| Network Working Group | G. Fairhurst, Ed. |
| Internet-Draft | University of Aberdeen |
| Intended status: Informational | B. Trammell, Ed. |
| Expires: December 11, 2015 | M. Kuehlewind, Ed. |
| ETH Zurich | |
| June 09, 2015 |
Services provided by IETF transport protocols and congestion control mechanisms
draft-ietf-taps-transports-05
This document describes services provided by existing IETF protocols and congestion control mechanisms. It is designed to help application and network stack programmers and to inform the work of the IETF TAPS Working Group.
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This Internet-Draft will expire on December 11, 2015.
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Most Internet applications make use of the Transport Services provided by TCP (a reliable, in-order stream protocol) or UDP (an unreliable datagram protocol). We use the term “Transport Service” to mean the end-to-end service provided to an application by the transport layer. That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at design time, compile time, or run time, and may include guidance on whether a feature is required, a preference by the application, or something in between. Examples of features of Transport Services are reliable delivery, ordered delivery, content privacy to in-path devices, integrity protection, and minimal latency.
The IETF has defined a wide variety of transport protocols beyond TCP and UDP, including SCTP, DCCP, MP-TCP, and UDP-Lite. Transport services may be provided directly by these transport protocols, or layered on top of them using protocols such as WebSockets (which runs over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run over SCTP over DTLS over UDP or TCP). Services built on top of UDP or UDP-Lite typically also need to specify additional mechanisms, including a congestion control mechanism (such as a windowed congestion control, TFRC or LEDBAT congestion control mechanism). This extends the set of available Transport Services beyond those provided to applications by TCP and UDP.
Transport protocols can also be differentiated by the features of the services they provide: for instance, SCTP offers a message-based service providing full or partial reliability and allowing to minimize the head of line blocking due to the support of unordered and unordered message delivery within multiple streams, UDP-Lite provides partial integrity protection, and LEDBAT can provide low-priority “scavenger” communication.
The following terms are defined throughout this document, and in subsequent documents produced by TAPS describing the composition and decomposition of transport services.
[EDITOR’S NOTE: we may want to add definitions for the different kinds of interfaces that are important here.]
This section provides a list of known IETF transport protocol and transport protocol frameworks.
[EDITOR’S NOTE: Contributions to the subsections below are welcome]
TCP is an IETF standards track transport protocol. [RFC0793] introduces TCP as follows: “The Transmission Control Protocol (TCP) is intended for use as a highly reliable host-to-host protocol between hosts in packet-switched computer communication networks, and in interconnected systems of such networks.” Since its introduction, TCP has become the default connection-oriented, stream-based transport protocol in the Internet. It is widely implemented by endpoints and widely used by common applications.
TCP is a connection-oriented protocol, providing a three way handshake to allow a client and server to set up a connection, and mechanisms for orderly completion and immediate teardown of a connection. TCP is defined by a family of RFCs [RFC4614].
TCP provides multiplexing to multiple sockets on each host using port numbers. An active TCP session is identified by its four-tuple of local and remote IP addresses and local port and remote port numbers. The destination port during connection setup has a different role as it is often used to indicate the requested service.
TCP partitions a continuous stream of bytes into segments, sized to fit in IP packets. ICMP-based PathMTU discovery [RFC1191][RFC1981] as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821] are supported.
Each byte in the stream is identified by a sequence number. The sequence number is used to order segments on receipt, to identify segments in acknowledgments, and to detect unacknowledged segments for retransmission. This is the basis of TCP’s reliable, ordered delivery of data in a stream. TCP Selective Acknowledgment [RFC2018] extends this mechanism by making it possible to identify missing segments more precisely, reducing spurious retransmission.
Receiver flow control is provided by a sliding window: limiting the amount of unacknowledged data that can be outstanding at a given time. The window scale option [RFC7323] allows a receiver to use windows greater than 64KB.
All TCP senders provide Congestion Control: This uses a separate window, where each time congestion is detected, this congestion window is reduced. A receiver detects congestion using one of three mechanisms: A retransmission timer, detection of loss (interpreted as a congestion signal), or Explicit Congestion Notification (ECN) [RFC3168] to provide early signaling (see [I-D.ietf-aqm-ecn-benefits])
A TCP protocol instance can be extended [RFC4614] and tuned. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only, some are explicitly negotiated during connection setup.
By default, TCP segment partitioning uses Nagle’s algorithm [RFC0896] to buffer data at the sender into large segments, potentially incurring sender-side buffering delay; this algorithm can be disabled by the sender to transmit more immediately, e.g. to enable smoother interactive sessions.
