Session Initiation Protocol Proxy Research Articles

On Implementing IPTV Platform with IPv4 and IPv6 Devices

The end of IPv4 addresses is now a reality. Providers not updated to IPv6 will have to hurry up the IPv6 start in its own network. Introduction of IPv6 means not only change of main routers but also change of mentality in operators, applications’ programmers besides end users. Even when for the last years the core network is prepared for transferring IPv6 traffic, other built-in parts of the Internet limit the IPv6 start. Examples of these limitations we find in not IPv6-awareness of many applications and services. For instance, voice over IP service, which uses session initiation protocol (SIP) needs to implement IPv6 aware SIP proxies and IPv6 aware AAA (authentication, authorization and accounting) servers as well as adapting application programming interfaces to IPv6. Internet protocol television (IPTV) system includes many different hardware devices, which not always are IPv6 compatible. In this paper, we propose a global solution for integrating all the devices, these one working on IPv4 and these one working on IPv6, under the same IPTV platform. This solution allows end users to receive IPTV stream irrespective of IP protocol used. The proposed solution is particularly relevant for small IPTV systems, which, step by step, are adapting into IPv6.

Session Initiation Protocol Proxy in a Role of a Quality of Service Control Application in Software-Defined Networks

This article deals with quality of service (QoS) in internet protocol (IP) telephony by applying software-defined networking (SDN) tools. The authors develop a new design that deterministically classifies real-time protocol (RTP) streams based on data found in session initiation protocol (SIP) using SIP proxy as a mediator, and the concept making this possible is called SDN. Compared to traditional networks, SDN allows us to approach network configuration differently. SDN networks are programmable through software applications running on top of the SDN controller. One of the technologies that might benefit from this concept is IP telephony, which often needs an additional priority management configuration to ensure consistent quality of its real-time media exchange. Typically, a session protocol for real-time communications is SIP, and as such, its infrastructure may be used to classify the traffic in question and take advantage of the centralized approach of SDN networks to distribute the class information across the switching devices. Different approaches and possible applications are discussed in the conclusion. The contribution of this paper lies in the proposal of SDN-based QoS mechanisms. The entire design of the concept was implemented and validated in a laboratory environment. The results clearly demonstrate the efficiency of the proposed approach.

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.

SDVoIP—A Software-Defined VoIP Framework For SIP And Dynamic QoS

AbstractSoftware-defined networking (SDN) allows for the decoupling of the control and data planes, enabling more programmability and a global view of the network. Previous research indicates that traditional applications recreated using SDN principles allow for more granularity and customization. In this research, we extend the insights behind SDN to develop a Voice over Internet Protocol (VoIP) framework with the objective to enhance traditional Session Initiation Protocol (SIP) operation and quality of service (QoS) approaches. The contributions of this research are 2-fold: first, an SIP control application is implemented, which communicates with an SDN controller to provide VoIP call registration and call routing capabilities, thereby eliminating the need for specialized SIP proxy hardware devices; second, a dynamic QoS application is developed that provides the ability to make network-wide QoS decisions based on real-time network measurements of latency, bandwidth and packet loss. Functional validation of the framework is performed to verify its operation. The experiment results indicate that the proposed framework allows for enhancements to traditional QoS implementations.

The Devil is in the Detail: SDP-Driven Malformed Message Attacks and Mitigation in SIP Ecosystems

VoIP services, in general, and session initiation protocol (SIP) ones, in particular, continue to grow at a fast pace and have already become a key component of next-generation networks. Despite this proliferation, SIP-based services expose a large attack surface for perpetrators, especially those who seek to cause denial of service (DoS). While so far, a plethora of works in the literature have been devoted to the detection of DoS attacks in SIP ecosystems, the focus is on those which exploit SIP headers neglecting the message body. In an effort to fill this gap, this paper concentrates on the detection of DoS attacks, which, instead, capitalize on the session description protocol (SDP) part of SIP requests. To this end, we not only scrutinize this ilk of attacks and demonstrate their effect against the end-user but also develop an open-source extensible SDP parser module capable of detecting intentionally or unintentionally crafted SDP segments parasitizing in SIP requests. Following a firewall-based logic, currently, the parser incorporates 100 different rules organized in four categories (policies) based on the corresponding RFC 4566. Through extensive experimentation, we show that our scheme induces negligible overhead in terms of processing time when working as a software module in either the SIP proxy or a separate machine in front of the latter.

