In this thesis, we proposed a simple audio conferencing system including a new mechanism integrated into Peer-to-Peer network for setting up an audio conference and an improved conferencing media streaming control protocol for a pure Peer-to-Peer like media streaming control protocol-Mutualcast.
Setting up an audio conference is a pre-process in our proposed scheme even for all audio conferences, so establishing an audio conference will always take a while and participants are unable to do anything during this period of time. Eliminating or reducing the time wasted on setting up or speeding up the gap between users joining and starting to communicate with other participants is worthy to study, but setting up is ignored by most studies.
Our proposed scheme integrate three major conference modes, dial-in, dial-out (central server mixing) and P2P (end-user mixing), into a new mode with Peer-to-Peer network which is not utilized by previous three modes. In this new mode of setting up an audio conference three previous modes will interact with each other as well as with Peer-to-Peer network, they are not individually operating. It would reduce partial wasted time in certain scenarios.
An improved audio conferencing media streaming control protocol originated from the Mutualcast [15] is proposed. Our proposed protocol reduces the amount of packets a participant has to upload by deploying relay nodes which switch the delivery path of the mixed packets. Our scheme is evaluated by simulation in the aspects of the packet dropped and throughput.
Comparisons of the original Mutualcast and our proposed conferencing control protocol were presented in Chapter 4 including packet drop, throughput and packet delay. Although the method of original Mutualcast is not open to public and no module in NS-2 is available to simulate the protocol, we modified the necessary functions to accommodate our task. As a result, the simulation uses the NS-2.31 and the ns2VoIP module developed by the Computer Networking Group of Pisa University in Italy show that our proposed conferencing control protocol has a superior improvement comparing with the original Mutualcast, but our proposed scheme pays the cost of longer packets delay for using Relay nodes.
In the future, our proposed audio conferencing system faces many issues that should be solved or improved including NAT [25], firewall traversal and security. It is difficult to position the users’ location when a user connects to the network through NAT technology (one IP address allocated to many users). To solve NAT and firewall issue we suggest using the ICE (Interactive Connectivity Establishment) [26] when users connect to a super-node.
On the other hand, a distributed P2P architecture has an open security issues for a long time such as trust (privacy and confidentiality), malicious node behavior (call dropping) and DoS attacks. Hop-by-hop routing of request and responses at which each node changes the source identifier can be used to provide some confidentiality.
In our simulation scenario, it is just a small scope comparing to the original Mutualcast and our proposed conferencing control protocol. We will try to construct an integrated network environment and perform simulation for more accurate packet loss, packets delay and throughput. It will be tested to deliver the real voice on the simulation network and analyze the voice quality of service that end-users may experience in advance.
implement our proposed system in practice. It also should be added more simulation scenarios to enhance the authenticity of the simulation results of our proposed system.
References
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