| Network Working Group | G. Fairhurst, Ed. |
| Internet-Draft | University of Aberdeen |
| Intended status: Informational | B. Trammell, Ed. |
| Expires: June 10, 2016 | M. Kuehlewind, Ed. |
| ETH Zurich | |
| December 08, 2015 |
Services provided by IETF transport protocols and congestion control mechanisms
draft-ietf-taps-transports-08
This document describes transport services provided by existing IETF protocols. It is designed to help application and network stack programmers and to inform the work of the IETF TAPS Working Group.
This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
This Internet-Draft will expire on June 10, 2016.
Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
Internet applications make use of the Services provided by a Transport protocol, such as 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, and integrity protection.
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 NewReno, TFRC or LEDBAT). This extends the set of available Transport Services beyond those provided to applications by TCP and UDP.
The following terms are defined throughout this document, and in subsequent documents produced by TAPS describing the composition and decomposition of transport services.
Transport protocols can be differentiated by the features of the services they provide.
One fundamental feature is whether a transport offers a service that divides the data into transmission units based on network packets (known as a Datagram service), or whether it combines and segments data across multiple packets (e.g., the Stream service provided by TCP).
Another fundamental feature is whether a transport requires a control exchange across the network at setup (e.g., TCP), or whether it connection-less (e.g., UDP).
A transport service can also offer reliability, 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 and DCCP provide partial integrity protection.
A transport service can provide congestion control (see Section 3.1). TCP and SCTP provide congestion control for use in the Internet, whereas UDP leaves this function to the upper layer protocol that uses UDP. DCCP offers a range of congestion control approaches and LEDBAT can support low-priority “scavenger” communication, intending to defer use of capacity to other Internet flows sharing a congested bottleneck.
Transport services may be unidirectional or bidirectional, to a single a single endpoint, to one of multiple endpoints, or multicast simultaneously to multiple endpoints.
The service offered by transport protocols and frameworks can also be differentiated in many other ways.
Congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use a datagram protocol (e.g., UDP or UDP-Lite) need to employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic.
A variety of techniques are used to provide congestion control in the Internet. Each technique requires that the protocol provide a method for deriving the metric the congestion control algorithm uses to detect congestion and the property of a packet it uses to determine when to send. Given these relatively wide constraints, the congestion control techniques that can be applied by different transport protocols are largely orthogonal to the choice of transport protocols themselves. This section provides an overview of the congestion control techniques available to the protocols described in Section 4.
Most commonly deployed congestion control mechanisms use one of three mechanisms to detect congestion:
Protocols such as SCTP and TCP [RFC5681] that use sliding-window-based receiver flow control commonly use a separate congestion window for congestion control. Each time congestion is detected, this separate congestion window is reduced. Data in flight is capped to the minimum of the two windows. This approach is also used by DCCP CCID-2 for datagram congestion control.
Rate-based methods have also been defined based on the loss ratio and observed round trip time, such as TFRC [RFC5348] and TFRC-SP [RFC4828]. These methods utlise a throughput equation to determine the maximum acceptable rate. Such methods are used with DCCP CCID-3 [RFC4342] and CCID-4 [RFC5622], WEBRC [RFC3738], and other applications.
In addition, a congestion control mechanism may react to changes in delay as an indication for congestion. Delay-based congestion detection methods tend to induce less loss than loss-based methods, and therefore generally do not compete well with them across shared bottleneck links. However, such methods, such as LEDBAT [RFC6824], are are deployed in the Internet for scavenger traffic, which will use unused capacity but readily yield to presumably interactive or otherwise higher-priority, loss-based congestion-controlled traffic.
This section provides a list of known IETF transport protocols and transport protocol frameworks. It does not make an assessment about whether specific implementations of protocols are fully compliant to current IETF specifications.
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 negotiate features, 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. A similar approach is adopted by other IETF-defined transports. 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 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 Path MTU discovery [RFC1191][RFC1981] as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821] have been defined by the IETF.
