Session Initiation Protocol (SIP) is simply a means of using Voice over IP (VoIP) over multiple trunks with the SIP protocol. SIP is an application-layer protocol that creates, modifies, and terminates sessions with multiple participants. Applications interfacing with this protocol can create two-party (unicast) or multiparty (multicast) sessions.
SIP is different from the VoIP H.323 primarily in that the protocol focuses on point to point connections rather than using a central routing scheme. Think of it like this. VoIP is an IP version of a standard phone system. Each phone on the system is rather dumb and is activated only when the central system routs a call to it. SIP, on the other hand, can utilize a smaller internal structure because most of the work is done on a network’s periphery. With VoIP, it is the central system that handles much of the workload of compression, dialing, and connection work while SIP passes much of this work out to the network ends.
For a call center, this means that the requirements for a central voice server are much smaller while the requirements for the systems in use by the agents could increase. The benefit is that a call center’s server will be always be busy with a variety of other functions. On the other hand, the agents’ systems are generally overpowered for what they need to do. It is just impossible to purchase a computer that is just barely capable of running the agent’s required software. SIP moves much of the workload away from the busy server and puts it on the agents’ systems. By spreading out this workload, SIP based call centers are limited by their total bandwidth rather that the traditional limitations of routing VoIP calls through the central system.
