• Journal of KIISE:Computing Practices and Letters
• /
• v.14 no.1
• /
• pp.101-105
• /
• 2008

This paper proposes an analytical model to calculate VoIP (Voice of IP) capacity over wireless LANs with frequent bit errors. Since the traditional analytical models for VoIP capacity have not included the effect of bit errors, simulations ould only evaluate VoIP capacity over erroneous channels. For analytically accurate estimation of VoIP capacity over noisy channels, we extend the conventional model to include the effect of propagation errors, end-to-end delay, voice quality, the waiting time in AP(Access Point). The experiments show that our model predicts the VoIP capacity of a given network within the range from 3% to 9% difference comparing with the simulation results.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.16 no.11
• /
• pp.2405-2410
• /
• 2012

The 4th generation mobile communication system, LTE, does not support an additional core network to provide voice service, and it is merged into a packet network based on all IP. Although Voice service over LTE can be supported by VoIP, it will be provided by the existing 3G networks because of the discontinuity of LTE coverage. However, it is inevitable to adopt VoIP over LTE to provide high quality voice service. In this paper, we investigate the capacity of VoIP over LTE. Our results indicate that spectral efficiency can be significantly improved as channel bandwidth increases in terms of VoLTE capacity. In addition, we can achieve higher VoLTE capacity without decreasing control channel capacity.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.7 no.4
• /
• pp.747-752
• /
• 2012

In this paper we evaluated by simulation the effect of VoIP roaming and data traffic roaming on other VoIP calls. Regardless of MIPv4 or MIPv6, the quality of voice of all VoIP calls falls down quickly to the bottom level after the start of roaming by one voice terminal. That was caused by the excessive retransmission on downlink. Thus it seems that we need a kind of call admission control when we adopt the roaming service on VoIP calls. Data traffic degrades also the voice quality especially at the foreign agent side.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.3B
• /
• pp.161-167
• /
• 2007

In this paper, we evaluate the capacity of voice over internet protocol (VoIP) services over high-speed downlink packet access (HSDPA), in which frame-bundling (FB) is incorporated to reduce the effect of relatively large headers in the IP/UDP/RTP layers. Also, a modified proportional pair (PF) packet scheduler design supporting for VoIP service is provided. The main focus of this work is the effect of FB on system outage based on delay budget in radio access networks. Simulation results show that VoIP system performance with FB scheme is highly sensitive to delay budget. We also conclude that HSDPA is attractive for transmission of VoIP if compared to the circuit switched (CS) voice that is used in WCDMA (Release'99).

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2010.10a
• /
• pp.459-461
• /
• 2010

Ad-Hoc VoIP system is an important IP phone technique used in MANET networks. In small-scale networks, VoIP system could be built easily while keeping the Qos on a acceptable level. Although Ad-Hoc VoIP system is designed as 'Client-Server' model, due to the property of MANETs that no Base Stations working as servers, we have the SIP Server with promary funtions worked on mobile terminals instead. In this paper, we explored how to build VoIP systems in MANETs and achieved the construction from within linux.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.7 no.2
• /
• pp.243-248
• /
• 2012

In this paper we simulate the capacity of VOIP calls through WiFi network by computer simulations using OPNET modeler. The results show that sudden quality degradations occur on all VoIP calls when the number of call of an AP(Access Point) increases beyond a specific value. The reason of the quality degradation was turned out to be the queueing delay at the down link of AP. Under the IEEE 802.11b environments, the maximum number of VoIP calls of an AP maintaining the required voice quality (MOS > 2.5), was evaluated as 5, 12, and 27 when we use G.711, G.729a, and G.729a VAD codec, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.19 no.6
• /
• pp.1327-1332
• /
• 2015

In this paper, we studied several aspects that affect the quality of VoIP using WiFi network. It's clear that the background data traffic within an AP, the end-to-end delay and the traffic loss of TCP/IP network gives serious effects on the voice quality. A kind of access control for the VoIP connection within an AP should be done for the acceptable voice quality.

• Journal of Internet Computing and Services
• /
• v.8 no.5
• /
• pp.141-150
• /
• 2007

This paper proposes new Steganographic model for VoIP that has very effective method using low bit encoding. Most of Steganographic models using Low bit Encoding have two disadvantages; one is that the existence of hidden secret message can be easily detected by auditory, the other is that the capacity of stego data is low. To solve these problems, this method embed more than one bit in inaudible range, so this method can improve the capacity of the hidden message in cover data. The embedding bit position is determined by using a pseudo random number generator which has seed with remaining message length, so it is hard to detect the stego data produced by the proposed method. This proposed model is able to use not only to communicate wave file with hidden message in VoIP environment but also to hide vary information which is user basic information, authentication system, etc.

• Proceedings of the Korean Information Science Society Conference
• /
• 2007.06a
• /
• pp.177-178
• /
• 2007

• Proceedings of the Korean Information Science Society Conference
• /
• 2007.10d
• /
• pp.602-607
• /
• 2007

최근 4G 이동통신 시스템의 진화를 위한 단계로 기존의 이동통신 시스템을 유지하면서 효율 극대화를 통한 비용 절약 측면과 고 주파수 대역을 이용한 셀 간격의 협소화로 인해 나타나는 문제점에 대한 대안으로 커버리지 확대와 데이터 처리율 향상을 위한 IEEE 802.16j 표준화 작업이 진행 중에 있다. 향후 VoIP 서비스를 위한 업링크 스케줄링 기법으로 몇 가지가 IEEE 802.16j에 제시되어 있으나, 본 논문에서는 보이스 트래픽 특성 및 수학적 모델링을 통하여 기존 방법들보다 더 효율적인 대역 이용을 통해 같은 대역폭 대비 성능과 용량 면에서 개선된 스케줄링 기법인 eertPS를 제시하여 IEEE 802.16j Multi-Hop 환경하에서도 대역 효율적인 VoIP 서비스를 이용할 수 있도록 한다.