The SPEC Cloud (TM) IaaS2016 benchmark is the Standard Performance Evaluation Corporation's (SPEC) first benchmark suite to measure cloud performance. The benchmark suite's use is targeted at cloud providers, cloud consumers, hardware vendors, virtualization software vendors, application software vendors, and academic researchers. The benchmark addresses the performance of infrastructure-as-a-service (IaaS) cloud platforms. IaaS cloud platforms can either be public or private. The key metrics of the benchmark are scalability, elasticity, and mean instance provisioning time.

This paper introduces a novel approach to automating failure diagnostics in distributed systems by combining fault injection and data analytics. We use fault injection to populate the database of failures for a target distributed system. When a failure is reported from production environment, the database is queried to find “matched” failures generated by fault injections. Relying on the assumption that similar faults generate similar failures, we use information from the matched failures as hints to locate the actual root cause of the reported failures. In order to implement this approach, we introduce techniques for (i) reconstructing end-to-end execution flows of distributed software components, (ii) computing the similarity of the reconstructed flows, and (iii) performing precise fault injection at pre-specified executing points in distributed systems. We have evaluated our approach using an OpenStack cloud platform, a popular cloud infrastructure management system. Our experimental results showed that this approach is effective in determining the root causes, e.g., fault types and affected components, for 71-100 percent of tested failures. Furthermore, it can provide fault locations close to actual ones and can easily be used to find and fix actual root causes. We have also validated this technique by localizing real bugs that occurred in OpenStack.

Cloud providers sell identically configured virtual machines (VMs) for the same price. Customers purchasing these VMs expect that they perform similarly and are allocated the same amount of virtual resources. In practice, however, the real performance of identically provisioned VMs depends on the underlying hardware, i.e., how the hardware is configured, and how much shared resources are consumed by co-located VMs. As workloads often have different resource requirements (e.g., CPU or disk I/O bound), a physical machine can be a better host to one VM than another, and swapping the locations of the two VMs can improve the performance (or any other metrics) of both VMs. However, cloud providers are unlikely to offer a VM swapping service since it is a tacit admission of providing different quality of services while charging the same rate. We propose, VBoom, a cloud provider-agnostic system in which VMs can dynamically relocate themselves (via location swapping) in a cloud environment if the new location better serves the needs of the VM. We discuss technical and business benefits and challenges in building such a system. When location matters, we believe that through VBoom, real estate of virtual machines can be established in a completely market-driven fashion.

Open source cloud technologies such as OpenStack, CloudStack, OpenNebula, Eucalyptus, OpenShift, and Cloud Foundry have gained significant momentum in the last few years. For a researcher and practitioner, they present a unique opportunity to analyze, contribute, and innovate new services using these technologies. The first part of the tutorial will provide an overview of OpenStack and CloudStack, two open source infrastructure as a service (IaaS) cloud platforms. The second part of the tutorial will present a detailed analysis of different OpenStack components, namely, glance (image service), nova (compute service), keystone (security service), quantum (network service), and swift (object storage service). In particular, the tutorial will describe the scheduling and provisioning process in OpenStack, and how different configuration options lead to myriad provisioning performance. Further, the tutorial will describe how OpenStack has evolved over releases. Finally, the tutorial will describe weaknesses of OpenStack and highlight important areas where researchers can contribute.

Applying patches to operating systems, middleware, and applications is considered a major IT pain point due to several reasons. The operating systems and software are of myriad types, there is interdependency among the updates, operating system, and applications, there is lack of standardization among different enterprise customers, and finally testing the applications and operating systems post-update is challenging. As a result, human operator is involved in different stages of the patching process, making it costly and cumbersome. Cloud can help standardize various offerings to customers, and potentially remove human operators. However, it introduces other challenges such as VM time zones and restoring VMs from snapshots which are not present in traditional enterprise environments. We discuss the challenges of achieving patch automation in a Cloud, and then describe our solution.

