The 3G-324M protocol for conversational video telephony

2004

In this paper we survey the conferencing architectures in H.323 and SIP. We give a compact review of the H.323 framework and capture the current status of the emerging SIP conferencing architecture. Subsequently we present a qualitative comparison of the schemes under study. The evaluation criteria are scalability to large conference sizes and to large geographical dispersion of participants ,as well as adaptiveness to heterogeneity in media streams format and protocol implementation. Our findings suggest that both architectures have certain advantages and limitations. Wherever possible we propose measures to overcome the latter. Concluding, we propose future work to mitigate the identified gaps in the SIP conferencing framework.

2005

Abstract The Global Multimedia Collaboration System is designed to bridge H. 323, Session Initiation Protocol, Access Grid clients, and 2. SG/3G cellular phones in audio-visual collaborations. One of its key components is the H. 323 gateway, which lets H. 323 terminals interact with other clients, providing complete conferencing control services. This system is designed to help scientists and students who have access only to H. 323 and SIP endpoints or cellular phones with scientific collaboration in the high-end Access Grid environment.

Computer Communications, 2000

Internet Protocol (IP) based communication is fast becoming a viable alternative for voice communications. The Intelligent Network (IN) represents the world wide accepted basis for the uniform provision of advanced telecom services. Mobile Agents offer unique opportunities for structuring and implementing open distributed service architectures, facilitated by the dynamic downloading and movement of service code to specific network nodes. In this paper, a new service architecture for IP telephony, based on the ITU-T standard H.323, is proposed. The implementation uses Mobile Agents as an enabling technology and existing architectural concepts taken from IN. This IP service architecture enables telecom services deployed through mobile service agents on a per user basis, which results in several advantages when compared to centralized service architectures.

2000

There are currently two standards for signaling and control of Internet telephone calls, namely ITU-T Recommendation H.323 and the IETF Session Initiation Protocol (SIP). We describe how an interworking function (IWF) can allow SIP user agents to call H.323 terminals and vice versa. Our solution addresses user registration, call sequence mapping and session description. We also describe and compare various approaches for multi-party conferencing and call tranfer.

2012

Intellectual mobile terminals (or users) of next generation wireless networks are expected to initiate/Establish voice over IP (VoIP) calls using session set-up protocols like H.323 or SIP (Session Initialized Protocols). To provide quality metrics of video conferences, telemedicine and other voice over broadband telephony (VoBB) applications. In this work, we analyse the performance of the H.323 call setup procedure over the wireless link. We used to call modes of operation over heterogeneous networks. The proposed model application layers in the RTP Control Protocol and Real-Time Transport Control Protocol (RTCP) protocols used in two different modes of call established. Initiate services through VoIP used for H.323 control packets. Our analytical model provides that the VoIP call set-up performance, jitter and delay in peer to peer networks. Moreover, the call setup performance can be improved significantly using the robust in application link layer such as RTP/RCTP with a comparison of heterogeneous network proposed in our paper. The analytical results were validated by our experimental measurements

2002

Abstract This paper presents the architecture and design of an enhanced VoIP gateway based on the H. 323 protocol. The key characteristics of this gateway include: scalability by separating signaling from media transformation and use of media gateway control protocols such as MGCP/MEGACO for communication between the signaling part and the media gateway; lightweight property by using the transport protocol SCTP (stream control transmission protocol) to convey H.

2006

Abstract:-This paper presents a lightweight VoIP platform based on H. 323/MGCP interworking and QoS capabilities. The lightweight feature is accomplished by introducing two new schemes in the platform; the use of the SCTP as a transport protocol and the use of the Compressed-RTP as a compressed implementation of the RTP. The MGCP/H. 323 interworking provides the platform with the essential scalability so that many different VoIP scenarios can be implemented.

Mobile Information Systems, 2007