Randa El-Marakby

A Critique of Modern Internet Protocols: The Issue of Support for Multimedia

The provision and support of new distributed multimedia services are of prime concern for telecommunications operators and suppliers. Clearly, the potential of the latest Internet protocols to contribute communications components is of considerable interest to them. In this paper we present three of the major new protocols introduced into the Internet environment to support real-time and multimedia communications. These are respectively IPv6 (the new version of the Internet protocol), RSVP (Resource Reservation Protocol) and RTP (Real-time Transport Protocol) with its control protocol RTCP. We then attempt to review these protocols critically in order to assess their suitability for purpose.

A Scalability Scheme for the Real-time Control Protocol

Recently, some problems related to using the Real-time Control Protocol (RTCP) in very large dynamic groups have arisen. Some of these problems are: feedback delay, increasing storage state at every member, and ineffective RTCP bandwidth usage, especially for receivers that obtain incoming RTCP reports through low bandwidth links. In addition, the functionality of some fields (e.g. packet loss fraction) in the Receiver Reports (RRs) becomes questionable as, currently, an increasing number of real-time adaptive applications are using receiver-based rate adaptive schemes instead of rate adaptation schemes based on the sender.

Scalability improvement of the real-time control protocol (RTCP) leading to management facilities in the Internet

We address some of the problems concerning RTCP scalabilty in large multicast groups. The problems are summarised as the increased storage state maintained by every member about every other member in the group, the feedback delay, and the bandwidth usage problem especially for receivers that obtain incoming RTCP reports through low bandwidth links. We designed a scheme to tackle these problems and to improve the RTCP scalability. Our scheme is based on a hierarchical structure in which members are grouped dynamically, into local regions and receiver reports (RRs) are sent with limited scope. As a result, the load is distributed among members in the group. Each region has an aggregator (AG) that receives the RRs feedback, performs some aggregation statistics on this control data, then sends these statistics to a manager. The manager performs additional analysis to evaluate the network performance and to estimate regions which are suffering from significant congestion. Finally, we present the benefits of using our scheme.

Enhanced QoS for Real-time Multimedia Delivery over the Wireless Link using RFID Technology

Currently, there is an increasing demand for real-time multimedia applications running over the Internet. In a wired network, if congestion occurs, the quality of real-time multimedia transmission degrades severely. Rate-based adaptation schemes are being used to alleviate congestion. Packet loss incurred is used as the main indicator of network congestion. However, when running multimedia applications over wireless/mobile networks, packet loss can be attributed to different causes other than congestion. Packet loss can be due to the low bandwidth of the wireless link or the frequent interruptions in service due to mobility, handoff, or signal propagation effects, e.g. obstruction, attenuation. In this paper, we present the design of our sensor guided wireless adaptation scheme (SGWAS) that works in a micromobility domain and that infers the main reason of packet loss incurred by the mobile node in the cell. Consequently, it takes the appropriate action to improve the QoS of the transmission. Determining the reason of packet loss relies on information obtained from wireless sensors, specifically RFlD devices, to detect the location of the mobile node within the cell. On one hand, if packet loss is due to the low bandwidth of the wireless link, which can cause local wireless link congestion, then local transmission rate adaptation is applied in the cell. On the other hand, if packet loss is due to mobility or signal propagation effects, then other appropriate actions are taken. We conducted some simulation experiments to verify that we can determine the location of the mobile node when it is in the handoff region. The results demonstrate that SGWAS identifies the reason of packet loss when the mobile node is in the handoff region. Rate adaptation should not be performed in this case because packet loss is not due to congestion

