Service re-routing for service network graph: efficiency, scalability and implementation

The key to success in Next Generation Network is service routing in which service requests may need to be redirected as in the case of the INVITE request in Session Initiation Protocol. Service Path (SPath) holds the authentication and server paths along side with service information. As the number of hops in a redirection increases, the length of SPath increases. The overhead for service routing protocols which uses SPath increases with the length of SPath. Hence it is desirable to optimize SPath to ensure efficiency and scalability of protocols involving service routing. In this paper, we propose a re-routing strategy to optimize service routing, and demonstrate how this strategy can be implemented using SPath to enhance the efficiency and scalability of Service Network Graph (SNG)

VERIFIABLY SECURE SESSION INITIATION PROTOCOL REQUESTS

Techniques are described for reducing the amount of spam and congestion on Session Initiation Protocol (SIP) devices and endpoints to significantly improve customer User Experience (UX). This may be packaged as a web Application Programming Interface (API) that provides an “anti-spam as a service” for other web-based clients

Telephony Denial of Service Defense at Data Plane (TDoSD@DP)

The Session Initiation Protocol (SIP) is an application-layer control protocol used to establish and terminate calls that are deployed globally. A flood of SIP INVITE packets sent by an attacker causes a Telephony Denial of Service (TDoS) incident, during which legitimate users are unable to use telephony services. Legacy TDoS defense is typically implemented as network appliances and not sufficiently deployed to enable early detection. To make TDoS defense more widely deployed and yet affordable, this paper presents TDoSD@DP where TDoS detection and mitigation is programmed at the data plane so that it can be enabled on every switch port and therefore serves as distributed SIP sensors. With this approach, the damage is isolated at a particular switch and bandwidth saved by not sending attack packets further upstream. Experiments have been performed to track the SIP state machine and to limit the number of active SIP session per port. The results show that TDoSD@DP was able to detect and mitigate ongoing INVITE flood attack, protecting the SIP server, and limiting the damage to a local switch. Bringing the TDoS defense function to the data plane provides a novel data plane application that operates at the SIP protocol and a novel approach for TDoS defense implementation.Final Accepted Versio

A novel mechanism for anonymizing Global System for Mobile Communications calls using a resource-based Session Initiation Protocol community network

Considering the widespread adoption of smartphones in mobile communications and the well-established resource sharing use in the networking community, we present a novel mechanism to achieve anonymity in the Global System for Mobile Communications (GSM). We propose a Voice over Internet Protocol infrastructure using the Session Initiation Protocol (SIP) where a smartphone registers on a SIP registrar and can start GSM conversation through another smartphone acting as a GSM gateway, by using a SIP intermediate without an extra cost. The testbed that we developed for empirical evaluation revealed no significant quality of service degradation

Using an External DHT as a SIP Location Service

Peer-to-peer Internet telephony using the Session Initiation Protocol (P2P-SIP) can exhibit two different architectures: an existing P2P network can be used as a replacement for lookup and updates, or a P2P algorithm can be implemented using SIP messages. In this paper, we explore the first architecture using the OpenDHT service as an externally managed P2P network. We provide design details such as encryption and signing using pseudo-code and examples to provide P2P-SIP for various deployment components such as P2P client, proxy and adaptor, based on our implementation. The design can be used with other distributed hash tables (DHTs) also

Towards a scalable video interactivity solution over the IMS

Includes bibliographical references (leaves 72-76).Rapid increase in bandwidth and the interactive and scalability features of the Internet provide a precedent for a converged platform that will support interactive television. Next Generation Network platforms such as the IP Multimedia Subsystem (IMS) support Quality of Service (QoS), fair charging and possible integration with other services for the deployment of IPTV services. IMS architecture supports the use of the Session Initiation Protocol (SIP) for session control and the Real Time Streaming Protocol (RTSP) for media control. This study aims to investigate video interactivity designs over the Internet using an evaluation framework to examine the performance of both SIP and RTSP protocols over the IMS over different access networks. It proposes a Three Layered Video Interactivity Framework (TLVIF) to reduce the video processing load on a server

Automatic voice relay with open source Kiara

One way for Deaf people to communicate with hearing people over the telephone is to use a voice relay. The service is often provided with a human relay operator that relays text into voice, and vice versa, on behalf of the Deaf and hearing users. In developed countries, voice relay is frequently subsidised by governments or service providers. There is no such service in South Africa. We have built several automatic voice relay systems for a disadvantaged Deaf community in Cape Town. This paper describes how we augmented a general-purpose communication system for voice relay. Kiara is a fully open source Instant Messaging, voice and video over Internet Protocol communication system based on the Session Initiation Protocol. We integrated automatic speech recognition and text-to-speech technologies into Kiara to provide real-time automatic voice relay for relayed communication. As it stands, Kiara can also be used for standard voice and video relay with a human operator.Telkom, Cisco, THRIP, SANPADDepartment of HE and Training approved lis

Using SIP as P2P Technology

Nowadays peer-to-peer (p2p) technologies are widely adopted and used for building even more sophisticated services: from ubiquitous file-sharing systems to the even more popular Internet telephony. In addition, the Session Initiation Protocol (SIP) has been used for different purposes. Due to its intrinsic generality and flexibility, it could be adopted to build and manage also p2p applications. Moreover, the p2p philosophy could be applied to the existing SIP architecture, to cope with issues such as Denial of Service (DoS). In this paper, we survey the state of the art of the joint use of p2p and SIP. Some hints and examples in using SIP as a core technological component of the p2p world are also presented

Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way

Mobility Management in beyond 3G-Environments

Beyond 3G-environments are typically defined as environments that integrate different wireless and fixed access network technologies. In this paper, we address IP based Mobility Management (MM) in beyond 3G-environments with a focus on wireless access networks, motivated by the current trend of WiFi, GPRS, and UMTS networks. The GPRS and UMTS networks provide countrywide network access, while the WiFi networks provide network access in local areas such as city centres and airports. As a result, mobile end-users can be always on-line and connected to their preferred network(s), these network preferences are typically stored in a user profile. For example, an end-user who wishes to be connected with highest bandwidth could be connected to a WiFi network when available and fall back to GPRS when moving outside the hotspot area.\ud In this paper, we consider a combination of MM for legacy services (like web browsing, telnet, etc.) using Mobile IP and multimedia services using SIP. We assume that the end-user makes use of multi-interface terminals with the capability of selecting one or more types of access networks\ud based on preferences. For multimedia sessions, like VoIP or streaming video, we distinguish between changes in network access when the end-user is in a session or not in a session. If the end-user is not in a session, he or she needs to be able to start new sessions and receive invitations for new sessions. If the end-user is in a session, the session needs to be handed over to the new access network as seamless as possible from the perspective of the end-user. We propose an integrated but flexible solution to these problems that facilitates MM with a customizable transparency to applications and end-users