Abstract

IP telephony is on its way to succeeding the World Wide Web as the next breakthrough application for all-IP networks. Two standards are currently competing for dominance in IP telephony - the H.323 protocol suite by the International Telecommunication Union (ITU) and the session initiation protocol (SIP) by the Internet Engineering Task Force (IETF). Both protocols exchange multimedia data over the real time protocol (RTP), so the choice of protocol suite does not influence Internet telephony QoS. However, H.323 and SIP do differ in a range of implementation aspects. The article examines these differences. It finds that SIP has clear advantages over H.323. It is more powerful, modular and extendable. It has been gaining in popularity, especially in North America and with new entrants into the VoIP market. However, H.323 has the advantage of being implemented first, which resulted in many of the early VoIP adopters embracing it. This situation makes it hard to simply conclude on one winning technology over the other. Instead, there will be a need for interoperability between SIP and H.323 signaling, for interconnection between different existing VoIP networks and PSTN/ISDN legacy deployments.

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