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 the reliable, ordered delivery of data in a TCP 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.
TCP provides congestion control [RFC5681], described further in Section 3.1 below.
TCP protocol instances 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 reduce latency for interactive sessions.
TCP provides an “urgent data” function for limited out-of-order delivery of the data. This function is deprecated [RFC6093].
A mandatory checksum provides a basic integrity check against misdelivery and data corruption over the entire packet. Applications that require end to end integrity of data are recommended to include a stronger integrity check of their payload data. The TCP checksum does not support partial corruption protection (as in DCCP/UDP-Lite).
TCP supports only unicast connections.
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.
[RFC1122] describes extensions of the TCP/application layer interface for:
In API implementations derived from the BSD Sockets API, TCP sockets are created using the SOCK_STREAM socket type as described in the IEEE Portable Operating System Interface (POSIX) Base Specifications [POSIX]. The features used by a protocol instance may be set and tuned via this API. There are current no documents in the RFC Series that describe this interface.
The transport features provided by TCP are:
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:
The document also recommends the use of extensions defined for SCTP [RFC6458] (see next section) to support multihoming.
As an extension to TCP, MPTCP provides mostly the same features. 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 features provided by MPTCP in addition to TCP therefore are:
SCTP is a message-oriented IETF standards track transport protocol. The base protocol is specified in [RFC4960]. It supports multi-homing and path failover to provide resilience to path failures. An SCTP association has multiple streams in each direction, providing in-sequence delivery of user messages within each stream. This allows it to minimize head of line blocking. SCTP supports multiple stream scheduling schemes controlling stream multiplexing, including priority and fair weighting schemes.
SCTP is extensible. Currently defined extensions include mechanisms for dynamic re-configuration 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. It can also be used for other services, for example in the WebRTC framework for data channels. It 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. SCTP only supports unicast.
SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit errors and misdelivery of packets to an unintended endpoint. This is stronger than the 16-bit checksums used by TCP or UDP. However, 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 efficiency. For example, this bundling may be done by delaying user messages at the sender, similar to Nagle’s algorithm used by TCP. User messages which would result in IP packets larger than the MTU will be fragmented at the sender and reassembled at the receiver. 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 in [RFC4820] are supported.
[RFC4960] specifies TCP-friendly congestion control to protect the network against overload; see Section 3.1 for more. SCTP also uses sliding window flow control to protect receivers against overflow. Similar to TCP, SCTP also supports delaying acknowledgments. [RFC7053] provides a way for the sender of user messages to request the immediate sending of the corresponding acknowledgments.
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. 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 minimizes head of line blocking.
The base protocol defined in [RFC4960] does not allow interleaving of user- messages. Large messages on one stream can therefore block the sending of user messages on other streams. [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. This draft also specifies multiple algorithms for the sender side selection of which streams to send data from, supporting a variety of scheduling algorithms including priority based methods. The stream re- configuration extension defined in [RFC6525] allows streams to be reset during the lifetime of an association and to increase the number of streams, if the number of streams negotiated in the SCTP handshake becomes insufficient.
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 [RFC4960], similar to TCP behaviour. In addition to this reliable transfer, the partial reliability extension [RFC3758] allows a sender to abandon user messages. The application can specify the policy for abandoning user messages. Examples of these policies defined in [RFC3758] and [RFC7496] are:
SCTP supports multi-homing. Each SCTP endpoint 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. [I-D.ietf-tsvwg-sctp-failover] specifies a quicker failover operation reducing the latency of the failover.
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, such as un-ordered delivery or partial reliability. 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 methods for endpoints and middleboxes to provide support NAT 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 within the WebRTC context.
SCTP has a well-defined API, described in the next subsection.
[RFC4960] defines an abstract API for the base protocol. This API describes the following functions callable by the upper layer of SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message, Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status, Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure Threshold, Set Protocol Parameters, and Destroy. The following notifications are provided by the SCTP stack to the upper layer: COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST, COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE.
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, where 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.