Scalability improvement of the real time control protocol

Scalability problems arise when the Real Time Control Protocol (RTCP), which is the control protocol of the Real-time Transport Protocol (RTP), is used in large multicast groups. The problems include: increased feedback delay, increased storage state at every member, and ineffective RTCP bandwidth usage. We have designed a Scalable RTCP (S-RTCP) which is based on a hierarchical structure in which members are grouped into local regions. For every region, there is an Aggregator (AG) which receives the feedback Receiver Reports (RRs) sent by its local members. The AG summarizes important information in the RRs, derives some statistics, and sends them to a Manager. The Manager performs additional statistical analysis to monitor the transmission quality and to identify regions suffering massively from congestion. A simulation of S-RTCP using the Network Simulator (NS) showed that S-RTCP alleviates some of the RTCP scalability problems and makes effective use of RRs. Consequently, the feedback provides timely and useful QoS information required for network monitors and for adaptive applications. Details of the simulations and performance analysis are presented and described which show the advantages of using S-RTCP over the original RTCP.

AdamRTP: Adaptive Multi-flows Real-time Multimedia delivery over WSNs

Real-time multimedia applications are time sensitive and require extra resources from the network, e.g. large bandwidth and big memory. However, Wireless Sensor Networks (WSNs) suffer from limited resources such as computational, storage, and bandwidth capabilities. Therefore, sending real-time multimedia applications over WSNs can be very challenging. For this reason, we propose an Adaptive Multi-flows Real-time Multimedia Transport Protocol (AdamRTP) that has the ability to ease the process of transmitting real-time multimedia over WSNs by splitting the multimedia source stream into smaller independent flows using an MDC-aware encoder, then sending each flow to the destination using joint/disjoint path(s). AdamRTP uses also dynamic adaptation techniques, e.g. changing number of flows and rate adaptation. Simulations experiments demonstrate that AdamRTP enhances the Quality of Service (QoS) of the data transmission. Also, we show that in an ideal WSN, using multi-flows consumes less power per node than using a single flow and extends the life-time of the network.

An adaptive algorithm for Internet multimedia delivery in a differentiated services environment

We present an adaptive class switching algorithm (ACSA) that provides good quality with good price for real-time multimedia applications running in a differentiated services (DiffServ) environment. The algorithm selects the most suitable class by taking both QoS and price into account. Class switching is based on both the QoS feedback received and the highest user utility. The user utility reflects the service quality and price tradeoff. The simulation results demonstrate that ACSA can react fast to the current class state and can control the tradeoff between price and quality.

Spatial Clustering of Wireless Sensor Network Nodes Based on Packet Loss

This paper examines the problem of grouping wireless sensor nodes forming a network into a set of spatial clusters. The technique for grouping the nodes proposed here, exploits the spatial degradation in wireless signals to group the nodes into regions based on the packet loss rate. The technique is validated on datasets obtained from actual wireless sensor networks. The obtained results support the validity of the proposed approach.

Effect of RTCP Bandwidth on Feedback Delay

With the evolution of high-speed networks, many problems related to insufficient and width are being solved. This paper shows by using some additional bandwidth as provided in high- speed networks, it is possible to increase feedback frequency in a session. The real-time transport protocol (RTP) provides end-to-end transport functions suitable for real-time multimedia applications. RTP’s real time control protocol (RTCP) is based on periodic multicast transmission of control packets to all session members. It provides traffic monitoring by gathering certain control statistics and sending them as feedback on the quality of data distribution. It is also used by adaptive applications to reduce network congestion. The feedback should be sent periodically within acceptable time intervals. In a large session, this is not the case as feedback is sent rarely. This happens because, according to the current RTCP specifications, all RTCP reports multicast to all members must not consume more than a small fraction (nominally 5%) of the whole bandwidth assigned for the session. Consequently, RTCP reporting interval grows with session size resulting in infrequent feedback in large sessions. The experience of users during short intervals is not utilised and hence the feedback reports do not provide timely and useful information. We conducted two sets of simulations. The first set shows the increase of feedback delay in a large session due to limiting RTCP band-width. The second set shows that by allowing RTCP more bandwidth, the average number of feedback reports is increased in all sessions providing better and timely feedback.