[EDITOR’S NOTE: add URGENT and PUSH flag (note [RFC6093] says SHOULD NOT use due to the range of TCP implementations that process TCP urgent indications differently.) ]
A checksum provides an Integrity Check and is mandatory across the entire packet. The TCP checksum does not support partial corruption protection as in DCCP/UDP-Lite). This check protects from misdelivery of data corrupted data, but is relatively weak, and applications that require end to end integrity of data are recommended to include a stronger integrity check of their payload data.
A TCP service is unicast.
A User/TCP Interface is defined in [RFC0793] providing six user commands: Open, Send, Receive, Close, Status. This interface does not describe configuration of TCP options or parameters beside use of the PUSH and URGENT flags.
In API implementations derived from the BSD Sockets API, TCP sockets are created using the SOCK_STREAM socket type.
The features used by a protocol instance may be set and tuned via this API.
(more on the API goes here)
The transport protocol components provided by TCP are:
[EDITOR’S NOTE: discussion of how to map this to features and TAPS: what does the higher layer need to decide? what can the transport layer decide based on global settings? what must the transport layer decide based on network characteristics?]
Multipath TCP [RFC6824] is an extension for TCP to support multi-homing. It is designed to be as transparent as possible to middle-boxes. It does so by establishing regular TCP flows between a pair of source/destination endpoints, and multiplexing the application’s stream over these flows.
MPTCP uses TCP options for its control plane. They are used to signal multipath capabilities, as well as to negotiate data sequence numbers, and advertise other available IP addresses and establish new sessions between pairs of endpoints.
By default, MPTCP exposes the same interface as TCP to the application. [RFC6897] however describes a richer API for MPTCP-aware applications.
This Basic API describes how an application can - enable or disable MPTCP; - bind a socket to one or more selected local endpoints; - query local and remote endpoint addresses; - get a unique connection identifier (similar to an address–port pair for TCP).
The document also recommend the use of extensions defined for SCTP [RFC6458] (see next section) to deal with multihoming.
[AUTHOR’S NOTE: research work, and some implementation, also suggest that the scheduling algorithm, as well as the path manager, are configurable options that should be exposed to higher layer. Should this be discussed here?]
[AUTHOR’S NOTE: shouldn’t it be “service feature”?]
As an extension to TCP, MPTCP provides mostly the same components. By establishing multiple sessions between available endpoints, it can additionally provide soft failover solutions should one of the paths become unusable. In addition, by multiplexing one byte stream over separate paths, it can achieve a higher throughput than TCP in certain situations (note however that coupled congestion control [RFC6356] might limit this benefit to maintain fairness to other flows at the bottleneck). When aggregating capacity over multiple paths, and depending on the way packets are scheduled on each TCP subflow, an additional delay and higher jitter might be observed observed before in-order delivery of data to the applications.
The transport protocol components provided by MPTCP therefore are:
[AUTHOR’S NOTE: it is unclear whether MPTCP has to provide data bundling.] [AUTHOR’S NOTE: AF muliplexing? sub-flows can be started over IPv4 or IPv6 for the same session]
SCTP is a message oriented standards track transport protocol and the base protocol is specified in [RFC4960]. It supports multi-homing to handle path failures. An SCTP association has multiple unidirectional streams in each direction and provides in-sequence delivery of user messages only within each stream. This allows to minimize head of line blocking. SCTP is extensible and the currently defined extensions include mechanisms for dynamic re-configurations of streams [RFC6525] and IP-addresses [RFC5061]. Furthermore, the extension specified in [RFC3758] introduces the concept of partial reliability for user messages.
SCTP was originally developed for transporting telephony signalling messages and is deployed in telephony signalling networks, especially in mobile telephony networks. Additionally, it is used in the WebRTC framework for data channels and is therefore deployed in all WEB-browsers supporting WebRTC.
SCTP is a connection oriented protocol using a four way handshake to establish an SCTP association and a three way message exchange to gracefully shut it down. It uses the same port number concept as DCCP, TCP, UDP, and UDP-Lite do and only supports unicast.
SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit errors. This is stronger than the 16-bit checksums used by TCP or UDP. However, a partial checksum coverage as provided by DCCP or UDP-Lite is not supported.
SCTP has been designed with extensibility in mind. Each SCTP packet starts with a single common header containing the port numbers, a verification tag and the CRC32c checksum. This common header is followed by a sequence of chunks. Each chunk consists of a type field, flags, a length field and a value. [RFC4960] defines how a receiver processes chunks with an unknown chunk type. The support of extensions can be negotiated during the SCTP handshake.