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 permit SCTP-specific information to be sent without using ancillary data in the form of additional cmsgs. 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 features provided by SCTP are:
The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF standards track transport protocol. It provides a unidirectional datagram protocol that preserves message boundaries. It provides no 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 {{I-D.ietf-tsvwg- rfc5405bis}}. UDP is widely implemented and widely used by common applications, including DNS.
UDP is a connection-less protocol that maintains message boundaries, with no connection setup or feature negotiation. The protocol uses independent messages, ordinarily called datagrams. Each stream of messages is independently managed, therefore retransmission does not hold back data sent using other logical streams. It provides detection of payload errors and misdelivery of packets to an unintended endpoint, either of which result in discard of received datagrams, with no indication to the user of the service.
It is possible to create IPv4 UDP datagrams with no checksum, and while this is generally discouraged [RFC1122] [I-D.ietf-tsvwg-rfc5405bis], certain special cases permit this use. These datagrams rely on the IPv4 header checksum to protect from misdelivery to an unintended endpoint. IPv6 does not by permit UDP datagrams with no checksum, although in certain cases this rule may be relaxed [RFC6935]. The checksum support considerations for omitting the checksum are defined in [RFC6936].
UDP does not provide reliability and does not provide retransmission. This implies messages may be re-ordered, lost, or duplicated in transit. 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.
Because UDP provides no flow control, a receiving application that is unable to run sufficiently fast, or frequently, may miss messages. The lack of congestion handling implies UDP traffic may experience loss when using an overloaded path, and may cause the loss of messages from other protocols (e.g., TCP) when sharing the same network path.
On transmission, UDP encapsulates each datagram into an IP packet, which may in turn be fragmented by IP. Fragments are reassembled before delivery to the UDP receiver.
Applications that need to provide fragmentation or that have other requirements such as receiver flow control, congestion control, PathMTU discovery/PLPMTUD, support for ECN, etc need these to be provided by protocols operating over UDP [I-D.ietf-tsvwg-rfc5405bis].
[RFC0768] describes basic requirements for an API for UDP. Guidance on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis].
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) that bind 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 [I-D.ietf-tsvwg-rfc5405bis].
The transport features provided by UDP are:
The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an IETF standards track transport protocol. It provides a unidirectional, datagram protocol that preserves message boundaries. IETF guidance on the use of UDP- Lite is provided in [I-D.ietf-tsvwg-rfc5405bis].
Like UDP, UDP-Lite is a connection-less datagram protocol, with no connection setup or feature negotiation. It changes the semantics of the UDP “payload length” field to that of a “checksum coverage length” field, and is identified by a different IP protocol/next-header value. Otherwise, UDP-Lite is semantically identical to UDP. Applications using UDP-Lite therefore cannot make assumptions regarding the correctness of the data received in the insensitive part of the UDP-Lite payload.
In the same way 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 [I-D.ietf-tsvwg-rfc5405bis].
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, and IPv6 multicast, anycast and unicast.
There is no API currently specified in the RFC Series, but guidance on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis].
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 features 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 messages without providing reliability.
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].
DCCP 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 at each endpoint 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 service code [RFC5595], a mechanism that allows 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 maximum packet size. A DCCP interface allows 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. DCCP supports reduced checksum coverage, a partial integrity mechanism similar to UDP-Lite. There is also a Data Checksum option that when enabled, contains a strong CRC, to enable endpoints to detect application data corruption - similar to SCTP.
Receiver flow control is supported, which limits the amount of unacknowledged data that can be outstanding at a given time.
A DCCP protocol instance can be extended [RFC4340] and tuned using additional features. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only; and some are explicitly negotiated during connection setup.
DCCP service is unicast-only.
It supports negotiation of the congestion control profile, to provide plug- and-play congestion control mechanisms. Examples of specified profiles include “TCP-like” [RFC4341], “TCP-friendly” [RFC4342], and “TCP-friendly for small packets” [RFC5622]. Additional mechanisms are recorded in an IANA registry.
