Table of Contents
Definition SIP(Session Initiation Protocol)
SIP (Session Initiation Protocol), is a signaling and control protocol used mostly in IP Telephony systems, which was developed by the IETF (RFC 3261). This protocol allows us to create, modify, and end multimedia sessions with one or more participants and its most significant advantages lie in its simplicity and consistency.
To date, there have been multiple signaling protocols such as ITU H.323, Cisco SCCP, or MGCP. Still, it seems that SIP is gradually winning the battle of the standard: Cisco is progressively adopting SIP as the protocol in its IP telephony systems to the detriment of H.323 and SCCP. Microsoft has chosen SIP as the protocol for its new OCS (Office Communication Server), and operators (mobile and fixed) are also implementing [SIP] within their convergence strategy, taking advantage of the scalability and interoperability provided by the SIP protocol.
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The [SIP] protocol acts transparently, allowing name mapping and service redirection, thus offering the implementation of the IN (Intelligent Network) of the PSTN or RTC.
The SIP protocol has different functions. The most important are listed below
- User localization (SIP provides support for mobility).
- User capabilities (SIP allows parameter negotiation).
- And user availability
- Establishment and maintenance of a session.
In short, the [SIP] protocol allows interaction between devices, which we can achieve with different types of messages typical of the protocol covered in this section. These messages provide capabilities to register and invite a user to a session, negotiate the parameters of a session, establish a communication between two to more devices, and, finally, end sessions.
SIP protocol benefits compared to other protocols
Currently, the most used protocols in ToIP are three: SIP, H.323, and IAX2.
As the main features of H.323 we have:
- Implement QoS internally.
- Conference control
- IAX2 (Inter-Asterisk eXchange) is a protocol created and standardized by Asterisk. Some of its main characteristics are Media and signaling travel in the same data stream.
- Data encryption
The advantages of the protocol are by sending the “streaming” and the signaling through the same data flow, problems derived from NAT can get avoided. Thus, it is not necessary to open port ranges for RTP traffic. Finally, IAX2 allows us to do trunking so that we can send several conversations on the same stream, which means significant bandwidth savings.
Essential aspects regarding said protocol are listed as follows:
- Call control is stateless or stateless and provides scalability between phone devices and servers.
- [SIP] requires fewer CPU cycles to generate signaling messages so that a server can handle more transactions.
- A [SIP] call does depend on a connection at the transport layer.
- SIP supports caller and call authentication using HTTP mechanisms.
- Authentication, cryptography, and encryption are supported hop by hop over SSL / TSL. But SIP can use any transport layer or any HTTP security mechanism, like SSH or S-HTTP.
- A [SIP] proxy can control cell signaling and can branch to any number of devices simultaneously.
In short, we see that [SIP] is a protocol with high scalability, modular, and very apt to become the immediate future of VoIP.
The standard defines various SIP components, and there are several ways to implement them in a call control system.
- User Agent servers,
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