Recently problems with commercial IP telephony systems have been reported one after another, in Japan. One of the important causes is congestion in the control plane. It has been recognized that with the current Internet it is important to control not only congestion caused by overload of the data plane but also congestion caused by overload of the control plane. In particular, "retry traffic," such as repeated attempts to set up a connection, tends to cause congestion. In general, users make repeated attempt to set up connections not only when the data plane is congested but also when the control plane in the network is overloaded. The latter is caused by user behavior: an increase in the waiting time for the processing of connection establishment to be completed tends to increase his or her initiation of reattempts. Thus, it is important to manage both data plane and control-plane resources effectively. In this paper, we focus on RSVP-based communication services including IP telephony, and introduce a model that takes account of both data-plane and control-plane systems, and we examine the behavior of retry traffic. In addition, we compare the system stability achieved by two different resource management methods, the hard-state method and the soft-state method.

Grouping multiple voice frames into a single IP packet ("frame grouping") is a commonly mentioned approach to saving bandwidth in IP telephony. But little is known as to when, how, and how much frame grouping should be done in Internet environment. This paper explores the feasibility and the methods of frame grouping based on Internet delay measurement. Specifically, we propose an adaptive frame grouping method that minimizes the delay violation while reducing the bandwidth usage by as much as a factor of two under real Internet delay fluctuations. The performance of the method is evaluated as it is used against a single voice stream and then against multiple voice streams.

Voice over IP (VoIP) is a generic name for services, systems and technology for telephony over an IP network. It is also referred to as Internet telephony and IP (Internet Protocol) telephony. Internet telephone client software attracted attention 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 network traffic already exceeds telephone network traffic and is expected to further increase several-fold in the next few years. In future, the telephone network will be integrated into the IP network and telephony will become entirely VoIP. There are three expectations for VoIP. The first is inexpensive telephone service. The second expectation is for integrated telephony and IP network services such as a CTI (Computer Telephony Integration) system in which there is interworking with various Internet applications, such as e-mail and Web call-back for communication services of greater convenience rather than simple replacement of the telephone. The third expectation is for a platform for providing high-quality voice communication, multicast communication, and other such enhanced voice services that have a high degree of freedom. However, many problems remain to be overcome before the VoIP System is realized. The main problems are real-time transmission of voice that allows a smooth conversation, session control for providing a variety of services, and the proposal of new services. In this paper, we give an overview of VoIP and the problems that must be solved in order to realize it and propose some solutions regarding stream control and applications. We also describe session control and other topics that are being discussed in standardization forums.

IP telephony systems are expected to be deployed 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 improve 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, because 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 packets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing interval period into a single UDP packet. The advantage of this method is that no new additional header is required and the current 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 typical case, the bandwidth is cut by 40% for eight G. 723.1-encoded voice streams through header overhead reduction and the number of voice packets is also decreased to 1/8. Furthermore, this method can easily be enhanced to a general RTP packet multiplexing method that is applicable not only to an IP-GW but also to other RTP multiplexing and de-multiplexing applications.