote:
> Hi, new to javascript. I tried to pass two arguments into javascript,
>
> 
>
>
> In test.js, I tried to use argv[1]  to retrieve $1 and argv[2] to retrieve
> $2, however, the javascript test.js complained about argv[] as undefined
> variables. How to retrieve the passing arguments in a javascript same as the
> case above. Thanks,
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] SIP provider with extern rtp server

2009-10-28 Thread Eliot Gable
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold  wrote:
> Oh, what happened to it?
> Anyway, here is a new pb:
> http://pastebin.freeswitch.org/10867
> Ivan
> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>
>
> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold  wrote:
>>
>> Here is a debug log from a call from an internal phone out to an
>> external (my iPhone with nbr 91316356):
>> http://pastebin.freeswitch.org/108578
>>
>> Ivan
>>
> Uh... you wanna try that PB number again?
> -MC
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] SIP provider with extern rtp server

2009-10-27 Thread Eliot Gable
No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia SIP profiles involved.

On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold  wrote:
> The server is on a public IP, so there is no nat issue here.
>
> I can also see the rtp messages on wireshark starting just after the
> 183 Session Progress message on the server, but just in one direction,
> coming in to the server.
> So it looks like Freeswitch is stopping the rtp.
> Is this because the rtp originates from another ip than the  sip
> provider ip?
>
> Ivan
>
> Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:
>
>> Make sure you let their media IPs through your firewall. Also, if you
>> are behind a NAT, check you have things passing to the correct
>> internal address.
>>
>> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold 
>> wrote:
>>> I have used a SIP provider for more than a year. A few days ago, he
>>> said he was moving to a new server, and asked me to reconfigure. I
>>> did, and everything seemed to work fine, until I did an outgoing call
>>> to an external telephone. I found out I had no audio, in neither
>>> direction. Incoming calls was working fine.
>>>
>>> My provider said that the rtp is not going through the sip server, as
>>> it did earlier, but now through several other IP's.
>>>
>>> Do I have to do some special configuration to handle that?
>>>
>>> Ivan
>>>
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>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
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>
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] SIP provider with extern rtp server

2009-10-27 Thread Eliot Gable
Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold  wrote:
> I have used a SIP provider for more than a year. A few days ago, he
> said he was moving to a new server, and asked me to reconfigure. I
> did, and everything seemed to work fine, until I did an outgoing call
> to an external telephone. I found out I had no audio, in neither
> direction. Incoming calls was working fine.
>
> My provider said that the rtp is not going through the sip server, as
> it did earlier, but now through several other IP's.
>
> Do I have to do some special configuration to handle that?
>
> Ivan
>
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-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] how to config FS with two net interface?

2009-10-27 Thread Eliot Gable
Try setting ext-rtp-ip and ext-sip-ip on both profiles.

On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang  wrote:
> Hi all, I run FS on a machine with two net interface, each interface has a
> ip addr, one of the them connect to public network(has ip addr A), the
> other  connect to a private network(has ip addr B), FS server as a SIP
> server for public through A, all outbound call will bridge to a softswitch
> in private network through B. here is my sofia config file and diaplan
> config:
>
> sofia internal.xml
> 
> 
> 
>  
>
> sofia external.xml
> 
> 
> 
> 
>
> dialplan
> ..
> 
>     
>     
>      data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
>      data="effective_caller_id_number=xxx"/>  
>          data="sofia/external/${destination_numb...@x"/>
>   
>     
> .
>
> then call seq is
> sipAgent --> [internal -->(bridge)-->external] -->softswith
>   FREESWITCH
>
> the question is, when sipAgent make a outbound call, FS can't recevie the
> caller's up audio stream, I traced the SIP packets, found that FS has return
> addr B in SDP when ack the invite request from sipAgent, the ack packet is
> ===
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> From: "1000" ;tag=cb4d3c4e
> To: "65960581" ;tag=DtvSc0QX01yKN
> Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> CSeq: 2 INVITE
> Contact: 
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
> REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 245
>
> v=0
> o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;>>>>wrong this is the ip addr
> of the adapter connect to the private network
> s=FreeSWITCH
> c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the
> private network
> t=0 0
> m=audio 31066 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> 
> I think FS should return A in SDP, not the external binding addr (B), does
> somebody known how to solve this problem?
>
> --
> Lei.Tang
> lei.tl...@gmail.com
>
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>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] Problem with inbound call answered but no sound