DCCP uses a Connect packet to initiate a session, and permits half-connections that allow each client to choose the 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 middleboxes 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 DCCP 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 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. - Slow Receiver flow control at a receiver. - ability to detect a slow receiver at the sender.
There is no API currently specified in the RFC Series.
The transport features provided by DCCP are:
The Internet Control Message Protocol (ICMP) [RFC0792] for IPv4 and [RFC4433] for IPv6 are IETF standards track protocols.
ICMP is a connection-less unidirectional protocol that delivers individual messages, without error correction, congestion control, or flow control. Messages may be sent as unicast, IPv4 broadcast or multicast datagrams (IPv4 and IPv6), in addition to anycast datagrams.
ICMP is a connection-less unidirectional protocol that delivers individual messages. The protocol uses independent messages, ordinarily called datagrams. Each message is required to carry a checksum as an integrity check and to protect from misdelivery to an unintended endpoint.
ICMP messages typically relay diagnostic information from an endpoint [RFC1122] or network device [RFC1716] addressed to the sender of a flow. This usually contains the network protocol header of a packet that encountered a reported issue. Some formats of messages can also carry other payload data. Each message carries an integrity check calculated in the same way as for UDP, this checksum is not optional.
The RFC series defines additional IPv6 message formats to support a range of uses. In the case of IPv6 the protocol incorporates neighbor discovery [RFC2461] [RFC3971]} (provided by ARP for IPv4) and the Multicast Listener Discovery (MLD) [RFC2710] group management functions (provided by IGMP for IPv4).
Reliable transmission can not be assumed. A receiving application that is unable to run sufficiently fast, or frequently, may miss messages since there is no flow or congestion control. In addition some network devices rate-limit ICMP messages.
Transport Protocols and upper layer protocols can use received ICMP messages to help them take appropriate decisions when network or endpoint errors are reported. For example to implement, ICMP-based Path MTU discovery [RFC1191][RFC1981] or assist in Packetization Layer Path MTU Discovery (PMTUD) [RFC4821]. Such reactions to received messages need to protects from off-path data injection [I-D.ietf-tsvwg-rfc5405bis], avoiding an application receiving packets that were created by an unauthorized third party. An application therefore needs to ensure that all messages are appropriately validated, by checking the payload of the messages to ensure these are received in response to actually transmitted traffic (e.g., a reported error condition that corresponds to a UDP datagram or TCP segment was actually sent by the application). This requires context [RFC6056], such as local state about communication instances to each destination (e.g., in the TCP, DCCP, or SCTP protocols). This state is not always maintained by UDP-based applications [I-D.ietf-tsvwg-rfc5405bis].
Any response to ICMP error messages ought to be robust to temporary routing failures (sometimes called “soft errors”), e.g., transient ICMP “unreachable” messages ought to not normally cause a communication abort [RFC5461] [I-D.ietf-tsvwg-rfc5405bis].
ICMP processing is integrated into many connection-oriented transports, but like other functions needs to be provided by an upper-layer protocol when using UDP and UDP-Lite. On some stacks, a bound socket also allows a UDP application to be notified when ICMP error messages are received for its transmissions [I-D.ietf-tsvwg-rfc5405bis].
The transport features provided by ICMP 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, or DCCP.
The RTP standard [RFC3550] defines a pair of protocols, RTP and the Real Time Control Protocol, RTCP. The transport does not provide connection setup, instead relying on out-of-band techniques or associated control protocols to setup, negotiate parameters or tear down a session.
An RTP sender encapsulates audio/video data into RTP packets to transport media streams. The RFC-series specifies RTP media formats allow packets to carry a wide range of media, and specifies a wide range of multiplexing, error control and other support mechanisms.
If a frame of media data is large, it will be fragmented into several RTP packets. Likewise, several small frames may be bundled into a single RTP packet. RTP may run over a congestion-controlled or non-congestion-controlled transport protocol.
