Best Android tablet for little children

Our good old Samsung Galaxy Tab 3 7.0 Kids Tablet has finally died after over 3 years of everyday heavy use, so I needed a new solution. So far, here is the best combination that I could find:

This silicon case for Samsung Galaxy TAB A 7″ SM-T280 is a solid and protective piece, and it allows the kids hold the tablet with their little hands without slipping off. It also works as a stand, so it’s very convenient for watching videos.

The Samsung Galaxy Tab A (7″, 8GB, Metallic Black) fits perfectly into the protective case. The tablet is coming with preinstalled “Kids Mode” application, which is pretty neat, but very restrictive: the kid can watch only the videos on SD card that you mark as safe, and YouTube is not available. You can install kid-safe YouTube wrappers from the Play market, but it’s a bit too much hassle to my taste.

So, instead of the Samsung Kids Mode, I installed Kids Place by kiddoware. With a little payment, you get a good child protection mode, and you can enable YouTube directly on the child screen. The payment is also transferable to other devices under your account.

Also, this portable Bluetooth speaker works as a stand for a tablet, and it produces a much better sound than the tablet’s own speaker. Unfortunately the silicon case is too thick for this stand, but it’s a minor issue, and the speaker can easily be placed behind the tablet.

 

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udev rules for ttyUSB devices

In my particular case, there are two physical USB devices that are represented as TTY devices in the kernel: a Gobi2000 3G modem, and a 4-port USB-to-serial adapter. The modem is presented by two ttyUSB devices, and the USB-to-serial adapter adds four more. At the machine boot, these devices get assigned random numbers ttyUSB0 to ttyUSB5, and this assignment changes between reboots.

So, this needs udev rules which would assign symlinks to these devices, and the symlinks should remain valid between the reboots.

As there’s only one physical device of each type attached to the host, we can base our udev rules on idVendor and idProduct attributes. If you need to distinguish between multiple physical devices of the same type, you have to match serial numbers in your udev rules. Read the rest of this entry »

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FreeSWITCH startup for FusionPBX

If you install FreeSWITCH 1.6 on Debian 8 from official .deb packages, and then add FusionPBX on top, the server boot sequence needs a modification: now FreeSWITCH configuration depends on the presence of Postgresql server, and it would load an empty configuration if the database service is not available at the moment of start.

This fixup adds a dependency on FreeSWITCH systemd service, so that it launches only after Postgresql has started:

mkdir /etc/systemd/system/freeswitch.service.d/
cat  >/etc/systemd/system/freeswitch.service.d/fusionpbx.conf <<'EOT'
[Unit]
After=syslog.target network.target local-fs.target postgresql.service
EOT

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tcpkali, TCP load generator

tcpkali is a lightweight and  easy-to-use tool that allows you to generate a traffic load with multiple TCP sessions. You push the load in one or both directions at the same time. Also the tool works easily over a NAT’ed connection. This tool is great if you need to test QoS for VoIP applications.

Here’s an example of a bidirectional load test:

# listening machine: listen on tcp port 8000, send traffic, and use 4 threads.
# the program will exit in 1 hour.
tcpkali -l 8000  --listen-mode=active -m X -T 1h -w 4

# connecting machine: send traffic using 4 threads and 10 simultaneous sessions
# for 1 minute
tcpkali 192.168.1.109:8000 -m Y -c 10 -T1m -w 4

The above test between directly connected PC Engines APU2 boards has shown 1Gbps of traffic, and the average CPU load was about 50%.

Also here are the packaging instructions for Debian, and a 64-bit binary package for Debian 8.

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Novatel E371 WWAN LTE modem for APU

Novatel E371 (also known as Dell DW5804) is sold for less than $30 at Aliexpress, and it’s so far the cheapest 4G/LTE WWAN card suitable for PC-Engines APU.

The initialization is fairly simple, although it was tricky to find the right command (AT$NWQMICONNECT=,,).

cat >/etc/chatscripts/lte_on.E371 <<'EOT'
ABORT BUSY
ABORT 'NO CARRIER'
ABORT ERROR
TIMEOUT 10
'' ATZ
OK 'AT+CFUN=1'
OK 'AT+CMEE=1'
OK 'AT\$NWQMICONNECT=,,'
OK
EOT

cat >/etc/chatscripts/lte_off.E371 <<'EOT'
ABORT ERROR
TIMEOUT 5
'' AT\$NWQMIDISCONNECT OK
AT+CFUN=0 OK
EOT

cat >/etc/network/interfaces.d/wwan0 <<'EOT'
allow-hotplug wwan0
iface wwan0 inet dhcp
    pre-up /usr/sbin/chat -v -f /etc/chatscripts/lte_on.E371 >/dev/ttyUSB0 </dev/ttyUSB0
    post-down /usr/sbin/chat -v -f /etc/chatscripts/lte_off.E371 >/dev/ttyUSB0 </dev/ttyUSB0
EOT

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One more 3G modem (Gobi2000) and a watchdog script

Qualcomm Gobi 2000 is quite old (released 2009), but decent 3G modem, able to deliver up to 7Mbps in downstream in PPP mode. These modems in mini-pcie packaging are available at Aliexpress for less than $10, and make up a great option for 3G connectivity for PC Engines APU boards.