Design, implementation and performance evaluation of a proactive overload control mechanism for networks of SIP servers

The extent and diversity of systems, provided by IP networks, have made various technologies approach integrating different types of access networks and convert to the next generation network (NGN). The session initiation protocol (SIP) with respect to facilities such as being in text form, end-to-end connection, independence from the type of transmitted data, and support various forms of transmission, is an appropriate choice for signalling protocol in order to make connection between two IP network users. These advantages have made SIP be considered as a signalling protocol in IP multimedia subsystem (IMS), a proposed signalling platform for NGNs. Despite having all these advantages, SIP protocol lacks appropriate mechanism for addressing overload causing serious problems for SIP servers. SIP overload occurs when a SIP server does not have enough resources to process messages. The fact is that the performance of SIP servers is largely degraded during overload periods because of the retransmission mechanism of SIP. In this paper, we propose an advanced mechanism, which is an improved method of the windows based overload control in RFC 6357. In the windows based overload control method, the window is used to limit the amount of message generated by SIP proxy server. A distributed adaptive window-based overload control algorithm, which does not use explicit feedback from the downstream server, is proposed. The number of confirmation messages is used as a measure of the downstream server load. Thus, the proposed algorithm does not impose any additional complexity or processing on the downstream server, which is overloaded, making it a robust approach. Our proposed algorithm is developed and implemented based on an open source proxy. The results of evaluation show that proposed method could maintain the throughput close to the theoretical throughput, practically and fairly. As we know, this is the only SIP overload control mechanism, which is implemented on a real platform without using explicit feedback.

High Load Control Mechanism for SIP Servers

To start voice, image, instant messaging, and generally multimedia communication, session communication must begin between two participants. SIP (session initiation protocol) that is an application layer control induces management and terminates this kind of sessions. As far as the independence of SIP from transport layer protocols is concerned, SIP messages can be transferred on a variety of transport layer protocols including TCP or UDP. Mechanism of Retransmission that is embedded in SIP could compensate for the missing packet loss, in case of need. This mechanism is applied when SIP messages are transmitted on an unreliable transmission layer protocol like UDP. Also, while facing SIP proxy with overload, it could cause excessive filling of proxy queue, postpone increase of other contacts, and add to the amount of the proxy overload. In the present work, while using UDP as transport layer protocol, invite retransmission timer (T1) was appropriately regulated and SIP functionality was improved. Therefore, by proposing an adaptive timer of invite message retransmission, attempts were made to improve the time of session initiation and consequently improve the performance. Performance of the proposed SIP was implemented and evaluated by SIPP software in a real network environment and its accuracy and performance were demonstrated.

SIP Flooding Attack Detection with a Multi-Dimensional Sketch Design

The session initiation protocol (SIP) is widely used for controlling multimedia communication sessions over the Internet Protocol (IP). Effectively detecting a flooding attack to the SIP proxy server is critical to ensure robust multimedia communications over the Internet. The existing flooding detection schemes are inefficient in detecting low-rate flooding from dynamic background traffic, or may even totally fail when flooding is launched in a multi-attribute manner by simultaneously manipulating different types of SIP messages. In this paper, we develop an online detection scheme for SIP flooding attacks, by integrating a novel three-dimensional sketch design with the Hellinger distance (HD) detection technique. In our sketch design, each SIP attribute is associated with a two-dimensional sketch hash table, which summarizes the incoming SIP messages into a probability distribution over the sketch table. The evolution of the probability distribution can then be monitored through HD analysis for flooding attack detection. Our three-dimensional design offers the benefit of high detection accuracy even for low-rate flooding, robust performance under multi-attribute flooding, and the capability of selectively discarding the offending SIP messages to prevent the attacks from bringing damages to the network. Furthermore, we design a scheme to control the distribution of the normal traffic over the sketch. Such a design ensures our detection scheme’s effectiveness even under the severe distributed denial of service (DDoS) scenario, where attackers can flood over all the sketch table entries. In this paper, we not only theoretically analyze the performance of the proposed detection techniques, but also resort to extensive computer simulations to thoroughly examine the performance.

