Hechmi Khlifi

IMS Application Servers: Roles, Requirements, and Implementation Technologies

The IP multimedia subsystem (IMS) defines a generic architecture to support communication services over a Session Initiation Protocol (SIP) infrastructure. In the IMS architecture, application servers host and execute the IMS service logic. These servers can be SIP application servers, open services architecture (OSA) application servers, or a customized applications for mobile networks using enhanced logic (Camel) service environment. Some technologies used in telephony and voice-over-IP (VoIP) application servers are also applicable to IMS application servers, but such servers have some unique requirements that could limit the extent to which these technologies can meet them.

A framework to use SIP in ad-hoc networks

A rapid increase in wireless communication services has occurred during the last years. Infrastructure-less ad-hoc networks emerge as an interesting option to provide pervasive wireless services, although several challenges must be resolved first. Our paper focuses on issues related to signalling and presence management in mobile wireless ad-hoc networks. We present a framework to use SIP in this kind of networks. The objective of this framework is to allow the users of ad-hoc networks to communicate with each other and exchange instant messages. This framework permits the discovery of participants, the establishment, the publishing and the ending of two-party or multiparty sessions by providing SIP (session initiation protocol) extensions. To cope with the bandwidth limitation problem, it is proposed to unify the network layer routing protocols and the application layer SIP REGISTER methods. Scalability of the framework is improved by introducing a clustering approach that group nodes together into clusters.

IMS for Enterprises

We discuss how the IMS architecture is applicable to the enterprise environment, the relevance it has for enterprise users, different options for integrating private and public IMS infrastructures, and the impact of those options on enterprises and IMS service providers. Similarly, we review IMS architecture, discuss the requirements of enterprise communications systems, and present four possible architectures for the use of IMS by enterprises. We also discuss the possibility of using the service oriented architecture (SOA) to integrate IMS-based enterprise communications systems within enterprise information technology (IT) infrastructure. We have built a softswitch prototype based on the IMS architecture. We describe the implementation and the lessons we learned from this experience.

Low-complexity offline and online clock skew estimation and removal

Packet delay traces are important sources of measurements for analyzing end-to-end performance of computer networks. Due to the lack of tight synchronization between the clocks of end systems, these measurements can be quite inaccurate. Therefore, detection, estimation and removal of clock skew from delay traces is a critical operation to obtain precise measurements of network latencies. In this paper, we propose two new techniques to detect, estimate and remove the clock skew in delay traces. The first technique, named average technique, derives the clock skew estimate by calculating the average of the difference between consecutive packets delay. The second technique, named direct skew removal technique, proceeds by iteratively evaluating a set of possible skew values until the best value is reached.Compared with existing techniques such as linear programming and Convex_Hull, the average technique reduces the complexity of the skew estimation operation. The direct skew removal is more accurate and allows us to obtain delay after skew removal with the same precision as the original traces. Applied to traces that contain clock resets, the direct skew removal also reduces the time complexity of the operation.Clock skew is also present in online delay measurements that are used by real-time endpoints such as audio and video terminals, to determine the buffering delay of received packets and to synchronize streams from different sources. The problem is more obvious when communication sessions last for a long time. Contrary to the offline skew removal, little work has been done on the online skew removal problem. In this paper, we propose two simple algorithms to remove the clock skew from online delay measurements. The first algorithm, named sliding window algorithm, tracks the skew by continually evaluating the variation of the minimum measured delay. The second algorithm, named the combined algorithm, is a mixed approach of the sliding window and the Convex_Hull algorithm proposed elsewhere.

