Sip login and password. SIP protocol messages

This section provides a description of the Session Initiation Protocol - SIP, its principles, addressing, architecture, and a comparison with the H323 protocol. The basis is taken from chapter 7 of the book by B.S. Goldstein IP Telephony.

SIP protocol messages

This version of the SIP protocol defines six types of requests. Each of them is designed to perform a fairly wide range of tasks, which is a clear advantage of the SIP protocol, since thanks to this the number of messages exchanged between terminals and servers is reduced to a minimum. Using requests, the client reports the current location, invites users to participate in communication sessions, modifies already established sessions, terminates them, etc. The server determines the type of request received by the name specified in the start line. In the same line, in the Request-URI field, the SIP address of the equipment to which this request is addressed is indicated. The contents of the To and Request-URI fields may differ, for example, the To field may indicate the subscriber's published address, and the Request-URI field may indicate the user's current address.

SIP requests

Request INVITE invites the user to participate in a communication session. It usually contains a description of the communication session, which indicates the type of information being received and the parameters (a list of possible parameter options) necessary to receive the information, and may also indicate the type of information that the called user wishes to transmit. The response to an INVITE request specifies the type of information that will be received by the called user, and may also indicate the type of information that the called user is going to transmit (possible information transfer parameters).

This message may also contain data necessary for subscriber authentication, and, consequently, client access to the SIP server. If it is necessary to change the characteristics of already organized channels, an INVITE request is sent with a new description of the communication session. The INVITE message is also used to invite a new participant to an already established connection.

Request ASK confirms receipt of the response to the INVITE request. It should be noted that the ACK request is used only in conjunction with the INVITE request, i.e. With this message, the calling user's equipment indicates that it has received a final response to its INVITE request. The ACK message may contain the final description of the communication session transmitted by the calling user.

Request CANCEL cancels the processing of previously transmitted requests with the same values ​​as in the CANCEL request, the values ​​of the Call-ID, To, From and CSeq fields, but does not affect those requests whose processing has already been completed. For example, the CANCEL request is used when the proxy server multiplies requests to search for a user in several directions and finds him in one of them. The server cancels processing of requests sent in all other directions with a CANCEL message.

By request BYE The called or calling user's equipment terminates the connection. The party receiving the BYE request must stop transmitting voice (multimedia) information and confirm its execution with a 200 OK response.

Using a request like REGISTER the user reports his current location. This message contains the following fields:

• Field That contains address information that must be saved or modified on the server;
• Field From contains the address of the registration initiator. A user can register either himself or another person, for example, a secretary can register his boss;
• Field Contact contains the new user address to which all future INVITE requests should be sent. If there is no Contact field in the REGISTER request, then the registration remains the same. If registration is cancelled, a symbol is placed here;
• In field Expires indicates the time in seconds during which the registration is valid. If this field is absent, then the default time is assigned - 1 hour, after which registration is canceled. Registration can also be canceled by sending a REGISTER message with the Expires field set to O and the corresponding Contact field.

By request OPTIONS the called user requests information about the functionality of the called user's terminal equipment. In response to this request, the called user's equipment provides the required information. The use of the OPTIONS request is limited to those cases where it is necessary to learn about the functionality of the equipment before establishing a connection. This type of request is not used to establish a connection.

After testing the SIP protocol in real networks, it turned out that the above six types of requests are not enough to solve a number of problems. Therefore, it is possible that new messages will be introduced into the protocol. Thus, the current version of the SIP protocol does not provide a method for transmitting connection control information or other information during a communication session. To solve this problem, a new type of request was proposed - INFO. It can be used in the following cases:

• To transfer PSTN/ISDN/cellular signaling messages between gateways during a conversation session;
• To carry DTMF signals during a conversation session;
• To transfer billing information.

Having completed the description of SIP protocol requests, let us consider, as an example, a typical INVITE type request (Fig. 6). INVITE sip: [email protected] SIP/2.0 Via: SIP/2.0/UDP kton.bell-tel.com From: A. Bell To: T. Watson Call-ID: [email protected] Cseq: 1 INVITE Content-Type: application/sdp Content-Length: ... v=0 o=bell 53655765 2353687637 IN IP4 12&.3.4.5 C=IN IP4 kton.bell-tel.com m=audio 3456 RTP/ AVP 0345

Rice. 6 Example INVITE request

In this example, the user Bell ( [email protected]) calls the user Watson ( [email protected]). The request is sent to the proxy server (boston.bell-tel.com). In the To and From fields, the address is preceded by the entry that the calling user wishes to display on the called user's display. In the body of the message, the calling user's equipment indicates in SDP protocol format that it can receive voice information on port 3456, packaged in RTP packets and encoded using one of the following encoding algorithms: 0 - PCMU, 3 - GSM, 4 - G.723 and 5 - DVI4.

