Session Initiation Protocol Communication 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.

A Performance Evaluation of In-Memory Databases Operations in Session Initiation Protocol

Real-time communication has witnessed a dramatic increase in recent years in user daily usage. In this domain, Session Initiation Protocol (SIP) is a well-known protocol found to provide trusted services (voice or video) to end users along with efficiency, scalability, and interoperability. Just like other Internet technology, SIP stores its related data in databases with a predefined data structure. In recent, SIP technologies have adopted the real advantages of in-memory databases as cache systems to ensure fast database operations during real-time communication. Meanwhile, in industry, there are several names of in-memory databases that have been implemented with different structures (e.g., query types, data structure, persistency, and key/value size). However, there are limited resources and poor recommendations on how to select a proper in-memory database in SIP communications. This paper provides recommended and efficient in-memory databases which are most fitted to SIP servers by evaluating three types of databases including Memcache, Redis, and Local (OpenSIPS built-in). The evaluation has been conducted based on the experimental performance of the impact of in-memory operations (store and fetch) against the SIP server by applying heavy load traffic through different scenarios. To sum up, evaluation results show that the Local database consumed less memory compared to Memcached and Redis for read and write operations. While persistency was considered, Memcache is the preferable database selection due to its 25.20 KB/s for throughput and 0.763 s of call–response time.

Efficient IPv4-IPv6 translation mechanism for IMS using SIP proxy

Next-generation IP Multimedia Subsystem (IMS) has a promise future in internet technologies. Technically, IMS utilises Session Initiation Protocol (SIP) for call communication. Potentially, SIP clients might reside in coexistence network family like IPv6 and/or IPv4. The coexistence of SIP clients considered a serious issue in SIP communication and IPv6 migration in general. In this paper a SIP-Proxy IP Translator (SPIPT) is proposed to translate SIP call between from IPv6 to IPv4 and vice versa. The proposed solution comes to insure successive connectivity in term of IP version compatibility. The experiment shows acceptable CPU usage, throughput, and Call Response Time parameters. SIP proxy hits negligible amount (1.93%) of CPU usage when 100 calls load. In addition, throughput also stays with the safety level with 1.9 Kbps in the same 100 calls. In sum, the solution aims to reduce call setup complexity and leverages user experience which in turn accelerates IPv6 transition.

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.

SECURING VOIP COMMUNICATIONS IN AN OPEN NETWORK

Channeling voice calls over IP networks has brought many advantages to enterprise communications, but it also creates some security risks. In this paper a technique to enhance the security of vocal and text communication is proposed. This technique is a combination of multipath routing and secret sharing scheme. Information is conveyed securely from one person to another over an open network using the secret sharing scheme. The original data is divided into two shares and each of these shares is sent over the open network through multiple different paths using the multipath routing technique. In case of a man-in-the-middle attack the person will not obtain any information of the original form with only one of the shares. Only if the person collects both the shares he/she can reconstruct the original information. A mechanism to secure SIP (Session Initiation Protocol) and text communication is also discussed in this paper.

End to End Encryption Architecture for Voice over Internet Protocol

Session Initiation Protocol (SIP) is an open standard protocol and constitutes the provisioning of services like Internet Telephony and Instant Messaging. Vulnerabilities in SIP have made it possible to exploit it and launch many of the known internet attacks and also few specific attacks, thus affecting the services deploying SIP for session management. To maintain the confidentiality and integrity of voice data security mechanisms need to be deployed. Currently available security measures do not take into account real time nature of data and are generic i.e. not optimized for VoIP technology. This paper presents new end to end encryption architecture for securing the VoIP calls which use SIP to establish their session taking into account the real time nature of data. General Terms SIP Vulnerability, VoIP Security, Threats and attacks, SIP Messages, SIP Communication

A Survey of Open Source Products for Building a SIP Communication Platform

The Session Initiation Protocol (SIP) is a multimedia signalling protocol that has evolved into a widely adopted communication standard. The integration of SIP into existing IP networks has fostered IP networks becoming a convergence platform for both real-time and non-real-time multimedia communications. This converged platform integrates data, voice, video, presence, messaging, and conference services into a single network that offers new communication experiences for users. The open source community has contributed to SIP adoption through the development of open source software for both SIP clients and servers. In this paper, we provide a survey on open SIP systems that can be built using publically available software. We identify SIP features for service development and programming, services and applications of a SIP-converged platform, and the most important technologies supporting SIP functionalities. We propose an advanced converged IP communication platform that uses SIP for service delivery. The platform supports audio and video calls, along with media services such as audio conferences, voicemail, presence, and instant messaging. Using SIP Application Programming Interfaces (APIs), the platform allows the deployment of advanced integrated services. The platform is implemented with open source software. Architecture components run on standardized hardware with no need for special purpose investments.

VOIP in Peer-to-Peer using Session Initiation Protocol

42 ABSTRACT Session Initiation Protocol (SIP) is an application layer protocol for multimedia sessions developed by IETF. It is a signaling protocol for voice over IP (VoIP). Originally, SIP was specified as a client server protocol to set up multimedia communications. However, recent proposals suggest using SIP in a peer-to-peer setting. The initiators of P2P SIP claim higher robustness against failure as well as easier configuration and maintenance as in the main motivation for peer-to-peer SIP. Clearly, a peer-to-peer setting imposes new security threats to SIP communications; design decisions that affect security, which include node-ID computation, overlay routing, authentication of nodes, SIP message semantics, and representation of identity. For instance, the lack of a central authority makes authentication of peers a hard problem. Without authentication, adversary nodes can spoof identity and falsify messages in the network. The peer-to-peer network may be affected by attacks like bootstrapping, identity enforcement, free riding, and anonymity. The requirements for secure network are secure node-ID assignment, secure routing table maintenance and secure message forwarding. In our project, we focus on providing secure node-IDs. Solely the IP address (without port) is used for node-ID generation. It is suggested as a better choice than using a combination of IP address and port which is being originally used. VOIP in PEER-TO-PEER using Session Initiation Protocol

Security challenges for peer-to-peer SIP

Recent research activities have proposed using a peer-to-peer network for user registration and user location in session initiation protocol (SIP)-based voice-over-IP (VoIP) networks. The main motivation for peer-to-peer (P2P) SIP is higher robustness as well as easy configuration and maintenance (compared to client-server SIP). However, these advantages come at the price of security. In this article the security challenges of using a P2P network as a substrate for SIP communication are explored. After a short introduction to SIP and structured P2P networks, the different approaches that have been proposed for using P2P technology in conjunction with SIP are compared. We focus on design issues that affect security in order to identify the security implications of using a structured overlay network for SIP registration and location lookup. Finally, possible solutions for securing P2P SIP are examined and explicit suggestions on how to improve the security of P2P SIP are given