Packet loss and delay are the major network impairments for transporting real-time voice over IP networks. In the proposed system, multiple descriptions of the speech are used to take advantage of the packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a multi-stream playout buffer algorithm, together with an adaptive parameter adjustment scheme, that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that, compared to FEC-protected single-path transmission, the proposed multi-stream transmission scheme achieves significant reductions in delay and packet loss rates as well as improved speech quality.

This paper quantitatively analyzes the maximum UDP (User Datagram Protocol) throughput for two-way flows in wireless string multi-hop networks. The validity of the analysis is shown by the comparison with the simulation and the experiment results. The authors also clarify the difference fundamental characteristics between a one-way flow and a two-way flow in detail based on the simulation results. The result shows that collisions at both ends' nodes are decisive in determining the throughput for two-way flows. The analyses are applicable to the estimation of VoIP (Voice over Internet Protocol) capacity for string multi-hop networks represented by WLAN (Wireless Local Area Network) mesh networks.

This paper describes a QoS evaluation method for VoIP communications using a self-organizing neural network. Based on measurements in real environments, evaluation results confirmed that our method can effectively display total QoS level composed of several QoS-related factors such as PSQM+ and end-to-end delay.

This paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class packet telephony network for NGN (Next Generation Networks). Until recently, fone of the greatest challenges in the migration from existing PSTN (Public Switched Telephone Network) toward NGN is to build a carrier class packet telephony network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new packet telephony technology. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class packet telephony network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP (Voice over IP) calls in carrier class packet telephony network. In this paper, we derive the buffer size, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class packet telephony network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we derive the formulas of buffer size, queueing delay, and delay variation for the non-preemptive priority queue by queueing theory respectively. Besides, some numerical examples of buffer size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistence with simulation results.

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.