Quincy Wu

Information Hiding in Real-Time VoIP Streams

The real-time speech hiding is to hide the secret speech into a cover speech in real-time communication systems. By hiding one secret speech into the cover speech, we can get a stego speech, which sounds meaningful and indistinguishable from the original cover speech. Therefore, even if the attackers catch the audio packets on Internet, they would not notice that there is another speech hidden inside it. In this paper, we propose a scheme for speech hiding in a real-time communication system such as voice over Internet Protocol (VoIP). We propose a novel design of real-time speech hiding for G.711 codec, which is widely supported by almost every VoIP device. Experimental results show that the processing time for the proposed algorithm takes only 0.257 ms, which is suitable for real-time VoIP applications.

SIP Terminal Mobility for both IPv4 and IPv6

Session Initiation Protocol (SIP) supports application layer mobility during a session. In this paper the architecture design on the protocol stack implementation of SIP terminal mobility is described, and the performance of SIP user agents developed with open-source libraries are measured from empirical experiments. The experiments are performed in both IPv4 and IPv6 environment. In the best case, the delay of SIP mobility only takes 38ms in SIP signaling exchange, for both IPv4 and IPv6. Therefore, SIP mobility is suitable for supporting seamless handover in VoIP communications.

Enhancing Teredo IPv6 tunneling to traverse the symmetric NAT

By tunneling IPv6 packets over IPv4 UDP, Teredo supports IPv4/IPv6 dual-stack nodes in private IPv4 networks behind network address translation (NAT) to access IPv6 networks. However, the current Teredo protocol does not work with symmetric NAT. This letter proposes SymTeredo, an extension of Teredo with capability to traverse the symmetric NAT. Our extension preserves the Teredo architecture, and offers backward compatibility with the original Teredo protocol.

Relay placement problem in smart grid deployment

In this paper, we give an overview of power grid, smart grid, Advanced Metering Infrastructure (AMI), and the deployment cost analysis step by step. The importance between Relay Placement Problem (RPP) and the deployment cost in an AMI system is highlighted. Additionally, a decision supporting system in a pilot AMI system funded by National Science Council (NSC) is briefly described where RPP is solved by an approximation algorithm.

SIP mobility and IPv4/IPv6 dual-stack supports in 3G IP multimedia subsystem

In the Universal Mobile Telecommunications System (UMTS), session initiation protocol (SIP) and IPv6 are the default protocols for IP multimedia core network subsystem (IMS). However, a user equipment (UE) may not be allowed to roam or hand off from UMTS to a private-IPv4 GPRS network. In this paper, we utilize SIP mobility and an automatic IPv6 tunneling mechanism, called Teredo, to support roaming/handoff of a UE between different networks. We have developed the first non-commercial Linux-based Teredo mechanism, and compared our solution with other Teredo implementations in the public domain. Our study indicates that our solution can reduce the tunneling overhead and transmission delay over two other implementations by 44–74%. Copyright

OpenFlow based hybrid routing in Wireless Sensor Networks

Wireless Sensor Networks (WSNs) have been widely deployed in many fields and commonly adopted in many applications. There are several routing protocols for WSNs, such as Ad hoc On-Demand Distance Vector Routing (AODV) and Sensor Protocols for Information via Negotiation (SPIN). However, one major problem of these routing protocols is that they cannot provide a global view for network topologies, as they run in a distributed way. Moreover, for a distributed routing algorithm, troubleshooting always takes lots of time because the administrators must collect the routing information from individual routing nodes. OpenFlow is an emerging technology which makes network elements (such as routers or switches) programmable via a standardized interface. With OpenFlow, novel routing algorithms can be quickly deployed. In this paper, we proposed a new routing approach that integrates OpenFlow with link-state routing in WSNs. The approach allows administrators to maintain a global view on each routing path, as well as simplifies the procedure of network provisioning and troubleshooting.

Porting VoIP applications to DCCP

In the digital era, the increase of network bandwidth and the ubiquitous wireless access facilitates the creation of more and more innovative network services. Among these services, Voice over Internet Protocol (VoIP) is surely one of the most popular and successful real-time multimedia services on the Internet. For decades, User Datagram Protocol (UDP) has been adopted to transport the voice streams of VoIP applications on the network. In this paper, we presented how the Datagram Congestion Control Protocol (DCCP), recently developed by Internet Engineering Task Force (IETF), can be utilized to transport realtime audio stream through Real-time Transport Protocol (RTP). We proposed the design and architecture of a VoIP application running on DCCP, and take Linphone, an open-source Internet VoIP phone on Linux, as an example to illustrate how to apply DCCP to establish a bidirectional Session Initiation Protocol (SIP) communication.

Development and Deployment of IPv6-Based SIP VoIP Networks

This paper presents an IPv6-based SIP VoIP network deployed in Taiwan. This deployment project is supported by NICI IPv6 R&D Division. The major contributions of this paper are exercising the ENUM deployment on IPv6 SIP network, developing the IPv6 SIP User Agent and the IPv6 SIP Analyzer.

Prepaid mechanism of VoIP and messaging services

Session initiation protocol (SIP) is used for voice over IP (VoIP) signaling and call control. Billing mechanisms for messaging and VoIP are typically deployed for postpaid services. Prepaid mechanisms for SIP-based VoIP calls toward public switched telephone network (PSTN) are seldom discussed. In this paper, we propose a prepaid mechanism that can simultaneously process messaging and VoIP calls. This prepaid mechanism is implemented in a VoIP platform developed under National Science & Technology Program for Telecommunications (NTP). Unlike previous VoIP prepaid approaches which require modifications to end points, our mechanism can be easily integrated to a VoIP platform without modifying existing network components.

A Proof of MITM Vulnerability in Public WLANs Guarded by Captive Portal

In this paper, A lot of public areas provide the WLAN service for nomadic users so that they can finish the tasks even when they are out of office. Therefore, the security of public WLANs is more important than past. Nowadays many public WLANs service providers the Captive Portal to authenticate users. The Captive Portal uses a webpage to request a user to authenticate himself by providing his own username and password. This security mechanism proved to be simple and effective because users cannot access Internet before they get authenticated. However, in this paper, we shall illustrate that for public WLANS which are guarded by Captive Portal, will be vulnerable to man-in-themiddle attacks. Therefore, a hacker can careful send out some spoofing packets and take advantage of the public WLAN to access Internet without being authenticated. We show the vulnerability by both protocol analysis and a real implementation in C programs.