2009-10-26 Thread Eliot Gable
FYI, it generally makes debugging easier if you do this:

sofia profile external siptrace on
sofia profile internal siptrace on

That way you can see the actual signaling and it is usually more clear
what is going on. In most cases, you will probably be able to figure
it out yourself just looking at the signaling.


 On Mon, Oct 26, 2009 at 10:29 PM, Lars Zeb  wrote:
> I have tried to update (make current) twice since 15183. All inbound calls
> are picked up but the caller hears nothing but a couple of clicks. The most
> recent version I’ve tried is 15241.
>
>
>
> Any ideas on what may be causing this?
>
>
>
> http://pastebin.freeswitch.org/10843
>
>
>
> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
> i386 GNU/Linux
>
>
>
> Thanks Lars
>
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>

-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] Estimating Call Capacity

2009-10-26 Thread Eliot Gable
Although, FYI, I just benchmarked mod_xml_curl on a separate web app
server from FS with FS on a Dell R710 with their current best
processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
GB memory. The web app server is less than half the power of the R710.
I maxed the web app server at 300 calls per second (both setting up
and tearing down) and the R710 running FS was 65% idle. No audio was
being proxied through FS, though. If I were running the web app server
on an equivalent R710, they probably would have been on-par with each
other in performance. Extrapolating, I expect that in such a case I
should be able to get at least 650 CPS out of FS, though for
production I would probably limit it to 400 CPS or less so I leave
room for miscellaneous tasks. I maxed out the R710 at over 16,000
simultaneous calls (again, no audio proxying) but the only reason I
couldn't do more was because I hit some sort of thread creation limit
in Linux. There was about 17 GB of memory used for this many calls.
This should give you some ballpark idea of what you can accomplish
with FS.

At some point, I will track down and resolve the thread creation
issue, at which time I believe call limits will be limited either by a
complex combination of available memory, the speed of the processor,
the cost of thread context switching, calls per second setup rate, and
call duration.

--
Eliot Gable

> -Original Message-
>
> From: freeswitch-users-boun...@lists.freeswitch.org 
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni 
> Maruzzelli
>
> Sent: Monday, October 26, 2009 4:56 PM
>
> To: freeswitch-users@lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] Estimating Call Capacity
>
>
>
> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
>
>  wrote:
>
> > Here are a few benchmarks that I had stumbled upon.
>
> > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
>
>
>
> Please remember NO benchmarks are endorsed by the FS community or
>
> developers, because there are just too many variables, and a simple
>
> figure is just useful for marketing hype, not for real dimensioning.
>
>
>
> You MUST do your own benchmarking, so you get an idea about how to
>
> dimension for your own use case and hardware.
>
>
>
>
>
> > Thanks,
>
> > Vinuth.
>
> >
>
> > On Tue, Oct 27, 2009 at 1:43 AM, Brian West  wrote:
>
> >>
>
> >> I highly doubt it... You can wait for someone to post their results
>
> >> but in the end you'll have to do your own load testing because not
>
> >> everyone's numbers will jive with your use case.  Which is the reason
>
> >> the project never posts or endorses a set call count.
>
> >>
>
> >> /b
>
> >>
>
> >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>
> >>
>
> >> > Are there any benchmarking test results available publicly?
>
> >> > 
>
> >>
>
> >>
>
> >> ___
>
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>
> >
>
> >
>
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>
> > FreeSWITCH-users mailing list
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>
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>
> >
>
> >
>
>
>
>
>
>
>
> --
>
> Sincerely,
>
>
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
>
>
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