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Reusing HTTP connections in client-server applications

Sat, 11/16/2013 - 02:16

I’m working on a clientserver application which uses HTTP as a transport protocol for API requests, and sometimes there are occasions with a need to execute a few hundreds requests, such as data import or synchronization.

With default Apache HTTP server settings and default LWP::UserAgent options, every new request would result in a new HTTP session, and each time a DNS query is sent out. So, a synchronization process with a thousand object floods the DNS service with the same requests for the HTTP server name. This results in delays, and some public DNS servers apply rate limits which cause DNS lookup failures (had this with a domain hosted at Godaddy name servers).

HTTP 1.1 protocol supports reusing of persistent connections, but it’s not enabled by default in Apache and in the client.

In Apache HTTP server, the following options need to be configured:

  KeepAlive On   MaxKeepAliveRequests 500

In the Perl client program, LWP::UserAgent needs the keep-alive option:

  my $ua = LWP::UserAgent->new(keep_alive => 1);

With these modifications, the DNS queries are only sent on every 500th API request, and the HTTP connection is reused between the requests, which saves CPU time on the server. This speeds up the whole process significantly, and also prevents the DNS failures caused by rate limiting.


Filed under: Networking, Programming Tagged: network api, perl

Minimal FreeSWITCH configuration

Wed, 11/13/2013 - 14:15

It’s always a bit of an effort to remove unneeded features from the default FreeSWITCH configuration. So, I made the minimal configuration which still allows to start the server, but does completely nothing. It’s now much easier to start a new server configuration for any new project.

The configuration is placed at Github. It’s very straightforward to use with FreeSWITCH Debian packages, and can also be used if you compile it from sources:

cd /etc git clone https://github.com/xlab1/freeswitch_conf_minimal.git freeswitch

The configuration contains a number of empty “stub.xml” files in order to make the XML pre-processor happy.

It also makes sense to start using Git for your own FreeSWITCH configurations :)


Filed under: Networking Tagged: freeswitch, pbx, sip, voip, xlab1

FreeSWITCH performance on Intel Atom CPU

Sun, 10/13/2013 - 00:22

There are multiple low-power, fanless  appliances on the market, and most of them are powered by Intel Atom processors. I needed an estimation how well an Atom would perform for a FreeSWITCH PBX application.

In this test, I use two Acer Aspire One notebooks with different processors:

  • atom01: Atom N2600 (2 cores, 4 virtual CPUs, 512KB cache and 600MHz per virtual CPU, 12768.02 BogoMIPS)
  • atom02: Atom N570 (2 cores, 4 virtual CPUs, 512KB cache and 1000MHz per virtual CPU, 13302.08 BogoMIPS)

Both notebooks are running 32-bit Debian 7 Wheezy (Kernel version 3.2.0-4-686-pae), and FreeSWITCH version 1.2.13 from pre-built Debian packages.

Test results summary

All calls in this test used transcoding between G.711alaw and G.722. The bottleneck in performance was always at the N2600 (atom01), because of slower CPU. In general, N570 can handle approximately 30% higher load than N2600.

With 10 concurrent calls (21 channels on atom01 and 20 channels on atom02), there is no voice distortion and new call processing does not disturb the ongoing calls. Each virtual CPU is busy at 20-25%

With 20 concurrent calls (41 channels on atom01 and 40 cannels on atom02), there is some minor voice distortion, especially during incoming calls, but quality s still acceptable.

With 27 concurrent calls (55 and 54 channels), voice distortions were too high and not acceptable. Every virtual CPU on atom01 was busy at around 50%, which means full load for the whole CPU.

With 20 concurrent calls without transcoding (PCMA only in all call legs), each CPU core on atom01 was utilized at around 9-10%. So, theoretically the platform can handle up to 40-50 simultaneous calls in non-transcoding mode.

Only the voice quality was tested. CPS was not tested, and it depends heavily on the complexity of the dialplan. But the overall response of the system was quite acceptable.

