Simple sip server for windows. Deciding on a SIP server

SIP telephony can significantly reduce telephone communication costs. By using the services of IP providers, we save money and get the opportunity to make calls at reduced rates. tariff plans from anywhere in the world. This type communication is also used to organize intra-office telephony - for this you need to install a SIP server on one of the computers and connect software and hardware phones to it. In this review we will compare the most popular SIP servers, including free ones:

  • Asterisk;
  • Kamailio;
  • OfficeSIP Server;
  • sipX.

Let's look at these servers in more detail and find out how to start a SIP server with your own hands.

We will begin this review by considering one of the most famous servers for IP telephony - the Asterisk SIP server. It is aimed at organizing office telephony and is very popular.

Asterisk SIP server

Asterisk can be called a freely distributed solution, but it still contains licensed modules. The program runs on Linux operating systems and is available in several distributions that differ in functionality, web interfaces and sets of additional modules. This is not to say that this is a solution for novice users.– rather, this is a more professional solution. The Asterisk SIP server has the following capabilities:

  • Call forwarding and transfer;
  • Call holding and waiting (with background music);
  • Call interception and parking (functions allow you to answer calls from other devices or continue conversations started on other devices);
  • Conference calling;
  • Video communication;
  • Call center functions;
  • Integration of traditional telephone lines;
  • Administration via web interface;
  • Billing functions.

We can say that using the Asterisk SIP server will allow you to solve a problem of any complexity. Scalability, the presence of additional modules, a huge number of supported protocols - all this can be called the advantages of the program. As for the disadvantages, it is difficult to configure for novice users and the presence of a dual license.

Despite the fact that this server is free, it may contain modules distributed based on licensed code - sometimes this causes some problems.

Kamailio SIP server

This project was once called the OpenSER SIP server, but in 2008 it was renamed Kamailio. But it cannot be called the most famous when compared with such monsters as 3CX or Asterisk. The server has decent functionality and is most often used in a professional environment. That's why it is not suitable for solving simple problems.

In the list of its advantages we can include support large number all kinds of modules that expand its functionality. The disadvantages included the difficulty of setting up.

SIP server sipX

This is another one free product, running Linux systems. The sipX server is simple and focused on office use. The developers have endowed it with decent functionality, providing a large number of functions for managing voice calls. When using suitable equipment, the sipX SIP server allows you to solve even the most complex problems.

Its advantages include stability, simplicity and minimum dimensions. SipX allows you to deploy local SIP networks in a matter of hours, which is used for quick telephone installation of offices. This server is also free. As for the disadvantages, the most negative point is that for all functions to work, you need advanced phones and VoIP gateways.

SIP servers for Windows

Linux systems have the highest stability and excellent performance. But they require certain knowledge and cannot be called user-friendly. ordinary users. Therefore, more understandable SIP servers for Windows have appeared in the software world. Of course, here too, users and system administrators may face various difficulties, but it is much easier to get around them.

3CX SIP server

Among the most advanced SIP servers, we can highlight the Voip-PBX 3CX Phone System for Windows. This solution is designed for organizing corporate communications of any scale, even if individual offices are located on different ends of the planet. Server advantages:

  • Full voice functionality;
  • Support for a large number of clients (including our own software for various platforms);
  • Web conferencing support;
  • Integration of services of third-party SIP providers and traditional telephony operators.

Using the 3CX Phone System server allows you to minimize communication costs and make office telephony more convenient. The developer provides users with a variety of training materials, conducts training events, and provides comprehensive user support. Customers can choose from a standard free version, as well as a commercial version that features support for additional functions.

The free trial version is quite functional and can be used as a basic option for organizing IP telephony.

This product has many advantages. First of all, it is necessary to highlight that the 3CX Phone System server runs on the Windows operating system. It is extremely flexible in settings and has great functionality. If you need regular telephony, and not a whole call center, then the free version will be enough for you. Disadvantages - it is impossible to add something of your own to the system, since the source code is closed. However, this cannot be considered a significant drawback.

Office SIP Server

Free SIP Server OfficeSIP Server is free software for Windows. This server is so simple that even the most inexperienced user can handle its installation and configuration. Installing and launching the program takes a couple of minutes, after which you can begin creating local user accounts.