The modem needs a binary firmware to be loaded at the start. Numerous sources in Internet describe the ways to retrieve these files. The kernel driver in Debian 8 recognizes the modem as generic Qualcomm one, and sets up a QMI device (wwan0). But this model does not support packet mode, and you need to run PPP over ttyUSB1 device.

apt-get install -y gobi-loader wvdial
mkdir /lib/firmware/gobi
cd /lib/firmware/gobi
wget --no-check-certificate -nd -nc https://www.nerdstube.de/lenovo/treiber/gobi/{amss.mbn,apps.mbn,UQCN.mbn}

cat >/etc/wvdial.conf <<'EOT'
[Dialer Defaults]
Init1 = ATZ
Init2 = ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0
Init3 = AT+CGDCONT=1,"IP","internet"
Phone = *99#
New PPPD = yes
Modem = /dev/ttyUSB1
Dial Command = ATDT
Baud = 9600
Username = ''
Password = ''
Ask Password = 0
Stupid Mode = 1
Compuserve = 0
Idle Seconds = 0
ISDN = 0
Auto DNS = 1 
EOT

cat >/etc/network/interfaces.d/ppp0 <<'EOT'
auto ppp0
iface ppp0 inet wvdial
EOT

Also this script is useful for 3G connections, because with some providers, the Internet connection gets stalled every few days and needs to be re-connected.

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Quality Assurance for VoIP calls: integration scripts

The scripts for integrating FreeSWITCH with Sevana AQuA software are now available at github: https://github.com/voxserv/fsqa

More details on what they are doing are available in this older post: https://txlab.wordpress.com/2015/06/02/quality-assurance-for-voip-calls-2/

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Quick IP prefix calculation

It’s a quite common task that you need to translate an IP address into a prefix — for example, when creating an IP prefix list from a set of addresses. Here’s a simple Perl script that helps it:

sudo apt-get install libnetaddr-ip-perl
cat >getprefix.pl <<'EOT'
use strict;
use warnings;
use NetAddr::IP;
if( scalar(@ARGV) == 0 ) {
    die("Usage: $0 PREFIX ...");
}
foreach my $pref (@ARGV) {
    my $ip = NetAddr::IP->new($pref) or
        die("Cannot create NetAddr::IP from $pref");
    print $ip->network()->cidr(), "\n";
}
EOT

# testing
cat >/tmp/x <<'EOT'
10.1.1.1/23
192.168.5.3/28
EOT

cat /tmp/x | xargs perl getprefix.pl | awk '{print "set ", $1}'
set  10.1.0.0/23
set  192.168.5.0/28


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Quality Assurance for VoIP calls

UPD: the FreeSWITCH integration scripts are available at https://github.com/voxserv/fsqa

A customer has requested to set up a QA service that would continuously monitor the voice quality in their telephony infrastructure. They use a number of telephony carriers, and a set of applications on top of Plivo and FreeSWITCH. Also the conference module in FreeSWITCH is actively used.

Measuring jitter and packet loss, like it’s done in VoIPmonitor, is not sufficient, as we need to monitor end-to-end performance, including that of the FreeSWITCH server itself. So, there has to be a software component that compares the source audio with the recording on the other end of a call.

There are currently two major player on the market for voice quality measurements:

  1. ITU-T PESQ algorithm is proposed as an ITU recommendation P.862. Its source code is available at the ITU website and on Github. But the algorithm is patented, and the source code license does not allow any production use. The evaluation went quite smoothly, and the algorithm was able to detect even minor distortions, like one 20ms frame loss in a 2-minute call. The PESQ algorithm is designed and calibrated to be used for audio files of 6 to 20 seconds in length. Processing of a 2-minute recording takes approximately 5 seconds on a modern Xeon CPU. Commercial software is provided by OPTICOM and PsyTechnics.
  2. Sevana.biz is an Estonian company that provides their own algorithms and software product for voice quality assessment. Their AQuA (Audio Quality Analyzer) software provides a fast and reliable way to compare the audio files: processing of a 2-minutes recording took about half a second on a modern CPU. Sevana has kindly provided a 10-days evaluation license and a fully functional software package, and the customer decided to go ahead with purchasing the license.

The simplest single-server license for Sevana AQuA allows running only one AQuA process at a time, so we wrapped its execution into a Perl script that utilizes a simple exclusive locking mechanism and performs audio file processing one at a time.

AQuA produces two scores in each measurement: the similarity percentage, and the MOS score. Both metrics are useful for quality analysis (for example, a 20ms frame added or lost inside of a silent pause influences the similarity score more significantly than MOS). It also takes a number of command-line options which can increase its tolerance to certain types of distortions, such as frequencies outside of G.711 range.

