List of actions before using VoIP equipment
To set up VoIP phones or VoIP gateways, you need to perform a number of preliminary steps:

1. Get the password for each account SIP. Passwords are required to configure the equipment. To obtain passwords, follow these steps:

• Change account password SIP records and print the settings when changing the password.
• Send yourself current SIP settings via e-mail.

• Link a SIP account to the product;
• Link a SIP account to an IP address.

Obtaining a SIP account password
Before using a VoIP phone or VoIP gateway, the administrator should obtain an electronic or printed SIP account password. The password is required for subsequent configuration of the equipment.
You can obtain a password in one of the following ways:

1. Changing the password and printing the new password.
2. Sends the current password to the specified email addresses.

Changing your SIP account password

To change the password for your SIP account and print the password, you need to do the following:

Sending a password by email

To send yourself Current Password to your SIP account via email, follow these steps:

Linking a SIP account to a product
To link a SIP account to a product, follow these steps:

Linking a SIP account to an IP address

Once you have selected the entity to which the SIP account will be associated, you must also associate the account with an IP address or subnet. Binding to an IP address is configured in the “Binding” parameter group:

Note
Options to bind a SIP account to an IP address are not available when the “Do not change” checkbox is selected.

1. Select binding type:

• No - the SIP account will not be associated with any IP address.

Attention!
Selecting the “No” option may compromise security by allowing the SIP account to be hacked. Binding to an IP address makes it impossible to access the account from other IP addresses.

• Auto - the SIP account is linked automatically. This usually happens when you first register for an account.
• Yes - when selecting this option, you need to add the required number of IP addresses or subnet in the IP address input field. The field becomes available only when you select the “Yes” option. Enter the IP address and click Add. Then enter the following IP address or network mask of the form 1.2.3.4/24, followed by / to indicate the network mask.
  Add all the necessary IP addresses in this way.

Group binding of SIP accounts to product and IP address

For your convenience, the system provides the ability to simultaneously link several accounts to a product or IP address. To group account linking, follow these steps:

SIP/2.0 100 Trying- The request is being processed, for example, the server is accessing databases, but the location of the called user is in currently undefined.

SIP/2.0 180 Ringing- The location of the called user is determined. He is given a signal about an incoming call.

SIP/2.0 181 Call Is Being Forwarded- The proxy server redirects the call to another user.

SIP/2.0 182 Queued- The called user is temporarily unavailable, but incoming call queued. When the called user becomes available, he will send the final response.

SIP/2.0 200 OK- The command was completed successfully. An INVITE request means that the called equipment agrees to participate in the communication session; the body of the response indicates the functionality of this equipment; In response to a BYE request, the communication session is terminated; the response body does not contain any information; For a CANCEL request, the search is canceled; the response body does not contain any information; to the REGISTER request means that registration was successful; On an OPTION request, it is used to transmit information about functionality equipment, this information is contained in the body of the response.

SIP/2.0 300 Multiple Choices- The called user is available at several addresses. The caller can select any of them. The response specifies several SIP addresses where the called user can be found, and the calling user is asked to select one of them.

SIP/2.0 301 Moved Permanently- The user has changed his location, his new address is indicated in the Contact field.

SIP/2.0 302 Moved Temporarily- The user has temporarily changed his location (the period of time can be specified in the Expires field), his new address is indicated in the Contact field.

SIP/2.0 305 Use Proxy- The called party can only accept an incoming call if it goes through a proxy server. The calling party is recommended to contact the proxy server whose address is specified in the Contact field. The response is transmitted only by the terminal equipment (UAS).

SIP/2.0 380 Alternative Service- The call did not reach the recipient, but an alternative service option exists, which is indicated in the response body. For example, a call can be forwarded to a voice mailbox.

SIP/2.0 400 Bad Bequest- The request is not understood due to the presence of syntax errors.

SIP/2.0 401 Unauthorized- The request requires a user authentication procedure. Exist different variants authentication, and the response may indicate which one to use in a given case.

SIP/2.0 402 Payment Required- Advance payment for services is required.

