Analysis of Connectivity Model and Encoding Standards on IP Interconnection Implementation in Indonesia (Study Case: Low Data Rate up to 72 Mbps)

Saat ini Indonesia dihadapkan pada permasalahan dimana lalu lintas data, termasuk OTT di dalamnya, mendominasi layanan telekomunikasi yang menyebabkan pendapatan interkoneksi semakin menurun. Padahal, biaya pemeliharaan jaringan cenderung naik. Kemunculan teknologi IP dapat memberikan keuntungan, baik terhadap Operator dalam scissor effect maupun menaikkan tingkat loyalitas pelanggannya. Namun, saat ini regulasi Interkoneksi  di Indonesia masih menggunakan Time Division Multiplexing (TDM). Oleh karena itu, diperlukan suatu rekomendasi mengenai standarisasi pengkodean dan model interkoneksi IP. Dalam penelitian ini, aspek teknis dari model interkoneksi IP dianalisis dengan menggunakan perbandingan model, yaitu Peering dan Hubbing dengan metode no-transcoding pada 6 jenis codec(G.711a, G.711u, GSM, G.723, G.729, dan G.722) dengan pemberian berbagai beban trafik, (0 Mbps, 15 Mbps, 40 Mbps, dan 72 Mbps). Hasil performansi QoS berupa delay, Mean Opinion Score, packet loss, dan throughput yang diperoleh dari hasil simulasi masing-masing model dan kombinasi codec dianalisis dengan  menggunakan server VOIP Asterisk 11 dan Microsip 3.17.3 untuk SIP phone juga Wireshark 2.2.4 dianalisis untuk mengetahui performansinya. Nilai one way delay QoS mengacu pada standar nilai pada ITU-T G.1010. Dari hasil simulasi diperoleh bahwa secara keseluruhan dengan beban trafik sampai 72 Mbps, model Peering merupakan alternatif model interkoneksi IP yang terbaik. Selain itu, penggunaan codec G729 menghasilkan performansi paling baik dengan nilai delay paling minimum dan MOS paling besar, sehingga paling direkomendasikan untuk digunakan dalam implementasi interkoneksi IP. *****Currently, Indonesia is faced with problems where data traffic including OTT dominates the telecommunications services lead to interconnection revenue declining. In the other hand, the cost of network maintenance tend to increase. The emergence of IP technology may provide benefit to the operators in handling the scissor effect and improving the level of customer’s loyalty. However, the current interconnection regulations  in Indonesia are still using TDM. Therefore, a recommendation on standardization of IP encoding and interconnection model is required. In this research, technical aspect analysis of IP interconnect model is analyzed using comparison model, that is Peering and Hubbing with no-transcoding method on 6 types of codec (G.711a, G.711u, GSM, G.723, G.729, G.722) and loading of various traffic loads (0 Mbps, 15 Mbps, 40 Mbps, 72 Mbps). The results of QoS performance (delay, Mean Opinion Score, packet loss, throughput) obtained from the simulation results of each model and combination of codec are analyzed using VOIP server Asterisk 11 and Microsip 3.17.3 for SIP phone also Wireshark 2.2.4 to assess the performance. One-way delay QoS value refers to the standard in ITU-T G.1010. From the simulation results, it is obtained that for overall traffic load up to 72 Mbps, Peering model is the best alternative IP interconnect model. The usage of G.729 codec was the best performance codec with the minimum delay value and the biggest MOS, thus it was the most recommended for used in the IP interconnection implementation

The Influence of SIP Call Control Signalling on VoIP Quality of Experience

The rapid growth in subscribers and usage of multimedia services enlarges the volume of Session Initiation Protocol (SIP) call control signalling creating a need to understand Quality of Experience (QoE) in this case and improve it. This paper provides an analysis of influence of SIP call control signalling on QoE for Voice over Internet Protocol (VoIP) service. The aim was to investigate whether SIP call control signalling load has the influence on the human perception of SIP signalling performances and QoE, and to identify the importance of distinct SIP-based signalling performance metrics. Moreover, the intention was to determine whether SIP call control signalling load changes its impact if previously proposed algorithm for differentiated treatment of SIP messages is activated, and quantify mutual relationships of considered user perceptions and QoE. The findings show that SIP call control signalling load has a strong and negative impact on dependent variables and that the proposed algorithm improves QoE and human perception of SIP signalling performances

IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH

Voice over Internet Protocol (VoIP) is a technology that allows the transmission of voice packets over Internet Protocol (IP). Recently, the integration of VoIP and Wireless Local Area Network (WLAN), and known as Voice over WLAN (VoWLAN), has become popular driven by the mobility requirements ofusers, as well as by factor of its tangible cost effectiveness. However, WLAN network architecture was primarily designed to support the transmission of data, and not for voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS) for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11 standards that support Link Adaptive (LA) technique. However, LA leads to having a network with multi-rate transmissions that causes network bandwidth variation, which hence degrades the voice quality. Therefore, it is important to develop an algorithm that would be able to overcome the negative effect of the multi-rate issue on VoIP quality. Hence, the main goal ofthis research work is to develop an agent that utilizes IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned negative effect. This could be expected from the interaction between Medium Access Control (MAC) layer and Application layer, where the proposed agent adapts the voice packet size at the Application layer according to the change of MAC transmission data rate to avoid network congestion from happening. The agent also monitors the quality of conversations from the periodically generated Real Time Control Protocol (RTCP) reports. If voice quality degradation is detected, then the agent performs further rate adaptation to improve the quality. The agent performance has been evaluated by carrying out an extensive series ofsimulation using OPNET Modeler. The obtained results of different performance parameters are presented, comparing the performance ofVoWLAN that used the proposed agent to that ofthe standard network without agent. The results ofall measured quality parameters hav

Mobile Networks

The growth in the use of mobile networks has come mainly with the third generation systems and voice traffic. With the current third generation and the arrival of the 4G, the number of mobile users in the world will exceed the number of landlines users. Audio and video streaming have had a significant increase, parallel to the requirements of bandwidth and quality of service demanded by those applications. Mobile networks require that the applications and protocols that have worked successfully in fixed networks can be used with the same level of quality in mobile scenarios. Until the third generation of mobile networks, the need to ensure reliable handovers was still an important issue. On the eve of a new generation of access networks (4G) and increased connectivity between networks of different characteristics commonly called hybrid (satellite, ad-hoc, sensors, wired, WIMAX, LAN, etc.), it is necessary to transfer mechanisms of mobility to future generations of networks. In order to achieve this, it is essential to carry out a comprehensive evaluation of the performance of current protocols and the diverse topologies to suit the new mobility conditions