LAN Traffic Capture Applications Using the Libtins Library

Capturing traffic and processing its contents is a valuable skill that when put in the right hands makes diagnosing and troubleshooting network issues an approachable task. Apart from aiding in fixing common problems, packet capture can also be used for any application that requires getting a deeper understanding of how things work under the hood. Many tools have been developed in order to allow the user to study the flow of data inside of a network. This paper focuses on documenting the process of creating such tools and showcasing their use in different contexts. This is achieved by leveraging the power of the C++ programming language and of the libtins library in order to create custom extensible sniffing tools, which are then used in VoIP (Voice over IP) and IDS (Intrusion Detection System) applications.

Simulation of inorganic Al/Pt platforms

The understanding of semaphores is an important issue. In this work, we disprove the understanding of voice-over-IP, which embodies the technical principles of cryptoanalysis. We introduce a novel algorithm for the study of access points, which we call BlandTrub

Zeroize: A New Method to Improve Utilizing 5G Networks when Running VoIP over IPv6

5G technology propagation curve is ascending rapidly. 5G will open up the horizon to improve the performance of many other IP-based services such as voice over IP (VoIP). VoIP is a worldwide technology that is expected to rule the telecommunication world in the near future. However, VoIP has expended a significant part of the 5G technology bandwidth with no valuable use owing to its lengthy packet header. This issue even worsens when VoIP works in IPv6 networks, where the wasted bandwidth and airtime may reach 85.7% of 5G networks. VoIP developers have exerted many efforts to tackle this snag. This study adds to these efforts by proposing a new method called Zeroize (zero sizes). The main idea of the Zeroize method is to use superfluous fields of the IPv6 protocol header to carry the digital voice data of the packet and, thus, reduce or zeroize the VoIP packet payload. Although simple, the Zeroize method achieves a considerable reduction of the wasted bandwidth of 5G networks, which also directly affects the consumed airtime. The performance analysis of the Zeroize method shows that the consumed bandwidth is saved by 20% with the G.723.1 codec. Thus, the Zeroize method is a promising solution to reduce the wasted bandwidth and airtime of 5G networks when running VoIP over IPv6.

Towards Standardized Prevention of Unsolicited Communications and Phishing Attacks

The world of communication technology is changing fast and the means of communication are moving towards a packet switched transmission systems such as Voice over IP (VoIP). Formerly call identity spoofing of the displayed number in circuit switched (CS) networks was too difficult to perform so that people could be sure that when receiving a call on their mobile phone or at home, the displayed number is the one as it is supposed to be. Nowadays this is not the case anymore, voice communication from the internet with VoIP is cheap and spam calls can be easily realized without any costs, also it is getting easier to perform spoofed calls with wrong display name or number. The mobile network operators have no mechanisms to tackle those threats, but standardization activities are already in place within the security group SA3 of 3GPP. This paper provides an overview of the current status of the standards activities and shows the most promising solutions that are proposed up to now. The proposed solutions detect unsolicited communications and spoofed calls by tracing back to the displayed number used in the attack.

Improved Jitter Buffer Management for WebRTC

This work studies the jitter buffer management algorithm for Voice over IP in WebRTC. In particular, it details the core concepts of WebRTC’s jitter buffer management. Furthermore, it investigates how jitter buffer management algorithm behaves under network conditions with packet bursts. It also proposes an approach, different from the default WebRTC algorithm, to avoid distortions that occur under such network conditions. Under packet bursts, when the packet buffer becomes full, the WebRTC jitter buffer algorithm may discard all the packets in the buffer to make room for incoming packets. The proposed approach offers a novel strategy to minimize the number of packets discarded in the presence of packet bursts. Therefore, voice quality as perceived by the user is improved. ITU-T Rec. P.863, which also confirms the improvement, is employed to objectively evaluate the listening quality.