SCTP provides a message-oriented service. Multiple small user messages can be bundled into a single SCTP packet to improve the efficiency. For example, this bundling may be done by delaying user messages at the sender side similar to the Nagle algorithm used by TCP. User messages which would result in IP packets larger than the MTU will be fragmented at the sender side and reassembled at the receiver side. There is no protocol limit on the user message size. ICMP-based path MTU discovery as specified for IPv4 in [RFC1191] and for IPv6 in [RFC1981] as well as packetization layer path MTU discovery as specified in [RFC4821] with probe packets using the padding chunks defined the [RFC4820] are supported.
[RFC4960] specifies a TCP friendly congestion control to protect the network against overload. SCTP also uses a sliding window flow control to protect receivers against overflow.
Each SCTP association has between 1 and 65536 uni-directional streams in each direction. The number of streams can be different in each direction. Every user-message is sent on a particular stream. User messages can be sent un-ordered or ordered upon request by the upper layer. Un-ordered messages can be delivered as soon as they are completely received. Only all ordered messages sent on the same stream are delivered at the receiver in the same order as sent by the sender. For user messages not requiring fragmentation, this minimises head of line blocking. The base protocol defined in [RFC4960] doesn’t allow interleaving of user-messages, which results in sending a large message on one stream can block the sending of user messages on other streams. [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. Furthermore, [I-D.ietf-tsvwg-sctp-ndata] specifies multiple algorithms for the sender side selection of which streams to send data from supporting a variety of scheduling algorithms including priority based ones. The stream re-configuration extension defined in [RFC6525] allows to reset streams during the lifetime of an association and to increase the number of streams, if the number of streams negotiated in the SCTP handshake is not sufficient.
According to [RFC4960], each user message sent is either delivered to the receiver or, in case of excessive retransmissions, the association is terminated in a non-graceful way, similar to the TCP behaviour. In addition to this reliable transfer, the partial reliability extension defined in [RFC3758] allows the sender to abandon user messages. The application can specify the policy for abandoning user messages. Examples for these policies include:
SCTP supports multi-homing. Each SCTP end-point uses a list of IP-addresses and a single port number. These addresses can be any mixture of IPv4 and IPv6 addresses. These addresses are negotiated during the handshake and the address re-configuration extension specified in [RFC5061] in combination with [RFC4895] can be used to change these addresses in an authenticated way during the livetime of an SCTP association. This allows for transport layer mobility. Multiple addresses are used for improved resilience. If a remote address becomes unreachable, the traffic is switched over to a reachable one, if one exists. Each SCTP end-point supervises continuously the reachability of all peer addresses using a heartbeat mechanism.
For securing user messages, the use of TLS over SCTP has been specified in [RFC3436]. However, this solution does not support all services provided by SCTP (for example un-ordered delivery or partial reliability), and therefore the use of DTLS over SCTP has been specified in [RFC6083] to overcome these limitations. When using DTLS over SCTP, the application can use almost all services provided by SCTP.
[I-D.ietf-tsvwg-natsupp] defines a methods for end-hosts and middleboxes to provide for NAT support for SCTP over IPv4. For legacy NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP-packets. Alternatively, SCTP packets can be encapsulated in DTLS packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The latter encapsulation is used with in the WebRTC context.
Having a well defined API is also a feature provided by SCTP as described in the next subsection.
[RFC4960] defines an abstract API for the base protocol. An extension to the BSD Sockets API is defined in [RFC6458] and covers:
For the following SCTP protocol extensions the BSD Sockets API extension is defined in the document specifying the protocol extensions:
Future documents describing SCTP protocol extensions are expected to describe the corresponding BSD Sockets API extension in a Socket API Considerations section.
The SCTP socket API supports two kinds of sockets:
One-to-one style sockets are similar to TCP sockets, there is a 1:1 relationship between the sockets and the SCTP associations (except for listening sockets). One-to-many style SCTP sockets are similar to unconnected UDP sockets as there is a 1:n relationship between the sockets and the SCTP associations.
The SCTP stack can provide information to the applications about state changes of the individual paths and the association whenever they occur. These events are delivered similar to user messages but are specifically marked as notifications.
A couple of new functions have been introduced to support the use of multiple local and remote addresses. Additional SCTP-specific send and receive calls have been defined to allow dealing with the SCTP specific information without using ancillary data in the form of additional cmsgs, which are also defined. These functions provide support for detecting partial delivery of user messages and notifications.
