Research on the algorithm for identifying VoIP session

WiFi-enabled mobile handheld devices are being increasingly used for Voice over IP (VoIP) as well as supporting conventional data applications such as email, file transfer and web access. Wireless LANs (WLANs) are thus increasingly required to support Quality of Service (QoS)-centric applications, which are delay sensitive and require a certain level of throughput. While 802.11e goes some way towards meeting this need, severe congestion leading to unacceptable delays and packet loss can still occur. In the existing 802.11e standard, all VoIP sessions contend within the same prioritization Access Category (AC), despite potentially having very different, and varying one-way (M2E - Mouth to Ear) delays. In this paper we provide a detailed analysis on simulations that demonstrate how VoIP endpoints that are time synchronized can help optimize 802.11e EDCA in order to prioritize VoIP sessions that have relatively large M2E delays and thus distinguish between VoIP sessions. Using the NS-3 Network Simulator, we quantify the benefits achievable through synchronization of an 802.11e network handling multiple VoIP calls in the presence of other TCP traffic. We present a heuristic EDCA tuning algorithm which uses the ITU-T E-Model R-Factor QoS planning tool as the basis of control.

Voice over IP (VoIP) is one of the most popular Internet applications. Many VoIP protocols (e.g. SIP) are based on a central-server scheme in which a caller needs to query the central server first for the callee's information (e.g. IP address) before it can connect to them. However, such a centralized architecture has some drawbacks, such as an unbalanced load and a single point of the failure. As a result, some researchers have proposed the use of peer-to-peer (P2P) techniques for VoIP communication. However, compared to a centralized approach, setting up a VoIP connection over multiple hops could potentially take a longer time and might discourage users from utilizing this service. In this paper, we set out to study the performance of existing P2P protocols for VoIP applications in a more realistic setting based on traces collected from the Skype network. The Skype traces are used as an input to model the VoIP network topology. We evaluated the performance of three types of distribute hash table(DHT) protocols, namely, flat DHT, hierarchical DHT and proximity DHT, in terms of their lookup latency for the connection setup phase of VoIP applications. In addition, to understand the feasibility of using infrastructure-less VoIP in the real world, we adopted pair comparison techniques to evaluate the quality of the user experience. Finally, we proposed a new hybrid protocol for the infrastructure-less VoIP communication. We concluded that current P2P protocols for VoIP are not satisfactory as compared to the traditional centralized approach and there is still a lot of room for improvement.

Voice over internet protocol (VoIP) is being widely used, while the integration of voice and data provide several opportunities. Lower cost and more flexibility are the main advantages of the VoIP which derived the attention of the enterprises to it. Unfortunately in the other extreme, some security issues may come across with the extensive use of VoIP. The purpose of this review paper is to encourage the VoIP industry to consider the importance of the security matters of their application and to find the gaps by discussing the pros and cons of each application and performing the fair comparison. This article begins by discussing the motivation of security measures in VoIP applications and a list of possible threats such as denial-of-service (DoS), Call Hijacking and Interception. These security issues are the most well known attacks that can be carried out in current VoIP deployment. Next, we discuss about two VoIP applications, which are Skype and GTalk, and their security levels have been measured briefly by highlighting some of the security issues in them. These security measurements are used to perform the comparison between the studied VoIP applications. Finally some methods have been proposed to improve the security in the VoIP applications in future state.

Voice over Internet Protocol (VoIP) is no longer only feasible in wired networks, but also applicable through wireless networks as prevalence of wireless networks grows with the technology maturity and rising demand. While making a VoIP call over WiFi wireless networks is applicable, this is not a service with guaranteed quality. With limited bandwidth, an overloaded AP may cause wireless users to experience delayed packets, high packet loss rate, and even low throughput. IEEE 802.11 wireless Local Area Network (WLAN) was originally designed only for data traffic transmission, and therefore deficiency in real-time traffic transmission, such as VoIP applications, is inevitable. Not being able to timely transmit real-time packets not only fails to meet the demand of users, but also increases unnecessary loading in the wireless network. Industries and academies have made lots of efforts on providing QoS for real-time voice traffic which usually coexists with other classes of traffic. In this thesis, a call admission control mechanism is proposed which not only timely monitors the wireless networks, but also dynamically adjusts the number of admissible clients to wireless networks. New VoIP users not only know the network condition before entering it, but also accurately predict transmission quality through sending probing packets. With the mechanism, existing VoIP users in the wireless networks have their service protected, and potential VoIP users can expect good quality of service once admitted.

