One advantage of the traditional landline services is that electrical power is sent over the telephone wires so your phone service is isolated from your house power. This meant that your phone service would continue to work if your house power went out. However, with VoIP, power is used not only for the ATA, or the IP phone, but it is also used for your Internet modem and router devices. No power also typically means no Internet service.
What makes SIP so popular is not only that it's deep and flexible, but also because it was purpose-built to engage in multimedia (meaning not just audio but also video and even text) communications over TCP/IP networks. For VoIP calls, SIP can set up calls using a number of IP-related protocols, including the Stream Control Transmission Protocol (SCTP), the Transmission Control Protocol (TCP), and the User Datagram Protocol (UDP), among others. But it can also handle other functions, including session setup (initiating a call at the target endpoint—the phone you're calling), presence management (giving an indicator of whether a user is "available," "away," etc.), location management (target registration), call monitoring, and more. Despite all that capability, SIP is simple compared to other VoIP protocols primarily because it's text-based and built on a simple request/response model that's similar in many ways to both HTTP and SMTP. Yet, it's still capable of handling the most complex operations of business-grade PBXes.
Typically, price is one of the most important reasons people opt for residential VoIP. One of the most attractive is the "triple play" sales pitch we mentioned above made by almost every regional residential cable company and internet provider: Get your Internet, TV, and phone service all rolled into one monthly charge. Not only is that usually an attractive number, it also means a technician will hook everything up for you including your phone, and you'll probably be able to use the same phone you're using now instead of having to migrate to a VoIP phone.
While understanding the basics of VoIP and SIP is important, setting one of these systems up will require some general network knowledge, too. For the best quality, you will need to meet a minimum upstream and downstream data throughput requirement. In addition, you'll also need to meet a minimum latency number (that is, the time between when a signal leaves a remote computer and when your system receives it), typically measured in milliseconds. It is possible to test your network connection to see if it will support a VoIP service. RingCentral offers this service from their website, other vendors like to have their service engineers do it for you.