Comparative Study of Various Voip Applications In 802.11A Wireless Network Scenario

Today, Voice over Wireless Local Area Network (VOWLAN) is the most accepted Internet application. There are a large number of literatures regarding the performance of various WLAN networks. Most of them focus on simulations and modeling, but there are also some experiments with real networks. This paper explains the comparison of performance of two different VOIP (Voice over Internet Protocol) applications over the same IEEE 802.11a wireless network. Radio link standard 802.11a have maximum transmission rate of 54Mbps. First protocol is session initiation protocol (SIP) and second is H.323 protocol. First one has an agent called SIP proxy. Second have a gateway reflects the characteristics of a Switched Circuit Network (SCN). With this comparison we have required to obtain a better understanding of wireless network suitability for voice communication in IP network.

The Overload Reduction in SIP Servers through Exact Regulation of the Retransmission Timer of the Invite Message

To initiate voice, image, instant messaging and general multimedia communication, the Session communication must initiate between two participants. SIP (Session initiation protocol) is an application layer control, which task is creating management, and terminating this kind of Sessions. With regard to the independence of SIP from the Transport layer protocols, the SIP messages can be transferred on a variety of Transport layer protocols such as TCP or UDP. The mechanism of Retransmission, which has been embedded in SIP, is able to compensate the missing Packet loss, if needed. The application of this mechanism is when SIP messages are transmitted on an unreliable transmission layer protocol such as UDP. This mechanism, while facing with SIP proxy with overload, causes excessive filling of proxy queue, delays the increase of other contacts and adds the amount of the proxy overload. We in this article, while using UDP, as the Transport layer protocol, by regulating the Invite Retransmission Timer appropriately (T1), have improved the SIP functionality. Therefore, by proposing an Adaptive Timer of Invite message retransmission, we have tried to improve the time of Session initiation and as a result, improving the performance. The performance of the proposed SIP, by the SIPP software in a real network environment has been implemented and evaluated and its accuracy and performance has been demonstrated.

The Impact of TLS on SIP Server Performance: Measurement and Modeling

Securing Voice over IP (VoIP) is a crucial requirement for its successful adoption. A key component of this is securing the signaling path, which is performed by the Session Initiation Protocol (SIP). Securing SIP can be accomplished by using Transport Layer Security (TLS) instead of UDP as the transport protocol. However, using TLS for SIP is not yet widespread, perhaps due to concerns about the performance overhead. This paper studies the performance impact of using TLS as a transport protocol for SIP servers. We evaluate the cost of TLS experimentally using a testbed with OpenSIPS, OpenSSL, and Linux running on an Intel-based server. We analyze TLS costs using application, library, and kernel profiling and use the profiles to illustrate when and how different costs are incurred. We show that using TLS can reduce performance by up to a factor of 17 compared to the typical case of SIP-over-UDP. The primary factor in determining performance is whether and how TLS connection establishment is performed due to the heavy costs of RSA operations used for session negotiation. This depends both on how the SIP proxy is deployed and what TLS operation modes are used. The cost of symmetric key operations such as AES, in contrast, tends to be small. Network operators deploying SIP-over-TLS should attempt to maximize the persistence of secure connections and will need to assess the server resources required. To aid them, we provide a measurement-driven cost model for use in provisioning SIP servers using TLS. Our cost model predicts performance within 15% on average.