When sending SIP messages packaged within UDP signaling messages, there is a possibility that the size of the request or response will be larger than the maximum allowed for the given network, and packet fragmentation will occur. To avoid this, a compressed format for the main header names is used, similar to what is done in the SDP protocol. Below is a list of such headers (Table 3).

Table 3. Compressed SIP header names

When the header names are written in compressed form, the INVITE message shown earlier in Figure 6 would look like this (Figure 7): INVITE sip: [email protected] SIP/2.0 v: SIP/2.0/UDP kton.bell-tel.com f: A. Bell t: T. Watson i: [email protected] Cseq: 1 INVITE from: application/sdp l: ... v=0 o=bell 53655765 2353687637 IN IP4 128.3.4.5 C=IN IP4 kton.bell-tel.com m=audio 3456 RTP/AVP 0345

Rice. 7 Example of an INVITE request with shortened headers

Request type	Request Description
INVITE	Invites the user to a communication session. Contains SDP session description
ASK	Confirms receipt of the final response to the INVITE request
BYE	Ends the communication session. Can be transmitted by any of the parties involved in the session
CANCEL	Cancels processing of requests with the same Call-ID, To, From, and CSeq headers as the CANCEL request itself.
REGISTER	Transfers address information for user registration on the location server
OPTION	Requests information about terminal functionality

Replies to SIP requests

After receiving and interpreting the request, the recipient (proxy server) transmits a response to this request. The content of the responses varies: confirmation of connection establishment, transmission of the requested information, information about faults, etc. The structure of responses and their types were inherited by the SIP protocol from the HTTP protocol.

Six types of responses have been identified that carry different functional loads. The response type is coded as a three-digit number. The most important is the first digit, which determines the class of the answer; the remaining two digits only complement the first. In some cases, the equipment may not even know all the response codes, but it must interpret the first digit of the response.

All answers are divided into two groups: informational and final. Informational responses indicate that the request is being processed. They are encoded with a three-digit number starting with one - 1xx. Some informational answers, e.g. 100 Trying, are designed to set timers to zero that are started in the equipment that sent the request. If a response to a request is not received by the time the timer expires, then the request is considered lost and can (at the discretion of the manufacturer) be retransmitted. One of the common answers is 180 Ringing; its purpose is identical to the signal in the PSTN and means that the called user receives a signal about an incoming call.

Final responses are encoded as three-digit numbers starting with the numbers 2, 3, 4, 5 and 6. They indicate the completion of request processing and contain, when necessary, the result of request processing. The purpose of each type of final response is discussed below.

Answers 2xx indicate that the request was successfully processed. Currently, of all the 2xx type responses, only one is defined -200 OK. Its meaning depends on what request it answers:

• answer 200 OK to 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;
• answer 200 OK to a request, BYE means the end of the communication session, the response body does not contain any information;
• answer 200 OK for a CANCEL request means cancellation of the search, the response body does not contain any information;
• answer 200 OK to the REGISTER request means that registration was successful;
• answer 200 OK When responding to an OPTION request, it is used to transmit information about the functionality of the equipment; this information is contained in the body of the response.

Answers Zhh inform the calling user's equipment of the new location of the called user or carry other information that can be used for a new call:

• in the answer 300 Multiple Choices several SIP addresses are indicated at which the called user can be found, and the calling user is asked to select one of them;
• answer 301 Moved Permanently means that the called user is no longer at the address specified in the request, and requests must be sent to the address specified in the Contact field;
• answer 302 Moved Temporary means that the user is temporarily (the period of time can be specified in the Expires field) at a different address, which is specified in the Contact field.

Answers 4xx informs that an error has been detected in the request. After receiving such a response, the user should not send the same request without modifying it:

• answer 400 Bad Request means that the request is not understood due to the presence of syntax errors;
• answer 401 Unauthorized means that the request requires a user authentication procedure. There are different authentication options, and the response may indicate which one to use in a given case;
• answer 403 Forbidden means that the server understood the request but refused to serve it. A repeated request should not be sent. The reasons may be different, for example, requests from this address are not serviced, etc.;
• answer 485 Ambiguous means that the address in the request does not uniquely identify the called user;
• answer 486 Busy Here means that 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.

