• Proceedings of the IEEK Conference
• /
• 2003.07a
• /
• pp.430-433
• /
• 2003

In this paper, we propose the method of voice packet transformation for the improvement of voice quality in embedded VoIP system terminals. For this purpose, it was analyzed about the RTP header in the voice packet receiving side and designed about the handling method of voice packets by using jitter buffer. Through this analyzed and designed results, by implementing the voice packet transformation method and testing on the self-product VoIP system, we reached that a conclusion is satisfied with performance of VoIP services.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.12
• /
• pp.1841-1854
• /
• 1993

In this paper, the packet voice protocol for the transmission of voice signal onto ethernet is implemented in a personal computer (PC). The packet voice protocol used is a modified one from CCITT G.764 packetized voice protocol. The hardware system to facilitate the voice communication onto ethernet is divided into telephone interface, speech processing, PC interface and controllers. The software structure of the protocol is designed according to the OSI seven layer architecture and is divided into three routines : ethernet device driver, telephone interface, and processing routine of the packet voice protocol. Experiments through ethernet with telephone interface show that this packet voice communication achieves satisfactory quality when the network traffic is light.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.13 no.2
• /
• pp.171-181
• /
• 1988

In this study, a packet voice/data terminal(PVDT) that services both voice and data in the packet-switched network is implemented. The software structure of the PVDT is designed according to the OSI 7 layer architecture. The discrimination of voice and data is made in the link layer. Voice packets have priority over data packets in order to minimize the transmission delay, and are serviced by a simple protocol so that the overhead arising form the retransmission of packets may be minimized. The hardware structure of the PVDT is divided into five modules; a master control module, a speech proessing module, a speech activity detection module, a telephone interface module, and an input/output interface module. In addition to the hardware implementation, the optimal reconstruction delay of voice packets to reduce the influence of delay variance is analyzed.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.3B
• /
• pp.423-430
• /
• 2000

Future microcellular systems will require distributed network control. A packet-switched network is suitable for this requirement. The packet reservation multiple access(PRMA) is a Reservation-ALOHA like protocol for wireless terminals to transmit packet speech to a base station. It allows spatially distributed users in cellular systems to transmit packeted voice and data to a common base station using a shared channel. In the existing PRMA, the problem is that the voice packets may collide with the data packets due to simultaneous channel access. the problem may be a major performance degradation factor to a voice and data mixed system. We propose a new PRMA method that integrates voice and data traffic efficiently by resolving the collision problem between data and voice packets. The proposed PRMA method gives a performance improvement than the existing PRAMA method in terms of voice packet dropping probability and data delay characteristic. From analytic results, we can confirm that the proposed PRMA method show a performance improvement than the existing PRMA protocol.

• The Journal of the Acoustical Society of Korea
• /
• v.33 no.1
• /
• pp.75-80
• /
• 2014

The quality of real-time Voice over Internet Protocol (VoIP) network is affected by network impariments such as delays, jitters, and packet loss. This paper proposes a packet loss concealment algorithm based on voice classification for enhancing VoIP speech quality. In the proposed method, arriving packets are classified by an adaptive thresholding approach based on the analysis of multiple features of short signal segments. The excellent classification results are used in the packet loss concealment. Additionally, linear prediction-based packet loss concealment delivers high voice quality by alleviating the metallic artifacts due to concealing consecutive packet loss or recovering lost packet.

• Journal of information and communication convergence engineering
• /
• v.8 no.5
• /
• pp.513-518
• /
• 2010

This paper proposes a medium access control protocol for integrated voice and data services in slotted CDMA systems. The proposed protocol, which is named as RCSSMA (Reservation Code and Status Sensing Multiple Access), adopts a code reservation and status sensing schemes. RCSSMA protocol gives higher access priority to the voice traffic than data traffic for reducing the packet dropping probability. The voice terminal reserves an available spreading code to transmit voice packets during a talkspurt, whereas the data terminal transmits a packet over one of the available spreading codes that are not reserved by the voice terminals. In this protocol, the voice packets never contend with the data packets. Packet dropping probability and average data packet transmission delay are analyzed using a Markov chain model.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.5 no.5
• /
• pp.916-922
• /
• 2001

A medium access control protocol is proposed for integrated voice and data services in the packet CDMA network with a small coverage. Uplink channels are composed of time slots and multiple spreading codes for each slot. This protocol gives higher access priority to the delay-sensitive voice traffic than to the data traffic. During a talkspurt, voice terminals reserve a spreading code to transmit multiple voice packets. On the other hand, whenever generating a data packet, data terminals transmit a packet based on the status Information of spreading codes in the current slot, which is received from base station. In this protocol, voice packet does not come into collision with data packet. Therefore, this protocol can increase the maximum number of voice terminals.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.40 no.9
• /
• pp.373-383
• /
• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.6 no.4
• /
• pp.1006-1025
• /
• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• The KIPS Transactions:PartC
• /
• v.19C no.2
• /
• pp.119-126
• /
• 2012

This paper covers the service quality of packet-based voice service that is provided over wireless broadband (WiBro) and high speed downlink packet access (HSDPA) systems. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over wireless networks, a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement results, the service quality of the voice service is supposed to be quite good over both wireless systems. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.