SIP/2.0 403 Forbidden- The request will not be serviced by the server and should not be retransmitted.

SIP/2.0 404 Not Found- The server did not find the called user in the domain specified in the Request-URI field.

SIP/2.0 405 Method Not Allowed- It is not permitted to send a request of this type to the address specified in the Request-URI field. The Allow field of the response indicates the allowed types of requests

SIP/2.0 406 Not Acceptable- Responses generated by the callee will not be understood by the caller.

SIP/2.0 407 Proxy Authentication Required- The client must confirm its right to access the proxy server.

SIP/2.0 408 Request Timeout- The server cannot send a response, for example, indicating the location of the called user, within the period of time specified in the Expires field of the request. The caller can resubmit the request after some time.

SIP/2.0 409 Conflict- Processing of the REGISTER request could not be completed due to a conflict between the action specified in the action parameter of the request and the current state of the resources.

SIP/2.0 410 Gone- The server no longer has access to the requested resource and does not know where to forward the request.

SIP/2.0 411 Length Required- You must specify the length of the message body in the Content-Length field.

SIP/2.0 413 Request Entity Too Large- The request size is too large to process.

SIP/2.0 414 Request-URI Too Large- The address specified in the Request-URI field was too large to be interpreted.

SIP/2.0 415 Unsupported Media Type- The request contains an unsupported message body format.

SIP/2.0 420 Bad Extension- The server did not understand the protocol extension specified in the Require field.

SIP/2.0 480 Temporarily not available- The called user is temporarily unavailable.

SIP/2.0 481 Call Beg/Transaction Does Not Exist- Sent in response to receiving a BYE request that is not related to current connections, or a CANCEL request that is not related to current requests.

SIP/2.0 482 Loop Detected- The server has detected that the request it has accepted is being transmitted along a closed route (the Via field already contains the address of this server).

SIP/2.0 483 Too Many Hops- The server detected in the Via field that the request it accepted went through more proxies than allowed in the Max-Forwards field.

SIP/2.0 484 Address Incomplete- The server accepted a request with an incomplete address in the To or Request-URI field. Additional address information required.

SIP/2.0 485 Ambiguous- The address of the called user is ambiguous. The Contact header of the response may contain a list of addresses to which this request can be sent.

SIP/2.0 486 Busy Here- The called user is currently unable to receive an incoming call at this address. The answer does not exclude the possibility of contacting the user at another address or, for example, leaving a message in the voice mailbox.

SIP/2.0 500 Internal Server Error- The server is unable to service the request due to an internal error. The client may try to resend the request after some time.

SIP/2.0 501 Not Implemented- The server does not implement the functions necessary to service this request. The response is sent, for example, when the server cannot recognize the type of request.

SIP/2.0 502 Bad Gateway- A server operating as a gateway or proxy server receives an incorrect response from the server to which it sent the request.

SIP/2.0 503 Service Unavailable- The server cannot currently service the call due to overload or maintenance.

SIP/2.0 504 Gateway Timeout- A server operating as a gateway or proxy server has not received a response within a specified time interval from the server (for example, a location server) that it contacted to complete processing of the request.

SIP/2.0 505 SIP Version not supported- The server does not support this version of the SIP protocol.

SIP/2.0 600 Busy Everywhere- The called user is busy and does not want to accept the call at the moment. The answer may indicate the appropriate time to call

SIP/2.0 603 Decline- The called user is unable or unwilling to receive incoming calls. The response may indicate a suitable time for the call.

SIP/2.0 604 Does not exist anywhere- The called user does not exist.

SIP/2.0 606 Not Acceptable- The called user cannot accept the incoming call due to the fact that the type of information specified in the description of the communication session in SDP format, bandwidth, etc. unacceptable.

SIP is a free IP telephony standard. Widely used by both commercial VoIP operators and free services such as Google Voice, Voxalot, Ekiga.net, Sipnet.ru, etc., supported by many soft/hard phones and adapters, supports video telephony. Unlike Skype, it is an approved, generally accepted and open signaling protocol for VoIP systems.

