Relevant bibliographies by topics / QoS, VoIP

Academic literature on the topic 'QoS, VoIP'

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Published: 30 May 2022
Last updated: 31 May 2022

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Journal articles on the topic "QoS, VoIP"

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Susiani Pande, Putu Sintia, Pande Ketut Sudiarta, and I. Made Oka Widyantara. "PENGUKURAN KINERJA VOIP DENGAN CODEC G.711?, G.711a DAN G.729 DI MEDIA TRANSMISI NIRKABEL BERBASIS SIP DAN IAX." Jurnal SPEKTRUM 5, no. 1 (June 25, 2018): 21. http://dx.doi.org/10.24843/spektrum.2018.v05.i01.p04.

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Voice over Internet Protocol (VoIP) is a technology that can send real-time data with IP-based networks (Internet Protocol). In VoIP technology with wireless network has several problems that cause the performance of the network to be varied due to the QoS (Quality of Service) include delay, jitter, packet loss and MOS that affect the wireless network. This research uses G.711?, G.711a and G.729 codec based on SIP and IAX server on wireless network which then the QoS result from each codec compared with ITU-T standard which become the reference of whether the network is good or not so that later can realized on campus. In the research results, QoS on wireless IEEE 802.11 b has linear results, whereas QoS wireless in VoIP has fluctuating results because the use of codecs in VoIP on each codec has a large bitrate and different coding techniques and is a feature of wireless networks. The QoS comparison of three codecs produced the best G711 Q7S codecs because the G.711 codec has a bitrate that conforms to the 64 kbps voice communication standard and uses voice coding techniques that match the digital signal encoding technique of PCM (Pulse Code Modulation).
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Vakilinia, Shahin, Mohammadhossein Alvandi, Mohammadreza Khalili Shoja, and Iman Vakilinia. "Cross-Layered Secure and QoS Aware Design of VOIP over Wireless Ad-Hoc Networks." International Journal of Business Data Communications and Networking 9, no. 4 (October 2013): 23–45. http://dx.doi.org/10.4018/ijbdcn.2013100102.

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In this paper, Cross-layer design has been used to provide quality of service (QoS) and security at the same time for VOIP over the wireless ad-hoc network. In this paper the authors extend their previous work (i.e. Multi-path Multi-Channel Protocol Design for Secure QoS-Aware VOIP in Wireless Ad-Hoc Networks) by adding transport and application layers considerations. The goal of this paper is to support QoS and security of VOIP simultaneously. Simulation results shows that the proposed cross-layered protocol stack design significantly improve QoS parameters of the VOIP calls under the jamming or Denial-of-service attacks.
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Daramola, Oladunni Abosede. "QUALITY OF SERVICE ISSUES IN WIRELESS VOICE OVER INTERNET PROTOCOL." International Journal of Advanced Research in Computer Science and Software Engineering 7, no. 10 (October 30, 2017): 57. http://dx.doi.org/10.23956/ijarcsse.v7i10.386.

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Voice over Internet Protocol (VoIP) is a significant application of the converged network principle where the voice traffic is routed over Internet Protocol shared traffic networks. VoIP traffic was modelled over wireless network and a simulation of the traffic was transmitted over the network. E-model technique was used to analyze the traffic data and also to rate VoIP QoS parameters. The result achieved was mapped to the Mean Opinion Scale to determine the Quality of Service of VoIP over wireless networks. The results shows that QoS in the VoIP communications is significantly impacted by these parameters and the impact varies according to the parameters and also the communication aspects selected for the VoIP traffic analysis.Keywords: VoIP, QoS, E-Model and Mean Opinion Scale
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Núñez Cuadrado, Marcelo David, Carlos Andres Jativa Huilcapi, and Román Alcides Lara Cueva. "Performance evaluation of VoIP technology in an extended service set, in concordance with IEEE 802.11g." Sistemas y Telemática 15, no. 42 (October 19, 2017): 85–100. http://dx.doi.org/10.18046/syt.v15i42.2541.