Answers 5xxx informs that the request cannot be processed due to server failure:

• answer 500 Server Internal Error means that the server is unable to service the request due to an internal error. The client may try to resend the request after some time;
• answer 501 Not Implemented means that 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;
• answer 502 Bad Gateway informs that the server operating as a gateway or proxy server has received an incorrect response from the server to which it sent the request;
• answer 503 Service Unavailable indicates that the server cannot currently service the call due to overload or maintenance.

Answers 6xxx informs that a connection with the called user cannot be established:

• answer 600 Busy Everywhere reports that the called user is busy and cannot receive a call at the moment at any of his available addresses. The response may indicate a time appropriate to call the user;
• answer 603 Decline means that the called user is unable or unwilling to accept the incoming call. The response may indicate a suitable time for the call;
• answer 604 Does Not Exist Anywhere means that the called user does not exist.

Requests and responses to them form a SIP transaction. It occurs between the client and the server and includes all messages from the first request to the final response. When using TCP as a transport, all requests and responses related to one transaction are transmitted over one TCP connection.

Figure 8 shows an example response to an INVITE request:

SIP/2.0 200 OK Via: SIP/2.0/UDP kton.bell-tel.com From: A. Bell To: ; Call-ID: [email protected] Cseq: 1 INVITE Content-Type: application/sdp Content-Length: ... v=0 o=watson 4858949 4858949 IN IP4 192.1.2.3 t=3149329600 0 c=IN IP4 bostcon.bell-tel.com m=audio 5004 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000

Rice. 8 Example of SIP response 200 OK

This example shows a Watson user's response to a call invitation from a Bell user. We discussed the most likely format of the invitation earlier (Fig. 7). The called party informs the calling party that it can receive voice information encoded in accordance with PCMU and GSM encoding algorithms at port 5004. The From, To, Via, Call-ID fields are taken from the request shown in Figure 7. From the example you can see that this is a response to an INVITE request with the CSeq:1 field.

After we have examined the requests and responses to them, it can be noted that the SIP protocol provides different connection establishment algorithms. It is worth noting that the same answers can be interpreted differently depending on the specific situation. Table 5 summarizes all responses to requests defined by the SIP protocol.

Table 5. Responses to SIP Requests

Response code	Explanation	Purpose
100	Trying	The request is being processed, for example, the server is accessing databases, but the location of the called user is currently undefined
180	Ringing	The location of the called user is determined. He is given a signal about an incoming call
181	Call Is Being Forwarded	The proxy server redirects the call to another user
182	Queued	The called user is temporarily unavailable, but an incoming call is queued. When the called user becomes available, he will send the final response
200	OK	Command completed successfully
300	Multiple Choices	The called user is available at several addresses. The caller can choose any of them
301	Moved Permanently	The user has changed his location, his new address is indicated in the Contact field
302	Moved Temporarily	The user has temporarily changed his location, his new address is indicated in the Contact field
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)
380	Alternative Service	The call did not reach the destination, but an alternative service option exists, which is indicated in the response body. For example, a call can be forwarded to a voice mailbox
400	Bad Bequest	There was a syntax error in the request
401	Unauthorized	User authorization procedure is required
402	Payment Required	Advance payment for services required
403	Forbidden	The request will not be serviced by the server and should not be retransmitted
404	Not Found	The server did not find the called user in the domain specified in the Request-URI field
405	Method Not Allowed	It is not permitted to send this type of request to the address specified in the Request-URI field. The Allow field of the response indicates the allowed types of requests
406	Not Acceptable	Responses generated by the callee will not be understood by the caller
407	Proxy Authentication Required	The client must confirm its right to access the proxy server
408	Request Timeout	The server cannot send a response, such as the location of the called user, for the period of time specified in the Expires field of the request. The caller can resubmit the request after some time
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
410	Gone	The server no longer has access to the requested resource and does not know where to forward the request
411	Length Required	You must specify the length of the message body in the Content-Length field
413	Request Entity Too Large	Request size is too large to process
414	Request-URI Too Large	The address specified in the Request-URI field was too large to be interpreted
415	Unsupported Media Type	The request contains an unsupported message body format
420	Bad Extension	The server did not understand the protocol extension specified in the Require field
480	Temporarily not available	The called user is temporarily unavailable
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.
482	Loop Detected	The server detected that the request it accepted was transmitted along a closed route (the Via field already contains the address of this server)
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
484	Address Incomplete	The server accepted a request with an incomplete address in the To or Request-URI field. Additional address information required
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
486	Busy Here	The called user is currently unwilling or unable to accept a call to this address. The answer does not exclude the possibility of contacting the user at another address.
500	Internal Server Error	Internal Server Error
501	Not Implemented	The server does not implement the functions necessary to service the request. A response is sent when the server cannot recognize the type of request it has received.
502	Bad Gateway	A server acting as a gateway or proxy receives an incorrect response from the server to which it sent the request
503	Service Unavailable	The server is currently unable to service the call due to overload or maintenance.
504	Gateway Timeout	A server acting as a gateway or proxy 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
505	SIP Version not supported	The server does not support this version of the SIP protocol
600	Busy Everywhere	The called user is busy and does not want to accept the call at this time. The answer may indicate the appropriate time to call
603	Decline	The called user is unable or unwilling to accept incoming calls. The response may indicate a suitable time for the call.
604	Does not exist anywhere	The called user does not exist
606	Not Acceptable	The called user cannot accept the incoming call due to the fact that the type of information specified in the communication session description in SDP format, bandwidth, etc. unacceptable

