SIP Functions and Features

When SIP was developed, it was designed to support five specific elements of setting up and tearing down communication sessions. These supported facets of the protocol are:

• User location, where the endpoint of a session can be identified and found, so that a session can be established
• User availability, where the participant that’s being called has the opportunity and ability to indicate whether he or she wishes to engage in the communication
• User capabilities, where the media that will be used in the communication is established, and the parameters of that media are agreed upon
• Session setup, where the parameters of the session are negotiated and established
• Session management, where the parameters of the session are modified, data is transferred, services are invoked, and the session is terminated

Although these are only a few of the issues needed to connect parties together so they can communicate, they are important ones that SIP is designed to address. However, beyond these functions, SIP uses other protocols to perform tasks necessary that allow participants to communicate with each other.

User Location

The ability to find the location of a user requires being able to translate a participant’s username to their current IP address of the computer being used. The reason this is so important is because the user may be using different computers, or (if DHCP is used) may have different IP addresses to identify the computer on the network. The program can use SIP to register the user with a server, providing a username and IP address to the server. Because a server now knows the current location of the user, other users can now find that user on the network. Requests are redirected through the proxy server to the user’s current location. By going through the server, other potential participants in a communication can find the user, and establish a session after acquiring their IP address.

User Availability

The user availability function of SIP allows a user to control whether he or she can be contacted. Users can set themselves as being away or busy, or available for certain types of communication. If available, other users can then invite the user to join in a type of communication (e.g., voice or videoconference), depending on the capabilities of the program being used.

User Capabilities

Determining the user’s capabilities involves determining what features are available on the programs being used by each of the parties, and then negotiating which can be used during the session. Because SIP can be used with different programs on different platforms, and can be used to establish a variety of single-media and multimedia communications, the type of communication and its parameters needs to be determined. For example, if you were to call a particular user, your computer might support video conferencing, but the person you’re calling doesn’t have a camera installed. Determining the user capabilities allows the participants to agree on which features, media types, and parameters will be used during a session.

Session Setup

Session setup is where the participants of the communication connect together. The user who is contacted to participate in a conversation will have their program “ring” or produce some other notification, and has the option of accepting or rejecting the communication. If accepted, the parameters of the session are agreed upon and established, and the two endpoints will have a session started, allowing them to communicate.

Session Management

Session management is the final function of SIP, and is used for modifying the session as it is in use. During the session, data will be transferred between the participants, and the types of media used may change. For example, during a voice conversation, the participants may decide to invoke other services available through the program, and change to a video conferencing. During communication, they may also decide to add or drop other participants, place a call on hold, have the call transferred, and finally terminate the session by ending their conversation. These are all aspects of session management, which are performed through SIP.

SIP URIs

Because SIP was based on existing standards that had already been proven on the Internet, it uses established methods for identifying and connecting endpoints together. This is particularly seen in the addressing scheme that it uses to identify different SIP accounts. SIP uses addresses that are similar to e-mail addresses. The hierarchical URI shows the domain where a user’s account is located, and a host name or phone number that serves as the user’s account. For example, SIP: myaccount@madeupsip.com shows that the account myaccount is located at the domain madeupsip.com. Using this method makes it simple to connect someone to a particular phone number or username.

Because the addresses of those using SIP follow a username@domainname format, the usernames created for accounts must be unique within the namespace. Usernames and phone numbers must be unique as they identify which account belongs to a specific person, and used when someone attempts sending a message or placing a call to someone else. Because the usernames are stored on centralized servers, the server can determine whether a particular username is available or not when a person initially sets up an account.

URIs also can contain other information that allows it to connect to a particular user, such as a port number, password, or other parameters. In addition to this, although SIP URIs will generally begin with SIP:, others will begin with SIPS:, which indicates that the information must be sent over a secure transmission. In such cases, the data and messages transmitted are transported using the Transport Layer Security (TLS) protocol