The SCTP socket API allows a fine-grained control of the protocol behaviour through an extensive set of socket options.
The SCTP kernel implementations of FreeBSD, Linux and Solaris follow mostly the specified extension to the BSD Sockets API for the base protocol and the corresponding supported protocol extensions.
The transport protocol components provided by SCTP are:
[EDITOR’S NOTE: update this list.]
The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF standards track transport protocol. It provides a uni-directional, datagram protocol which preserves message boundaries. It provides none of the following transport features: error correction, congestion control, or flow control. It can be used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), in addition to unicast (and anycast) datagrams. IETF guidance on the use of UDP is provided in[RFC5405]. UDP is widely implemented and widely used by common applications, especially DNS.
UDP is a connection-less protocol which maintains message boundaries, with no connection setup or feature negotiation. The protocol uses independent messages, ordinarily called datagrams. The lack of error control and flow control implies messages may be damaged, re-ordered, lost, or duplicated in transit. A receiving application unable to run sufficiently fast or frequently may miss messages. The lack of congestion handling implies UDP traffic may cause the loss of messages from other protocols (e.g., TCP) when sharing the same network paths. UDP traffic can also cause the loss of other UDP traffic in the same or other flows for the same reasons.
Messages with bit errors are ordinarily detected by an invalid end- to-end checksum and are discarded before being delivered to an application. There are some exceptions to this general rule, however. UDP-Lite (see [RFC3828], and below) provides the ability for portions of the message contents to be exempt from checksum coverage. It is also possible to create UDP datagrams with no checksum, and while this is generally discouraged [RFC1122] [RFC5405], certain special cases permit its use [RFC6935]. The checksum support considerations for omitting the checksum are defined in [RFC6936]. Note that due to the relatively weak form of checksum used by UDP, applications that require end to end integrity of data are recommended to include a stronger integrity check of their payload data.
On transmission, UDP encapsulates each datagram into an IP packet, which may in turn be fragmented by IP. Applications concerned with fragmentation or that have other requirements such as receiver flow control, congestion control, PathMTU discovery/PLPMTUD, support for ECN, etc need to be provided by protocols other than UDP [RFC5405].
[RFC0768] describes basic requirements for an API for UDP. Guidance on use of common APIs is provided in [RFC5405].
A UDP endpoint consists of a tuple of (IP address, port number). Demultiplexing using multiple abstract endpoints (sockets) on the same IP address are supported. The same socket may be used by a single server to interact with multiple clients (note: this behavior differs from TCP, which uses a pair of tuples to identify a connection). Multiple server instances (processes) binding the same socket can cooperate to service multiple clients– the socket implementation arranges to not duplicate the same received unicast message to multiple server processes.
Many operating systems also allow a UDP socket to be “connected”, i.e., to bind a UDP socket to a specific (remote) UDP endpoint. Unlike TCP’s connect primitive, for UDP, this is only a local operation that serves to simplify the local send/receive functions and to filter the traffic for the specified addresses and ports [RFC5405].
The transport protocol components provided by UDP are:
The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an IETF standards track transport protocol. UDP-Lite provides a bidirectional set of logical unicast or multicast message streams over a datagram protocol. IETF guidance on the use of UDP-Lite is provided in [RFC5405].
UDP-Lite is a connection-less datagram protocol, with no connection setup or feature negotiation. The protocol use messages, rather than a byte-stream. Each stream of messages is independently managed, therefore retransmission does not hold back data sent using other logical streams.
It provides multiplexing to multiple sockets on each host using port numbers. An active UDP-Lite session is identified by its four-tuple of local and remote IP addresses and local port and remote port numbers.
UDP-Lite fragments packets into IP packets, constrained by the maximum size of IP packet.
UDP-Lite changes the semantics of the UDP “payload length” field to that of a “checksum coverage length” field. Otherwise, UDP-Lite is semantically identical to UDP. Applications using UDP-Lite therefore can not make assumptions regarding the correctness of the data received in the insensitive part of the UDP-Lite payload.
As for UDP, mechanisms for receiver flow control, congestion control, PMTU or PLPMTU discovery, support for ECN, etc need to be provided by upper layer protocols [RFC5405].
Examples of use include a class of applications that can derive benefit from having partially-damaged payloads delivered, rather than discarded. One use is to support error tolerate payload corruption when used over paths that include error-prone links, another application is when header integrity checks are required, but payload integrity is provided by some other mechanism (e.g. [RFC6936].
A UDP-Lite service may support IPv4 broadcast, multicast, anycast and unicast.
