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SIP (Session Initiation Protocol) is a communication protocol through computer networks that allows the transmission of multimedia information either via the Internet or through a local network. It requires the use of a computer that has the SIP server role. It first appeared in 1996 as a protocol for teleconferences. SIP has an important place in online telephony because it does not bind the user to a specific provider, such as Skype, if it has the knowledge to use it individually, or through the hundreds of VoIP providers with SIP support.
In order to make simple use of SIP-based technology, a SIP (Voice Phone with SIP support) needs a SIP-enabled phone where it is possible to connect independently to the modem or router on our ADSL line or other fast internet provisioning.
Then it is possible to select a provider for the passage of phone calls over the internet. There are also sip devices that allow the entry of simple phone devices (PSTN network) called ATA.

SIP is the Session Initiation Protocol.  In IP and traditional telephony, network engineers have always made a clear distinction between two different phases of a voice call.  The first phase is "call setup," and includes all of the details needed to get two telephones talking.  Once the call has been setup, the phones enter a "data transfer" phase of the call using an entirely different family of protocols to actually move the voice packets between the two phones.  In the world of VoIP, SIP is a call setup protocol that operates at the application layer.  You may have also heard of H.323, an ITU protocol with similar function.

 

SIP is a very flexible protocol that has great depth.  It was designed to be a general-purpose way to set up real-time multimedia sessions between groups of participants.   For example, in addition to simple telephone calls, SIP can also be used to set up video and audio multicast meetings, or instant messaging conferences.  In this document, we'll focus on SIP's capabilities for VoIP, and how it sets up calls that then use RTP (the Real-time Transport Protocol) to actually send the voice data between phones.

 

SIP also has a wide range since it does nothing more than handle call settings. The following table shows the five main functions within the SIP in terms of VoIP.
To give you an idea of how simple SIP is, we have included a SIP message here:

 

INVITE sip:This email address is being protected from spambots. You need JavaScript enabled to view it. SIP/2.0

Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK77ds

Max-Forwards: 70

To: 21022XXXXX <sip:This email address is being protected from spambots. You need JavaScript enabled to view it.>;

From: 00357XXXXXXXX <sip:This email address is being protected from spambots. You need JavaScript enabled to view it.>;;tag=1928301774

Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.

CSeq: 314159 INVITE

Contact: <sip:This email address is being protected from spambots. You need JavaScript enabled to view it.>;

Content-Type: application/sdp

Content-Length: 142