TCP has traditionally been considered inappropriate for real-time applications. Nonetheless, popular applications such as Skype use TCP since UDP packets cannot pass through restrictive network address translators (NATs) and firewalls. Motivated by this observation, we study the delay performance of TCP for real-time media flows. We develop an analytical performance model for the delay of TCP. We use extensive experiments to validate the model and to evaluate the impact of various TCP mechanisms on its delay performance. Based on our results, we derive the working region for VoIP and live video streaming applications and provide guidelines for delay-friendly TCP settings. Our research indicates that simple application-level schemes, such as packet splitting and parallel connections, can reduce the delay of real-time TCP flows by as much as 30% and 90%, respectively.

We present vDelay, a tool for measuring the capture-to-display latency (CDL) and frame-rate of real-time video applications such as video chat and conferencing. vDelay allows measuring CDL and frame-rate without modifying the source code of these applications. Further, it does not require any specialized hardware. We have used vDelay to measure the CDL and frame-rate of popular video chat applications such as Skype, Windows Live Messenger, and GMail video chat. vDelay can also be used to measure the CDL and frame-rate of these applications in the presence of bandwidth variations.

We present vDelay, a tool for measuring the capture-to-display latency (CDL) and frame-rate of real-time video applications such as video chat and conferencing. Real-time video chat applications have three key software components: a video encoder that compresses the video captured from the camera, a video decoder that decompresses the video received over the network, and a playout buffer that smooths the playout of received video due to network delay variations. These software components impact capture-to-display latency (CDL) and frame-rate of the real-time video played at a receiver application. Capture-to-display latency is the total time to encode and decode a video frame, playout buffer time, and latency of the network path. Along with bit-rate, these two metrics provide quick insights into the performance of a real-time video application.

This paper presents a demonstration proposal about an implementation of P2PSIP in the mobile networks. The implementation demonstrates establishment of SIP multimedia sessions between mobile phones with little or no need for centralized servers of any kind. It allows a mobile phone user to use standard phone UI to make voice calls using resources of P2P overlay. A mobile phone user may freely use standard call manager and contact list in a mobile phone. According to our knowledge, P2P SIP has not yet been demonstrated on a mobile platform by anyone.

TCP has traditionally been considered unfriendly for real-time applications. Nonetheless, popular applications such as Skype use TCP since UDP packets cannot pass through many NATs and firewalls. Motivated by this observation, we study the delay performance of TCP for real-time media flows. We develop an analytical performance model for the delay of TCP. We use extensive experiments to validate the model and to evaluate the impact of various TCP mechanisms on its delay performance. Based on our results, we derive the working region for VoIP and live video streaming applications and provide guidelines for delay-friendly TCP settings. Our research indicates that simple application-level schemes, such as packet splitting and parallel connections, can reduce the delay of real-time TCP flows by as much as 30% and 90%, respectively.

In this paper, we examine the feasibility of sending real-time CBR workloads over TCP. This is motivated by the friendliness of NATs and firewalls towards TCP as opposed to UDP as well as by recent improvements in Internet's bandwidth and loss rates. Traditionally, TCP has been considered undesirable for real-time CBR workloads. We evaluate this assertion by developing a novel analytical tool that yields TCP's sender-to-receiver socket delay distribution for CBR workloads. A key insight gained is that the use of smaller than MSS-sized packets in CBR workloads can exploit the TCP's ACK counting mechanism thereby limiting the delay impact of congestion window variations. We leverage this insight to provide a heuristic and system-level guidelines for reducing TCP transport delays.

Skype is a peer-to-peer VoIP client developed by KaZaa in 2003. Skype claims that it can work almost seamlessly across NATs and firewalls and has better voice quality than the MSN and Yahoo IM applications. It encrypts calls end-to-end, and stores user information in a decentralized fashion. Skype also supports instant messaging and conferencing. This report analyzes key Skype functions such as login, NAT and firewall traversal, call establishment, media transfer, codecs, and conferencing under three different network setups. Analysis is performed by careful study of Skype network traffic.

Despite recent advancements in cellular phone network infrastructure, survivability of the calls is still an open issue. Important business calls and tele-conferences cannot benefit from mobility because landlines are preferred owing to their reliability and tolerance to failures. In this paper, we present a framework to improve call survivability by monitoring the user's location and intimating him about the possibility of loss of connectivity, interference, and room schedules that might disrupt conversation. In addition, if a call drops because of unforeseeable circumstances like battery failure or cell-phone damage then the call is switched to the nearest available device capable of streaming voice