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In this paper, we evaluate the performance in function of the metrics associated to Quality of Service [QoS] and Quality of user Experience [QoE] in an experimental way in the VoIP service for G.711 and G.729 códecs. This was performed over an extended service set based on Wi-Fi technology in concordance with IEEE 802.11g standard using embedded systems. QoS related metrics are obtained by using the intrusive traffic injection technique. In addition, we assessed the QoE using the MOSc [Mean Opinion Score conversational] analysis. The best results were obtained for G.729, reaching up to 25 simultaneous injections with optimal delay, jitter and packet loss values according to the ITU-T recommendation for VoIP. However, the G.711 codec presented a better throughput. On the other hand, QoE evaluation indicates a slight superiority of G.729 in the MOSc appreciation. Finally, we conclude that packet loss and delay are the most influential metrics in VoIP service degradation.
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Liang, Jin Hua, and Xuan Zen. "Projects Improving QoS of the Voice Real-Time Data Based on IP Network." Applied Mechanics and Materials 220-223 (November 2012): 2471–74. http://dx.doi.org/10.4028/www.scientific.net/amm.220-223.2471.

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The reasons influencing the VoIP QoS include transmission delay, jitter and data packet drop. The main measures improving the VoIP QoS are the integrated services and the differentiated services. But the integrated service is only suitable for the small-scale network, and the differentiated services can’t guarantee QoS from the source end to the destination end for every IP data stream alone. The paper sets forth a kind of mixed model combining integrated Services with differentiated Services to support the VoIP QoS on the bases analysis of their defects.
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Alvianto, Richard, Samuel Hutagalung, and Franciscus Ati Halim. "RANCANG BANGUN MEKANISME QUALITY OF SERVICE TERHADAP PROTOKOL RTP DAN SIP PADA ARSITEKTUR OPENFLOW." Ultima Computing : Jurnal Sistem Komputer 11, no. 1 (August 30, 2019): 9–15. http://dx.doi.org/10.31937/sk.v11i1.1093.

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Pada beberapa tahun terakhir, angka dari pengguna Voice Over Internet Protocol (VoIP) terus meningkat, dengan teknologi VoIP yang berkomunikasi melalui satu medium dalam jaringan. Hal ini tentu menimbulkan beberapa dampak terhadap VoIP seperti penggunaan bandwidth tidak terbagi dengan rata sesuai dengan kebutuhan masing-masing paket, dengan tuntutan VoIP yang membutuhkan delay, jitter, packet loss yang seminimal mungkin, untuk menjamin kualitas suara dan memberikan kenyamanan kepada pengguna VoIP. Pada penelitian ini dengan mekanisme Quality of Service (QoS) untuk memberikan prioritas terhadap protokol Real-time Transport Protocol (RTP) dan Session Initiation Protocol (SIP) dalam jaringan dirancang supaya kualitas VoIP tetap terjaga dan menghindari terjadi kemacetan terhadap paket RTP maupun SIP dalam proses antrian dalam jaringan. Analisis dalam penelitian ini dilakukan implementasikan pada emulator mininet dan diuji dengan beberapa parameter QoS, pada skenario mengujian jaringan tersebut dialiri paket dengan kecepatan 100 Mbps untuk menciptakan kondisi trafik yang padat dalam jaringan tersebut dan secara bersamaan dialiri juga trafik RTP, SIP dan data yang merupakan paket yang akan diukur nilai dari delay, jitter, packet loss. Hasil pengukuran dalam jaringan setelah diterapkan QoS menunjukan nilai dari delay, jitter, packet loss dapat berkurang dan juga memenuhi standar ITU-T G.1010 sehingga trafik VoIP dapat terjaga stabilitas dalam jaringan dan pengguna juga merasa nyaman, sedangkan pada kondisi jaringan tidak menerapkan QoS, trafik VoIP memperoleh nilai delay, jitter, packet loss yang cukup tinggi dan juga tidak memenuhi standar dari ITU-T G.1010 menyebabkan pengguna VoIP akan terganggu dengan keterlambatan dan terbuang paket VoIP yang membuat suara yang hilang dalam sebuah percakapan.
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Tanoyo, Suryo Aji, Eva Yovita Dwi Utami, and Eva Yovita Dwi Utami. "Unjuk Kerja QoS (Quality of Services) Jaringan Voice over Internet Protocol Berbasis SIP yang Diimplementasikan pada Jaringan Ethernet Gedung FEB-UKSW." Techné : Jurnal Ilmiah Elektroteknika 15, no. 01 (April 1, 2016): 17–26. http://dx.doi.org/10.31358/techne.v15i01.137.