Voice over Internet Protocol (VoIP) is one of the most popular technologies nowadays that facilitate the user by providing different features as instant messages, phone calls, video calls, and voicemails. Basic VoIP protocols were designed to be efficient instead of secure. After numerous attacks on these protocols several solutions were proposed to prevent against these threats. In this paper, we focus on the security of Session Initiation Protocol (SIP) that is used to initiate, modify, and terminate the VoIP sessions. The paper presents the design and implementation of secure layered architecture for SIP, which adds a new layer to the standard SIP layer model and entitled as Security layer. The Security layer provides authentication, authorization, adaptable feature, and secure key exchange, based on our newly designed protocol, named as Session Initiation Protocol using Single Sign-On (SIPSSO). In order to implement the secure layered architecture based on SIPSSO, we have developed an Android Secure Call application and extend the open source Asterisk accordingly. After the designing and implementation phases, we have verified the SIPSSO protocol formally by using an automated security verification tool, Scyther. Our analysis results reveal that by adding Security layer, we ensured protection against different SIP attacks such as Eavesdropping, Man In The Middle (MITM) attack, Message Tampering, Replay attack, Session Teardown, and Spam over Internet Telephony (SPIT).

An objective no-reference video quality assessment method is proposed to evaluate the video quality in voice over Internet protocol (VoIP) applications under network distortion. In order to assess the distortions of videos in VoIP applications caused by codec and packet loss, the method evaluates the quality of received videos by calculating the block artifacts and blurring. Video blocking artifacts are measured by splitting the picture into small blocks and calculating the difference of the pixels around each border while video blurring is measured by getting edge information through Sobel operator, and counting the gradient histogram. The weighted sum of block artifacts scores and blurring scores can be calculated using linear regression. In order to eliminate the impact of different video contents, the scores which represent the quality of videos need to be normalized. The experimental results show that the objective quality scores have a strong correlation with the subjective quality scores, and the algorithm can be used in video quality assessment in VoIP applications.

Recent voice over Internet protocol (VoIP) technology is attracting more and more attention and interest in the industry. VoIP applications such as IP telephony systems involve sending voice transmissions as data packets over private or public IP networks as well as reassembled and decoded on the receiving side. Security concerns are always the major barrier that prevents many businesses from employing VoIP technologies. In this paper, we discuss various VoIP security threats and possible approaches to tackle the threats in VoIP applications

Recently, with the expansion and advent of the wireless Internet such as WiFi, mobile WiMax, and mobile peer-to-peer (P2P) voice over IP (VoIP) communication through P2P overlay network in all-IP area has been deeply investigated as a research area. When the connected peer terminal leaves hot spot of all-IP area, it will be lost the communication session. In this paper, we propose seamless voice connection over P2P mobile VoIP and conventional wireless network. In this case, the concept of vertical handoff is not the handoff of IP layer but that of session layer to create new VoIP session in latency before the previous session ends. We have two kind of wireless communication network; one is all-IP and the other is CDMA cellular and assumes that the mobile node has the functionality to communicate with each other network. We certain the dynamically adjusted trigger method is better than the fixed trigger in the success of new call setup processing and the decrease of unnecessary signaling in cellular network. We think that this work is able to achieve seamless and low-cost communication between P2P VoIP and cellular network.

Service providers and enterprises expect voice over Internet Protocol (VoIP)to enable third-party application development, allowing them to mix-and-match best-of-breed applications from multiple vendors. By taking advantage of applicable techniques employed for Web-based services, service broker functionality deployed in the network will provide a framework for specifying VoIP application interaction rules. However, each unique VoIP deployment will require development of a complex, customized, domain-specific set of interaction rules for the service brokers. The complexity of VoIP application interaction rule development in a multi-vendor environment will provide Lucent with an opportunity to sell application integration services to enterprises and service providers.

With the exponential increase of mobile users and internet subscribers, the utilization of voice over internet protocol (VoIP) application is increasing dramatically. People exploit different VoIP applications for effective communication, for example, Google Meet, Microsoft Skype, Zoom video conferencing applications, etc. The single-ended speech quality metrics are employed for measuring and monitoring the quality of speech. However, different types of degradations present in the surroundings distort the quality of speech. In order to meet the desired quality of experience (QoE) level of end-user while using VoIP applications, it is necessary to reduce VoIP degradations and obtain the optimized speech quality. Along that line, this paper investigates the conjunction of filtering of silence and noise as a pre-processing block with the single-ended speech quality metric under various common occurring degradations encountered during VoIP communication. This can help the internet service providers in understanding the potential root cause of decrement in quality of speech and then applying the QoE management service to fulfill desired human QoE level. Results demonstrate that the deployment of joint pre-processing on speech samples under various VoIP degradations improves the quality of speech to a great extent.