Testing details

I took my minimal FreeSWITCH configuration for the tests, and extended it as follows:

atom01

sip_profiles/external/itsp.xml registers at my vPBX, so that I could initiate the calls. Incoming calls are sent to extension 500.

sip_profiles/external/atom02.xml defines the gateway that points to atom02:

<include>   <gateway name="atom02">     <param name="register" value="false"/>     <param name="proxy" value="192.168.1.61:5080"/>     <param name="ping" value="27"/>   </gateway> </include>

dialplan/public/15_bulkmatch.xml consists of 100 identical conditions, like shown below. It does not do anything useful, and it’s only used to make FreeSWITCH  busy processing the dialplan:

<include>   <extension name="bmatch" continue="true">     <condition field="destination_number" expression="^(\d+)$">       <action application="set" data="xxxxx=$1"/>     </condition>   </extension>   <extension name="bmatch" continue="true">     <condition field="destination_number" expression="^(\d+)$">       <action application="set" data="xxxxx=$1"/>     </condition>   </extension> ......

dialplan/public/20_perftest.xmltakes the call at extension 500 and plays delayed echo. When I press *1, it makes a new call leg to atom02 in G711alaw codec, and atom02 makes a new call leg back to atom01 in G.722:

<include>   <extension name="500">     <condition field="destination_number" expression="^500$">       <action application="answer"/>       <action application="bind_meta_app" data="1 a si transfer::501 XML ${context}"/>       <action application="delay_echo" data="1000"/>     </condition>   </extension>   <extension name="501">     <condition field="destination_number" expression="^501$">       <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>       <action application="unbind_meta_app" data=""/>       <action application="bridge" data="{absolute_codec_string=PCMA}sofia/gateway/atom02/600"/>     </condition>   </extension>  </include> atom02

sip_profiles/external/atom01.xml defines the gateway pointing to atom01:

<include>   <gateway name="atom01">     <param name="register" value="false"/>     <param name="proxy" value="192.168.1.60:5080"/>     <param name="ping" value="27"/>   </gateway> </include>

dialplan/public/15_bulkmatch.xml is identical to that on atom01.

dialplan/public/20_perftest.xml answers the call at extension 600 and places a new call to extension 500 at atom01:

<include>     <extension name="600">     <condition field="destination_number" expression="^600$">       <action application="answer"/>       <action application="bridge" data="{max_forwards=65}{absolute_codec_string=G722}sofia/gateway/atom01/500"/>     </condition>   </extension> </include>

When non-transcoding tests are made, the codec string is changed to PCMA.


Filed under: Networking Tagged: freeswitch, linux, pbx, sip, testing, voip, xlab1

How I bought Microsoft Visio Pro for Office 365

Tue, 10/08/2013 - 02:01

I needed to have MS Visio Professional on my new Win8 notebook, but paying $900 for the license was somewhat uncomfortable. Also relatively recently, Microsoft started offering Office 365 subscriptions where the software is offered as a monthly or yearly subscription instead of a one-off purchase.

It appears that MS Visio is also offered as subscription, but for some reason it’s not so easy to find: on the product page, you see only the full license for purchasing. But if you click to “Try or buy”, you have a “Learn more” link under the Visio Pro for Office 365 title.

Then, I could not find anywhere, which Office 365 subscription is needed to add Visio to it. So i thought, maybe I should just buy one, so I ordered the one which seemed the right one for my purposes, Office 365 Small Business Premium.

When I tried to add Visio Pro for Office 365 to my account, I got an error that this product is incompatible with my current subscription, and they tried to make me create a new subscription instead.

So, I opened a support request and they explained me that Visio can only be added to Office 365 Enterprise subscriptions!

I then asked them to cancel my subscription and promised to open an Enterprise one.

But: to say the truth, I actually had a valid Office 2007 license, and I only needed Visio. So, I made a new subscription for Visio only. Also funny, that even after deleting my first subscription, I could not use the same account name, and had to come up with a new name.

Actually that was not the end of the fun: after buying it, it took a while to find the installation link in the Office 365 Admin panel. Also when I clicked and downloaded something, it was not an installer in its traditional way. It was a self-extracting archive with a command line utility in it, and a sample XML file. Luckily this XML file had already an example for Visio, so I only needed to uncomment the relevant parts of it, and then use the command-line tool to download and set up the software. It was not difficult, but kind of surprising :)

So, finally I got Visio 2013 working, and Office 2007 has installed and activated smoothly. But I lost about hour or so because of:

  1. obscure product information on MS website
  2. strange incompatibility in subscription plans
  3. command-line installer with an XML that needed manual editing (!)