Overload control in SIP networks using no explicit feedback: A window based approach

The Session Initiation Protocol (SIP) has gained momentum and is being widely used both in the Internet and Next Generation Telecommunications networks as the core signaling protocol. SIP operation relies on SIP servers which are responsible for routing SIP messages. It has been shown that the performance of SIP servers is largely degraded during overload periods due to the built in message re-transmission mechanism of SIP. In this paper we propose a distributed and end-to-end adaptive window based overload control algorithm, which does not use explicit feedback from the downstream server. Upstream servers use call establishment delay as a measure of the amount of load on the downstream server. Therefore, the proposed algorithm imposes no additional complexity or processing on the downstream server which is overloaded, making it a very robust approach. Using simulations we show that our proposed method achieves higher throughput than a commonly used overload control algorithm and is also fair among different upstream servers under different network latencies. To the best of our knowledge, fairness under different network latencies has not been previously addressed in the context of SIP overload control. In addition, compared with approaches using explicit feedback, our scheme is less sensitive to network latency. The proposed overload control algorithm is also implemented in the OpenSIPS open source SIP proxy and shown to perform as expected under various conditions.

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.

Approach to stress tests in SIP environment based on marginal analysis

In the field of performance metrics and measurements of SIP (Session Initiation Protocol) Proxy and B2BUA (Back-to-Back User Agent) no standardized methodology has been presented yet. This gap results in a problematic determination of a hardware, the performance of which would be cost-effective and sufficient for the running the SIP Server in a given environment. Today practice relies on the administrator's skills and experience with the needs of the telephony infrastructure. From this and the increasing usage of SIP based VoIP technologies come the main reasons for creating a methodology that would allow administrators to precisely measure the SIP Server performance and compare it to other software and hardware platform. This work also utilizes SIP Server performance measurements to comparison the results taken when transcoding was in use and when it was not and provides the means for comparison of B2BUAs platform independently.

Session Initiation Protocol firewall for the IP Multimedia Subsystem core

As the deployment of Session Initiation Protocol (SIP) accelerates, there is an accompanying need to secure the SIP infrastructure. One way to do so is through a SIP firewall, which is loosely defined as a device that blocks attacks mounted via SIP. Using this definition, a firewall is indistinguishable from a session border controller (SBC), also used by SIP service providers to secure their networks. SIP firewalls and SBCs are often deployed by SIP service providers at the periphery of the network to impose some manner of order on the SIP traffic before allowing it to enter the network. In this vein, a SIP firewall needs to effectively block many SIP attacks and distinguish a distributed denial of service (DDoS) attack from a classic overload traffic arrival rate. But what exactly is a firewall and what features should it provide in its role of inspecting SIP traffic bound for the service provider's network? What are the economic and technical tradeoffs necessary for ubiquitous deployment? In this paper, we define the role of a firewall in protecting the IP Multimedia Subsystem (IMS) or SIP-based core network, distinguish it from an SBC, and characterize the specific threats to SIP messages at the L2 (data link layer), L3 (network layer), L4 (transport layer), and L5 (session layer). We show how a SIP firewall can thwart these attacks and we propose an implementation based on a simplified, but fully hardware accelerated SIP proxy as a front end SIP firewall. Such a system naturally blocks most attacks and implements many defense mechanisms.

