Due to its wide compatibility between devices and because it is an open and flexible protocol, SIP, considered the standard protocol, is the main star when it comes to VoIP (Voice over IP).
Developed by IETF (Internet Engineering Task Force), in 1999, as an alternative to H.323 implemented by the ITU (International Telecommunication Union) in 1996, SIP, or Session Initiation Protocol, is a communication protocol used to initiate, modify, maintain and terminate voice, video and even instant messaging sessions over IP networks.
The flexibility of integration with other communication protocols was a major advantage over H.323. This advantage led to the growth of SIP systems, becoming a standard in the world of voice communications, thus creating better interoperability between other systems. Another major advantage was its scalability, as it was designed to handle multiple users and sessions simultaneously, and also the ability to handle various types of networks (public, corporate, or virtual private networks (VPN).
SIP is based on a client-server architecture operating at the application/session layer, performing signaling between the two devices. It is important to remember that SIP is only responsible for signaling, which is the responsibility of the RTP (Real-time Transport Protocol) the transmission of data media, i.e., audio, video, etc. For signaling to occur smoothly, the SIP client sends a request (message) to the server that locates the recipient of the call, thus establishing a session.
According to RFC 3261, the initial message must contain the following header fields mandatory:
1. From: You must identify the origin of the message;
2. To: Identifies the destination of the message;
3. Call-ID: This is a unique identifier for each SIP session (used to correlate the initial message with the following responses);
4. CSeq: The sequence number of messages during the session;
5. Max-Forwards: Specifies the maximum number of “hops” that the message can travel;
6. Via: Records the path taken by the message;
7. Contact: Specifies the sender's direct address.
There are other optional headers used in the protocol, however, understanding these mandatory header fields helps us understand and analyze possible signaling errors.
Another type of analysis that can be performed is validating the SIP message type. There are different message types in the protocol, the most common of which are:
REGISTER: Used to register a user's address on the server.
INVITE: Placed on the “Request-Line”, it is with this that it is possible to generate SIP sessions.
ACK: An abbreviation for acknowledge, used to confirm receipt of a response.
CANCEL: Used to cancel a pending request.
BYE: Message to end an existing session.
OPTIONS: Similar to the PING command, used to test the availability and responsiveness of a target.
In short, SIP is one of the pillars of the network RCTS VoIP, is a fundamental protocol for unified communications and other real-time communication applications over the internet. Its use allows for cost reduction compared to traditional telephony systems. Due to its implementation flexibility, it not only becomes the standard protocol for VoIP but also becomes customizable to meet the specific needs of each organization.