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Jaringan komputer yang diimplementasikan di dalam suatu perkantoran yang lebih banyak dimanfaatkan untuk layanan data dapat dioptimalkan dengan penambahan layanan voice berbasis IP. Voice over Internet Protocol (VoIP) menghemat resource jaringan dibandingkan dengan PSTN (Public Switched telephone Network). Namun demikian implementasi VoIP harus memperhatikan kualitas layanan atau Qualitiy of Service (QoS). Parameter kualitas layanan VoIP antara lain throughput, delay, jitter, dan packet loss. Teknologi VoIP telah dikembangkan dengan menciptakan berbagai macam protocol seperti SIP, H.323, MGCP dan codec seperti G.711, G.723.1, G.726, G.728, G.729 dengantujuan untuk memperbaiki kualitas layanan VoIP. Penelitian ini bertujuan menganalisis kinerja QoS dengan membandingkan variasi codec G.711, G.723.1 dan G.726 pada sebuah rancangan jaringan VoIP berbasis SIP di gedung FEB-UKSW, dengan parameter QoS adalah Throughput, delay, packet loss, jitter. Komunikasi VoIP yang dilakukan terdiri atas komunikasi internal dan komunikasi eksternal. Komunikasi internal mencakup simulasi komunikasi hardphone ke PC. Komunikasi eksternal mencakup simulasi hardphone ke PC eksternal. Dari hasil penelitian, secara umum didapatkan bahwa codec G.711 memiliki kualitas paling baik untuk simulasi komunikasi internal ataupun eksternal dengan menghasilkan rata-rata delay, jitter, packet loss paling rendah.
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ANDRIANTO, HERI, DANIEL SETIADIKARUNIA, and HENDRY RAHARJO. "Evaluasi Kinerja GSM VoIP Gateway pada Sistem IP PBX." ELKOMIKA: Jurnal Teknik Energi Elektrik, Teknik Telekomunikasi, & Teknik Elektronika 9, no. 3 (July 9, 2021): 731. http://dx.doi.org/10.26760/elkomika.v9i3.731.

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ABSTRAKGSM VoIP Gateway digunakan untuk menghubungkan jaringan VoIP dengan jaringan GSM sehingga memungkinkan VoIP client melakukan komunikasi dengan VoIP client lain melalui jaringan GSM sehingga biaya komunikasi dapat ditekan. Pada penelitian ini, telah dirancang dan direalisasikan sistem IP PBX yang dihubungkan ke jaringan GSM menggunakan GSM VoIP Gateway. Evaluasi kinerja GSM VoIP Gateway pada sistem IP PBX dilakukan dengan mengamati nilai parameter Quality of Service (QoS). Komunikasi antara VoIP client dengan GSM VoIP Gateway dikategorikan pada kualitas layanan VoIP yang baik karena memiliki nilai rata-rata jitter ≤ 5,7 ms, packet loss ≤ 0,18% dan delay ≤ 9,41 ms. Komunikasi antara softphone SIPdroid dengan GSM VoIP Gateway memiliki nilai rata-rata jitter 22,58 ms, paket loss 48,68%, dan delay 14,54 ms, hal ini disebabkan karena komunikasi VoIP menggunakan koneksi WiFi. Selain itu perbedaan spesifikasi perangkat keras dan perangkat lunak juga turut mempengaruhi nilai parameter QoS.Kata kunci: GSM VoIP Gateway, IP PBX, VoIP ABSTRACTGSM VoIP Gateway is used to connect the VoIP network to the GSM network, allowing VoIP clients to communicate with other VoIP clients via the GSM network therefore the communication costs can be reduced. In this research, an IP PBX system connected to a GSM network using a GSM VoIP Gateway has been designed and realized. Performance evaluation of the GSM VoIP Gateway on the IP PBX system is carried out by observing the value of the Quality of Service (QoS) parameter. Communication between the VoIP client and GSM VoIP Gateway is categorized as a good quality VoIP service because it has an average value of jitter ≤ 5.7 ms, packet loss ≤ 0.18% and delay ≤ 9.41 ms. Communication between the SIPdroid softphone and the GSM VoIP Gateway has an average jitter value of 22.58 ms, a packet loss of 48.68%, and a delay of 14.54 ms, due to VoIP communication uses a WiFi connection. In addition, differences on hardware and software specifications also affect the value of QoS parameters.Keywords: GSM VoIP Gateway, IP PBX, VoIP
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Musbah, Esra Musbah Mohammed, Khalid Hamed Bilal, and Amin Babiker A. Nabi Mustafa. "Comparison of QoS Performance Over WLAN, VoIP4 and VoIP6." International Research Journal of Management, IT & Social Sciences 2, no. 11 (November 1, 2015): 42. http://dx.doi.org/10.21744/irjmis.v2i11.80.