There is no current API specified in the RFC Series, but guidance on use of common APIs is provided in [RFC5405].
The interface of UDP-Lite differs from that of UDP by the addition of a single (socket) option that communicates a checksum coverage length value: at the sender, this specifies the intended checksum coverage, with the remaining unprotected part of the payload called the “error-insensitive part”. The checksum coverage may also be made visible to the application via the UDP-Lite MIB module [RFC5097].
The transport protocol components provided by UDP-Lite are:
Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF standards track bidirectional transport protocol that provides unicast connections of congestion-controlled unreliable messages.
[EDITOR’S NOTE: Gorry Fairhurst signed up as a contributor for this section.]
The DCCP Problem Statement describes the goals that DCCP sought to address [RFC4336]. It is suitable for applications that transfer fairly large amounts of data and that can benefit from control over the trade off between timeliness and reliability [RFC4336].
It offers low overhead, and many characteristics common to UDP, but can avoid “Re-inventing the wheel” each time a new multimedia application emerges. Specifically it includes core functions (feature negotiation, path state management, RTT calculation, PMTUD, etc): This allows applications to use a compatible method defining how they send packets and where suitable to choose common algorithms to manage their functions. Examples of suitable applications include interactive applications, streaming media or on-line games [RFC4336].
DCCP is a connection-oriented datagram protocol, providing a three way handshake to allow a client and server to set up a connection, and mechanisms for orderly completion and immediate teardown of a connection. The protocol is defined by a family of RFCs.
It provides multiplexing to multiple sockets on each host using port numbers. An active DCCP session is identified by its four-tuple of local and remote IP addresses and local port and remote port numbers. At connection setup, DCCP also exchanges the the service code [RFC5595] mechanism to allow transport instantiations to indicate the service treatment that is expected from the network.
The protocol segments data into messages, typically sized to fit in IP packets, but which may be fragmented providing they are less than the A DCCP interface MAY allow applications to request fragmentation for packets larger than PMTU, but not larger than the maximum packet size allowed by the current congestion control mechanism (CCMPS) [RFC4340].
Each message is identified by a sequence number. The sequence number is used to identify segments in acknowledgments, to detect unacknowledged segments, to measure RTT, etc. The protocol may support ordered or unordered delivery of data, and does not itself provide retransmission. There is a Data Checksum option, which contains a strong CRC, lets endpoints detect application data corruption. It also supports reduced checksum coverage, a partial integrity mechanisms similar to UDP-lIte.
Receiver flow control is supported: limiting the amount of unacknowledged data that can be outstanding at a given time.
A DCCP protocol instance can be extended [RFC4340] and tuned. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only, some are explicitly negotiated during connection setup.
DCCP supports negotiation of the congestion control profile, to provide Plug and Play congestion control mechanisms. examples of specified profiles include [RFC4341] [RFC4342] [RFC5662]. All IETF-defined methods provide Congestion Control.
DCCP use a Connect packet to start a session, and permits half-connections that allow each client to choose features it wishes to support. Simultaneous open [RFC5596], as in TCP, can enable interoperability in the presence of middleboxes. The Connect packet includes a Service Code field [RFC5595] designed to allow middle boxes and endpoints to identify the characteristics required by a session. A lightweight UDP-based encapsulation (DCCP-UDP) has been defined [RFC6773] that permits DCCP to be used over paths where it is not natively supported. Support in NAPT/NATs is defined in [RFC4340] and [RFC5595].
Upper layer protocols specified on top of DCCP include: DTLS [RFC5595], RTP [RFC5672], ICE/SDP [RFC6773].
A DCCP service is unicast.
A common packet format has allowed tools to evolve that can read and interpret DCCP packets (e.g. Wireshark).
API characteristics include: - Datagram transmission. - Notification of the current maximum packet size. - Send and reception of zero-length payloads. - Set the Slow Receiver flow control at a receiver. - Detect a Slow receiver at the sender.
There is no current API specified in the RFC Series.
The transport protocol components provided by DCCP are:
RTP provides an end-to-end network transport service, suitable for applications transmitting real-time data, such as audio, video or data, over multicast or unicast network services, including TCP, UDP, UDP-Lite, DCCP.
[EDITOR’S NOTE: Varun Singh signed up as contributor for this section. Given the complexity of RTP, suggest to have an abbreviated section here contrasting RTP with other transports, and focusing on those features that are RTP-unique.]
