Real-time voice transmission based on source rate control and audio transmission using erasure code in packet networks

Abstract

Recently, the technique to transmit not only non-real-time data services but also voice and multimedia services over packet networks and the Internet is widely used. Many researches have been done about Asynchronous Transfer Mode(ATM) and Internet Protocol(IP) based Quality of Service(QoS) guarantee mechanism using admission control, bandwidth reservation, rate control, and loss recovery. QoS-supporting protocols, such as IP version 6(IPv6), ReSource reserVation Protocol(RSVP), differentiated services in IPv4, and Real-time Transport Protocol(RTP) are widely used for this purpose. The popular standards for transmitting multimedia in packet networks are International Telecommunication Union(ITU) Recommendation H.323 and Session Initiation Protocol(SIP). Both of them use IP/UDP/RTP encapsulation for audio. RTP is a generic mechanism for supporting the integration of voice, video, and data. RTP headers provide the media type and the information, like sequence number and time stamp, needed to reassemble a real-time stream from received packets. RTP is widely used in the enterprise network and the Internet for real-time data transmission including voice and audio. To design the transmission algorithm for real-time data transmission over packet networks, the resource reservation is the most important point to be considered. Rate control and admission control method are used in case where network resource is guaranteed, and loss recovery and resilient methods are used to improve service quality for real-time data transmission over the best effort service networks like the Internet. Therefore, this thesis deals with the source rate control in the bandwidth guaranteed packet networks and then considers loss recovery in the best effort service networks like the Internet. In the QoS guaranteed networks, a rate control method to acquire more statistical gain and lower loss rate is exist. It controls a source code rate dynamically according to the quantity of incoming tra...