ny options for advanced telephony proach (inefficient architecture) features Slow moving Good velocity Industry Established now, primarily system Rapidly growing industry momentum acceptance level (system level) Few if any H:323 base telephones Growing interest in SIP phones and soft clients End-user primarily driven by Micro- soft (NetMeeting), Intel, etc. Undoubtedly, SIP is poised as the most appropriate protocol to enable PS video teleph- ony in UTMS. At this writing, technical bodies are debating the final outcome. From the author's point of view, it seems evident that SIP would lead to better results and widespread usage of video telephony. 6.4.1.5 Layer Structure Enabling for Multimedia - MEGACO/H.248 MEdia GAteway COntrol (MEGACO) or H.248 is part of the protocols that will facili- tate the control of video telephony on the PS side. Megaco/248 jointly developed by ITU TG-16 and IETF, covers all gateway applications moving information streams from IP networks to PSTN, ATM and others. These include: PSTN trunking, gateways, Service Components in UMTS ATM interfaces, analog line and telephone interfaces, announcement servers, IP phones, and many others. The Megaco IP phone master/slave approach is entirely compatible with peer-level call control approaches such as SIP and H.323. It acts orthogonal to the last two protocols. Megaco/H.248 allows: profiles to be defined, i.e. permits application level agreements on gateway organi- zation and behaviour to be made for specific application types, thereby reducing complexity; allows support of multiple underlying transport types (e.g. ALF reliability layer over UDP, TCP), and both text and binary encoding; the latter enables more appro- priate support for a broader range of application scales (e.g. big VS. small gateways) and more direct support for existing systems. 6.4.1.6 IETF Signalling Transport (SIGTRAN) SIGTRAN develops an essential Simple Control Transmission Protocol (SCTP), which we view as a layer between the SCTP user application and an unreliable end-t