NORM is an IETF standards track protocol specified in [RFC5740]. The protocol was designed to support reliable bulk data dissemination to receiver groups using IP Multicast but also provides for point-to-point unicast operation. Its support for bulk data dissemination includes discrete file or computer memory-based “objects” as well as byte- and message-streaming. NORM is designed to incorporate packet erasure coding as an inherent part of its selective ARQ in response to receiver negative acknowledgements. The packet erasure coding can also be proactively applied for forward protection from packet loss. NORM transmissions are governed by TCP-friendly congestion control. NORM’s reliability, congestion control, and flow control mechanism are distinct components and can be separately controlled to meet different application needs.
[EDITOR’S NOTE: needs to be more clear about the application of FEC and packet erasure coding; expand ARQ.]
The NORM protocol is encapsulated in UDP datagrams and thus provides multiplexing for multiple sockets on hosts using port numbers. For purposes of loosely coordinated IP Multicast, NORM is not strictly connection-oriented although per-sender state is maintained by receivers for protocol operation. [RFC5740] does not specify a handshake protocol for connection establishment and separate session initiation can be used to coordinate port numbers. However, in-band “client-server” style connection establishment can be accomplished with the NORM congestion control signaling messages using port binding techniques like those for TCP client-server connections.
NORM supports bulk “objects” such as file or in-memory content but also can treat a stream of data as a logical bulk object for purposes of packet erasure coding. In the case of stream transport, NORM can support either byte streams or message streams where application-defined message boundary information is carried in the NORM protocol messages. This allows the receiver(s) to join/re-join and recover message boundaries mid-stream as needed. Application content is carried and identified by the NORM protocol with encoding symbol identifiers depending upon the Forward Error Correction (FEC) Scheme [RFC3452] configured. NORM uses NACK-based selective ARQ to reliably deliver the application content to the receiver(s). NORM proactively measures round-trip timing information to scale ARQ timers appropriately and to support congestion control. For multicast operation, timer-based feedback suppression is uses to achieve group size scaling with low feedback traffic levels. The feedback suppression is not applied for unicast operation.
NORM uses rate-based congestion control based upon the TCP-Friendly Rate Control (TFRC) [RFC4324] principles that are also used in DCCP [RFC4340]. NORM uses control messages to measure RTT and collect congestion event (e..g, loss event, ECN event, etc) information from the receiver(s) to support dynamic rate control adjustment. The TCP-Friendly Multicast Congestion Control (TFMCC) [RFC4654] used provides some extra features to support multicast but is functionally equivalent to TFRC in the unicast case.
NORM’s reliability mechanism is decoupled from congestion control. This allows alternative arrangements of transport services to be invoked. For example, fixed-rate reliable delivery can be supported or unreliable (but optionally “better than best effort” via packet erasure coding) delivery with rate-control per TFRC can be achieved. Additionally, alternative congestion control techniques may be applied. For example, TFRC rate control with congestion event detection based on ECN for links with high packet loss (e.g., wireless) has been implemented and demonstrated with NORM.
While NORM is NACK-based for reliability transfer, it also supports a positive acknowledgment (ACK) mechanism that can be used for receiver flow control. Again, since this mechanism is decoupled from the reliability and congestion control, applications that have different needs in this aspect can use the protocol differently. One example is the use of NORM for quasi-reliable delivery where timely delivery of newer content may be favored over completely reliable delivery of older content within buffering and RTT constraints.
The NORM specification does not describe a specific application programming interface (API) to control protocol operation. A freely-available, open source reference implementation of NORM is available at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented API is provided for this implementation. While a sockets-like API is not currently documented, the existing API supports the necessary functions for that to be implemented.
The transport protocol components provided by NORM are:
Transport Layer Security (TLS) and Datagram TLS are IETF protocols that provide several security-related features to applications. TLS is designed to run on top of TCP, DTLS is designed to run on top of UDP. At the time of writing, the current version of TLS is 1.2; it is defined in [RFC5246]. DTLS provides nearly identical functionality; it is defined in {RFC6347}} and also at version 1.2.
While older versions of TLS and DTLS are still in use, they provide weaker security guarantees. [RFC7457] outlines important attacks on TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document that describes secure configurations for TLS and DTLS to counter these attacks. The recommendations are applicable for the vast majority of use cases.
[NOTE: The Logjam authors (weakdh.org) give (inconclusive) evidence that one of the recommendations of [RFC7525], namely use to DHE-1024 as a fallback, may not be sufficient in all cases to counter an attacker with the resources of a nation-state. It is unclear at this time if the RFC is going to be updated as a result or whether there will be an RFC7525bis.]
