We apply some of the existing web server redundancy techniques for high service availability and scalability to the relatively new IP telephony context. The paper compares various failover and load sharing methods for registration and call routing servers based on the Session Initiation Protocol (SIP). In particular, we consider the SIP server failover techniques based on the clients, DNS (Domain Name Service), database replication and IP address takeover, and the load sharing techniques using DNS, SIP identifiers, network address translators and servers with same IP addresses. Additionally, we present an overview of the failover mechanism we implemented in our test-bed using our SIP proxy and registration server and the open source MySQL database.

interactive voice response applications that are accessible from telephones as well as IP phones. We also describe how we have used sipvxml in our multi-party multimedia conferencing server. We propose other applications and extensions that can benefit from this technology in our IP telephony test bed. (keywords: Internet telephony; interactive voice response; SIP; VoiceXML; sipvxml) I.

We describe the architecture and implementation of our comprehensive multi-platform collaboration framework known as Columbia InterNet Extensible Multimedia Architecture (CINEMA). It provides a distributed architecture for collaboration using synchronous communications like multimedia conferencing, instant messaging, shared web-browsing, and asynchronous communications like discussion forums, shared files, voice and video mails. It allows seamless integration with various communication means like telephones, IP phones, web and electronic mail. In addition, it provides value-added services such as call handling based on location information and presence status. The paper discusses the media services needed for collaborative environment, the components provided by CINEMA and the interaction among those components.

The public switched telephone network (PSTN) provides ubiquitous availability and very high scalability of more than a million busy hour call attempts per switch. If large carriers are to adopt Internet telephony, then Internet telephony servers should offer at least similar quantifi-able guarantees for scalability and reliability using metrics such as call setup latency, server call handling capacity, busy hour call arrivals, mean-time between failures and mean-time to recover. This thesis presents a reliable, scalable and interoperable Internet telephony architecture for user registration, call routing, conferencing and unified messaging using commodity hardware. The results extend beyond Internet telephony to encompass multimedia communication in general. The architecture presented in this thesis deals with two aspects: at least PSTN-grade re-liability and scalability of the Internet telephony servers, and interoperable Internet telephony services such as conferencing and voice mail using existing protocols. We describe the archi-tecture and implementation of our Session Initiation Protocol (SIP)-based enterprise Internet telephony architecture known as Columbia InterNet Extensible Multimedia Architecture (CIN-EMA). It consists of a SIP registration and proxy server, a multi-party conferencing server, a gateway for interworking SIP with ITU’s H.323, an interactive voice response system and a

We apply some of the existing web server redundancy techniques for high service availability and scalability to the relatively new IP telephony context. The paper compares various failover and load sharing methods for registration and call routing servers based on the Session Initiation Protocol (SIP). In particular, we consider SIP server failover techniques based on the clients, DNS (Domain Name Service), database replication and IP address takeover, and load sharing techniques using DNS, SIP identifiers, network address translators and servers with same IP addresses. We describe our two-stage reliable and scalable SIP server architecture in which the first stage proxies the request to one of the second stage server group based on the destination user identifier. We quantitatively evaluate the performance improvement of the load sharing architecture using our SIP server. We quantitatively compare the effect of SIP server architecture such as event-based and thread pool. Additionally, we present an overview of the failover mechanism we implemented in our test-bed using the open source MySQL database.

Abstract—Placing voice traffic on the data network exposes it to the same attacks that plague the existing Internet infrastructure. Traditional perimeter security solutions cannot cope with the complexity of VoIP protocols at carrier-class performance. We have implemented a large-scale, SIP-aware application layer firewall capable of filtering VoIP traffic at carrier-class rates. We have also built an integrated testing and analysis tool and a testbed that validates its functionality and performance. Our testing tool is unique in its use of finer granularity measurements of pinhole opening and closing delays of the system under test (SUT) than previously available. The tool is further enhanced with an array of ten standard Pentium 3GHz processors to distribute VoIP traffic load generation, reaching the desired target performance rates. This paper describes our novel and original application layer firewall solution, the testing methodology and the testbed used to benchmark our solution. P I.