Session Initiation Protocol (SIP) Server Overload Control: Design and Evaluation

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

Abstruct- This paper addresses the design and implementation of Session Initiation Protocol (SIP) server components required for providing Internet telephony services. We first describe a family of SIP servers necessary for Internet telephony, and discuss overall functionality for each SIP server, Then we present our implementations of SIP servers including proxy server and registrar server, along with details of processing routines. We also discuss possible organization of SIP server components for providing Internet telephony services. We will also mention about our experiences on interoperability tests with other products.

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.

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.

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.

To meet the network requirement of the heterogeneous characteristics and future development of the smart home, a system architecture based on the Session Initiation Protocol(SIP) for smart home gateway is proposed. A middleware is designed in this paper to solve the data interaction problem between standard SIP and device services and provide the Interactive gateway application services for the external user of the family network based on sip. In the meantime, to settle the frequent access problem of the gateway database in the smart home, a database middleware is designed as well based on the mechanism of the database connection pool. Moreover, the feasibility of this scheme is proved by platform implementation and experiment verification based on the Linux embedded system.

In this paper we propose a novel architecture for converged multimedia SIP (session initiation protocol) services in wireline and 3G networks. The work on the future evolution of 3G systems is taking momentum with ongoing research on access to the IMS (IP Multimedia Subsystem) from any broadband access network. In particular WLAN (wireless local area network) access to the 3G services located in the IMS is being standardized though major points still need to be addressed. Wireline access to 3G services is not yet standardized and no deployable solution exists today. We propose in this paper a new architecture that allows an xDSL (digital subscriber line) access to 3G SIP based services, enabling a new generation of converged fixed/mobile services. This would also allow network operators to mutualise equipments and therefore reduce costs. In this paper, we analyze the high level requirements of this architecture, and we propose to introduce two new functional entities in the xDSL network: a SIP proxy and a border gateway. We study the impact of this new architecture with several transversal requirements: connectivity, QoS and security

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.