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VoIP stands for voice over internet protocol. It is one of the most widely used technologies. It enables users to send and transmit media over IP network. The transition from IPv4 to IPv6 provides many benefits for internet IPv6 is more efficient than IPv4. This paper presents a performance analysis of VoIP over WLAN using IPv4 and IPv6 and OPNET software program to simulate the protocols and to investigate the QoS parameters such as jitter, delay variation, packet send, and packet received and throughputs for IP4 and IP6 and compare between them.
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Luca, Robert, Petrica Ciotirnae, and Florin Popescu. "Influence of the QoS Measures for VoIP Traffic in a Congested Network." International Journal of Computers Communications & Control 11, no. 3 (March 24, 2016): 405. http://dx.doi.org/10.15837/ijccc.2016.3.2558.

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The paper revolves around the subject regarding quality of service (QoS) n a telecommunication network. The chosen scenario is based on the transmission of ata and voice packets using a WAN connection, which has a limited bandwidth and mphasize the need of implementing QoS mechanisms in order to fulfill the quality equirements of the traffic, especially for VoIP. This topology will outline the impact nd importance of the QoS implementation, illustrated by the desired quality resulted hrough VoIP traffic simultaneously with maintaining the data conectivity using a ower bandwidth for applications which require a smaller amount of QoS properties, uch as FTP.
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Dissertations / Theses on the topic "QoS, VoIP"

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Islam, Mohammad Shahidul, and Syed Nasir Mehdi. "How Different QoS Mechanisms Affect VoIP QoS Metrics." Thesis, Högskolan i Halmstad, Sektionen för Informationsvetenskap, Data– och Elektroteknik (IDE), 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:hh:diva-15337.

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Voice over Internet Protocol (VoIP) has become a key technology of communication. Our work has been a practical implemenation of different scenarios to show that VoIP voice quality can be improved by adopting certain Quality of Service(QoS) measures such as classification, marking or queuing. It has been discussed that different QoS metrics like delay, packet loss and jitter could affect the voice quality of VoIP. To reduce the negative affects, one option is to implement certain QoS mechanisms with some set of configurations. For this purpose, Cisco IP phones have been configured in our topology with routers, switches, traffic generators, end stations and VoIP quality monitoring software called VQmanager. Tests have been divided into two sets. In one test a fixed bandwidth of 70 kbps is set while in the other test a random bandwidth is set with trafic generators unleashing packets of traffic. In both these tests further scenarios with configurations are worked out. They include no QoS, Auto Qos and Customized Qos mechanisms. Results have been indicative of top performance by the Customized QoS mechanism, in both sets of tests, followed by Auto QoS and no QoS mechanisms. It has been observed that a customized scenario could be a particular configuration to any organization’s needs and that will have the lowest delay, jitter and packet loss which are the main QoS metrics that impact the voice quality of VoIP. It  can be fundamentally composed of classification of voice, data or web-traffic, marking and queuing depending upon the need of the organization. It is finally suggested to carry more tests in companies to get more data for analysis
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Liu, Mingkuan. "QoS Improvement Schemes for Real-Time Wireless VoIP." Diss., The University of Arizona, 2006. http://hdl.handle.net/10150/193858.