SIP does not have a single global management and registration node. There are many different service providers - registrars. You can draw an analogy with email, or more precisely, jabber servers.
Additionally, any user or organization can run their own server.

Any SIP address (sip uri) consists of a user login and server address and has the form, for example, " ". In principle, letters are also allowed in the login, but I would advise limiting it to numbers only, so that later you will not have problems dialing such a number from phones without an alphabetical keyboard. You can have an unlimited number of SIP accounts on one or different servers, for different purposes.
Since SIP is an open standard, there are many software or hardware SIP clients, all of them compatible with each other and with any servers. Clients do not need to be on the same server to establish communication. At the same time, during a conversation they will be automatically connected directly to each other.

To connect to SIP telephony, you need to select a registrar and a client. For beginners, we can recommend sipnet.ru or comtube.ru. Windows users Sipnet offers its softphone with presets. It can only work with this recorder, but it requires almost no configuration. Another popular softphone for this OS is X-lite.
For *Nix systems, a good option is Ekiga or SFLphone with the same recorders. (Ekiga is also available for Windows)

Do not use the ekiga.net registrar offered by Ekiga. It has been tested and found problems with incoming calls. Opt out of it the first time you launch it, then select “add a SIP account” in the account manager.

Different registrars may provide different sets of services and amenities. Some are more focused on calls to telephone network(PSTN termination), some - to different online services. Choose what suits you best or use several at once; this is quite normal practice in SIP. The geographic location of the registrar does not matter much, since media traffic between clients in most cases will still go directly.
Registration is usually free, does not oblige you to anything and is done directly from the site, after which you receive a number, password, connection data (or download a program with presets) and can immediately use your SIP account. Calls within the network, to and from other networks, and from the regular telephone network through gateways are free. You do not need to indicate any means of payment during registration if you do not intend to call regular landline/cellular numbers, contain a personal direct access number, or use any other exotic services.

At manual setting the main parameters are your number (login), server address for connection and password. Often a STUN server is also specified. Do not neglect it; in some cases, work without it is impossible.

Once you've successfully connected, you'll want to test how your client works. Almost all registrars have service numbers for verification, a list of which is on their website. In this case, the most useful is the “echo test” or answering machines with recording and subsequent playback. Often such numbers work only within the network, but there are also open ones that are accessible from anywhere, for example:

In conclusion, a few words about cryptography.
SIP telephony includes 2 protocols - signal sip (control, dialing and information about connection status) and transport rtp (direct audio/video streams). If both clients support stream encryption (SRTP/zRTP), then the conversation can be conducted over an encrypted channel. If the server and client support TLS, then the signaling traffic will be secure.

The most current SIP softphones:

Ekiga (GPL, *nix/windows) http://ekiga.org/
Qutecom (GPL, *nix/windows/osx) http://www.qutecom.org/
SIP Communicator (LGPL, java) http://www.sip-communicator.org/
SFLphone (GPL, *nix) http://www.sflphone.org/
Linphone (GPL, *nix/windows/osx/android/iphone) http://www.linphone.org/
SipDroid (GPL, android) http://sipdroid.org/
X-Lite (proprietary, windows) http://www.counterpath.com/x-lite.html
fring (freeware, mobile devices) http://www.fring.com/

Hardware example SIP solutions: a gateway that allows you to connect 2 regular phones to 2 independent SIP accounts and use them without connecting to a PC. A very convenient and rich thing. Supports CallerID. Allows you to use SipBroker regardless of the registrar (dial plan support).
http://voips.ru/Linksys-by-Cisco-PAP2T.html

The full range of SIP-compatible equipment produced can be found, for example, here: http://www.sipnet.ru/orderandpay/hardware.php
(For regular user, probably the most useful categories will be "SIP phones" and "VoIP gateways"). When choosing, you should give preference to well-known and proven brands (Linksys, D-link, Cisco, Grandstream, etc.)

