SIP is currently the most commonly implemented signaling protocol for VoIP services. Its popularity and wide distribution make it a de facto standard when dealing with VoIP signaling protocols. One alternative protocol would be H.323, which has a much smaller distribution and a more historical value.

The utilization of SIP, however, is not limited to telephony. SIP can be utilized anywhere, where the signaling for data streaming is necessary or desired.

SIP provides only a part of the functions needed for VoIP (the signaling, which contains the start and end of a call), is however a basic function. For example, SIP negotiates at call start which technology is available for use for the actual call.

SIP is the basis for enabling Voice over IP. Therefore it is possible to use the infrastructure of the internet for telephony. Thus an ideal solution could be achieved, avoiding the classical telephone infrastructure (PSTN). With such an ideal solution, calls could be made between VoIP end points (for example, between Branch offices, field offices, home offices) solely via the internet, avoiding all telephone costs.

From conventional telephone networks (PSTN), SIP users can be contacted via gateways, and vice versa SIP users can reach PSTN users who do not use VoIP technology.

The sound quality for VoIP has increased so well, that you have most likely been called from a SIP user, and not noticed andy difference.