VoIP-based Air Traffic Controller Training

Extending VoIP beyond the Internet telephony, we propose a case study of applying the technology outside of its intended domain, to solve a real-world problem. This work is an attempt to understand an analog hardwired communication system of the U.S. Federal Aviation Administration (FAA), and effectively translate it into a generic, standards-based VoIP system that runs on their existing data network. We develop insights into the air traffic training and weigh on the design choices for building a soft real-time data communication system. We also share our real-world deployment and maintenance experiences, as the FAA Academy has been successfully using this VoIP system in five training rooms since 2006 to train the future air traffic controllers of the U.S. and the world

The Virtual Device: Expanding Wireless Communication Services Through Service Discovery and Session Mobility

We present a location-based, ubiquitous service architecture, based on the Session Initiation Protocol (SIP) and a service discovery protocol that enables users to enhance the multimedia communications services available on their mobile devices by discovering other local devices, and including them in their active sessions, creating a 'virtual device.' We have implemented our concept based on Columbia University's multimedia environment and we show its feasibility by a performance analysis

Using an External DHT as a SIP Location Service

Peer-to-peer Internet telephony using the Session Initiation Protocol (P2P-SIP) can exhibit two different architectures: an existing P2P network can be used as a replacement for lookup and updates, or a P2P algorithm can be implemented using SIP messages. In this paper, we explore the first architecture using the OpenDHT service as an externally managed P2P network. We provide design details such as encryption and signing using pseudo-code and examples to provide P2P-SIP for various deployment components such as P2P client, proxy and adaptor, based on our implementation. The design can be used with other distributed hash tables (DHTs) also

Next Generation Emergency Call System with Enhanced Indoor Positioning

The emergency call systems in the United States and elsewhere are undergoing a transition from the PSTN-based legacy system to a new IP-based system. The new system is referred to as the Next Generation 9-1-1 (NG9-1-1) or NG112 system. We have built a prototype NG9-1-1 system which features media convergence and data integration that are unavailable in the current emergency calling system. The most important piece of information in the NG9-1-1 system is the caller's location. The caller's location is used for routing the call to the appropriate call center. The emergency responders use the caller's location to find the caller. Therefore, it is essential to determine the caller's location as precisely as possible to minimize delays in emergency response. Delays in response may result in loss of lives. When a person makes an emergency call outdoors using a mobile phone, the Global Positioning System (GPS) can provide the caller's location accurately. Indoor positioning, however, presents a challenge. GPS does not generally work indoors because satellite signals do not penetrate most buildings. Moreover, there is an important difference between determining location outdoors and indoors. Unlike outdoors, vertical accuracy is very important in indoor positioning because an error of few meters will send emergency responders to a different floor in a building, which may cause a significant delay in reaching the caller. This thesis presents a way to augment our NG9-1-1 prototype system with a new indoor positioning system. The indoor positioning system focuses on improving the accuracy of vertical location. Our goal is to provide floor-level accuracy with minimum infrastructure support. Our approach is to use a user's smartphone to trace her vertical movement inside buildings. We utilize multiple sensors available in today's smartphones to enhance positioning accuracy. This thesis makes three contributions. First, we present a hybrid architecture for floor localization with emergency calls in mind. The architecture combines beacon-based infrastructure and sensor-based dead reckoning, striking a balance between accurately determining a user's location and minimizing the required infrastructure. Second, we present the elevator module for tracking a user's movement in an elevator. The elevator module addresses three core challenges that make it difficult to accurately derive displacement from acceleration. Third, we present the stairway module which determines the number of floors a user has traveled on foot. Unlike previous systems that track users' foot steps, our stairway module uses a novel landing counting technique. Additionally, this thesis presents our work on designing and implementing an NG9-1-1 prototype system. We first demonstrate how emergency calls from various call origination devices are identified, routed to the proper Public Safety Answering Point (PSAP) based on the caller's location, and terminated by the call taker software at the PSAP. We then show how text communications such as Instant Messaging and Short Message Service can be integrated into the NG9-1-1 architecture. We also present GeoPS-PD, a polygon simplification algorithm designed to improve the performance of location-based routing. GeoPS-PD reduces the size of a polygon, which represents the service boundary of a PSAP in the NG9-1-1 system

On SIP Server Clusters and the Migration to Cloud Computing Platforms

This thesis looks in depth at telephony server clusters, the modern switchboards at the core of a packet-based telephony service. The most widely used de facto standard protocols for telecommunications are the Session Initiation Protocol (SIP) and the Real Time Protocol (RTP). SIP is a signaling protocol used to establish, maintain, and tear down communication channel between two or more parties. RTP is a media delivery protocol that allows packets to carry digitized voice, video, or text. SIP telephony server clusters that provide communications services, such as an emergency calling service, must be scalable and highly available. We evaluate existing commercial and open source telephony server clusters to see how they differ in scalability and high availability. We also investigate how a scalable SIP server cluster can be built on a cloud computing platform. Elasticity of resources is an attractive property for SIP server clusters because it allows the cluster to grow or shrink organically based on traffic load. However, simply deploying existing clusters to cloud computing platforms is not good enough to take full advantage of elasticity. We explore the design and implementation of clusters that scale in real-time. The database tier of our cluster was modified to use a scalable key-value store so that both the SIP proxy tier and the database tier can scale separately. Load monitoring and reactive threshold-based scaling logic is presented and evaluated. Server clusters also need to reduce processing latency. Otherwise, subscribers experience low quality of service such as delayed call establishment, dropped calls, and inadequate media quality. Cloud computing platforms do not guarantee latency on virtual machines due to resource contention on the same physical host. These extra latencies from resource contention are temporary in nature. Therefore, we propose and evaluate a mechanism that temporarily distributes more incoming calls to responsive SIP proxies, based on measurements of the processing delay in proxies. Availability of SIP server clusters is also a challenge on platforms where a node may fail anytime. We investigated how single component failures in a cluster can lead to a complete system outage. We found that for single component failures, simply having redundant components of the same type are enough to mask those failures. However, for client-facing components, smarter clients and DNS resolvers are necessary. Throughout the thesis, a prototype SIP proxy cluster is re-used, with variations in the architecture or configuration, to demonstrate and address issues mentioned above. This allows us to tie all of our approaches for different issues into one coherent system that is dynamically scalable, is responsive despite latency varations of virtual machines, and is tolerant of single component failures in cloud platforms

sipc, a multi-function SIP user agent

Abstract. Integrating multiple functions into one communication user agent can introduce many innovative communication services. For example, with networked appliance control, a user agent can turn off the stereo when receiving an incoming call. With location sensing, a user agent can automatically reject a call if it knows the location preference is ’quiet’. Multi-function interactions enable services that are otherwise impossible. In this paper, we first present the new services introduced by the integration, then introduce our SIP user agent, SIPC, which handles these new services in a programmable way. SIPC integrates multimedia call setup, networked appliance control, presence handling, Internet TV, instant messaging, location sensing, networked resource discovery, third-party call control, real-time multimedia streaming, emergency call handling, and conference floor control into one application. We analyze the relationship among these functions and propose different approaches for function integration. SIPC uses the Session Initiation Protocol (SIP) for multimedia call setup and the Language for End System Services (LESS) for service programming