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There is a tremendous demand on real-time multimedia delivery over wireless Internet due to the dramatic increase in wireless communication and the growth of the Internet. However, real-time multimedia over wireless Internet poses many challenges. First of all, the inherent best-effort characteristic of packet-switched networks makes it difficult to provide guaranteed QoS for real-time multimedia delivery. Secondly, wireless channels have much higher packet-loss rate, bit-error rate, and channel instability compared to wired channels due to noise, path loss, multi-path fading and shadowing, which result in fluctuating communication channel statistics. Thirdly, the real-time communication demands strict time limitations on the network end-to-end delay and delay jitter.In this dissertation, an intelligent application architecture and several QoS improvement mechanisms are proposed to timely estimate the current wireless network statistics and dynamically take smart actions to improve the overall performance of a real-time wireless Internet telephony system. An online network traffic modeling method based on time series analysis was used to estimate the dynamic wireless network statistics such as end-to-end packet delay and delay jitters. Using this real-time updated information, the application's sender side can take some adaptive actions such as voice codec selection and forward error-correction schemes for packet-loss concealment to improve the QoS under current available network resources. Also, a novel adaptive playout jitter buffer adjustment algorithm is proposed. The proposed algorithm achieved 11%-15% performance improvement compared to traditional adaptive playout adjustment algorithms using the ITU-E model measurement metric.
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Simoni, Chiara. "Monitoraggio della QOS in interfacce wireless per voip." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/1917/.

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Saburova, S. A., Gmati Abdulbari, and О. I. Kadatskaja. "Methods Of Control Quality Of Services VoIP Over LTE." Thesis, ХНУРЕ, 2021. https://openarchive.nure.ua/handle/document/19003.

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Methods of control of providing users with VoIP over LTE network services are considered for the basic classes of LTE QoS in UMTS / 3GPP with the availability of the following traffic classes conversational, interactive and streaming.Shown testing, calculation, analysis and evaluation of quality parameters for VoIP over LTE network services, the behavior of VoLTE MOS vs. packet loss Ppl and Pjitter, VoLTE MOS vs. effective packet loss VoLTE.
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Konečný, Zbyněk. "Mapování QoS požadavků na síťové prostředí." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-218914.

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The issue of converged networks is to ensure the sufficient quality of services along the entire length of the communication transmission. This issue is closely connected to the real-time services, such as VoIP (Voice over Internet Protocol) and videoconferencing. These services require strict adherence to quality parameters, otherwise their function is not guaranteed. This problem particulary resolves subsystem IMS (IP Multimedia Subsystem), which concluded on the basis of user profiles can provide the required quality of service. Therefore the theoretical part deals with the description of the structure of the system and protocols designed to signal the network. Various mechanisms to support quality of services in access and backbone networks are also described. The following section explains the principle of provision of quality requirements of end-user networks. In the practical part is this theoretical knowledge used for designing and configuration of the network consisting of various technologies. The resulting model is then simulated in Opnet Modeler program, which is used for designing and testing of packet networks. Each simulation shows the effect of mapping quality requirements in the different access network on technologies, which are supported in the backbone. The outcome of this work is detailed network analysis and comparison of mechanisms for implementing quality of service. The conclusion summarises all simulation outcomes.
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Perera, Bandaralokuge Earl Shehan. "VoIP and best effort service enhancement on fixed WiMAX." Thesis, University of Canterbury. Electrical & Computer Engineering, 2008. http://hdl.handle.net/10092/1575.

