Automatic Addition and Deletion of Clients in VoIP Conferencing
نویسندگان
چکیده
In a conference, considering the packets only f i o m a set of selected clients can reduce the degradation of the quality of speech because mixing packets from all clients can lead to lack of clarity in the speech of any participants. The automatic selection should be smooth arid should not cause frequent interruptions. A method of selecting the clients f o r mixing is suggested here based on a new quantifier of the voice activity called Loudness Number (LN). The dependence of the Loudness Number on the amplitude of the packet at present and the past activity is clearly brought out. The structure of the packet used has been explained. A method to avoid echo and enhance the quality of the conference is presented. The contributions of the paper are expected to aid in the implementation of H.323 recommendations fo r the Multipoint Processors (MP). A working prototype based on the proposed Loudness Number is already functional.
منابع مشابه
A Scalable Distributed VoIP Conferencing Using SIP
Session Initiation Protocol seems to be the preferred standard for Voice over IP. In this paper, we consider the limitations of existing SIP conferencing methods and propose a distributed architecture using Controllers (SIP Proxy Servers) and Conference Servers which facilitates the control and media handling of a VoIP conference. The Conference Servers are designed on the basis of H.323’s Mult...
متن کاملA Scalable Architecture for VoIP Conferencing
1: Real-Time services are traditionally supported on circuit switched network. However, there is a need to port these services on packet switched network. Architecture for audio conferencing application over the Internet in the light of ITU-T H.323 recommendations is considered. In a conference, considering packets only from a set of selected clients can reduce speech quality degradation becaus...
متن کاملService oriented architecture for VoIP conferencing
Voice/Video over IP (VoIP) systems to date have been either highly centralized or dependent on the IP multicast in nature. Global Multimedia Collaboration System is a scalable, integrated and service-oriented VoIP conferencing system, based on the XGSP collaboration framework and NaradaBrokering messaging middleware. This system can provide media and session services to heterogeneous endpoints ...
متن کاملA Group Synchronization Algorithm for VoIP conferencing
Real Time audio conferencing in IP based packet network is one of the popular applications in these days. There are some serious synchronization problems in this technology, some of them are inter stream, intra stream and group synchronization. This paper deals with the problem of audio group synchronization in distributed multiparty group conferencing. When more than one participant in an audi...
متن کاملTandem - Free VoIP Conferencing : A Bridge to Next - Generation Networks
This article surveys approaches to teleconferencing in voice over IP networks. The considerations for conferencing include perceived quality, scalability, control, and compatibility. Architectures used for conferencing range from centralized bridges to full mesh. Centralized conference bridges used with compressed speech degrade speech quality when multiple talkers are mixed and subjected to ta...
متن کامل