 


Filed under: Weird things Tagged: microsoft, office 365, visio

Out-of-business greeting with FreeSWITCH

Sat, 08/17/2013 - 23:22

After I got my US number at Callcentric, I got several wrong calls in the following days. The calls were quite late at night, and most of them dropped after few seconds, before I could take up the handset. And another ring around 3am was really long and loud, and it dropped anyway before I could come up and pick the call.

First, I needed to record my own greeting. In “default” dialplan, I added a new extension. It takes a recording and plays it back :

  <!-- 7396: Record a greeting -->   <extension name="app_7396">     <condition field="destination_number" expression="^7396$">       <action application="answer"/>       <action application="sleep" data="500"/>       <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>       <action application="set" data="playback_terminators=#"/>       <action application="set" data="record_waste_resources=true"/>       <action application="set" data="recfilename=$${base_dir}/recordings/greeting_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>       <action application="record" data="${recfilename}"/>       <action application="sleep" data="700"/>       <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>       <action application="playback" data="${recfilename}"/>       <action application="hangup"/>     </condition>   </extension>

As I’m using a Gigaset 610IP handset with G722 codec, the produced recording had 16KHz sampling rate, and needed to be resampled, because inbound external calls are G711 only:

sox recordings/greeting_2013-08-17-11-46-35.wav sounds/dvop/8000/ssinyagin_oob.wav rate 8000

Then my extension in “public” dialplan is modified to play the greeting unless the call is between 7am and 11pm. It also plays MOH for 5 seconds before bridging the call, and continues playing MOH while ringing my phone.  This gives the mistaken caller another chance to realize that something is wrong and drop the call.

  <extension name="pub_ssinyagin">     <condition field="destination_number" expression="^ssinyagin$" break="on-false">       <action application="set" data="timezone=Europe/Zurich" inline="true"/>     </condition>     <!-- 7:00 - 23:00 -->     <condition minute-of-day="420-1000" break="on-true">  <action application="answer"/> <action application="set" data="playback_timeout_sec=5"/> <action application="playback" data="$${hold_music}"/> <action application="set" data="ringback=$${hold_music}"/>       <action application="transfer" data="7110 XML default"/>     </condition>     <condition>       <action application="answer"/>       <action application="sleep" data="1000"/>       <action application="playback" data="$${sounds_dir}/dvop/ssinyagin_oob.wav"/>       <action application="hangup"/>     </condition>   </extension>
Filed under: Networking Tagged: freeswitch, pbx, voip

Free US number and Caller ID manipulation

Sat, 08/03/2013 - 17:46

Callcentric offers US numbers in NY area code, with zero recurring costs. This is very convenient if you want your USA customers to connect to your PBX. After ordering a free number, you create a SIP account and route the DID to it. The service allows to register multiple free numbers. Forwarding to SIP URI is not supported.

Upon receiving a call, the caller ID in From field will look like 16313335447, with the country code without any leading symbols. The following piece of FreeSWITCH configuration (in public context) alters the caller ID in order to look like a normal number in a European dial plan:

  <extension name="intl_normalize" continue="true">     <!-- remove Swiss country code -->     <condition field="${caller_id_number}" expression="^41(\d+)" break="on-true">       <action application="set" data="effective_caller_id_number=0$1"/>       <action application="set" data="effective_caller_id_name=0$1"/>     </condition>     <!-- add 00 in front of country code -->     <condition field="${caller_id_number}" expression="^[1-9]" break="on-true">       <action application="set" data="effective_caller_id_number=00${caller_id_number}"/>       <action application="set" data="effective_caller_id_name=00${caller_id_number}"/>     </condition>   </extension>

It is important to set both effective_caller_id_number and effective_caller_id_name variables. If only effective_caller_id_number is set, the effective_caller_id_name still keeps the original caller ID number, and if the call is bridged to a local extension, the SIP phone may want to use it for displaying the caller.