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Fixed Broadband Wireless Access (BWA) for the last mile is a promising technology which can offer high speed voice, video and data service and fill the technology gap between Wireless LANs and wide area networks. This is seen as a challenging competitor to conventional wired last mile access systems like DSL and cable, even in areas where those technologies are already available. More importantly the technology can provide a cost-effective broadband access solution in rural areas beyond the reach of DSL or cable and in developing countries with little or no wired last mile infrastructure. Earlier BWA systems were based on proprietary technologies which made them costly and impossible to interoperate. The IEEE 802.16 set of standards was developed to level the playing field. An industry group the WiMAX Forum, was established to promote interoperability and compliance to this standard. This thesis gives an overview of the IEEE 802.16 WirelessMAN OFDM standard which is the basis for Fixed WiMAX. An in depth description of the medium access control (MAC) layer is provided and functionality of its components explained. We have concentrated our effort on enhancing the performance of Fixed WiMAX for VoIP services, and best effort traffic which includes e-mail, web browsing, peer-to-peer traffic etc. The MAC layer defines four native service classes for differentiated QoS levels from the onset. The unsolicited grant service (UGS) class is designed to support real-time data streams consisting of fixed-size data packets issued at periodic intervals, such as T1/E1 and Voice over IP without silence suppression, while the non-real-time polling service (nrtPS) and best effort (BE) are meant for lower priority traffic. QoS and efficiency are at opposite ends of the scale in most cases, which makes it important to identify the trade-off between these two performance measures of a system. We have analyzed the effect the packetization interval of a UGS based VoIP stream has on system performance. The UGS service class has been modified so that the optimal packetization interval for VoIP can be dynamically selected based on PHY OFDM characteristics. This involves cross layer communication between the PHY, MAC and the Application Layer and selection of packetization intervals which keep the flow within packet loss and latency bounds while increasing efficiency. A low latency retransmission scheme and a new ARQ feedback scheme for UGS have also been introduced. The goal being to guarantee QoS while increasing system efficiency. BE traffic when serviced by contention based access is variable in speed and latency, and low in efficiency. A detailed analysis of the contention based access scheme is done using Markov chains. This leads to optimization of system parameters to increase utilization and reduce overheads, while taking into account TCP as the most common transport layer protocol. nrtPS is considered as a replacement for contention based access. Several enhancements have been proposed to increase efficiency and facilitate better connection management. The effects of proposed changes are validated using analytical models in Matlab and verified using simulations. A simulation model was specifically created for IEEE 802.16 WirelessMAN OFDM in the QualNet simulation package. In essence the aim of this work was, to develop means to support a maximum number of users, with the required level of service, using the limited wireless resource.
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Niemelä, Markus. "Estimating Internet-scale Quality of Service Parameters for VoIP." Thesis, Linköpings universitet, Programvara och system, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-127360.

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With the rising popularity of Voice over IP (VoIP) services, understanding the effects of a global network on Quality of Service is critical for the providers of VoIP applications. This thesis builds on a model that analyzes the round trip time, packet delay jitter, and packet loss between endpoints on an Autonomous System (AS) level, extending it by mapping AS pairs onto an Internet topology. This model is used to produce a mean opinion score estimate. The mapping is introduced to reduce the size of the problem in order to improve computation times and improve accuracy of estimates. The results of testing show that estimating mean opinion score from this model is not desirable. It also shows that the path mapping does not affect accuracy, but does improve computation times as the input data grows in volume.
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Rana, Bilal Zahid, and Shahid Ali. "OPNET Analysis of VoIP over MPLS VPN with IP QoS." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-3404.

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There are many disadvantages (cost, lack of security, difficult to manage large networks, support to non-sensitive applications, delay, etc.) associated with traditional networking, IP network, ATM and Frame relay networking. To solve this, an MPLS-based VPN networking is introduced that can work with existing deployed backbones and allow organizations to interconnect the dispersed sites and remote workers through secure links by using public internet. In this thesis, we are trying to build a better understanding to MPLS VPN and we researched to analyze the behavior of OSPF and RIPv2 based MPLS-BGP VPN architectures by using intense VoIP traffic. Then it comes with an OPNET simulation process and scenarios for MPLS-BGP VPN. At last, the conclusion is made: OSPF based MPLS-BGP VPN architecture has lower VPN delay, background traffic Flow delay, LSP delay and point-to-point Queuing delay, and has better performance in VPN load and VPN throughput that can acquire customer satisfaction and confidence as compared to the RIPv2 based MPLS-BGP VPN architecture.
Det finns många nackdelar (kostnader, bristande säkerhet, svåra att hantera stora nätverk, stöd till icke-känsliga tillämpningar, delay, etc.) i samband med traditionella nätverk, IP-nätverk, ATM och Frame Relay nätverk. För att lösa detta, är ett MPLS-baserat VPN nätverk införs som kan arbeta med befintliga sättas samman och låter organisationer för att förbinda de spridda platser och distansarbetare genom säkra länkar genom att använda publika Internet. I denna avhandling försöker vi bygga en bättre förståelse för MPLS VPN och vi forskat för att analysera beteendet hos OSPF och RIPv2 baserad MPLS-VPN BGP arkitekturer med hjälp av intensiv VoIP-trafik. Då kommer med en OPNET simulering process och scenarier för MPLS-BGP VPN. Äntligen är den slutsatsen: OSPF bygger MPLS-VPN BGP arkitektur har lägre VPN dröjsmål bakgrund trafikflödet dröjsmål, LSP dröjsmål och punkt-till-punkt Queuing dröjsmål, och har bättre prestanda i VPN-belastning och VPN som kan få kunden tillfredsställelse och förtroende jämfört med RIPv2 baserad MPLS-VPN BGP arkitektur.
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Paulsen, Stefan [Verfasser], and Bernd [Akademischer Betreuer] Wolfinger. "QoS/QoE-Modelle für den Dienst Voice over IP (VoIP) / Stefan Paulsen. Betreuer: Bernd Wolfinger." Hamburg : Staats- und Universitätsbibliothek Hamburg, 2016. http://d-nb.info/1106404505/34.

