摘要 :
Voice over IP (VoIP) is a generic name for ser- vices, systems and technology for telephony over an IP network. It is also referred to as Internet telephony and IP (Internet Pro- tocol) telephony. Internet telephone client softwar...
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Voice over IP (VoIP) is a generic name for ser- vices, systems and technology for telephony over an IP network. It is also referred to as Internet telephony and IP (Internet Pro- tocol) telephony. Internet telephone client software attracted at- tention when it first appeared in 1995. Since that, VoIP has rapidly matured into a practical technology, propelled by the popularization and rapid development of the Internet. IP net- work traffic already exceeds telephone network traffic and is ex- pected to further increase several-fold in the next few years.
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摘要 :
Progress being made in the development of multimedia network environments with net- worked multimedia PCs is leading to the day when LAN-based real time voice communication systems can be achieved. The major problems with LANs are...
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Progress being made in the development of multimedia network environments with net- worked multimedia PCs is leading to the day when LAN-based real time voice communication systems can be achieved. The major problems with LANs are delay, jitter and packet loss due to their best effort transfer characteristics; accordingly, the major issues of LAN-based systems are how to reduce end-to-end delay, absorb jitter and recover packet loss in order to provide high quality bi-directional real time voice communication. In this paper, a LAN-based real time voice communication system using IP protocols with adaptive jitter control is proposed. The advantageous characteristics of this system are the introduction of an adaptive jitter con- trol mechanism that minimizes end-to-end delay by optimizing jitter buffer size, the adoption of small algorithm delay hardware codecs, and the use of a TCP (UDP)/IP environment for voice communication in order to integrate data and voice applications. Results obtained in tests on a prototype system show that delay of less than 100 ms is achieved, which satisfies the ITU-T G.114 Recommendation of 150 ms as an asceptable range for bi-directional realtime voice communication. In addition, stable voice quality is ashieved that is little affected by the disturbance caused by dynamic changes in network load.
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摘要 :
We propose a P2P file sharing system that enables flexible and intuitive information sharing and management among large group of users. The proposed system (NRBS: Network Resource Browsing System) supports a virtual directory that...
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We propose a P2P file sharing system that enables flexible and intuitive information sharing and management among large group of users. The proposed system (NRBS: Network Resource Browsing System) supports a virtual directory that allows users to organize and manage distributed files based on simple and intuitive user interface. The system has a central management server that controls each user client in the system, which allows centralized security management and strict content control. In this paper, we compare conventional approach for managing P2P file sharing, and then propose a system based on virtual directory. We explain the mechanism for associating links on the virtual directory to actual file data stored in user clients' terminals. We evaluate the system based on usability, manageability, and security. The result shows that using virtual directory improves the usability and manageability while providing strict file security.
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摘要 :
IP telephony systems are expected to be de- ployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network ...
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IP telephony systems are expected to be de- ployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to im- prove bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, be- cause there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IP-GWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice pack- ets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing in- terval period into a single UDP packet. The advantage of this method is that no new additional header is required and the cur- rent well-defined H.323 and RTP standards can be applied with minimum changes. We implemented and tested the system. The results show that the proposed method is effective at reducing both the header overhead and the number of packets. In a typi- cal case, the bandwidth is cut by 40% for eight G.723.1-encoded voice streams through header overhead reduction and the num- ber of voice packets is also decreased to 1/8. Furthermore, this method can easily be enhanced to a general RTP packet multi- plexing method that is applicable not only to an IP-GW but also to other RTP multiplexing and de-multiplexing applications.
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摘要 :
VoIPサーゼスの提供開始に伴い，電話以外のサーゼスと連携した付加サービスの提案が望まれている．付加サーゼスを実現する方法の一つに，呼制御機能を持つサーバから端末間の通話を確立する3rd Party Call Control がある．本研究ではIETFで標準化が...
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VoIPサーゼスの提供開始に伴い，電話以外のサーゼスと連携した付加サービスの提案が望まれている．付加サーゼスを実現する方法の一つに，呼制御機能を持つサーバから端末間の通話を確立する3rd Party Call Control がある．本研究ではIETFで標準化が進められているSIPによる3rd Party Call Controlサーバと，これを用いた多者通話システムの実装を行い，有用性を確認した．
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摘要 :
VoIPサーゼスの提供開始に伴い，電話以外のサーゼスと連携した付加サービスの提案が望まれている．付加サーゼスを実現する方法の一つに，呼制御機能を持つサーバから端末間の通話を確立する3rd Party Call Control がある．本研究ではIETFで標準化が...
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VoIPサーゼスの提供開始に伴い，電話以外のサーゼスと連携した付加サービスの提案が望まれている．付加サーゼスを実現する方法の一つに，呼制御機能を持つサーバから端末間の通話を確立する3rd Party Call Control がある．本研究ではIETFで標準化が進められているSIPによる3rd Party Call Controlサーバと，これを用いた多者通話システムの実装を行い，有用性を確認した．
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摘要 :
VoIP service has been taken off. However, many people want to subscribe additional services such as WWW integrated VoIP service, unified messaging service and so on. In order to implement additional service, 3pcc (3rd Party Call C...
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VoIP service has been taken off. However, many people want to subscribe additional services such as WWW integrated VoIP service, unified messaging service and so on. In order to implement additional service, 3pcc (3rd Party Call Control) by SIP standardized in IETF is investigated. In 3pcc, call control server establishes call session between two clients. We implemented 3pcc server and showed the usefulness by applying it to multipoint call control system.
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摘要 :
VoIP service has been taken off. However, many people want to subscribe additional services such as WWW integrated VoIP service, unified messaging service and so on. In order to implement additional service, 3pcc (3rd Party Call C...
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VoIP service has been taken off. However, many people want to subscribe additional services such as WWW integrated VoIP service, unified messaging service and so on. In order to implement additional service, 3pcc (3rd Party Call Control) by SIP standardized in IETF is investigated. In 3pcc, call control server establishes call session between two clients. We implemented 3pcc server and showed the usefulness by applying it to multipoint call control system.
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