Performance Of Session Initiation Protocol Research Articles

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.

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.

Analytical Model to Create Proxy Server Sessions in Multimedia Networks

One of the most popular and widely applied protocols on multimedia networks is the Session Initiation Protocol (SIP) to create, modify, and terminate the sessions. SIP is the platform of Next Generation Networks (NGN). In this way, SIP should be able to respond to the needs of such a largely-used network. One of the major problems in SIP networks is overload. This challenge creates a sharp drop in quality of service for NGN users. In this regard, many studies have been conducted on the effectiveness of this protocol, especially under overload. A new analytical model is developed that prioritizes the SIP message processing. An analytical approach is proposed based on the Mean Value Analysis (MVA) algorithm in queue theory. Considering some appropriate assumptions customizing MVA as to implement this proposed model and to cope with the limitations of the MVA is highly essential. The output of the analytical model is compared with the standard SIP model obtained from the simulator and the results confirm that prioritizing original messages would enhance the SIP performance at different load conditions. Prioritization of original messages is advantageous, and outperforms the normal SIP. Nevertheless, prioritizing the repeated messages not only has no advantage, but also its performance is less than the normal SIP.

Detection And Countermeasure Scheme For Call-Disruption Attacks On SIP-Based Voip Services

Owing to its simplicity and flexibility, the session initiation protocol (SIP) has been widely adopted as a major session-management protocol for Internet telephony or Voice-over IP (VoIP) services. However, SIP has faced various types of security threats. Call-disruption attacks are some of the most severe threats they face, and can greatly inconvenience consumers. In this paper, we analyze such SIP call-disruption attacks, and propose a method for detecting and counteracting them by extending the SIP INFO method with authentication. Using the proposed method, both the target user and the SIP server can detect the existence of a call-disruption attack on a user and counteract the attack. We demonstrate the effectiveness of the proposed method from the viewpoint of computational complexity by configuring a test-bed with an Asterisk SIP proxy server and an SIP performance (SIPp) emulator.

Selective parsing to enhance SIP performance

The session initiation protocol (SIP) is used as the signaling protocol in the IP multimedia subsystem (IMS) and the signaling is becoming computing intensive comparing to the current telecommunication network. The SIP is a text-based protocol with characteristics of unordered and verbose headers, variable-size message, and case-insensitive keyword. It imposes challenges for an efficient message processing. The property of SIP elements being able to process SIP messages quickly is critical for the performance of IMS networks. This article investigates the performance of SIP message processed in SIP servers, mainly focusing on improving message parsing by introducing a method named selective parsing for SIP message (SP4SIP). By modeling and analyzing a SIP server with a tandem Jackson network, it is concluded that parsing messages is the bottleneck of a SIP server performance, i.e., it is the most processing intensive activity in the system. To validate the approach, it has been implemented in a high-performance SIP server in the authors' lab. The results show that selective parsing for SIP message can indeed reduce processing time.

On SIP performance

The performance characteristics of Session Initiation Protocol (SIP) network elements determine what services can be supported by SIP networks. These elements must be able to provide customers with appropriate response times for new services, e.g., push-to-talk service. In IP-telephony converged networks, SIP-based signaling elements must meet the requirements of telephony signaling — that is, SIP-based servers will be required to process messages according to real-time constraints. Performance of SIP network elements also determines how much SIP networks cost. To support large customer bases, service providers must provide adequate performance on economical components. In this paper, we examine the main processing requirements of SIP elements and describe a number of performance metrics. We support our analyses with measurements of four different implementations of selected SIP functions.

Analysis of SIP-based mobility management in 4G wireless networks

Providing seamless mobility support is one of the most challenging problems towards the system integration of fourth generation (4G) wireless networks. Because of the transparency to the lower layer characteristics, application-layer mobility management protocol like the Session Initiation Protocol (SIP) has been considered as the right candidate for handling mobility in the heterogeneous 4G wireless networks. SIP is capable of providing support for not only terminal mobility but also for session mobility, personal mobility and service mobility. However, the performance of SIP, operating at the highest layer of the protocol stack, is only as good as the performance of the underlying transport layers in such a heterogeneous environment. In this paper we analyze the handoff performance of SIP in a IP-based 4G network with Universal Mobile Telecommunication System (UMTS) and Wireless LAN (WLAN) access networks. Analytical results show that the handoff to a UMTS access network introduces a minimum delay of 1.4048 s for 128 kbps channel, while for handoff to a WLAN access network the minimum delay is 0.2 ms. In the former case the minimum delay is unacceptable for streaming multimedia traffic and requires the deployment of soft-handoff techniques in order to reduce the handoff delay to a desirable maximum limit of 100 ms.