Performance Analysis of WebRTC and SIP-based Audio and Video Communication Systems

Fowdur, Tulsi Pawan; Ramkorun, N.; Chiniah, P. K.

doi:10.1007/s42979-020-00380-z

Cited by 8 publications

(2 citation statements)

References 16 publications

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“…Fowder conducted a performance analysis of the WebRTC and SIP protocols in [ 51 ]. WebRTC provided better-quality video communication than SIP with respect to the signal-to-noise ratio (SNR), whereas SIP outperforms WebRTC in audio transmission in the context of segmental SNR.…”

Section: Related Workmentioning

confidence: 99%

Performance analysis: Securing SIP on multi-threaded/multi-core proxy server using public keys on Diffie–Hellman (DH) in single and multi-server queuing scenarios

Bhatti,

Sidrat,

Saleem

et al. 2024

PLoS ONE

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The rapid replacement of PSTN with VOIP networks indicates the definitive phase-out of the PBX/PABX with smartphone-based VOIP technology that uses WLAN connectivity for local communication; however, security remains a key issue, regardless of the communication coverage area. Session initiation protocol (SIP) is one of the most widely adopted VOIP connection establishment protocols but requires added security. On the Internet, different security protocols, such as HTTPS (SSL/TLS), IPSec, and S/MIME, are used to protect SIP communication. These protocols require sophisticated infrastructure and some pose a significant overhead that may deteriorate SIP performance. In this article, we propose the following: i) avoid using Internet bandwidth and complex Internet protocols for local communication within an organization, but harness WLAN connectivity, ii) use multi-threaded or multicore computer systems to handle concurrent calls instead of installing hardware-based SIP servers, and iii) run each thread in a separate core. Cryptography is a key tool for securely transmitting confidential data for long- and short-range communication, and the Diffie-Hellman (DH) protocol has consistently been a popular choice for secret key exchanges. Primarily, used for symmetric key sharing, it has been proven effective in generating public/private key pairs, sharing public keys securely over public channels, and subsequently deriving shared secret keys from private/public keys. This key exchange scheme was proposed to safeguard VOIP communication within WLANs, which rely on the SIP for messaging and multimedia communication. For ensuring an efficient implementation of SIP, the system was rigorously analyzed using the M/M/1 and M/M/c queuing models. We analyze the behavior of SIP servers with queuing models with and without end-to-end security and increase users’ trust in SIP security by providing a transparent sense of end-to-end security as they create and manage their private and public keys instead of relying on the underlying SIP technology. This research implements instant messaging, voice conversation, and secret key generation over DH while implementing and observing the role of multi-threading in multiqueue systems that serve incoming calls. By increasing the number of threads from one to two, the SIP response time improved from 20.23809 to 0.08070 min at an arrival rate of 4250 calls/day and a service rate of three calls/min. Similarly, by adding one to seven threads, the queue length was reduced by four calls/min. Implementing secure media streaming and reliable AES-based signaling for session confidentiality and integrity introduces a minor 8-ms tradeoff in SIP service performance. However, the advantages of implementing added security outweigh this limitation.

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Section: Related Workmentioning

confidence: 99%

Performance analysis: Securing SIP on multi-threaded/multi-core proxy server using public keys on Diffie–Hellman (DH) in single and multi-server queuing scenarios

Bhatti,

Sidrat,

Saleem

et al. 2024

PLoS ONE

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“…This enabled us to optimize the camera's capturing and encoding capabilities for efficient streaming. [62] Additionally, we leveraged the UV4L streaming server's two-way audio/video intercom recorder (WebRTC) feature, which utilizes advanced codecs and protocols specifically designed for low latency and real-time communication. By incorporating WebRTC, we significantly reduced latency and achieved near-instantaneous audio and video transmission between the Raspberry Pi and the Flutter app.…”

Section: A Block Diagrammentioning

confidence: 99%

Design and Development of swarm AGV’s alliance for Search and Rescue operations

Pyla,

Pandalaneni,

Raju

et al. 2023

Journal of Robotics and Control

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Rapid response is essential for saving lives in search and rescue operations since the amount of time is critical. In this project, a swarm of autonomous ground vehicles (AGVs) equipped with ROS-based software architecture will be designed and built for rapid search and rescue missions. The swarm of AGVs will function autonomously to navigate through challenging areas and be outfitted with a variety of sensors, including cameras, and LIDAR. The proposed system will be capable of performing 2D mapping, live video surveillance, autonomous navigation, victim/object detection, and two-way audio communication. The goal of the project is to reduce the risk to human life in dangerous areas by providing a quick and efficient response system for search and rescue operations. A centralized management system with a de-centralized module will be created as part of the project to keep an eye on and manage the AGV horde. However, there will be a functionality to take control of a specific robot in the swarm network when needed. In difficult areas where it might not be safe for humans to operate, the suggested method will enable quick and efficient search and rescue operations.

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A systematic review on WebRTC for potential applications and challenges beyond audio video streaming

Mahmoud,

Abozariba

2024

Multimed Tools Appl

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Video conferencing and live streaming are being used in various industries, such as healthcare, gaming, telecommunication, manufacturing and others. As technology progresses, the need for real-time data transmission with minimal latency has increased. Web Real-Time Communication () addresses this need effectively. WebRTC is a technology designed to provide real-time communication through web and mobile browsers. Its low latency and P2P communication capabilities make it a convenient technology for secure, efficient communication in real-time applications. This paper reviews the key features of WebRTC, discusses its strengths and weaknesses and investigates a detailed analysis of 83 existing studies. Moreover, It evaluates all use cases that can be adopted by WebRTC by examining their descriptions, problem statements, and research gaps based on literature to date. Finally, It highlights the open research directions for the emerging technologies and enhancements of WebRTC. to identify their potential applications.

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Performance Analysis of WebRTC and SIP-based Audio and Video Communication Systems

Cited by 8 publications

References 16 publications

Performance analysis: Securing SIP on multi-threaded/multi-core proxy server using public keys on Diffie–Hellman (DH) in single and multi-server queuing scenarios

Performance analysis: Securing SIP on multi-threaded/multi-core proxy server using public keys on Diffie–Hellman (DH) in single and multi-server queuing scenarios

Design and Development of swarm AGV’s alliance for Search and Rescue operations

A systematic review on WebRTC for potential applications and challenges beyond audio video streaming

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