FreeSWITCH performance test on PC Engines APU

This test is analogous to the one I described for Intel Atom CPU.This time it’s the new APU board from PC Engines, the maker of famous ALIX and WRAP boards. APU is a fanless appliance board, with a dual-core 1GHz AMD G series CPU. The overall performance is comparable to that of Intel Atom.

In these tests, FreeSWITCH was forwarding the call to itself on request by pressing *1. Each such forwarding resulted in creating four new channels in G722 and G711, thus resulting in transcoding to G711 and back. For example, if “show channels” shows 5 channels, it’s equivalent to 2 simultaneous calls with transcoding.

Test result: 57 channels were running completely fine, 65 channels had slight distortions, and with 85 channels the speech was still recognizable, but with significant distortions. With Speex instead of G722, distortions were quite annoying at 25 channels. Thus, the APU platform can easily be used as a small-to-medium business PBX for  20-30 simultaneous calls if there’s not too much transcoding.

Test details follow.

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PC Engines APU board: installing Debian on mSATA drive

PC Engines started shipping its new APU board in 2014. It can boot from an SD card (slow on writes), and it can also have an mSATA drive and boot from it (fast read-write, and more write cycles). Voyage Linux is well optimized for SD card.

Here I started my scripts for building a Debian CD and installing it on APU’s mSATA drive: https://github.com/ssinyagin/pcengines-apu-debian-cd

 

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Call forwarding/redirection in FreeSWITCH

Consider you have two different contexts in your dialplan for inbound and outbound calls: the “public” context transfers the calls into “XXX_inbound” (XXX being your organization name), and the user directory has “XXX_outbound” as “user_context” variable.

Having two contexts, you have more flexibility in defining the short dial strings and outbound destinations.

But there’s a little problem: if the SIP client redirects the ringing call, or if the user makes an attended transfer, FreeSWITCH would initiate a new outbound leg in the same context where the call was bridged toward the SIP client.

As a solution, you need to define a new extension in your XXX_inbound context which would match PSTN outbound numbers. The channel will already have all custom variables which were set before bridging toward the SIP client, so you can set an additional condition criteria to make sure that this is the redirected call. This example would be placed at the bottom of the inbound context, and “directory_ext” is the variable that was earlier in the same context before the call was bridged to the SIP client:

    <extension name="call_forward">
      <condition field="destination_number" expression="^\d+$"/>
      <condition field="${directory_ext}" expression="^70\d$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=false"/>
        <action application="bridge" data="${outgw}/${destination_number}"/>
      </condition>        
    </extension>

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Reusing HTTP connections in client-server applications

I’m working on a client-server 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.

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Minimal FreeSWITCH configuration

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 :)

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FreeSWITCH performance on Intel Atom CPU

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.

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How I bought Microsoft Visio Pro for Office 365

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 (!)

 

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Out-of-business greeting with FreeSWITCH

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>

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Free US number and Caller ID manipulation

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.

 

 

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voxserv.ch and Twitter Bootstrap site templates

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

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