Also It is possible to connect to a third-party IP provider for calls around the world. Great program for small offices that need office telephony. Benefits of the program:

  • Ease of settings;
  • Working in a Windows environment;
  • Easy to connect new subscribers;
  • Having a connection with the outside world.

Disadvantages of the program:

  • Lack of many convenient office and voice functions;
  • Inability to scale;
  • There is no possibility of connecting to “your” PBX from anywhere in the world (only local connections).

However, this is an extremely affordable and free SIP server for small offices.

SIP telephony server service

By turning on this service, you get the opportunity to use SIP telephony server (PBX) based on Asterisk within your home network.

You will be able to register your smartphone or computer with a SIP client in this telephone exchange and call your relatives and friends who are also registered in this server.

Hint! In addition, you can configure your SIP telephony server using these instructions

Example of use and settings

Everything is very simple.

1. On the applications page you need activate the SIP telephony server service, which will act as a single point of registration for your smartphones, computers and other devices using the SIP protocol. This server will switch your phone calls within your distributed network.

The server address on your network is 172.16.255.14

After starting the server, check its availability by running the command ping 172.16.255.14

2. If the ping was successful, then register your devices. To do this, on our website, indicate the desired phone number and password for this device, and then configure your device, as shown in the example below.

2.1. On the SIP telephony server page, specify the desired phone numbers and passwords for connection.

This example shows two telephone numbers- 10 and 11 with password 1111 each.

2.2. Set up your device. This example shows two connection implementations - with a standard SIP client of the Android OS and using the Zoiper application located on a PC running Windows 8

So Android. It has a built-in SIP telephony client.

First, create a new SIP server account

We indicate the previously selected phone number with which we will register on the SIP telephony server (in our example 10), password and server address

After saving account the phone will try to register with the SIP server.

There are also various settings, and most likely you will need to select "Accept incoming" so that the phone is in communication with the SIP server and is waiting for an incoming call. Actually that's all.

Now let's create a contact for the person we are going to call through our SIP telephony server. To do this, let's go to the notebook and add a new Contact , which we will call "Dacha". But there is a nuance... we need to indicate the number of the “Dachi” and this must be done in the field called “Call via the Internet”.

On the main contact screen this field is not present, so you need to scroll down to "Add another field" and then a new window will open with a selection of fields, among which will be “Call via the Internet”

Now the last thing left is to indicate the number in this field. It is indicated as shown in the figure below -

With this our client on Android is ready. Let's add settings on the second side of our future telephone connection.

2.2 In the role of the second party we will have Windows 8 PC with Zoiper SIP telephony client installed.

After installation, go to settings and add new account with SIP protocol.


In your account settings, specify your username and server address in the following format: This email address is being protected from spambots. You must have JavaScript enabled to view it..4 and password. Then check the box " Skip auto detection"


After saving the settings, go to the settings again and click the Register button. A status entry should appear in the right corner - Registred.

If everything registered successfully, you can try and call. Close this window. From the Home screen, select Dialpad and type Android number - 10.


We hope that your Android is ringing and you can check the quality of the connection.

That's all, actually.

Technical features

Your SIP telephony server located at 172.16.255.14 is only a SIP server and no longer contains any data other than the numbers you entered.

Service testing period

We plan that the testing period for the SIP telephony server service will take about a month.

Cancellation of the service

You can cancel the service at any time. In this case, your device registrations will be deleted and the SIP telephony server will be stopped.

Despite the development various systems exchange of information such as Email and instant messaging services, the regular telephone will remain the most popular means of communication for a long time. A key event in the history of telecommunications and the Internet was the advent of voice over IP networks, so the very concept of a telephone has changed in recent years. The use of VoIP is modern, convenient, and cheap, since you can combine remote offices without even resorting to the services of operators telephone communication. What other reasons are needed to set up your IP telephony server?

Project Asterisk

Asterisk is present in the package repositories of most distributions. So, in Ubuntu the command is sudo apt-cache search Asterisk gives a decent list of packages, after installing which you can immediately start configuring. But installing from a repository has one drawback - as a rule, it contains the version Asterisk is significantly behind the current one, which can be downloaded from the official website. Therefore, let's consider universal method installation using the example of Ubuntu, although everything said (with rare exceptions) also applies to other distributions.