Both TLS and DTLS provide the same security features and can thus be discussed together. The features they provide are:
[Note: Both TLS and DTLS provide replay protection, although it is optional in DTLS. The TLS RFC discusses this only in the security considerations and thus views it as a feature that is implicit in the ones listed above. DTLS mentions it as an explicit feature.]
The authentication of the peer entity can be omitted, although this is a rare use case. In many use cases (e.g. the Web), authentication is not mutual, however (e.g. only the Web server is authenticated, but not the client). It is important to note that TLS itself does not specify how a peering entity is to be authenticated. This is part of the application logic; i.e. the authentication decision rests with the application. As an example, in the common use case of authentication by means of an X.509 certificate, it is the application’s decision whether the certificate of the peering entity is acceptable for the purposes of the application or whether the handshake should be aborted.
As DTLS is used over the unreliable UDP transport, it needs to add three features to provide the same security guarantees as TLS: * Message fragmentation * Message reordering * Message loss
As a result, DTLS provides features that UDP lacks.
[EDITOR’S NOTE: Need to describe how this is achieved?]
TLS is commonly used with a socket-like interface, although details can vary between implementations. This is particularly true for the choice which cryptographic algorithms to use, see below.
[TODO: DTLS interface]
Both TLS and DTLS allow to employ a multitude of cipher suites for encryption, hashing and applying message integrity. It is no easy task to choose safe settings here. [RFC7525] provides guidance.
[TODO: list the RFCs?] [TODO: more detail?] ### Transport Protocol Components
Both TLS and DTLS employ a layered architecture. The lower layer is commonly called the record protocol. It is responsible for fragmenting messages, applying message authentication codes (MACs), encrypting data, and sending it via the underlying transport protocol. Several essential protocols run on top of the record protocol in order to carry out the handshake and establish a secure session.
[EDITOR’S NOTE: TLS can also compress, but this has been found to be a security weakness. It is not described here.]
Hypertext Transfer Protocol (HTTP) is an application-level protocol widely used on the Internet. Version 1.1 of the protocol is specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] [RFC7235], and version 2 in [RFC7540]. Furthermore, HTTP is used as a substrate for other application-layer protocols. There are various reasons for this practice listed in [RFC3205]; these include being a well-known and well-understood protocol, reusability of existing servers and client libraries, easy use of existing security mechanisms such as HTTP digest authentication [RFC2617] and TLS [RFC5246], the ability of HTTP to traverse firewalls which makes it work with a lot of infrastructure, and cases where a application server often needs to support HTTP anyway.
Depending on application’s needs, the use of HTTP as a substrate protocol may add complexity and overhead in comparison to a special-purpose protocol (e.g. HTTP headers, suitability of the HTTP security model etc.). [RFC3205] address this issues and provides some guidelines and concerns about the use of HTTP standard port 80 and 443, the use of HTTP URL scheme and interaction with existing firewalls, proxies and NATs.
Though not strictly bound to TCP, HTTP is almost exclusively run over TCP, and therefore inherits its properties when used in this way.
Hypertext Transfer Protocol (HTTP) is a request/response protocol. A client sends a request containing a request method, URI and protocol version followed by a MIME-like message (see [RFC7231] for the differences between an HTTP object and a MIME message), containing information about the client and request modifiers. The message can contain a message body carrying application data as well. The server responds with a status or error code followed by a MIME-like message containing information about the server and information about carried data and it can include a message body. It is possible to specify a data format for the message body using MIME media types [RFC2045]. Furthermore, the protocol has numerous additional features; features relevant to pseudotransport are described below.
Content negotiation, specified in [RFC7231], is a mechanism provided by HTTP for selecting a representation on a requested resource. The client and server negotiate acceptable data formats, charsets, data encoding (e.g. data can be transferred compressed, gzip), etc. HTTP can accommodate exchange of messages as well as data streaming (using chunked transfer encoding [RFC7230]). It is also possible to request a part of a resource using range requests specified in [RFC7233]. The protocol provides powerful cache control signalling defined in [RFC7234].
HTTP 1.1’s and HTTP 2.0’s persistent connections can be use to perform multiple request-response transactions during the life-time of a single HTTP connection. Moreover, HTTP 2.0 connections can multiplex many request/response pairs in parallel on a single connection. This reduces connection establishment overhead and the effect of TCP slow-start on each transaction, important for HTTP’s primary use case.