Contracting VoIP Packet Payload Down to Zero

The inefficient use of the IP network bandwidth is a fundamental issue that restricts the exponential spreading of Voice over IP (VoIP). The primary reason for this is the big header size of the VoIP packet. In this paper, we propose a method, called Short Voice Frame (SVF), that addresses the big header size of the VoIP packet. The main idea of the SVF method is to make effective use of the VoIP packet header fields that are unneeded to the VoIP technology. In particular, these fields will be used for temporarily buffering the voice frame (VoIP packet payload) data. This will make the VoIP packet payload short or even zero in some cases. The performance evaluation of the proposed SVF method showed that the use of the IP network bandwidth has improved by up to 28.3% when using the G.723.1 codec.

Pemaksimalan Unified Commucation Untuk Kestabilan Pendidikan Di Masa Pandemi Covid- 19

Pendidikan merupakan kunci pembangunan sumber daya manusia di setiap negara, tidak terkecuali di Indonesia. Pandemi covid-19 yang saat ini terjadi, mengakibatkan gangguan dalam sistem pembelajaran bagi pendidikan tinggi di Indonesia, salah satunya adalah mengharuskan sistem pembelajaran dengan menggunakan sistem pembelajaran jarak jauh. Dirasa saat ini sistem pembelajaran jarak jauh di Indonesia, khususnya DKI Jakarta masih dapat disempurnakan. Salah satunya dengan menggunakan teknologi UC. Unified Communication merupakan gabungan dari layanan telepon dengan Komputer yang dapat terintegrasi seperti voice over ip, pesan instan , konferensi untuk membantu suatu kegiatan menjadi lebih produktif. Penggunaan Unified Communication dalam dunia pendidikan universitas dalam penerapannya seperti menggunakan zoom , webex , Google meet yang saling terhubung secara virtual dalam waktu bersamaan. Unified communication sangat diperlukan dalam dunia pendidikan pada saat ini , karna pada masa covid -19 ini tidak memungkinkan adanya pertemuan tatap muka secara langsung. Maka dari itu sistem Unified Communication memberikan kemudahan di dunia pendidikan saat ini.Oleh sebab itu, penulis memilih topik ini dengan tujuan dapat mengetahui bagaimana cara memaksimalkan UC di bidang pendidikan. Penulis akan melakukan studi literatur serta menyebarkan kuesioner kepada koresponden yang merupakan pengajar dan pelajar. Dimana mereka menggunakan sistem pembelajaran jarak jauh pada pendidikan tingkat tinggi, dalam hal ini kami mengkhususkan untuk S1. Harus bisa mendukung kegiatan dengan sistem Unified Communication guna untuk mencegah cobud-19 dan juga harus membiasakan diri beradaptasi dengan sistem UC ini , sehingga dosen dengan mahasiswa bisa menjalankannya Tidak perlu bertemu langsung dan meminimalkan kemungkinan siswa tertular virus corona

Steganography and Steganalysis in Voice over IP: A Review

The rapid advance and popularization of VoIP (Voice over IP) has also brought security issues. VoIP-based secure voice communication has two sides: first, for legitimate users, the secret voice can be embedded in the carrier and transmitted safely in the channel to prevent privacy leakage and ensure data security; second, for illegal users, the use of VoIP Voice communication hides and transmits illegal information, leading to security incidents. Therefore, in recent years, steganography and steganography analysis based on VoIP have gradually become research hotspots in the field of information security. Steganography and steganalysis based on VoIP can be divided into two categories, depending on where the secret information is embedded: steganography and steganalysis based on voice payload or protocol. The former mainly regards voice payload as the carrier, and steganography or steganalysis is performed with respect to the payload. It can be subdivided into steganography and steganalysis based on FBC (fixed codebook), LPC (linear prediction coefficient), and ACB (adaptive codebook). The latter uses various protocols as the carrier and performs steganography or steganalysis with respect to some fields of the protocol header and the timing of the voice packet. It can be divided into steganography and steganalysis based on the network layer, the transport layer, and the application layer. Recent research results of steganography and steganalysis based on protocol and voice payload are classified in this paper, and the paper also summarizes their characteristics, advantages, and disadvantages. The development direction of future research is analyzed. Therefore, this research can provide good help and guidance for researchers in related fields.