Install the packages required for compilation:

$ sudo apt-get install build-essential automake
autoconf bison flex libtool libncurses5-dev libssl-dev

Additionally, it is highly recommended to install libpri even if you do not need Primary Rate ISDN support. This can be done either through the repository: sudo apt-get install libpri1.2, or using the sources:

$ wget -c downloads.digium.com/pub/libpri/libpri-1.4-current.tar.gz

Compilation of the library is standard, so we won’t dwell on this.

Now download the source texts from the site Asterisk and configure:

$ wget -c downloads.digium.com/pub/Asterisk/Asterisk-1.4.11.tar.gz
$ tar xzvf Asterisk-1.4.11.tar.gz
$cd Asterisk-1.4.11
$ ./configure --prefix=/usr

At the end of the script, we will see the project logo and some information about the settings in the console.

$make
$ sudo make install

Note: if you are installing version 1.2, then to support the mp3 format you should enter “make mpg123” before the make command; version 1.4 no longer responds to this command.

After compilation, among other things, the following executable files will be installed:

  1. /usr/sbin/Asterisk - server daemon Asterisk, which provides all the work;
  2. /usr/sbin/safe_Asterisk - script for starting, restarting and checking the operation of the server Asterisk;
  3. /usr/sbin/astgenkey – a script for creating private and public RSA keys in PEM format, which are necessary for operation Asterisk.

To install configuration file templates and documentation, type:

$ sudo make samples

Example configuration files will be copied to /etc/ Asterisk. If there are already configuration files in this directory, they will be renamed with the ".old" prefix. To build the documentation you will need the doxygen package; if it is not there, install it:

$ sudo apt-get install doxygen
$ sudo make progdocs

Install the extension package in the same way. Asterisk-addons (this step is optional, you can safely skip it). Many of the modules included in this set are experimental. They should be installed only if you need to record information in the database, support mp3 files and the ooh323c protocol (Objective Systems Open H.323 for C):

$ wget -c downloads.digium.com/pub/Asterisk/Asterisk-addons-1.4.2.tar.gz
$ tar xzvf Asterisk-addons-1.4.2.tar.gz
$cd Asterisk-addons-1.4.2
$./configure; make; sudo make install; sudo make samples

Installation Asterisk finished. It is recommended to first start the server in debug mode and look at the output for errors:

$ sudo /usr/sbin/Asterisk -vvvgc

If we receive the message “ Asterisk Ready" and the management console prompt, then everything is in order. We leave:

*CLI> stop now

Now you can proceed to further configuration.

Setting up interface card support

If you plan to connect a server Asterisk using special interface boards to conventional telephone networks, care should be taken to ensure that appropriate drivers are available, implemented as a kernel module. But even if there are no such devices on your computer, it is still recommended to install these drivers. The fact is that all Zaptel devices have a timer, and for the full operation of the IP telephony server it is necessary. But if you don’t have a Zaptel device at hand, you can use a special driver - ztdummy - to emulate it.

From the repository we install the zaptel, zaptel-source packages and assemble modules for our system:

$ sudo apt-get install zaptel zaptel-source
$sudo module-assistant prepare
$ sudo m-a -t build zaptel

The zaptel-modules-*_i386.deb package will appear in /usr/src, install it using dpkg. After this, we check the operation of the kernel modules:

$ sudo depmod -a
$ sudo modprobe ztdummy

And if you need device support:

$ sudo modprobe zaptel
$ sudo modprobe wcfxo

To provide them automatic download, run the following command:

$ echo "ztdummy\nzaptel\nwcfxo" >> /etc/modules

Create rules for UDEV:

$ sudo mcedit /etc/udev/rules.d/51-zaptel.rules

KERNEL="zapctl", NAME="zap/ctl"
KERNEL="zaptimer", NAME="zap/timer"
KERNEL="zapchannel", NAME="zap/channel"
KERNEL="zappseudo", NAME="zap/pseudo"
KERNEL="zap0-9*", NAME="zap/%n"

You can also use the source code or the CVS version of the driver. If you compile yourself, you will need the kernel header files (or source code):

$ sudo apt-get install linux-headers-`uname -r`

$ sudo ln -s /usr/src/linux-headers-2.6.20-15-generic /usr/src/linux-2.6

Now we get latest version drivers:

$ cd /usr/src
$ wget -c downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz

Compile and install:

$ tar xzvf zaptel-1.4-current.tar.gz
$ cd /usr/src/zaptel-1.2.17.1
$./configure
$ make
$ sudo make install

And so as not to manually create configuration files:

$ sudo make config

After this command a script will be created for automatic start modules included in Zaptel, and the /etc/default/zaptel (or /etc/sysconfig/zaptel) config, which will indicate which modules need to be loaded. I recommend leaving only what is necessary in this file. Let's try to load the module:

$ sudo modprobe ztdummy
$lsmod | grep ztdummy
ztdummy 6184 0
zaptel 189860 1 ztdummy

Everything is fine. After installation, two more files will appear on the system:

  1. /etc/zaptel.conf – describes the configuration hardware;
  2. /etc/Asterisk/zapata.conf - server settings Asterisk for the Zap channel driver to work.

Detailed instructions for all types of devices are given in the documentation. In Russian, you can read about this in the document “Asterisk%0A+config+zaptel.conf">Zaptel kernel driver configuration." But we don’t stop there, we still have a lot of work ahead. After configuration, we check the work with the ztcfg -vv command.

User registration

If you now look in the /etc/ directory Asterisk, can be detected a large number of files. But the size of a journal article will allow us to get to know only some of them. So, in Asterisk.conf specifies the directories that will be used Asterisk during operation, the location and owner of the socket used to connect the remote management console, as well as the default server startup parameters. Some directories are not created during installation; this will have to be done manually:

$ sudo mkdir -p /var/(run,log,spool)/Asterisk
$ sudo adduser --system --no-create-home Asterisk
$ sudo addgroup --system Asterisk

Let's add a user Asterisk to the audio group:

$ sudo adduser Asterisk audio
$ sudo chown Asterisk:Asterisk /var/run/Asterisk
$ sudo chown -R Asterisk:Asterisk /var/(log,spool)/Asterisk

Next we are interested in the sip.conf file, where the servers and SIP clients with which ours will be friends are defined Asterisk. Each of them is presented in the file as a separate block, which begins with a table of contents enclosed in square brackets. There are quite a lot of parameters in sip.conf, we will limit ourselves to adding a SIP account:

$ sudo mcedit /etc/Asterisk/sip.conf


type=friend
host=dynamic
; defaultip=192.168.1.200
username=grinder
secret=password
language=en
nat=no
canreinvite=no
context=office
callerid=grinder<1234>
mailbox=1234@office
; Before using the allow parameter, you must disable all codecs
disallow=all
; The order of the codecs does not matter
allow=ulaw
allow=alaw

The type field specifies what this client can do. If the value is user, he will only be allowed to receive incoming calls, with peer he will only be able to make calls, and friend means all actions at once, that is, user+peer. The host field specifies the IP address from which this client is allowed to connect. If it can connect from any address, specify host=dynamic. And in this case, in order to call the client when it is not yet registered, you should write down the IP address in defaultip, where it can always be found. In username and secret we indicate the login and password used by the client when connecting. The Language parameter specifies the greeting language code and specific phone tone settings, which are defined in the indications.conf file. When the client is running behind NAT, the corresponding field must be set to yes. Disabling canreinvite forces all RTP voice traffic to pass through Asterisk. If clients support SIP re-invites, they can be allowed to connect directly by specifying canreinvite=yes. The context field defines the dial plan into which calls coming from this client fall, and callerid is the string that will be displayed when a call comes from the client. By default, the default context is used, which takes all settings from the demo context. The latter is intended for demonstration purposes only, working system you need to create your own context. The mailbox field indicates voice box 1234 in the context of office. Other clients are configured in the same way.
After defining SIP accounts, our clients can register on the server Asterisk and make outgoing calls. In order for them to be able to receive calls, they must refer to the extensions.conf file, which describes the dialplan that distributes calls in the system. All allowed extensions are also indicated here.

$ sudo mcedit /etc/Asterisk/extensions.conf


include => default
exten => 1234,1,Dial(SIP/grinder,20)
exten => 1234,2,Voicemail(grinder)

Everything is simple here. We assign the number 1234 to the grinder user, and if he does not answer the call, he can be left a message in voicemail. The number after the number means priority, which determines the sequence of tasks. Now if Asterisk running, you should connect to its console by running on the same machine Asterisk-r, and use the reload command to force it to re-read the configuration files. There are also commands to reload a specific file. For example, the dial plan is reread with the extensions reload command.