Several well-known sip voip providers:
http://www.sipnet.ru
http://www.comtube.ru
http://zadarma.com/ru
http://www.voxalot.com
http://www.ideasip.com
http://www.voipbuster.com

Comparison of prices for calls to the telephone network in different directions:
http://www.voipratetracker.com/compare_rates
http://www.voip-catalog.com/voip_routes.html

Possible problems.
If your client registered on the server normally, but calls in one or both directions do not go through or there is one-way audibility, the source of the problems is most likely a poorly configured firewall or a tightly closed NAT that does not support transparent work with such services out of the box. First, make sure that you have everything in order with the mixer and microphone, as described earlier. Try disabling the firewall. Check if STUN is specified. Below are several links that describe all this better than can be done in this article.
Separately, we can highlight the situation when incoming calls work immediately after connection, but stop going through after several minutes of inactivity. In this case, the solution comes down to selecting the keepalive time in the client and is also well described in the first link.
After each step, do not forget to reconnect to the server (if it is a softphone, just restart it)
http://wiki.sipnet.ru/index.php/Connection_via_router_with_NAT
http://wiki.sipnet.ru/index.php/Broadcast_network_addresses_%28NAT%29_and_SIP

Make cheap calls to all countries using the SIP protocol. Low tariffs for intercity and international telephone conversations. Today, IP telephony provides excellent opportunities to make cheap phone calls, gradually replacing traditional communications.

The operating principle of SIP calls is based on VoIP technologies, which allow voice transmission over IP networks (Internet or local networks). The main advantage of telephone calls using the SIP protocol is significant savings in money on international and long-distance telephone calls. the site will tell you in more detail about all the subtleties and secrets that will help you make cheap calls anywhere in the world.

What is SIP telephony and how does it work?

SIP stands for Session Initiation Protocol, namely a protocol that transfers data. In order to understand at least a little the meaning of such a service and how it works, we will take a short excursion into SIP. To begin with, what you should know is that communication is carried out using IP telephony. Protocols, methods and special technologies are used to conveniently provide ordinary and familiar numbers to us, make calls and carry out two-way communication.

SIP telephony is a protocol for transmitting messages, audio-video data, over the Internet. It can be used even with minimal Internet speed and the connection will remain of good quality.

This telephony method is considered one of the most popular ways to make calls at a low cost. You can make and receive calls through a computer or smartphones, where the following processes take place:

• when making a call, your or the client’s voice is compressed into a digital signal;
• after, the signal is transmitted to your device;
• a communication connection is made using the IP address and a conversation session using the SIP protocol begins;
• after conversion to analog signal, You will hear a normal voice and can speak fully.

Making cheap calls using the SIP protocol becomes possible when using softphones(softphones) that are installed on a computer, laptop or smartphone. You can also purchase an IP phone or simply connect your analogue landline phone to the Internet using an IP gateway. When connecting equipment, you need to configure network access and enter all SIP account settings.

If the user already has a SIP account connected to a third-party (external) server, then he must simply provide his data to set up forwarding. And if the client does not have a SIP account, and he plans to use the company’s server (internal server), then in this case the subscriber will create an account and be provided with data for setting up software or hardware equipment.

Advantages of SIP telephony

Every day, millions of subscribers around the world are convinced of all the undeniable advantages of this type of communication. Using calls over the SIP protocol allows you to fully experience all the benefits of IP telephony, makes it possible to reduce the cost of calls and reduce the current costs of paying bills, and also allows you to improve the quality and efficiency of the company’s employees and opens up unlimited opportunities for communication.

High quality of communication and the ability to make cheap calls anywhere there is an Internet connection, and not be tied to traditional telephone lines, – these are the main advantages of IP telephony. Calls between two SIP accounts will be absolutely free, as will forwarding calls from virtual numbers to SIP.

Using SIP telephony will greatly facilitate many tasks and also has the following advantages:

• the ability to organize your own automatic telephone exchange and manage it independently;
• cheap use without overpayments for international and long-distance calls;
• you can monitor the work of employees and improve efficiency;
• no reference to geographic location;
• good connection quality even at low Internet speeds;
• multi-channel numbers;
• a full range of services, such as: answering machine, forwarding, voice mail, PBX.