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Paulsen, Stefan Verfasser], and Bernd E. [Akademischer Betreuer] [Wolfinger. "QoS/QoE-Modelle für den Dienst Voice over IP (VoIP) / Stefan Paulsen. Betreuer: Bernd Wolfinger." Hamburg : Staats- und Universitätsbibliothek Hamburg, 2016. http://nbn-resolving.de/urn:nbn:de:gbv:18-79080.

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Books on the topic "QoS, VoIP"

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Detken, Kai-Oliver. Echtzeitplattformen fu r das Internet: Grundlagen, Lo sungsansa tze der sicheren Kommunikation mit QoS und VoIP. Mu nchen: Addison-Wesley, 2002.

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Lipovac, Vlatko. Testing Integrated QoS of VoIP: Packets to Perceptual Voice Quality. AUERBACH, 2008.

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Book chapters on the topic "QoS, VoIP"

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Fischer, Jörg, and Christian Sailer. "Netze, QoS, Pakete und Bandbreite." In VoIP Praxisleitfaden, 89–139. München: Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.003.

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Liu, Xiaojun, and Chunxia Tu. "An VoIP Application Design with Dynamic QoS Control." In Communications in Computer and Information Science, 605–12. Berlin, Heidelberg: Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-24999-0_84.

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Lonkar, Smita Avinash, and K. T. V. Reddy. "Relative QoS Investigation of VoIP Over LTE Networks." In Nanoelectronics, Circuits and Communication Systems, 71–80. Singapore: Springer Singapore, 2020. http://dx.doi.org/10.1007/978-981-15-7486-3_8.

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Hruby, Martin, Michal Olsovsky, and Margareta Kotocova. "Solving VoIP QoS and Scalability Issues in Backbone Networks." In Lecture Notes in Electrical Engineering, 537–49. Dordrecht: Springer Netherlands, 2013. http://dx.doi.org/10.1007/978-94-007-6190-2_41.

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Tran, Hung Tuan, Thomas Ziegler, and Fabio Ricciato. "QoS Provisioning for VoIP Traffic by Deploying Admission Control." In Architectures for Quality of Service in the Internet, 139–53. Berlin, Heidelberg: Springer Berlin Heidelberg, 2003. http://dx.doi.org/10.1007/3-540-45020-3_11.

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Ha, Eun-Ju, and Byeong-Soo Yun. "End-to-End QoS Management for VoIP Using DiffServ." In Computational Science and Its Applications – ICCSA 2004, 818–27. Berlin, Heidelberg: Springer Berlin Heidelberg, 2004. http://dx.doi.org/10.1007/978-3-540-24768-5_88.

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Miraz, Mahdi Hassan, Muzafar Aziz Ganie, Maaruf Ali, Suhail Ahmed Molvi, and AbdelRahman Hamza Hussein. "Performance Evaluation of VoIP QoS Parameters Using WiFi-UMTS Networks." In Transactions on Engineering Technologies, 547–61. Dordrecht: Springer Netherlands, 2015. http://dx.doi.org/10.1007/978-94-017-9804-4_38.