The server is ready to receive clients. At Asterisk%0AAsterisk%0A _softphone.html">www. Asterisk guru.com/tutorials/configuration_ Asterisk _softphone.html we select a soft client and try to connect. For example, I like the free version of the ZoIPer (formerly Idefisk) program, which is simple and easy to use. There are versions for Linux, Windows and Mac OS X. Another good and also multi-platform client is X-Lite.

If everything is fine, a message like “Registered SIP “grinder” at 192.168.0.1 port 5060” should appear in the console, dial the number and call.

We have set up Asterisk in a minimal configuration, but that's not all it can do. What remains behind the scenes is connecting to another IP telephony server, call parking, music while waiting, billing, using the GUI to administer the server, etc., but we will try to fill these gaps in the following articles.

A SIP server is a set software to launch IP telephony within an office or production. Traditional telephony is characterized by high call prices and does not provide any particular benefits for business. Deploying your own production or office PBX makes it possible to set up call distribution, reduce communication costs within the company and establish voice communication with clients.

Choosing an IP telephony server is not difficult - in our review you will find solutions for Windows and Linux. But they are increasingly being replaced by ready-made solutions from providers. In addition, the prices for launching office telephony are cheap. The client just has to choose a tariff, pay for communication services, connect the equipment to the network and carry out everything necessary settings.

We have one of the most popular SIP servers in the world for organizing office telephony. The project appeared in 1999 and was designed to replace expensive mini-PBXs. The server runs under operating control Linux systems, has all the necessary functionality:

  • Supports traditional telephony.
  • Able to manage the distribution and processing of telephone calls.
  • Supports video sessions.
  • Can be integrated into CRM systems.
  • Supports call encryption to prevent eavesdropping.

The functionality of the Asterisk SIP server can be expanded with additional software. It works with almost any IP telephony protocol and can solve even the most complex problems. Its main drawback is its complexity. Convenient Web interfaces have been developed to manage the server, but they do not solve the problem of the complexity of this software product.

Server from 3CX

The 3CX Phone System SIP server is designed for telephony for businesses of any size. These may be small companies or large corporations with dozens of branches, divisions and divisions. It supports the full functionality of office PBXs - work with calls, integration into CRM, conference calls, call center functions and much more. The product is notable for its comprehensive support from the developer. Working environment – ​​Windows operating system. It will not be possible to implement your own developments, as in Asterisk, due to the closed source code server.

sipXecs Server

Another software PBX for solving business problems. It does not support many protocols and only works with SIP. A web interface is used to manage telephony. There is support for most standard functions - call transfer/processing, speed dialing, conferences, hold and wait, multi-channel communication and much more. The server runs the Linux operating system.

OfficeSIP Server

Free application for organizing office telephone communication. Suitable for small and medium offices that do not require additional functions. For large enterprises with divisions and branches around the world, this SIP server is not suitable. But connecting the accounting department, directors, human resources department, several offices with access to intercity and international communications is always welcome.

The server is running under operating Windows system and does not create difficulties. It is free even for business clients, which determines some demand for this product. Installation is quick and without delays; registration of new subscribers is done in a couple of mouse clicks. If you are faced with the task of setting up a connection yourself, but you do not have much experience, use this simple and free solution.

Ready-made solutions from providers

Recently, businesses have switched to ready-made solutions. There are several reasons for this:

  • Reduced costs - connection is often free, only intercity costs, workplaces and some additional functions are paid.
  • Safety - self-configuration VoIP in the office will not give confidence in the security of the system from hacks and attacks. Providers have certified personnel doing this.
  • Convenience – the only additional equipment needed is computers and telephones. No separate hardware for IP servers.

Let's look at several solutions for organizing IP telephony for business.

Cloud PBX from Zadarma

This provider provides office telephony connections at prices starting from 10 kopecks/min, with premium voice quality. To the system administrator your office will not have to tinker with the equipment - just add subscribers to the system and set up call distribution. Advantages of Zadarma:

  • Free connection to IP telephony.
  • The provider offers multi-channel numbers in 90 countries and in many Russian cities.
  • Possibility of integration with the CRM used.
  • Full functionality of a cloud PBX.
  • Free calls within the company and its branches, regardless of the geographical location of the workplace.
  • Access to 8-800 numbers with the functionality of a full-fledged call center.
  • API interface for implementing your own business tasks.