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 SIP account. Passwords are required to configure the equipment. To obtain passwords, follow these steps:

• Change the password for the SIP account and print the settings when changing the password.
• Send yourself current SIP settings via email.

Set up a SIP account:

• 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 your current SIP account password 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.

Click .

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 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. For 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 the telephone network (PSTN termination), some - on various 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.

When setting up manually, 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/

An example of a hardware SIP solution: 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 the average user, the categories “SIP phones” and “VoIP gateways” will probably be the most useful). 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

While you are working with a virtual PBX from your operator - MSN Telecom, Mango, etc. - you don't need this article. But when transferring a number to another PBX or service, it will help.

What are we talking about?

All OnlinePBX clients use their own numbers in our virtual PBX. Since we don’t sell them and don’t count the minutes.

Someone will say that it is inconvenient to pay two bills: to the operator and to us. And our clients benefit from this: they find a cheap operator or call direction (within Moscow, intercity, abroad, etc.), connect to the PBX and save.

You can connect to almost any operator, the main thing is to get SIP settings from them. These are the parameters that are needed to connect. Since SIP is very similar to email, let’s make an analogy.

What is a SIP login?

For example, let's take a fictitious email on Yandex: [email protected] with password ******. There are actually three parameters:

1. User login: siplogin.
2. His password: ****** .
3. Server address, so as not to be confused with Mail.ru or Google: yandex.ru.

Those who have at least once set up an email client will immediately understand what we are talking about: you are asked to fill out all these parameters. Sometimes you also need to specify ports and separate servers for incoming and outgoing letters.

With SIP everything is the same, only bad with the names. SIP login is called differently: SIP ID, SIP Login, login, SIP account, SIP account, account, user, employee, manager, and so on. It consists of numbers and letters of the Latin alphabet, three characters minimum.

What is it for?

If mail is needed for sending and receiving letters, then a SIP login is for receiving incoming and making outgoing calls.

There is an important point: without an external number (trunk), you will not be able to call landlines and mobile phones, only to the same SIP logins.

What is a trunk in IP telephony?

Imagine a cable with many wires.

Each transaction is an incoming or outgoing communication channel. That's why the numbers are called multi-channel.

The entire cable is a trunk. You connect one “end” of the cable to the PBX, and the second operator connects it to his equipment. Electrical networks also work: if there is a cable, there is voltage.

Let's use an email analogy.

You write an email and send it to your mom. But mom doesn't have an email. A separate company is needed to deliver your letter. She will receive your email, print it on paper and put it in an envelope. Next, the company will find your mother’s address, write it on the envelope and deliver it to the Russian Post on your behalf. If mom responds, this company will receive it and write you an email.

Such a company performs the function of a trunk - connecting the electronic world with the real one.

pros: Mom doesn’t need a computer, the Internet, email and learning to use it. She uses familiar things - a pen, a piece of paper and an envelope. You also use what you are used to - email. And you don’t need to know your mother’s address.
Minuses: You will have to pay for the service.

That is, you need a trunk for calls to external numbers - landline and mobile. You can call SIP logins without a trunk, that’s what we’re talking about.

How to get a SIP login?

We learned that you need an external number (trunk) to make calls. And to connect to other services - a special address - SIP login.

You can obtain a SIP login and other parameters from your telecom operator. Some operators provide a SIP login directly for the trunk, these are MSN Telecom, Telfin, Arentel. Others are not, these are Mango, UISCOM and most other operators.

In the second case, you will have to use the operator’s PBX, then the SIP login is user/employee/extension. Create a new user, not tied to a person, and set a password for it. Remember these details: login, password and server.

Telephony for business with integration into any CRM

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 currently unknown.

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 the incoming call is 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; The OPTION request is used to transmit information about the functionality of the 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, the call may 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. There are different authentication options, 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.