An implementation of SIP servers for internet telephony

The Session Initiation Protocol (SIP) has gained momentum and is being widely used both in the Internet and Next Generation Telecommunications network as the core signaling protocol. SIP operation relies on SIP servers which are responsible for forwarding of SIP messages. It has been shown that the performance of SIP servers is largely degraded during overload conditions due to the built in message retransmission mechanism of SIP. This is an obstacle in wide deployment of SIP for a realistic situation. In this paper we propose a distributed and adaptive window based overload control algorithm, which controls the amount of calls that are forwarded to a downstream SIP server in an attempt to prevent it from being overloaded. Our scheme uses the call establishment delay as a measure of the amount of load on the downstream server and unlike other methods does not rely on explicit feedback. Using simulations we show that our proposed method performs better than a commonly used overload control algorithm and exhibits fast convergence and is stable.

Selective parsing to enhance SIP performance

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.

Nowadays Session Initiation Protocol (SIP) has become the most widely used signaling protocol by the vendors in the Voice over Internet Protocol (VoIP) communication. As the SIP is gaining popularity, open architecture of VoIP often makes SIP vulnerable to so many threats. The paper introduces SIP along with its different security mechanisms with the focus on attack signatures generation technique to identify the malformed SIP messages. It aims to generate various malicious SIP messages and to verify the limitations of the existing classical Intrusion Detection System (IDS) in identifying the malicious messages. Then it sets out to implement an improved detection framework using the rules as regular expression. The experiments were carried out using SER as an open source SIP server, Kphone and X-lite as SIP clients and the SIPSAK as an attacking tool on a testbed architecture which was in a form of virtual environment created with VMware. The platforms used were BackTrack7 and Windows 7. The test was first performed on SNORT, an open source IDS and later by adding the improved detection rules in the configuration file of the SIP server. The successful attacks on the classical attack-signatures generation framework proved that the existing IDSs couldn't identify logical errors in the malicious SIP messages. The proposed detection module identified and rejected all kind of malicious SIP messages with acceptable processing overheads. Moreover the associated signatures database can also be incorporated into other VoIP protocols and open source IDSs like SNORT. This paper has provided the effective mechanism in creating well-formed defense against malicious SIP message attacks to ensure the integrity and security of VoIP subsystems.

Funding Information: V. CONCLUSIONS This work proposed four classification models based on LSTM RNNs to classify SIP dialogs. The detection probability was evaluated based on experimental data. To detect abnormal SIP dialogs, we have adopted classification features computed from the output of the LSTM RNN model and two different classification schemes were proposed. A semi-supervised scheme is shown to reach higher performance, achieving a detection probability of 99.45%, thus confirming the effective utility of the proposed methodology to detect abnormal SIP sequences in a short period of time. ACKNOWLEDGEMENTS This work was funded by Fundac¸ão para a Ciência e Tecnologia, under the projects InfoCent-IoT (PTDC/EEI-TEL/30433/2017), CoSHARE (PTDC/EEI-TEL/30709/2017), and RFSense (UIDB/50008/2020).

Recently P2P-SIP (Peer-to-Peer Session Initiation Protocol) has been proposed to improve the scalability and reliability of the traditional SIP (Session Initiation Protocol) networks. However, P2P-SIP makes SIP service unreliable because large-scale P2P-SIP networks are probably more dynamic than traditional SIP networks and the service nodes are very likely to fail or leave the P2P-SIP networks. To deal with this issue, we propose a novel and lightweight algorithm, which replicates SIP transactions information among the nodes in P2P-SIP networks and selects one of the successors of the failed or departed node as the takeover server. Moreover, to reduce the retrieving delay of SIP transactions replicas, we optimize our algorithm by storing these replicas directly in the successors of the failed or departed node. The simulation results demonstrate that our algorithm maintains 99% dialogs correctly in the presence of nodes failure with acceptable overheads over the Internet.

A Session Initiation Protocol (SIP) server may be overloaded by emergency-induced call volume, “American Idol” style flash crowd effects or denial of service attacks. The SIP server overload problem is interesting especially because the costs of serving or rejecting a SIP session can be similar. For this reason, the built-in SIP overload control mechanism based on generating rejection messages cannot prevent the server from entering congestion collapse under heavy load. The SIP overload problem calls for a pushback control solution in which the potentially overloaded receiving server may notify its upstream sending servers to have them send only the amount of load within the receiving server’s processing capacity. The pushback framework can be achieved by either a rate-based feedback or a window-based feedback. The centerpiece of the feedback mechanism is the algorithm used to generate load regulation information. We propose three new window-based feedback algorithms and evaluate them together with two existing rate-based feedback algorithms. We compare the different algorithms in terms of the number of tuning parameters and performance under both steady and variable load. Furthermore, we identify two categories of fairness requirements for SIP overload control, namely, user-centric and provider-centric fairness. With the introduction of a new double-feed SIP overload control architecture, we show how the algorithms can meet those fairness criteria.KeywordsSession Initiation ProtocolSession Initiation Protocol ServerFeedback AlgorithmSession Initiation Protocol MessageOverload ControlThese keywords were added by machine and not by the authors. This process is experimental and the keywords may be updated as the learning algorithm improves.

Overload occurs in Session Initiation Protocol (SIP) networks when SIP servers have insufficient resources to handle all the SIP messages they receive. Even though the SIP protocol provides a limited overload control mechanism through its 503 (Service Unavailable) response code, SIP servers are still vulnerable to overload. This document defines the behavior of SIP servers involved in overload control and also specifies a loss-based overload scheme for SIP.

