With any new technology come new standards and protocols needed to make the products and services from different vendors interoperable.

The main focus of VoIP standards is on signaling protocols for making VoIP calls: H.323 and SIP (Session Initiation Protocol). For voice transport both use the same standard RTP/UDP/IP. The other players in the standard battle are: MGCP and Megaco (H.248)

Basically there are 2 approaches: One is modeled on the traditional PSTN network, the 'Smart Core/Dumb Edge' and the other is based on the data network model, the 'Dumb Core/Smart Edge.' In this approach, the IP 'cloud' carries call control messages and media streams between telephony applications housed in intelligent endpoints. In addition to call setup and control, the protocols must address QoS for multimedia applications as it provides elements such as timestamp and control well as protocols to introduce new value added services.

H.323 standard, developed by the International Telecommunications Union (ITU-T) came first and is currently most widely deployed. It is based on the Integrated Services Digital Network (ISDN) Q.931 protocol, which allows it to easily interoperate with legacy voice networks such as the PSTN or Signaling System 7 (SS7). It covers all aspects of the VoIP call setup, tear down and routing. The call-control devices in an H.323 network are called gatekeepers, which act as a single point for coordinating routing, resource allocation, billing etc.

Later a new protocol that emulated the switch-centric architecture (dubbed softswitch) of the Public Switched Network and would better permit service providers to mix and match the existing circuit-switched and the new packet technologies was developed by a joint activity of the ITU-T and the Internet Engineering Task Force (IETF). Initially this effort was called MGCP (Media Gateway Control Protocol) and later Megaco (also for Media Gateway Control). This work Softswitch is specially favored in Voice over Cable community.

The Session Initiation Protocol (SIP) is being developed by IETF from the ground up as a peer-to-peer protocol and is sufficiently lightweight to work effectively over the Internet or on LANs. SIP is now a key strategy for bringing Voice over IP to the enterprise and it will also play a central role in the architectures of many carriers. In the years ahead, it is possible that the softswitch protocols will merge with SIP for use in carrier deployments, as service providers choose the protocol that is most suitable for each specific set of infrastructure and service needs. SIP utilizes Internet protocols such as HTTP for message formatting and DNS/URL for naming, which makes it a popular choice going forward. Although similar in principle, SIP uses proxy servers rather than gatekeepers for call handling.

Another up and coming trend is VoIP over WiFi. The combination of the IP phone with SIP and mobility seems to be a natural.

The resources provide more details of the VoIP protocols.