The provider guarantees high quality voice transmission, supports customers by phone or via internal chat, offers low-cost calls within Russia and around the world. And all this without expensive equipment and settings. Order the service and receive a ready-made cloud PBX in 5 minutes. Configuration is carried out through a convenient web interface.

As customer reviews show, the Zadarma provider provides high-quality voice transmission and full functionality of office PBXs for large enterprises and small firms.

Cloud PBX from SIPNET

One of the oldest IP telephony providers. It works not only with individuals, but also with corporate clients. The starting tariff will cost only 1000 rubles. It will include three telephone workstations, a package of minutes to choose from (from 600 to 1500 minutes to numbers in Moscow and St. Petersburg, throughout Russia or to mobile phones). There is no connection fee. Clients also have access to options that expand the functionality, number of seats and provide the services of a personal manager. SIPNET is a full-fledged PBX for business, including call center functions.

Corporate use of SIP numbers often takes place under the Windows OS installed on most office PCs. Let's look at existing VoIP solutions for this system.

Web calls: what, how, where

Approved by senior management

Software aimed at senior management is usually designed to minimize the difference between online conferencing and physical proximity to a table at a meeting as much as possible. It is to eliminate the boundaries of the virtual and real worlds that Silicon Valley developers are struggling, in the hope of functionally “outperforming” the Asterisk server.

  • B-Force. Developed by the company of the same name in 2010, and since then it has been improved every day. Users of the Russian-language Wikipedia position the program as one of the few that meets security requirements even for use in government agencies.
  • 3CX Phone is multi-platform and can be used in conjunction not only with Windows, but also with Linux devices, as well as mobile operating systems - Android, i-/Mac-OS, etc. All features are available to subscribers for free, which is surprising, given the quality of services, technical support and user-friendliness of the interface. The latter, by the way, is recognized (according to Software Advice research) as the leader of the TOP-5 most comfortable sipphons to use.
  • Brosix. One of the most secure programs, working according to the US federal standard with 256-AES symmetric encryption. Corporations that choose to use Brosix Business will have to pay a license fee in exchange for the ability to create private, crypto-secure networks with the click of a few buttons. Individuals can legally use the program for free, but in the light version, which does not have whiteboard functions, desktop sharing, or conference calls.

Comfort of clerks is the key to stable operation of the company

But not only management, but also ordinary white-collar workers need high-quality communication. No matter how routine the work of clerks is, it is on it that the company’s activities are based, and therefore it is in the interests of management to simplify their actions as much as possible. Many offices make do with the functionality of programs such as Skype, Yahoo! Messenger and the like for internal communication, but in some cases the optimal solution would be to use special software.

  • Call Office. “Tailored” for working with large client bases. Maximizes the simplicity of calling, sending messages to (e-mail/SMS) and other mass notifications.
  • Ventrilo. A softphone walkie-talkie associated with gamer voice chats. Despite stereotypes, it is popular in companies where profit depends on the speed of reaction and dynamics - for example, in delivery services or closed offline exchanges.
  • Sippoint. A utility that supports a multi-user interface and allows you to configure multi-level contact databases. In addition, users can share files on a closed intra-office network. It is notable for the fact that it easily ports data from/to other systems - Google Talk, QIP and other popular instant messengers.
  • Jabbin. The main advantage of the softphone is the ability to make calls even without a provider SIP connection, only with a user web connection, including local intra-connections. But at the same time, alas, there is no way to call a landline or mobile number.

The best softphone for the best subscribers

Subscribers to the site will not have to be tormented by the dilemma of choice: there is a universal and at the same time simple program, available to all users - IP telephone server - YouMagic Softphone. In addition to the obvious advantages of working with the provider itself, the subscriber will receive the following “bonuses”:

  • virtual PBX with protection against flooding, spam and DDoS attacks on central nodes, which guarantees comfortable communication without interruptions;
  • The technical support service will answer any question in detail, and in case of problems, it will promptly solve them;
  • Each softphone user will be able to use several accounts and financial calculators for each of them, thereby automating the accounting of traffic expenses.

These and many other features make use as comfortable as possible on any platform, including Windows, Android and other OS.

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