Session Initiation Protocol (SIP) and Signaling System 7 (SS7) are the common protocols used for voice transmission over networks. Just how they work with VoIP .... or not .... it opens the door for both the concerns and opportunities.
Session Initiation Protocol (SIP) is a protocol developed by the IETF working group MMUSIC and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, games online and virtual reality. SIP is a text based, similar to HTTP and SMTP signaling, and its use to create, manage and terminate sessions in an IP-based network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session.
The entities that interact in a SIP scenario are called User Agents (UA) agents may operate in two modes -
or user Agent client (UAC): generates requests and send those to the server.
or User Agent Server (UAS): Gets demands, processes these requests and generating responses.
SIP works as follows: callers and callees are identified by SIP addresses. When making a SIP call, a caller first locates the appropriate server, and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended call, a SIP request may be redirected or may trigger a chain of new SIP requests by proxy. Users can register their location (s) with SIP servers.
Now ..... how this is different than the SS7 protocol?
Here is an explanation Simplied:
Signaling System 7 (SS7) is architecture for performing signaling in support of call-establishment, billing, routing and functions of information exchange of the PSTN network , while SIP is a protocol that is used for the maintenance of VOIP sessions.
SS7 are used to set the vast majority of PSTN phone calls from around the world, where as SIP used in the IP network.
A little 'more about the differences between SS7 and SIP.
SS7 uses a common channel signaling for call setup and tear down the information for the circuit-switched services. It is common to have hundreds or thousands of voice circuits controlled by a pair of links 64 kb / s signaling. SS7 was specifically designed for circuit switched although it has some very sophisticated call control features and additional transaction control.
SIP is an IP-based signaling solution that uses a signaling pathway in part, but is based on IP connectivity from the originator to a server, and then to the end termination. It is used for communications in the basic package and allows many different types of calls such as video, game interaction, etc. as well as voice.
As SIP is realized with the use of next-generation networks are both certain that we will see some very interesting network behavior, countless new technical issues as we iron the bugs out and probably new opportunities for fraud. They should be interesting times.
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