Many research activities have been provided to allow Session Initiation Protocol (SIP) based communication in Peer-to-Peer (P2P). We have contributed to these activities by proposing a scalable proxy topology building and routing in P2P SIP network where SIP location service is integrated in proxies that run over peer entities. Therefore as SIP cannot work efficiently through a NAT (Network Address Translation) device without using a specific NAT traversal mechanism, our proposed architecture needs an efficient NAT traversal technique which can work easily and make end-to-end communication effective in our framework by meeting our previously defining requirements. This paper provides a suitable technique to our P2P SIP framework for SIP signaling and media flow. Our NAT traversal solution proposes an implementation of a smart SIP registrar server which stores the client end device's public contact information (IP address and port) instead of its private address and maintains this public address in the NAT device table. This approach allows the proxy node to always send the SIP signaling message to the correct address by checking the Registrar server before each transmission. To enable media flow to be exchanged between the two SIP clients, we use a media relay located in the public network. The proxy server of the originator of the signaling message is allowed to replace suitably the private addresses in the message body by the public addresses of the media relay. We provide some relevant experimental results to show that the approach is feasible.

Extended flow records with application layer (L7) information allow for detection of various types of malicious traffic. Voice over IP (VoIP) is an example of technology that works on L7 and many attacks against it cannot be reliably detected using just basic flow information. Session Initiation Protocol (SIP), which is commonly used for VoIP signalling, is a frequent target of many types of attacks. This paper proposes and evaluates a novel algorithm for near real time detection of username scanning and password guessing attacks on SIP servers. The detection is based on analysis of L7 extended flow records.

The Session Initiation Protocol (SIP) is an application-layer control protocol for creating, modifying and terminating multimedia sessions. An open issue is the control of overload that occurs when a SIP server lacks sufficient CPU and memory resources to process all messages. We prove that the problem of overload control in SIP network with a set of n servers and limited resources is in the form of NP-hard. This paper proposes a Load-Balanced Call Admission Controller (LB-CAC), based on a heuristic mathematical model to determine an optimal resource allocation in such a way that maximizes call admission rates regarding the limited resources of the SIP servers. LB-CAC determines the optimal "call admission rates" and "signaling paths" for admitted calls along optimal allocation of CPU and memory resources of the SIP servers through a new linear programming model. This happens by acquiring some critical information of SIP servers. An assessment of the numerical and experimental results demonstrates the efficiency of the proposed method.

The emergence of the Voice over Internet Protocol (VoIP) services requires interconnection between different technologies. In this paper we present an embedded Session Initiation Protocol (SIP) Proxy Gateway (GW) to a Mobile Network with Forward Error Corrections (FEC). Our server registers, locates and forwards the calls of an end user providing intelligent routing for user tracking. In addition, FEC along with a Markov-Chain loss model has been integrated into the server to perform a range of tests. Results prove that under high packet loss rate the embedded SIP server GW improves the speech quality.

This paper introduces several novel load-balancing algorithms for distributing Session Initiation Protocol (SIP) requests to a cluster of SIP servers. Our load balancer improves both throughput and response time versus a single node while exposing a single interface to external clients. We present the design, implementation, and evaluation of our system using a cluster of Intel x86 machines running Linux. We compare our algorithms to several well-known approaches and present scalability results for up to 10 nodes. Our best algorithm, Transaction Least-Work-Left (TLWL), achieves its performance by integrating several features: knowledge of the SIP protocol, dynamic estimates of back-end server load, distinguishing transactions from calls, recognizing variability in call length, and exploiting differences in processing costs for different SIP transactions. By combining these features, our algorithm provides finer-grained load balancing than standard approaches, resulting in throughput improvements of up to 24% and response-time improvements of up to two orders of magnitude. We present a detailed analysis of occupancy to show how our algorithms significantly reduce response time.

This paper discusses the benefits of a proposed Session Initiation Protocol (SIP)-based device communication Service for deployment on a Mobile Open Services Gateway Initiative (OSGi) framework. Mobile OSGi deployment provides a gateway to a mobile user’s Personal Area Network (PAN). When the SIP Service is deployed in OSGi, it is possible to bridge multiple devices and gateways, offering device and service mobility as users traverse multiple mobile and home networks. Additionally, the SIP Service provides protocol bridging via bridging bundles and grants application layer mobility to OSGi services. In this paper, we discuss a novel architecture, based on the deployment of the SIP Service in a mobile OSGi framework, which effectively delivers PAN and home network device interoperability. Finally, we offer a use case scenario and analysis of its implementation using our proposed architecture.

In this paper we address the problem of tracking in an end user-accessible, secure and efficient manner the information about the location of objects, for example luggage items transported by airlines, over a geographically distributed area, using Radio-Frequency IDentification (RFID) technology. We analyze the use of a centralized, commercial software approach based on the Electronic Product Code-based (EPC) standard, and of a distributed, open approach based on the Session Initiation Protocol (SIP) standard. The challenges include, first the use of inexpensive RFID devices to continuously track the location of objects. Second the integration of these devices within the Internet addressing scheme, in order to enable fast development of the related end-user services. Third the creation or adaptation of the required infrastructure (e.g. object location databases) and of the access mechanisms (via HTTP or other protocols). Fourth the identification of the most appropriate ways to communicate the location and tracking information to people who may not be computer experts, but just casual users of PCs and smartphones. We provide an initial comparison between the two approaches in terms of architecture, openness versus commercial support, network load and overall performance.