PERBANDINGAN PERFORMANSI VOIP MENGGUNAKAN CODEC G.711 DAN GSM MELALUI VPN DALAM ARSITEKTUR JARINGAN IPV6

ABSTRAKSI: Dewasa ini pertumbuhan jumlah user internet semakin meningkat dikarenakan perkembangan teknologi internet dan tingkat kebutuhan manusia untuk melakukan pertukaran informasi juga meningkat. VoIP merupakan salah satu teknologi untuk melakukan pertukaran informasi berbentuk voice secara real time yang sedang berkembang. VoIP dibangun dengan memanfaatkan jaringan berbasis IP yang sekarang ini sedang mengalami proses migrasi dari IPv4 menuju IPv6. Dalam proses transimisinya, VoIP dikirim melalui jaringan publik seperti internet sehingga membutuhkan dukungan pada fungsi security untuk melewati jaringan publik tersebut. Virtual Private Network (VPN) merupakan salah satu solusi untuk mendukung teknologi VoIP dalam proses transmisinya. Dengan menggunakan VPN maka akan dimungkinkan untuk membangun jaringan private di atas jaringan publik (seperti internet). Implementasi VPN diharapkan dapat menjadi solusi terbaik untuk mendukung pembangunan VoIP dalam arsitektur jaringan IPv6. Dalam tugas akhir kali ini akan dibangun VoIP di atas jaringan VPN IPv6 dengan IPSec. Protokol VoIP yang digunakan adalah SIP dan menggunakan SER sebagai server VoIP. Codec yang digunakan adalah Codec G.711 dan GSM karena kedua codec tersebut bersifat free license. Hasil pembangunan VoIP yang telah dilakukan dengan codec G.711 dan GSM akan dianalisis, dan keluarannya adalah performansi dari penggunaan kedua codec tersebut masih memenuhi standar ITU (G.107), namun terdapat perbedaan performansi, performansi VoIP menggunakan codec G.711 lebih baik daripada GSM dikarenakan perbedaan bitrate saat penyamplingan. Codec G.711 64kbps sedangkan GSM 13kbps.Kata Kunci : VoIP, VPN, IPv4, IPv6, IPSec, SER, Codec G.711, Codec GSMABSTRACT: Nowadays, the amount of internet user is increasing significantly. It’s caused by increasing needs to get information using internet. VoIP is one of method to get audio information on real time trough IP networks. VoIP is becoming very important application in which will coexist on IP networks that is transitioning from IPv4 to IPv6. The transmission of VoIP is by unsecure public networks; so that VoIP needs security supports to pass the unsecure public networks. VPN is one of method to support the VoIP transmission; by using VPN it is possible to make a private network through the public network like internet. Adding VPN could be the best solution to secure VoIP connections. In this final assignment, VoIP will be building on IPv6 architecture by using VPN based on IPSec. SIP will be the VoIP protocol and SER will be VoIP server. The codec that will be use in this final assignment are G.711 and GSM codec because both of them are free license codec. The final result of this final assignment will be comparison of VoIP performance between G.711 codec and GSM codec and the final result still on the standard of ITU-T G.107 even though G.711 codec have better result than GSM codec does.Keyword: VoIP, VPN, IPv4, IPv6, IPSec, SER, Codec G.711, Codec GS

Design and Experimental Evaluation of a Route Optimisation Solution for NEMO

An important requirement for Internet protocol (IP) networks to achieve the aim of ubiquitous connectivity is network mobility (NEMO). With NEMO support we can provide Internet access from mobile platforms, such as public transportation vehicles, to normal nodes that do not need to implement any special mobility protocol. The NEMO basic support protocol has been proposed in the IETF as a first solution to this problem, but this solution has severe performance limitations. This paper presents MIRON: Mobile IPv6 route optimization for NEMO, an approach to the problem of NEMO support that overcomes the limitations of the basic solution by combining two different modes of operation: a Proxy-MR and an address delegation with built-in routing mechanisms. This paper describes the design and rationale of the solution, with an experimental validation and performance evaluation based on an implementation.Publicad

Comparison of IPv4 and IPv6 QoS implementations using Differentiated Services

Real-time applications such as VoIP place stringent demands on network QoS. However, IP is a best-effort service and is often unable to offer the levels of QoS required for real-time applications. One mechanism that has been commonly used to address this issue in IP networks is Differentiated Services (DiffServ). This paper describes the use of DiffServ in IPv4 and IPv6 networks, and implementation and evaluation of VoIP QoS within OPNET IT Guru. The simulation results demonstrated that DiffServ improved the performance of VoIP traffic in both IPv4 and IPv6, allowing previously congested networks to deliver VoIP with an acceptable QoS. However the simulations also showed that the performance of DiffServ in IPv6 is slightly worse than in IPv4. A number of possible reasons for this outcome are proposed along with recommendations for further research

Integrating Voice over IP Solution in IPv6 and IPv4 Networks to Increase Employee Productivity: A Case Study of Cameroon Telecommunications (Camtel), North-West

