SPDY is a protocol designed for low-latency transport of content over the World Wide Web. SPDY introduces two layers of protocol. The lower layer is a general purpose framing layer which can be used atop a reliable transport (likely TCP) for multiplexed, prioritized, and compressed data communication of many concurrent streams. The upper layer of the protocol provides HTTP-like RFC2616 semantics for compatibility with existing HTTP application servers.
- Making the web speedier and safer with SPDY
- SPDY performance on mobile networks
- Google Group: spdy-dev
TCP flows start with an initial congestion window of at most four segments or approximately 4KB of data. Because most Web transactions are short-lived, the initial congestion window is a critical TCP parameter in determining how quickly flows can finish. While the global network access speeds increased dramatically on average in the past decade, the standard value of TCP’s initial congestion window has remained unchanged. In this paper, we propose to increase TCP’s initial congestion window to at least ten segments (about 15KB). Through large-scale Internet experiments, we quantify the latency benefits and costs of using a larger window, as functions of network bandwidth, round-trip time (RTT), bandwidth delay product (BDP), and nature of applications. We show that the average latency of HTTP responses improved by approximately 10% with the largest benefits being demonstrated in high RTT and BDP networks. The latency of low bandwidth networks also improved by a significant amount in our experiments. The average retransmission rate increased by a modest 0.5%, with most of the increase coming from applications that effectively circumvent TCP’s slow start algorithm by using multiple concurrent connections. Based on the results from our experiments, we believe the initial congestion window should be at least ten segments and the same be investigated for standardization by the IETF.
Today’s web services are dominated by TCP flows so short that they terminate a few round trips after handshaking; this handshake is a significant source of latency for such flows. In this paper we describe the design, implementation, and deployment of the TCP Fast Open protocol, a new mechanism that enables data exchange during TCP’s initial handshake. In doing so, TCP Fast Open decreases application network latency by one full round-trip time, decreasing the delay experienced by such short TCP transfers. We address the security issues inherent in allowing data exchange during the three-way handshake, which we mitigate using a security token that verifies IP address ownership. We detail other fall-back defense mechanisms and address issues we faced with middleboxes, backwards compatibility for existing network stacks, and incremental deployment. Based on traffic analysis and network emulation, we show that TCP Fast Open would decrease HTTP transaction network latency by 15%and whole-page load time over 10% on average, and in some cases up to 40%
Packet losses increase latency for Web users. Fast recovery is a key mechanism for TCP to recover from packet losses. In this paper, we explore some of the weaknesses of the standard algorithm described in RFC 3517 and the non-standard algorithms implemented in Linux. We ﬁnd that these algorithms deviate from their intended behavior in the real world due to the combined eﬀect of short ﬂows, application stalls, burst losses, acknowledgment (ACK) loss and reordering, and stretch ACKs. Linux suﬀers from excessive congestion window reductions while RFC 3517 transmits large bursts under high losses, both of which harm the rest of the ﬂow and increase Web latency. Our primary contribution is a new design to control transmission in fast recovery called proportional rate reduction (PRR). PRR recovers from losses quickly, smoothly and accurately by pacing out retransmissions across received ACKs. In addition to PRR, we evaluate the TCP early retransmit (ER) algorithm which lowers the duplicate acknowledgment threshold for short transfers, and show that delaying early retransmissions for a short interval is eﬀective in avoiding spurious retransmissions in the presence of a small degree of reordering. PRR and ER reduce the TCP latency of connections experiencing losses by 3-10% depending on the response size. Based on our instrumentation on Google Web and YouTube servers in U.S. and India, we also present key statistics on the nature of TCP retransmissions.
Laminar is a new framework for TCP congestion control that separates transmission scheduling, which determines precisely when data is sent, from pure congestion control, which determines the total amount of data sent during each RTT. Laminar is expected to enable new advanced algorithms to more precisely regulate TCP traffic.
SSL & TLS
False Start is an optional behavior of TLS implementations. It affects only protocol timing, not on-the-wireprotocol data, and can be implemented unilaterally. The TLS False Start feature leads to a latency reduction of one round trip for certain handshakes.
Transport Layer Security (TLS) extension for application layer protocol negotiation. This allows the application layer to negotiate which protocol should be performed over the secure connection in a manner which avoids additional round trips and which is independent of the application layer protocols.
This draft defines an EDNS0 extension to carry information about the network that originated a DNS query, and the network for which a reply can be cached.