Design and performance of a stand-alone media server

We describe the design, implementation and performance of a stand-alone media server. This server is intended to be used in voice over IP systems for audio interaction with users, NAT traversal, call monitoring and media recording. Its advantage is that it is application independent, which means that it can be used by any service and for any signaling protocol. In this paper we describe the design of the server, of its control protocol and of its functionalities such as file playing, DTMF detection, RTP proxying and audio recording. For file playing, we propose an approach to schedule RTP packet transmission that overcomes the irregularity of event scheduling of the operating system and for conversation recording, we propose an original approach to synchronize media streams that allows to avoid voice overlap while recording a conversation. We have conducted many experiments on the media server. We evaluated the accuracy of the file playing timing under different load conditions, the effect of the RTP proxying on the media quality and the subjective quality of recorded conversations. The results of these experiments show the great impact of the load of the media server on the performance of its functionalities, the improvement that the packet scheduling algorithm brings to the precision of inter-departure timing, and the good subjective quality of the recorded conversation using our synchronizing approach.

A Peer to Peer Architecture for Enabling a Universal Presence Service

Presence service is used to publish and discover end-users’ context information. It is the basis of a wide range of end-user services. It is deployed using a wide range of technologies, which hampers interoperability. This paper proposes a peer to peer infrastructure for enabling a universal presence service, a service that will enable end-users with multiple presence accounts in domains with different technologies to publish and discover end-user context information through a single interface. Requirements are derived, an architecture is proposed and implementation aspects are discussed. The architecture is an overlay and the related procedures and protocols are introduced. New software architecture is proposed for the implementation and the related tool kits are identified.

ARTP: A buffer-aware rate control protocol for media streaming

We present adaptive real-time transport protocol (ARTP), a media streaming transport protocol that implements a congestion control mechanism. With this mechanism, the sender adapts its sending rate to network conditions and to the buffering capacity of the receiver. This adaptiveness takes into account the real-time constraints of media streaming. It aims at ensuring media playback continuity, and at achieving a low packet loss rate during media streaming sessions. ARTP ensures the continuity of media playback by buffering media packets during congestion-free periods and reduces the loss rate by reducing the transmission rate during congestion periods. This protocol considers the size of the buffer that the receiver dedicates to rate control in order to avoid overflow or underflow of the buffer. This approach allows limited memory devices such as cellular phones and PDAs to take advantage of rate control. ARTP is based on the feedback that the real-time control protocol (RTCP) reports give with the addition of two new parameters that we define in this paper: the steady state loss event rate and the duration to the next feedback report. It also requires that the real-time streaming protocol (RTSP) provide the server with the size of the buffer that the client dedicates to rate control. Our NS-2 simulations show that, besides buffer protection, ARTP significantly reduces the loss rate. Compared to Additive Increase Multiplicative Decrease (AIMD) rate control techniques, ARTP provides a better media quality by ensuring the continuity of media playback and, compared to equation-based rate control techniques, it achieves a better loss rate and reduces the bandwidth used for feedback.

A Modular Architecture for Providing Carrier-Grade SIP Telephony Services

We discuss the development of carrier-grade telephony services in SIP environments. We review available approaches to build SIP applications, introduce a new architecture to provide SIP telephony services and discuss available technologies to implement this architecture. The architecture addresses the modularity, flexibility and scalability requirements for mass-market deployment of telephony services. It is based on the concept of separation between application logic and network functionalities and, at the network level, between signaling and media processing. Parlay, a set of standard object-oriented and signaling protocol-neutral APIs, is used to allow applications to access network functionalities and SIP is used to invoke media capabilities. We have built a soft-switch prototype based on the architecture. We describe the implementation and the lessons we have learned from this experience.

Building Groupwares over Duplicated Object Systems

Groupware toolkits let developers build applications for synchronous and distributed computer-based conferencing. Duplicated object systems (or DoS), on the other hand, manage distributed objects over the Internet and, since they include high-level features such as communication facilities, load balancing, fault tolerance, and hierarchical messaging, may be leveraged as building blocks for rapidly developing groupware toolkits. This paper describes how we built such a groupware starting from a particular DoS. The system contains a run-time architecture that automatically manages the creation, interconnection, and communications of the distributed processes involved in the working sessions, a set of groupware facilities allowing users to collaborate, to take action on state changes, and to share relevant data, and a session management and user control mechanisms accommodating the groups working style.