Session Initiation Protocol System Research Articles

A NETWORK PERFORMANCE COMPARISON OF WEBRTC AND SIP AUDIO AND VIDEO COMMUNICA TIONS

Multimedia services are one of the internet needs with high data traffic network count since 2019. Two of the multimedia services, Web Real-Time Communications (WebRTC) and Session Initiation Protocol (SIP), have been widely used in applications for conducting video conferencing. The main objective of this research is to analyze network performance by an application with a client-server for audio and video communications developed with WebRTC and SIP protocols. The SIP system uses the FreePBX server, and the softphone application uses Bria. Whereas WebRTC uses JavaScript with servers on Ubuntu using Node.js. The analysis application uses a star topology and runs on a local network using Wi-Fi. After testing, the results show that the throughput, jitter, and packet loss of WebRTC are better than SIP. This result is caused by several factors including the type of codec, the type of platform, and the way of signaling and routing of each protocol.

Design of P2P-SIP System Network Structure Based on JXTA

Session Initiation Protocol (SIP) is one of the most important signaling connection protocols in the field of VoIP (Voice over Internet Protocol). However, the existing SIP network has some problems such as manual configuration, difficult maintenance and poor scalability. With the development of P2P (Peer-to-Peer) technology, the combination of SIP and P2P has become a trend. JXTA is an open peer-to-peer platform, which provides basic application services for building P2P network. The topic chooses JXTA development platform to build P2P network, and then constructs SIP soft terminal on it. Based on the in-depth study of JXTA structured network and SIP protocol, a P2P SIP system is designed and implemented. The system can maintain normal SIP communication without maintaining the central server. In this paper, the detailed design and implementation process of P2P-SIP system are given, including node join and leave, user join, user search and session establishment. This paper focuses on the implementation process of user search, that is, the collaborative process of discovery service, resolution service, aggregation service and endpoint routing service. Finally, the feasibility of the system is verified by functional test.

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

Video and audio communications have become an integral part of all spheres of life. Two commonly used real-time communication protocols for IP-based video and audio communications are the session initiation protocol (SIP) and web real-time communications (WebRTC). These two protocols have been widely used in softphone and video conferencing applications. The main aim of this paper is to make a comparative analysis of the performance of client server applications for video and audio communications developed by both SIP and WebRTC. The SIP system comprises of a desktop client developed in C#, a mobile client developed in Android studio and a FreeSwitch server. For the WebRTC application, the client was developed in JavaScript and the server in Node.js. The WebRTC application also included a speech translation API for real-time speech translation and employed two different codecs namely VP8 and VP9 via a modification of the Session Description Protocol (SDP) header. Results showed that WebRTC peer-to-peer audio communication provides a much lower quality in terms of SSNR than the SIP audio calls. However, for video communications, WebRTC provides better quality in terms of PSNR than SIP. The performance of the WebRTC application was also evaluated with the different codecs in terms of PSNR as well as the impact of the real-time speech translation delay.

On modeling of priority-based SIP request scheduling

Summary As the retransmission mechanism of the Session Initiation Protocol (SIP) significantly increases the amount of incoming traffic and makes an overloaded SIP server unstable, the models created for a stable SIP system cannot be effectively used to evaluate the performance of the overloaded server. To overcome this difficulty, in this paper, we present a fluid-flow model to analyze the priority-based request scheduling mechanism (PRSM) behavior in the overloaded SIP server equipped with an infinite buffer. The PRSM has been proposed in our earlier studies to prevent the overloading of SIP servers, where the original SIP request messages have strict priority over the retransmissions. The performance evaluation of the proposed model for the PRSM and the reference model of the conventional SIP has been reported under various test scenarios. The experiment results of the proposed model verify the effectiveness of the PRSM fluid-flow model as we already demonstrated in our previous studies using our simulation tool. Without loss of generality, the numerical results closely match with the simulation results, indicating that the proposed models can accurately capture the dynamic behavior of a SIP server with the PRSM.

On fluid‐flow modeling of priority based request scheduling for finite buffer SIP server

SummaryWith the widespread deployments of voice‐over‐internet protocol services, the existing session initiation protocol (SIP) design cannot scale up for large network sizes. Events triggering a demand burst or a server slowdown can cause SIP server overload, overload propagation, and crash, thus bringing down the whole SIP network. Since the SIP retransmission mechanism exacerbates the overload condition, existing models created for a stable SIP system cannot be effectively used to analyze an overloaded server. In this paper, we propose a fluid‐flow model to characterize the behavior of the finite buffer SIP server equipped with priority‐based request scheduling mechanism (PRSM). The model for the PRSM uses primary and secondary queues for the original request messages and the retransmitted requests, respectively. The performance metrics, namely, the failed call attempts and the response delay from sending INVITE request until receiving a 100‐Trying response, are derived using the arrival time‐slot tracking and the removal processes of the proposed fluid‐flow model. We conducted test cases under the heavy traffic conditions, where the overload is caused by bulk and bursty arrivals or server slowdown. The numerical results closely match with the simulation results for all experiments, indicating that the proposed model can accurately capture the dynamic behavior of an SIP server with the PRSM. The experiments demonstrate that the number of failed call attempts is close to 0 and the mean response delay is kept constant around 175 ms for the PRSM when the buffer size is higher than 1K while both metrics are significantly higher for the conventional SIP.