Telecommunications organizations have to follow the rapid innovation of technology if they want to face challenges raised by competition. The challenge to respond to the huge market demand of updated products and services from customers requires that the organization‘s working environment be equipped with tools and communication facilities that contribute to ameliorating productivity. Cameroon Telecommunications (Camtel) is facing a digital telephony and Internet Protocol strategic management challenge. Successful implementation cannot be achieved if the employees are still depending on the ageing public switched telephone network (PSTN) as their primary communication system, despite the frequent loss of dial tone experience in a day which can last up to a week, with serious repercussions on business activities and revenues. This study is designed to provide a solution to the telecommunications challenge. The fundamental question is how to integrate a digital telephony system that will provide telephony services in the existing IPv4 data network while prioritizing IPv6 traffic forwarding. This study proposes and implements solutions that integrate a Voice over IP solution with IPv6 as an alternative communication system that relies on the existing IPv4 data network. VoIP is deemed as one of the driving forces behind the adoption of IPv6. The purpose is to offer to workers an option that will free them from the poor Quality of Service (QoS) of their existing PSTN based solution, hopefully enhancing the overall productivity. This paper follows two research methodologies: Qualitative Research in Applied Situations and Engineering design process. The first part of this study reports the results of the evaluation of how much such a solution can enhance workers’ productivity. As it is important to provide an environment where IPv4 and IPv6 networks and applications/devices can interoperate in the context of VoIP; the second part describes practically a simulation environment where various configurations of network entities are done following a Dual-Stack transition approach. Document and records were used to gather information related to the structure, operations, and topological update of the Camtel’s existing IP data network. The findings demonstrated that VoIP can be an effective communication solution for Camtel and its implementation with IPv6 will be preferable. However, for this to be efficient there must be a provision of sufficient bandwidth and usage of types of equipment and transmission mediums that minimizes processing and propagation delays. Findings also reveal that better productivity will be achieved if workers are fully trained for the exploitation. This research article tries to highlight, discuss a required transition roadmap and extend the local knowledge and practice on IPv6. Future expansion of this research work will consist of deploying Dual-Stack VoIP in the remaining 9 regional offices for full integration in the corporate communication system of Camtel

A Survey on Handover Management in Mobility Architectures

This work presents a comprehensive and structured taxonomy of available techniques for managing the handover process in mobility architectures. Representative works from the existing literature have been divided into appropriate categories, based on their ability to support horizontal handovers, vertical handovers and multihoming. We describe approaches designed to work on the current Internet (i.e. IPv4-based networks), as well as those that have been devised for the "future" Internet (e.g. IPv6-based networks and extensions). Quantitative measures and qualitative indicators are also presented and used to evaluate and compare the examined approaches. This critical review provides some valuable guidelines and suggestions for designing and developing mobility architectures, including some practical expedients (e.g. those required in the current Internet environment), aimed to cope with the presence of NAT/firewalls and to provide support to legacy systems and several communication protocols working at the application layer

Mobile IP movement detection optimisations in 802.11 wireless LANs

The IEEE 802.11 standard was developed to support the establishment of highly flexible wireless local area networks (wireless LANs). However, when an 802.11 mobile node moves from a wireless LAN on one IP network to a wireless LAN on a different network, an IP layer handoff occurs. During the handoff, the mobile node's IP settings must be updated in order to re-establish its IP connectivity at the new point of attachment. The Mobile IP protocol allows a mobile node to perform an IP handoff without breaking its active upper-layer sessions. Unfortunately, these handoffs introduce large latencies into a mobile node's traffic, during which packets are lost. As a result, the mobile node's upper-layer sessions and applications suffer significant disruptions due to this handoff latency. One of the main components of a Mobile IP handoff is the movement detection process, whereby a mobile node senses that it is attached to a new IP network. This procedure contributes significantly to the total Mobile IP handover latency and resulting disruption. This study investigates different mechanisms that aim to lower movement detection delays and thereby improve Mobile IP performance. These mechanisms are considered specifically within the context of 802.11 wireless LANs. In general, a mobile node detects attachment to a new network when a periodic IP level broadcast (advertisement) is received from that network. It will be shown that the elimination of this dependence on periodic advertisements, and the reliance instead on external information from the 802.11 link layer, results in both faster and more efficient movement detection. Furthermore, a hybrid system is proposed that incorporates several techniques to ensure that movement detection performs reliably within a variety of different network configurations. An evaluation framework is designed and implemented that supports the assessment of a wide range of movement detection mechanisms. This test bed allows Mobile IP handoffs to be analysed in detail, with specific focus on the movement detection process. The performance of several movement detection optimisations is compared using handoff latency and packet loss as metrics. The evaluation framework also supports real-time Voice over IP (VoIP) traffic. This is used to ascertain the effects that different movement detection techniques have on the output voice quality. These evaluations not only provide a quantitative performance analysis of these movement detection mechanisms, but also a qualitative assessment based on a VoIP application