A study of performance and scalability metrics of a SIP proxy server – a practical approach

In recent years, Internet Protocol (IP) telephony has been a real alternative to the traditional Public Switched Telephone Networks (PSTN). IP telephony offers more flexibility in the implementation of new features and services. The Session Initiation Protocol (SIP) is becoming a popular signalling protocol for Voice over IP (VoIP) based applications. The SIP proxy server is a software application that provides call routing services by parsing and forwarding all the incoming SIP packets in an IP telephony network. The efficiency of this process can create large scale, highly reliable packet voice networks for service providers and enterprises. We established that the efficient design and implementation of the SIP proxy server architecture can enhance the performance characteristics of a SIP proxy server significantly. Since SIP proxy server performance can be characterised by its transaction states of each SIP session, we emulated the M / M / 1 performance model of the SIP proxy server and studied some of the key performance benchmarks such as average response time to process the SIP calls, and mean number of SIP calls in the system. We showed its limitations, and then studied an alternative M / M / c based SIP proxy server performance model with enhanced performance model and studied additional key performance characteristics such as server utilisation, queue size and memory utilisation. Provided the comparative results between the predicted results with the experimental results conducted in a lab environment.

An enhanced SIP proxy server for wireless VoIP in wireless mesh networks

The wireless mesh network (WMN) has emerged recently as a promising technology for next-generation wireless networking. In WMNs, it is important to provide high quality multimedia service in a flexible and intelligent manner. To address this issue in this article, we study the Session Initiation Protocol (SIP) for wireless voice over IP (VoIP) applications. Especially, we investigate the technical challenges in WMN VoIP systems and propose a design of an enhanced SIP proxy server to overcome them. An analysis of the signaling process and a study of simulation results have shown the advantages of our proposed approach.

Bandwidth Allocation and Session Scheduling using SIP

Session Initiation Protocol (SIP) is a new signaling protocol designed to establish multimedia sessions in telecommunication networks. In this paper, we suggest the extension of SIP functionalities to coordinate QoS mechanisms deployed in IP networks, and especially in DiffServ domain. Indeed, the interaction between small and big TCP sessions may have dramatic consequences on small TCP sessions. Hence, we use SIP to achieve QoS management on a session basis, in which the over all activity of the user during the session is considered. The suggested mechanisms deal with two issues: first, session scheduling based on session duration and/or volume, and second bandwidth allocation on a per-flow basis using equivalent bandwidth estimation techniques. The proposed mechanisms are implemented in the SIP proxy server as QoS management algorithms, and they are validated by simulations.

Improving performance and reliability of an IMS network by co-locating IMS servers

The IP Multimedia Subsystem (IMS) is an architecture for supporting multimedia services via a Session Initiation Protocol (SIP) infrastructure. IMS, specified by the 3rd Generation Partnership Project (3GPP), is increasingly adopted as the reference multimedia network architecture by providers of wireline and wireless infrastructure and associated multimedia application services. One of the characteristics of the IMS architecture is that many different and mandatory SIP proxies are used for each end-to-end SIP session. This may adversely impact the overall IMS call processing performance, which is already affected by the universal usage of SIP. This also makes each end-to-end SIP session dependent on the good functioning of many hosts, which decreases reliability while increasing significantly the end-to-end message latency. This paper analyzes the performance and quantifies the benefits of having the SIP servers running on the same host. It then explains which types of SIP calls in the IMS network can benefit from the co-location of IMS servers and shows how to design the IMS network to maximize IMS server co-location. Finally, it introduces Lucent Technologies' SIPia BUS software architecture, which implements the co-location capability by introducing a new SIP transport in a way that is both natural and non-disruptive for the SIP stack and the SIP user agent.

An extensible framework for constructing SIP user agents

A Session Initiation Protocol (SIP) user agent is an endpoint in a signaling network that can send or receive SIP messages. One can build a functional user agent in a few hundred lines of Java <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">∗</sup> code that sets up a call between two SIP phones. However, such a user agent will not fully comply with the protocol. Writing a compliant user agent is a complex undertaking involving thousands of lines of code. The iSURF framework greatly reduces the effort of this undertaking. iSURF uses a SIP transaction library called siptrans as its transaction processing layer. However, iSURF can use a different transaction library, and siptrans can be used in a different framework or even in a SIP proxy. In this paper, we describe the protocol requirements for a SIP user agent and how our framework facilitates building such an agent. We also describe the design and architecture of both iSURF and siptrans.