Usage control in SIP-based multimedia delivery

The Session Initiation Protocol (SIP) is an application layer signaling protocol for the creation, modification and termination of multimedia sessions and VoIP calls with one or more participants. While SIP operates in highly dynamic environments, in the current version its authorization support is based on traditional access control models. The main problem these models face is that they were designed many years ago, and under some circumstances they tend to be inadequate in modern highly dynamic environments. Usage Control (UCON), instead, is a model that supports the same operations as traditional access control models do, but it further enhances them with novel ones. In previous work, an architecture supporting continuous authorizations in SIP, based on the UCON model, was presented. In this article, an authorization support implementing the whole UCON model, including authorizations, obligations and conditions, has been integrated in a SIP system. Moreover, a testbed has been set up to experimentally evaluate the performance of the proposed security mechanism.

Modelling chaotic behaviour of SIP retransmission mechanism

The Session Initiation Protocol (SIP) retransmission mechanism can cause SIP network collapse with short-term overload. In this paper, we investigate with a fluid modelling approach the chaotic behaviour of the SIP retransmission mechanism in SIP networks. We capture the complex correlation structure in SIP systems through a detailed and novel queuing analysis. To dimension a buffer size which can avoid unnecessary message drop in a SIP server, we develop a sufficient condition for a stable SIP system analytically based on our fluid model. We also apply our fluid model to the simulation of a complex SIP system. We compare the simulation results achieved through our fluid models with those based on OPNET® event-driven approach to demonstrate the validity of our approach.

Research on P2P-SIP based VoIP system enhanced by UPnP technology

As voice over IP (VoIP) develops rapidly nowadays, session initiation protocol (SIP) has been the mainstream IP multimedia communication signal protocol in the real-time multimedia communications. But SIP system allocates heavy process and network load on the central server, thus bringing the bottleneck of the system. And it also has ‘single-point failure’ risk. While, peer-to-peer (P2P) overlays, especially distributed hash tables (DHTs) based structured P2P can provide features of decentralization and self-organization. So in our project, we design and implement a multimedia VoIP system exploiting P2P overlay approaches into SIP system. Moreover, we extend the terminal capability by universal plug and play (UPnP) technology so that the user can gain the best multimedia communication performance.

A Peer-to-Peer SIP System based on Service-Aware Transport Overlays

The Session Initiation Protocol (SIP) traditionally establishes and manages its sessions with centralized servers, which have been challenged by issues with TCP/IP networks right from its beginning, for instance, issues with NAT traversal or network congestions. On the other hand, there are peer-to-peer and overlay techniques that potentially can make SIP working better under various network conditions. Overlay networks can provide a good abstraction from the real network, thus hiding the presence of NATs. Some proposals on peer-to-peer SIP have been recently presented. While these proposals attempted to solve a subset of challenges faced by traditional SIP, such as removing the centralized SIP entities, there is to the best of our knowledge no single solution offering a comprehensive view of the peer-to-peer SIP architecture and operation. This paper presents a novel approach for a peer-to-peer SIP system, using overlay techniques for signalling and media transport in the Internet. This system is based on the Ambient Networks Service-Aware Transport Overlay (SATO) system. The proposed system replaces the traditional SIP proxy/registrar function with a distributed lookup mechanism, adding overlay functionality to the SIP signalling and to the RTP traffic. Moreover, different from previous proposals, our approach deliberately places media/packet relays into the SIP/RTP paths, which allows an efficient session management and media communication.

A framework for protecting a SIP-based infrastructure against malformed message attacks

This paper presents a framework that can be utilized for the protection of session initiation protocol (SIP)-based infrastructures from malformed message attacks. Its main characteristic is that it is lightweight and that it can be easily adapted to heterogeneous SIP implementations. The paper analyzes several real-life attacks on VoIP services and proposes a novel detection and protection mechanism that is validated through an experimental test-bed under different test scenarios. Furthermore, it is demonstrated that the employment of such a mechanism for the detection of malformed messages imposes negligible overheads in terms of the overall SIP system performance.

SIP message prioritization and its applications

Session Initiation Protocol (SIP) signaling is an integral part of the IP Multimedia Subsystem (IMS) to support services such as Voice over IP (VoIP), multimedia sessions, presence, and instant messaging (IM). All of these services share a common SIP signaling infrastructure. In spite of careful network engineering, SIP network elements may experience high-load situations, in which SIP messages may experience too much delay or even get dropped. This paper addresses how such undesired high-load situations can be handled effectively by introducing a message prioritization mechanism in SIP servers. To validate the mechanism, it has been implemented in a high-performance JAIN <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">∗</sup> -SIP-compliant Java <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">∗</sup> SIP stack. This SIP stack incorporates several other innovations, such as a very efficient SIP message parser, and it can be used for a variety of SIP-based applications. Its design enables service providers to define custom message prioritization and rejection strategies based on SIP message characteristics, system state, and statistics. Measurements show that high-priority emergency messages can indeed be serviced without significant delay or loss in a high-load situation. The concept of and techniques for SIP prioritization are the subjects of a Lucent Technologies patent application.