The Session Initiation Protocol (SIP) is a standard for managing IP multimedia sessions in the Internet. Identity management in SIP is a crucial security field that deals with identifying users in SIP networks and controlling their access to the corresponding resources. RFC 4474 describes a mechanism, based on certificates, for dealing with the SIP users identities. This RFC recommends the use of the RSA algorithm as it is currently the most popular public key cryptography system. The proliferation of small and simple devices as well as the need to increase the capacity of the SIP servers to handle the increasing VoIP traffic will make continued reliance on RSA more challenging over time. In this paper, we describe our implementation of the current RFC 4474, our integration of elliptic curves cryptography into this RFC and show that the corresponding performance is much more significant than the one where RSA is used. This paper can be considered as a first step in standardizing the use of elliptic curves in the identity management for SIP.

The session initiation protocol (SIP) is widely used for VoIP communication. Losses caused by network or server overload would cause retransmissions and delays in the session establishment and would hence reduce the perceived service quality of the users. In order to be able to take counter measures network and service planers require detailed models that would allow them to predict such effects in advance. This paper presents a theoretical model of SIP that can be used for determining various parameters such as the delay and the number of messages required for establishing a SIP session when taking losses and delays into account. The model is restricted to the case when SIP is transported over UDP. The theoretical results are then verified using measurements.

The main characteristics of ad hoc networks are the lack of predefined infrastructure and the dynamic topology. These characteristics present some new security vulnerabilities for this emerging networking paradigm. Usually, security in ad hoc networks is handled through authentication and encryption. This can be considered as a first line of defense, however, this remain inefficient against some other kind of attacks such as malicious packet dropping. The purpose of this work is to provide a mechanism for detecting malicious incorrect packet forwarding attacks. To this end, a trust model extending routing protocols and based on the reputation concept is developed. Our model provides two main functionalities: monitoring the behavior of the neighboring nodes in the network and computing their reputations based on the information provided by the monitoring. This paper also discusses how the reputation information is gathered, stored and exchanged between the nodes, and computed according to the different scenarios. Our mechanism is also validated with some simulation work showing its feasibility, performance and benefits.

This paper evaluates a self-organizing routing protocol for ad hoc network, called the neuron routing algorithm (NEURAL). NEURAL has been designed taking into account the learning and self-organizing abilities of the brain. More precisely, it was inspired by the synapses process between neurons, when a signal is propagated. Basically, the most significant characteristic of NEURAL is the uniform distribution of the information around the node's location based on the current changes in its neighborhood. Using a 2-hop acknowledgment mechanism, local information is monitored in order to be used for route selection method, classification procedures and learning algorithms.

We present a novel approach for realizing adaptive QoS control in the Internet called the adaptive load service (ALS). ALS was designed in a similar fashion to the available bit rate service (ABR) proposed for ATM. That is, the senders transmit control messages indicating their requirements, intermediate routers adjust the indicated values in accordance with their available resources and the receivers send the contents of the control messages back to the senders. The transmission behavior of the senders is then adjusted in accordance with the indicated values in the returned control messages. Unlike ABR, ALS is IP-based and was designed to scale for large multicast groups and accommodate the needs of heterogeneous receivers through the integration of sender-based and receiver-based adaptation mechanisms in addition to relying on the network feedback for adjusting the transmission behavior. Performance tests of ALS under different simulation topologies show its efficiency in terms of bandwidth utilization and loss reduction. The tests also show ALS to be fair towards competing TCP connections. Additionally, even though ALS uses a simple algorithm for determining the fair bandwidth share a connection should be using, ALS manages to fairly distribute available resources in accordance with the max-min fairness criterion.

For multimedia senders to benefit the most of dynamically changing availability of network resources they need to adjust their transmission behavior in accordance with the network congestion state. In this paper we investigate various mechanisms for realizing an additive increase and multiplicative decrease adaptation scheme based on the real-time transport protocol (RTP). Through analysis and simulations we identify optimal approaches for increasing and decreasing the transmission rate. Here, we will mainly concentrate on the aspects of TCP-friendliness, dynamical setting of the adaptation parameters, fair bandwidth distribution and stability in addition to the obligatory requirements of high bandwidth utilization and low losses in the network. Note that the main challenge here is in achieving these goals using the rather infrequent RTP feedback messages.

We present a scalable scheme called the direct adjustment algorithm (DAA) for adapting the transmission rate of multimedia applications to the congestion level of the network. The DAA is based on the end-to-end real time transport protocol (RTP). Results achieved through simulations and measurements suggest the efficiency of the scheme in utilizing the network resources and decreasing the loss rates. However, with no support from the network, adaptive applications tend to be unfair towards long distance connections and lead to the starvation of competing TCP connections.

Currently, a variety of the MBONE video tools provide video conferencing capabilities on different platforms and with a variety of compression algorithms. However most of these tools lack the ability to interact with other media agents that might be used during a conferencing session. Such interaction is required, for example, for achieving lip synchronisation between audio and video streams or for quality of service control. In this paper we present a new video tool, NEVIT. This tool provides the basic capabilities needed for video conferencing services such as video capturing, compression and decompression engines and multicasting and ATM network interfaces. To ease the interaction with other media agents, NEVIT incorporates a message handling facility to interact over a local conference bus with other media agents, a floor controller of the conference controller. Currently, we are working on adding lip synchronisation and quality-of-service control using this conference bus.

In previous work, we proposed to enhance TCP's congestion control mechanisms using binary congestion notification (BCN). With this combined scheme called BCN-TCP congestion is indicated both by packet losses and by having switches inform sources about their congestion state by setting a congestion bit. Unlike other studies, our approach maintains the basic elements of TCP's congestion control mechanisms and models the response of TCP to the congestion bit to that of the ATM available bit rate service (ABR). We compared the performance of BCN-TCP to the versions currently implemented in hosts (Tahoe and Reno TCP). However BCN-TCP only offered significant performance improvements if part of the path contained ATM ABR services. In this paper we present further results on the performance of BCN-TCP when used in heterogeneous environments, i.e., Internet paths where only some end systems support BCN-TCP. We also consider the fairness between long and short-haul connections and discover that BCN-TCP treats long-haul connections worse than Tahoe and Reno TCP. We investigate the cause of this unfairness and suggest possible remedies.