In this letter, we propose a shared adaptive packet loss concealment scheme for the high quality guaranteed Internet telephony service which connects multiple users. In order to recover packet loss efficiently in the all-IP based convergence environment, we provide a robust signal recovery scheme which is based on the shared adaptive both-side information utilization. This scheme is provided according to the average magnitude variation across the frames and the pitch period replication on the 1-port gateway (G/W) system. The simulated performance demonstrates that the proposed scheme has the advantages of low processing times and high recovery rates in the all-IP based ubiquitous environment.

This paper presents the characteristics and modeling of VoIP traffic for a real network. The new model, based on measured data, shows a significant difference from the previously proposed models in terms of parameters and their effects. It is found that the effects of background noise and ringing tones have essential influences on the model. The observed distributions of talkspurt and silent durations have long-tail characteristics and considerably differ from the existing models. An additional state called "Long burst", which represents the background noise at the talker's place, is added into the continuous-time Markov process model. The other three states, "Talk", "Short silence" and "Long silence", represent the normal behavior of the VoIP user. Models for conversational speech containing the communication during the dialogue are presented. In the case of the VoIP traffic aggregation, the simplified models, which neglect the conversation's interaction, are proposed. Depending on the occurrences of background noise during the speech, the model is classified as "noisy speech" or "noiseless speech". The measured data shows that the background noise typically increases the data rate by 60%. Simulation results of aggregated VoIP traffic indicate the self-similarity, which is analogous to the measured data. Results from the measurements support the fact that except the ringing duration the conversations from both the directions can be modeled in identical manner.

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.