VoIP (Voice over Internet Protocol) uses signaling protocol to communicate between the phone and other network devices. SIP and H.232 are mainly used protocols to transmit voice conversations over Internet.

Session Initiation Protocol (SIP) is a signaling protocol for Internet Telephony, widely used to establish voice and video calls in IP networks.

SIP Protocol was developed by the IETF as a real-time communication protocol for IP voice. SIP Protocol can be used for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality.

SIP Protocol performs basic call-control tasks such as setting up and tearing down (signaling) voice and video calls over the Internet. SIP controls signaling for calling features such as Call Transfer, Call Hold and Caller ID.

SIP works with Session Description Protocol (SDP) and Real-time Transport Protocol (RTP) for communication between multimedia devices.

H.323 is an ITU standard protocol for multimedia communications, which includes voice, video, and data conferencing, for use over packet-switched networks that defines how real-time audio, video and data information is transmitted. H.323 is commonly used in VoIP, Internet Telephony, and IP-based videoconferencing.

H.323 is based on the Internet Engineering Task Force (IETF) Real-Time Protocol (RTP) and Real-Time Control Protocol (RTCP), with additional protocols for call signaling, and data and audiovisual communications.

Both protocols have their advantages and disadvantages. Many vendors now support both protocols, as their customers want flexibility to choose based on different needs.