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Elsayed, Khaled M. F., Hassan Fadel, and Amin M. Nassar. "LMPS: Localized Multi-path Selection for QoS Routing in VoIP Networks." In Lecture Notes in Computer Science, 1072–83. Berlin, Heidelberg: Springer Berlin Heidelberg, 2004. http://dx.doi.org/10.1007/978-3-540-24693-0_88.

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Muezerie, André, Ioanis Nikolaidis, and Pawel Gburzyński. "Attaining VoIP–Grade QoS via Deflection: A Buffer Space Tradeoff Study." In NETWORKING 2005. Networking Technologies, Services, and Protocols; Performance of Computer and Communication Networks; Mobile and Wireless Communications Systems, 1457–60. Berlin, Heidelberg: Springer Berlin Heidelberg, 2005. http://dx.doi.org/10.1007/11422778_137.

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Lee, Kyu Ouk, Sang Soo Lee, and Tae Whan Yoo. "Functional Model and Service Scenarios for QoS Enabled Mobile VoIP Service." In Smart Spaces and Next Generation Wired/Wireless Networking, 389–97. Berlin, Heidelberg: Springer Berlin Heidelberg, 2010. http://dx.doi.org/10.1007/978-3-642-14891-0_34.

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Conference papers on the topic "QoS, VoIP"

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Ghiata, N., and M. Marcu. "VoIP communication QoS analysis system." In 2010 International Joint Conference on Computational Cybernetics and Technical Informatics. IEEE, 2010. http://dx.doi.org/10.1109/icccyb.2010.5491326.

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Cardeal, S., F. Neves, S. Soares, F. Tavares, and P. Assuncao. "ArQoS®: System to monitor QoS/QoE in VoIP." In IEEE EUROCON 2011 - International Conference on Computer as a Tool. IEEE, 2011. http://dx.doi.org/10.1109/eurocon.2011.5929310.

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Jiang, Chunlei, and Peng Huang. "Research of Monitoring VoIP Voice QoS." In information Services (ICICIS). IEEE, 2011. http://dx.doi.org/10.1109/icicis.2011.130.

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Weiwei Zhang, Yongyu Chang, Yitong Liu, and Yuan Tian. "Perceived QoS assessment for Voip networks." In 2013 15th IEEE International Conference on Communication Technology (ICCT). IEEE, 2013. http://dx.doi.org/10.1109/icct.2013.6820466.

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Malaney, Robert A., Ernesto Exposito, and Xun Wei. "Seeking VoIP QoS in physical space." In the 3rd ACM international workshop. New York, New York, USA: ACM Press, 2005. http://dx.doi.org/10.1145/1080730.1080735.

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Jia, Wen-Kang, Yi-Yu Chou, and Yaw-Chung Chen. "QoS Improvement of VoIP over SDN." In 2020 IEEE 17th Annual Consumer Communications & Networking Conference (CCNC). IEEE, 2020. http://dx.doi.org/10.1109/ccnc46108.2020.9045152.

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Kim, Keith, Petros Mouchtaris, Sunil Samtani, Rajesh Talpade, and Larry Wong. "Bandwidth broker architecture for VoIP QoS." In ITCom 2001: International Symposium on the Convergence of IT and Communications, edited by Petros Mouchtaris. SPIE, 2001. http://dx.doi.org/10.1117/12.434288.

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Liu, Ren, Glynn Rogers, and Jim Argyros. "IP/ATM QoS Solutions for VoIP Traffic." In 2006 Asia-Pacific Conference on Communications. IEEE, 2006. http://dx.doi.org/10.1109/apcc.2006.255772.

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Radhakrishnan, Kapilan, and Hadi Larijani. "A study on QoS of VoIP networks." In the 2010 Spring Simulation Multiconference. New York, New York, USA: ACM Press, 2010. http://dx.doi.org/10.1145/1878537.1878656.

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Radmand, Pedram, and Alex Talevski. "Impact of Encryption on Qos in Voip." In 2010 IEEE Second International Conference on Social Computing (SocialCom). IEEE, 2010. http://dx.doi.org/10.1109/socialcom.2010.112.

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You might also be interested in the extended bibliographies on the topic 'QoS, VoIP' for particular source types:
Journal articles Dissertations / Theses
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