The increasing use of Voice over Internet Protocol (VoIP) technology in telecom fraud has become a serious global concern. Its ability to spoof caller IDs and IP addresses, and the use of overseas or anonymized servers make VoIP-based scams difficult to trace and regulate. As a result, distinguishing VoIP calls from conventional mobile phone calls based on voice signal characteristics is crucial for enhancing anti-fraud measures. However, existing forensic techniques often struggle to accurately identify speech transmitted via VoIP. To address this challenge, we propose a dual-level 1D-CNN that leverages both frame and utterance features for effective VoIP detection. After evaluating a range of acoustic features, we primarily focus on short-frame Mel-Frequency Cepstral Coefficients (MFCCs) due to their effectiveness in capturing VoIP characteristics. Given the frame-based processing and transmission nature of VoIP, we employ a 1D-CNN, rather than the more commonly used 2D-CNN that treats spectrograms as image, to extract frame-level codec features. Finally, we propose a dual-level classification strategy: the frame-level classifier captures encoding discrepancies within individual frames, while the utterance-level classifier aggregates these frame-level features to learn global encoding patterns through global covariance pooling. Experimental results on the VoIP Phone Call Identification Database (VPCID) demonstrate that the proposed method consistently outperforms existing approaches, delivering superior accuracy and robustness across a wide range of challenging scenarios. Moreover, comprehensive ablation studies validate the effectiveness and rationale behind the design of the proposed model architecture.

New methods for voice quality evaluation for IP networks

Internet Protocol (IP) and Voice over Internet Protocol (VoIP) are technologies that are changing the way wired and wireless communication networks are developed. More recently, VoIP applications using the Internet are allowing people to communicate with one another without using traditional telephone networks. As these applications are incorporated into wireless devices, such as cordless phones and cellular phones, new wireless VolP applications as well as traditional IP applications are allowing telecommunication providers to develop services with greater flexibility and lower cost structures than traditional telephone service applications. This paper reviews industry and market information on VoIP, and discusses emerging standards that will influence the development of wireless IP applications. New customer premise equipment (CPE) devices and hand-sets are presented along with the emerging set of applications and services currently available and in development. A discussion of network architectures necessary to provide wireless IP applications including voice over IP is also presented.

Voice over Internet Protocol (VoIP) applications based on peer-to-peer (P2P) communications have been experiencing considerable growth in terms of number of users. To overcome filtering policies or protect the privacy of their users, most of these applications implement mechanisms such as protocol obfuscation or payload encryption that avoid the inspection of their traffic, making it difficult to identify its nature. The incapacity to determine the application that is responsible for a certain flow raises challenges for the effective management of the network. In this paper, a new method for the identification of VoIP sessions is presented. The proposed mechanism classifies the flows, in real-time, based on the speech codec used in the session. To make the classification lightweight, the behavioral signatures for each analyzed codec were created using only the lengths of the packets. Unlike most previous approaches, the classifier does not use the lengths of the packets individually. Instead, it explores their level of heterogeneity in real time, using entropy to emphasize such feature. The results of the performance evaluation show that the proposed method is able to identify VoIP sessions accurately and simultaneously recognize the used speech codec.

The fifth-generation new radio technology (5G NR) introduces improved functions to the air interface. In addition, the 5G NR non-standalone (NSA) will operate with long-term evolution, enabling vehicle-to-everything communications (V2X) for improved infotainment services. V2X includes four main classes of communications: vehicle-to-vehicle, vehicle-to-infrastructure, vehicle-to-pedestrian devices, and vehicle-to-network. However, the stringent transmission frequency, latency, and throughput requirements of infotainment applications constrain the transmitting packets of 5G-V2X-based NSA in highway scenarios. In this paper, the latency is improved by preventing the physical layer of gNodeB and the user equipment (UE) from sending redundant packets for service in a highway scenario. The proposed approach adopts an adaptive neuro-fuzzy inference system (ANFIS), a powerful modeling technique based on artificial neural networks ,and a fuzzy inference system. The performance of ANFIS is compared with that of the traditional 5G V2X NSA architecture in a simulation study using Voice over Internet Protocol (VoIP) traffic. The delays, throughputs, and packet losses of both architectures are determined in radio link control (RLC) and VoIP applications. The switch-modes, signal-to-interference-noise ratios (SINRs), hybrid automatic repeat request (HARQ) error rate, channel quality indicators (CQIs), served blocks, and transmission-state of gNodeB are computed for the two architectures for device-to-device (D2D), uplink (UL) and downlink (DL) traffic directions. The simulation results show comparable SINRs, CQIs, served blocks ,andswitchmodes in both scenarios, but the presented ANFIS model significantly outperforms the traditional architecture in delay by 66% in D2D, 29% in UL, 25% in DL, and packet loss by 21% in UL in RLC, the HARQ error rate by 9% in D2D, 30% in UL, 95% in DL, transmission-state in gNodeB by 29%, and the delay by 4% for UEs, and frame loss by 90%for UE in VoIP.