 

 


Filed under: Networking Tagged: freeswitch, pbx, sip, voip

voxserv.ch and Twitter Bootstrap site templates

Thu, 07/25/2013 - 23:48

Here’s a new website where I promote the VoIP integration services on Swiss market: http://www.voxserv.ch/

The website is built with the Twitter Bootstrap, and here are the templates for template-toolkit which separate the Bootstrap HTML from text content: https://github.com/ssinyagin/voxserv.ch/tree/master/builder


Filed under: Networking, Programming Tagged: linux, server management

Handling e164 numbers in FreeSWITCH

Sat, 07/06/2013 - 03:11

SIP clients installed on smartphones may pick up the destination number from the phone book, and it’s sometimes in e.164 format (+[countrycode][localdigits]).

The following piece of XML dialplan transforms such numbers into the standard form that is expected by most PSTN VoIP providers. This example assumes that the FreeSWITCH server is located in Switzerland and +41 is the e.164 prefix for in-land calls. It returns the call to the same context, making the switch traverse the whole context dialplan from the beginning. It makes sense to place this extension at the bottom of a context.

    <extension name="e164_pstn">       <condition field="destination_number" expression="^\+41(\d+)" break="on-true">         <action application="transfer" data="0$1 XML ${context}"/>       </condition>       <condition field="destination_number" expression="^\+(\d+)" break="on-true">         <action application="transfer" data="00$1 XML ${context}"/>       </condition>     </extension>
Filed under: Networking Tagged: freeswitch, pbx, voip

FreeSWITCH: Limiting the number of concurrent calls on multiple SIP accounts

Sun, 06/30/2013 - 02:52

The user has several SIP accounts on a vPBX, and he wants that maximum one call is possible at a time.

The limit application in FreeSWITCH allows to control the number of concurrent calls, but one should be careful with when this limit should be applied. The switch decrements the limit counter automatically when a channel is terminated. But if the limit is executed on a-leg, and b-leg is transferred, the limit counter decreases only when the a-leg finishes the call. As a result, the user may receive a call, transfer it to a new destination and hang up, but the new calls are not coming in because the limit counter is reset when the original call ends.

In order to reset the limit counter after the b-leg is transferred, the limit application needs to be executed on b-leg only. This is possible by exporting the execute_on_answer variable with nolocal modifier.

The example also shows how to retrieve user variables from the XML directory in the calls toward the user.

  <context name="moretti">     <extension name="common_variables" continue="true">       <condition>         <action inline="true" application="set" data="availability_username=moretti"/>       </condition>     </extension>     <extension name="pstn_out">             <condition field="destination_number" expression="^[01]" break="on-false">         <!-- For outbound calls, we only set the limit counters,              but do not limit the call -->         <action application="limit" data="hash ${domain_name} ${availability_username} -1"/>         <action application="set" data="hangup_after_bridge=true"/>         <action application="set" data="continue_on_fail=false"/>       </condition>       <condition>         <action application="bridge" data="${outgw}/${destination_number}"/>       </condition>     </extension>     <extension name="inbound_73x">       <condition field="destination_number" expression="^73(d)$" break="on-false">         <!-- retrieve variables from the user entry in the directory -->         <action application="set" data="directory_userid=70$1@${domain_name}"/>         <action application="set" data="call_timeout=${user_data(${directory_userid} var ring_timeout)}"/>       </condition>       <!-- check the limit -->       <condition field="${cond(${limit_usage(hash ${domain} ${availability_username})} > 0 ? true:false)}"                  expression="^true$" break="on-true">         <action application="hangup"/>       </condition>       <!-- group call to the SIP user and a mobile phone -->       <condition>         <action application="export" data="nolocal:execute_on_answer=limit hash ${domain} ${availability_username} -1"/>         <action application="set" data="ignore_early_media=true"/>         <action application="set" data="transfer_ringback=$${hold_music}"/>         <action application="set" data="hangup_after_bridge=true"/>         <action application="set" data="continue_on_fail=false"/>         <action application="bridge" data="user/${directory_userid},[leg_delay_start=10]${outgw}/0123456789"/>       </condition>     </extension>       </context>
Filed under: Networking Tagged: freeswitch, pbx, sip, voip

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