Today, the switching of telephony to VOIP (Voice over IP) is in full swing. As a basic protocol, SIP (Session Initial Protocol) is the one most often used. The versatility of the protocol can cause confusion – it can be connected to both ordinary subscriber terminals (SIP phones, SIP adapters, soft phones, etc.), and VoIP nodes (from IP PBX to Softswitch and Gateways with E1 etc.). The following are important connection characteristics for SIP.
The terminal and the node perform various functions in the VoIP network and such connections have different properties.
The type of connection (SIP trunk or SIP subscriber line) directly depends on the role of the SIP device on the network, the client devices can act as a terminal or node.
A terminal is a terminal device with which the user makes or receives calls, for example, a SIP phone, an ATA adapter or a softphone.
The SIP subscriber line has properties similar to the traditional subscriber line. For example, you can hardly accept more than two calls to one terminal at a time.
Mobility. Unlike the traditional analog line, which is physically attached to a specific location, for example, an apartment or office, the SIP subscriber line is mobile – in order to make a call, it is enough to have an Internet connection that meets the requirements for high-quality transmission of signal and voice information.
When the terminal is switched on, it sends a special registration packet containing all the necessary data for identification and authorization of the telephone line. With this package, the terminal reports the IP address of the SIP line to the server located on the service provider side. Now, when incoming calls, the server “knows” where to send the call addressed to this line. After a certain period of time, the terminal updates the registration, in the event that the IP address through which the device is registered changes, the information on the IP-telephony server is updated. Each time an outgoing call is made, the IP telephony server requests data for authentication (user name, login, and password).
A small amount of traffic from the terminal. Usually, the terminal can accept no more than two simultaneous calls since it is unlikely that the SIP phone user will need to receive 5 simultaneous calls.
The need for additional services. Since the terminal is a terminal device, additional services can be implemented for the SIP subscriber line. For example, through a web page on an IP telephony server, the user can configure services such as forwarding, voice mail, Follow Me, Find Me, Fax-to-email, and so on.
So, the general properties of the subscriber line: mobility and, as a consequence, the need to register, in general, up to two simultaneous calls from the line, often the need for additional services.
To connect a subscriber line via SIP, you need a logical account on the IP telephony server and the corresponding data that is written to the terminal configuration. This record is called a SIP account or SIP account or account (from a SIP account, English) or simply a SIP subscriber line.
This is an IP PBX or another VoIP device that serves as an intermediate in the VOIP network and is not a terminal device in the telephone network, for example, an IP PBX node to which the above-described terminals (e.g. SIP phones) are connected in the office.
Nodes have the following properties:
Stationarity or lack of mobility of the node. Usually, IP PBX has a static public IP address, which does not change. Therefore, the initial level of security can be provided by resolving incoming and outgoing calls for the IP address of the client node, provided that the caller (s) and the called number are correctly transmitted in SIP messages. Since the IP address of the node never changes, there is no need to register on the IP-telephony server; instead, a static record is created on the server. Often for reliability, additional password authorization is used for incoming and outgoing calls, this significantly increases the security of the connection.
Implementation of all additional services on the site. Usually, for this connection, the client does not need any additional services and services, because they are all implemented on the node (IP PBX) from the client to which the terminals are connected. Terminals (for example, SIP phones) in turn “take” services from the node.
A large number of calls between the node and the IP-telephony server. Since the node is not a terminal device and there may be a large number of terminals behind it (example: corporate IP PBX with internal subscribers). All of them ultimately generate a large amount of traffic, so the customer’s need can be tens or hundreds of simultaneous calls through such a node.
So, the node has the property of stationarity, all additional services are implemented on the client’s site, and not on the side of the service provider and generate a lot of calls (telephone traffic), the customer’s need can make tens or hundreds of simultaneous calls.
Such node connections to the network are called SIP Trunking, SIP trunk or static SIP trunk, SIP trunking ports. In other words, the SIP trunk is a logical record on the service provider equipment and the corresponding settings that it sends to the client to write them to the node configuration.