FreeSWITCH software is used as the SIP server for sending and terminating voice calls and for recording the received audio. It allows recording in several different formats: a) raw codec recording, done in the same thread as RTP processing; b) 16-bit signed PCM in WAV format, and file writing is done in a separate thread; c) compressed voice in a number of formats. The first two options produce similar results (raw codec recording had difficulties in the beginning). In case of raw codec recording, an additional step is required to convert the input files into 16-bit PCM WAV.

The call recording server requires to have a precise clock reference, so a baremetal hardware is required. Virtualized environments add up some uncontrollable imprecision to the virtual machines, although a thorough lab test is requires to verify this. It also depends on the type of hypervisor, as they implement the system clock differently.

The Linux kernel provides access to various clock sources. TSC is commonly used as default, and there is also HPET clock on modern hardware platforms. HPET is supposed to provide a more precise clock source, but it appears that it depends on CPU load: we accidentally discovered that audio recording in FreeSWITCH is significantly distorted when there’s some CPU activity is done in parallel (Debian package builder was working on the same 8-core machine). So far, TSC clock on a baremetal server provided the most reliable results.

The recording is done into a tmpfs mounted partition, in order to avoid any dependency on I/O load. The processing script performs the quality assessment on recorded files, and then moves or deletes them, depending on the measured score.

The SIP service was attached to an unusual UDP port, as port 5060 is frequently accessed by port scanners in public Internet. The DNS NAPTR and SRV records are used in order to use a universal SIP URI string, without having to reconfigure the remote servers if the IP address or UDP port changes.

Jitter buffer is disabled by default in FreeSWITCH, and it has to be activated whenever the calls are terminated on the server. In our case, the “jitterbuffer_msec” variable is set to “50:50” in the dialplan before answering and recording the call. With this, the jitter buffer is not allowed to grow dynamically above 50ms. So, we tolerate most of typical Internet-imposed jitter, but clock drift on the sending side would cause packet drop on the receiver.

The dialplan is designed to accept direct SIP calls from remote servers, and PSTN calls from telephony providers. If a remote server calls our QA service directly, it encodes the source name in the user part of the SIP URI. Also there are two options for a QA call: it can playback the test audio, or send silence. In case of PSTN calls, the caller ID is used as the source identifier. The dialplan activates audio recording into a WAV file on a tmpfs partition, and launches the processing script after the hangup.

The conference dialer is used for testing the conferencing performance on a production FreeSWITCH server. It requires a conferencing profile that does not play any greetings to conference participants. Also in case of more than two participants, only one has to be chosen as a speaker, and all others would be listeners. A dedicated SIP URI on the QA server is reserved to playback the test audio and not to perform any recording.

Each measurement result for QA calls is stored in an SQL database for further processing, and also sent to Syslog for real-time monitoring.

The test audio is a concatenation of speech samples from ITU-T Recommendation P.50 Appendix I, resampled from 16KHz to 8KHz and stored as 16-bit signed PCM audio.

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Simulating NAT with two Linux boxes

I needed to test some master-slave software in a situation that the master communicated to the slave over NAT (master’s IP address was replaced with the firewall’s external address), and then NAT would be removed, keeping master and slave addresses the same, but the slave would see the master directly.

This is the test scenario that worked on my desk, without having to add any routing to the LAN.

atom02 is the computer that emulates the slave system. It is connected back-to-back to alix102, and has only one IP address to communicate to:

ip link set dev eth0 up
ip addr add 192.168.1.50/31 dev eth0

alix102 is a Linux box with multiple Ethernet ports: eth0 is connected to my home LAN and has a DHCP address 192.168.1.142/24. Also eth1 (192.168.1.51/31) is connected directly to atom02.

The following configuration makes alix102 answer to ARP requests for 192.168.1.50 and forward packets to atom02, replacing the source address with 192.168.1.51. Also atom02 can make an SSH connection to 192.168.1.51:3022 and it will be connected to another box in the LAN that emulates the software master (192.168.1.147:22).

# enable IP forwarding
echo 1 > /proc/sys/net/ipv4/ip_forward
# Bring up eth1
ip link set dev eth1 up
ip addr add 192.168.1.51/31 dev eth1
# Enable proxy ARP on eth0
echo 1 > /proc/sys/net/ipv4/conf/eth0/proxy_arp
# Set up the NAT translation
iptables -t nat -A POSTROUTING -o eth1 -j SNAT --to 192.168.1.51
iptables -t nat -A PREROUTING -p tcp --dport 3022 -i eth1 -j DNAT --to 192.168.1.147:22

After that, atom02 can be re-connected directly into the LAN, keeping the address 192.168.1.50 with /24 network mask, and the software can be tested with direct communication. Alix102 has to be disconnected from the LAN, so that it does not pollute it with proxy ARP responses.

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