Tariffs for outgoing calls are significantly lower than the prices of landline and mobile communications for international and long-distance telephone calls. For example, calls to European and American countries will cost from 6 cents per minute, and to Asian countries - from 8 cents per minute. With such a call, the subscriber receiving the call will not even realize that the call is taking place over IP networks and his interlocutor is on the other side of the world.

How to connect SIP telephony and how it works?

To start using SIP telephony services, you need to follow a few simple steps:

1. Find a SIP telephony provider;
2. Using an IP phone connected to the Internet, a regular phone with access to an IP telephony gateway, or using phones, tablets, computers via special program, create a SIP account.
3. Configure all necessary settings;
4. Start making calls.

After installing the programs on your computer, PDA or laptop and making the settings (SIP login, SIP password, SIP server address), you can receive and make phone calls using the SIP protocol. Moreover, today there are mobile versions softphones for installation on a smartphone. In particular, they are designed for the operating room Android system and allow you to make and receive calls anywhere without being tied to a computer. The only condition in this case is a mandatory Internet connection.

SIP applications for cheap calls

The easiest way to make cheap phone calls without extra costs is to use SIP softphones, which can be downloaded for free online. There are many free SIP applications through which you can make calls:

• Zoiper - accessible program With user-friendly interface in Russian for Android, iPhone, Windows, WP8, Windows, Linux and Mac;
• X-lite- a program that supports transmission text messages, recording video and audio calls, archives, conference calls, video calls;
• Linphone- there are many functions, such as conference calling, address book, HD audio and video, call forwarding, call history, call recordings and other relevant functions. Supported by: operating systems like iOS, MAC, Linux, Windows, Android, and BlackBerry;
• PortSIP- you can install on a tablet, smartphone or computer this program With simple settings and get to work. It is also supported with almost all operating systems and has a user-friendly interface.

By topping up your account, you can get a SIP account and call anyone you want, there is no connection fee, and billing is per second, a rather convenient and profitable offer. At the same time, you yourself will regulate how much to pay for communication, just choose a tariff for yourself and according to it you will be connected to subscribers, the cost and quality will vary somewhat in different directions. But it is only your choice how much to pay and what quality of communication suits you. By the way, video calls can also be easily made via SIP, just like making calls.

An example of a Sip account, it looks like this:

• Host: sip.sipinout.net
• Username: Demo
• Password: Demo

Example of setting up a SIP program

Let's take "Xlite" as an example.

IP phones for making cheap calls

It is also possible to use IP phones for calls (they look like regular telephones, but contain additional functional buttons), and using network cable they connect to the Internet. Due to their ease of use, they are very popular in companies and call centers, because... are similar to a traditional telephone, provide the ability to display subscribers on a digital display, are quickly and easily configured, provide the ability to instantly respond to a call, forward the call to an employee’s device, mobile or home phone, work with line waiting modes, call queues, etc.

When making phone calls in this way, it is recommended to use a virtual phone number, which can also be connected to the company. In this case, the connected virtual telephone number will be indicated as the caller ID when making a call. Each client can buy as many direct city virtual telephone numbers, select your favorite number from one of 70 countries around the world and set up forwarding to your SIP account. If the user does not have a connected phone number, then a number consisting of a random dial of numbers will be indicated as the subscriber identifier.

Where to get a SIP account for cheap calls

There are many companies providing SIP telephony services where you can make cheap calls. An example of such a company is Freezvon. This is a reliable SIP telephony provider where you can purchase:

• SIP account and download free application to it (tariffs can be viewed on the website at https://freezvon.ru/pricing/calls-rates)
• mobile virtual numbers for SMS, calls, calls and SMS, calls and faxes;
• virtual PBX, which you can manage yourself;
• multi-channel numbers;
• toll free +800 numbers for free incoming calls for your clients;
• virtual numbers to different countries of the world;
• additional services such as: welcome message, voice mail, call listening, call statistics, etc.

Communicating via the Internet is not only convenient, but also profitable; you save significantly on mobile bills!