It is possible to combine HTTP with security mechanisms, like TLS (denoted by HTTPS), which adds protocol properties provided by such a mechanism (e.g. authentication, encryption, etc.). TLS’s Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used for HTTP version negotiation within TLS handshake which eliminates addition round-trip. Arbitrary cookie strings, included as part of the MIME headers, are often used as bearer tokens in HTTP.
Application layer protocols using HTTP as substrate may use existing method and data formats, or specify new methods and data formats. Furthermore some protocols may not fit a request/response paradigm and instead rely on HTTP to send messages (e.g. [RFC6546]). Because HTTP is working in many restricted infrastructures, it is also used to tunnel other application-layer protocols.
There are many HTTP libraries available exposing different APIs. The APIs provide a way to specify a request by providing a URI, a method, request modifiers and optionally a request body. For the response, callbacks can be registered that will be invoked when the response is received. If TLS is used, API expose a registration of callbacks in case a server requests client authentication and when certificate verification is needed.
World Wide Web Consortium (W3C) standardized the XMLHttpRequest API [XHR], an API that can be use for sending HTTP/HTTPS requests and receiving server responses. Besides XML data format, request and response data format can also be JSON, HTML and plain text. Specifically JavaScript and XMLHttpRequest are a ubiquitous programming model for websites, and more general applications, where native code is less attractive.
Representational State Transfer (REST) [REST] is another example how applications can use HTTP as transport protocol. REST is an architecture style for building application on the Internet. It uses HTTP as a communication protocol.
The transport protocol components provided by HTTP, when used as a pseudotransport, are:
HTTPS (HTTP over TLS) additionally provides the following components:
[EDITOR’S NOTE: Salvatore Loreto will contribute text for this section.]
The transport protocol components analyzed in this document which can be used as a basis for defining common transport service features, normalized and separated into categories, are as follows:
The next revision of this document will define transport service features based upon this list.
[EDITOR’S NOTE: this section will drawn from the candidate features provided by protocol components in the previous section – please discuss on taps@ietf.org list]
[EDITOR’S NOTE: Dave Thaler has signed up as a contributor for this section. Michael Welzl also has a beginning of a matrix which could be useful here.]
[EDITOR’S NOTE: The below is a strawman proposal below by Gorry Fairhurst for initial discussion]
The table below summarises protocol mechanisms that have been standardised. It does not make an assessment on whether specific implementations are fully compliant to these specifications.
| Mechanism | UDP | UDP-L | DCCP | SCTP | TCP |
|---|---|---|---|---|---|
| Unicast | Yes | Yes | Yes | Yes | Yes |
| Mcast/IPv4Bcast | Yes(2) | Yes | No | No | No |
| Port Mux | Yes | Yes | Yes | Yes | Yes |
| Mode | Dgram | Dgram | Dgram | Dgram | Stream |
| Connected | No | No | Yes | Yes | Yes |
| Data bundling | No | No | No | Yes | Yes |
| Feature Nego | No | No | Yes | Yes | Yes |
| Options | No | No | Support | Support | Support |
| Data priority | * | * | * | Yes | No |
| Data bundling | No | No | No | Yes | Yes |
| Reliability | None | None | None | Select | Full |
| Ordered deliv | No | No | No | Stream | Yes |
| Corruption Tol. | No | Support | Support | No | No |
| Flow Control | No | No | Support | Yes | Yes |
| PMTU/PLPMTU | (1) | (1) | Yes | Yes | Yes |
| Cong Control | (1) | (1) | Yes | Yes | Yes |
| ECN Support | (1) | (1) | Yes | TBD | Yes |
| NAT support | Limited | Limited | Support | TBD | Support |
| Security | DTLS | DTLS | DTLS | DTLS | TLS, AO |
| UDP encaps | N/A | None | Yes | Yes | None |
| RTP support | Support | Support | Support | ? | Support |
Note (1): this feature requires support in an upper layer protocol.
Note (2): this feature requires support in an upper layer protocol when used with IPv6.
This document has no considerations for IANA.
This document surveys existing transport protocols and protocols providing transport-like services. Confidentiality, integrity, and authenticity are among the features provided by those services. This document does not specify any new components or mechanisms for providing these features. Each RFC listed in this document discusses the security considerations of the specification it contains.
[Editor’s Note: turn this into a real contributors section with addresses once we figure out how to trick the toolchain into doing so]
Thanks to Karen Nielsen, Joe Touch, and Michael Welzl for the comments, feedback, and discussion. This work is partially supported by the European Commission under grant agreement FP7-ICT-318627 mPlane; support does not imply endorsement.
[EDITOR’S NOTE: add H2020-NEAT ack].
| [RFC0791] | Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981. |