An RTP receiver collects RTP packets from network, validates them for correctness, and sends them to the media decoder input-queue. Missing packet detection is performed by the channel decoder. The play-out buffer is ordered by time stamp and is used to reorder packets. Damaged frames may be repaired before the media payloads are decompressed to display or store the data.
RTCP is a control protocol that works alongside a RTP flow. Both the RTP sender and receiver can send RTCP report packets. This is used to periodically send control information and report performance. Based on received RTCP feedback, an RTP sender can adjust the transmission, e.g., perform rate adaptation at the application layer in the case of congestion.
An RTCP receiver report (RTCP RR) is returned to the sender periodically to report key parameters (e.g, the fraction of packets lost in the last reporting interval, the cumulative number of packets lost, the highest sequence number received, and the inter-arrival jitter). The RTCP RR packets also contain timing information that allows the sender to estimate the network round trip time (RTT) to the receivers.
The interval between reports sent from each receiver tends to be on the order of a few seconds on average, although this varies with the session rate, and sub-second reporting intervals are possible for high rate sessions. The interval is randomized to avoid synchronization of reports from multiple receivers.
There is no standard application programming interface defined for RTP or RTCP. Implementations are typically tightly integrated with a particular application, and closely follow the principles of application level framing and integrated layer processing [ClarkArch] in media processing [RFC2736], error recovery and concealment, rate adaptation, and security [RFC7202]. Accordingly, RTP implementations tend to be targeted at particular application domains (e.g., voice-over-IP, IPTV, or video conferencing), with a feature set optimised for that domain, rather than being general purpose implementations of the protocol.
The transport features provided by RTP are:
FLUTE/ALC is an IETF standards track protocol specified in [RFC6726] and [RFC5775]. Asynchronous Layer Coding (ALC) provides an underlying reliable transport service and FLUTE a file-oriented specialization of the ALC service (e.g., to carry associated metadata). The [RFC6726] and [RFC5775] protocols are non-backward-compatible updates of the [RFC3926] and [RFC3450] experimental protocols; these experimental protocols are currently largely deployed in the 3GPP Multimedia Broadcast and Multicast Services (MBMS) (see [MBMS], section 7) and similar contexts (e.g., the Japanese ISDB-Tmm standard).
The FLUTE/ALC protocol has been designed to support massively scalable reliable bulk data dissemination to receiver groups of arbitrary size using IP Multicast over any type of delivery network, including unidirectional networks (e.g., broadcast wireless channels). However, the FLUTE/ALC protocol also supports point-to-point unicast transmissions.
FLUTE/ALC bulk data dissemination has been designed for discrete file or memory-based “objects”. Transmissions happen either in push mode, where content is sent once, or in on-demand mode, where content is continuously sent during periods of time that can largely exceed the average time required to download the session objects (see [RFC5651], section 4.2).
Although FLUTE/ALC is not well adapted to byte- and message-streaming, there is an exception: FLUTE/ALC is used to carry 3GPP Dynamic Adaptive Streaming over HTTP (DASH) when scalability is a requirement (see [MBMS], section 5.6). In that case, each Audio/Video segment is transmitted as a distinct FLUTE/ALC object in push mode. FLUTE/ALC uses packet erasure coding (also known as Application-Level Forward Erasure Correction, or AL-FEC) in a proactive way. The goal of using AL-FEC is both to increase the robustness in front of packet erasures and to improve the efficiency of the on-demand service. FLUTE/ALC transmissions can be governed by a congestion control mechanism such as the “Wave and Equation Based Rate Control” (WEBRC) [RFC3738] when FLUTE/ALC is used in a layered transmission manner, with several session channels over which ALC packets are sent. However many FLUTE/ALC deployments target pre-provisioned networks and involve only Constant Bit Rate (CBR) channels with no competing flows, for which a sender-based rate control mechanism is sufficient. In any case, FLUTE/ALC’s reliability, delivery mode, congestion control, and flow/rate control mechanisms are distinct components that can be separately controlled to meet different application needs. Section 4.1 of [I-D.ietf-tsvwg-rfc5405bis] describes multicast congestion control requirements for UDP.
The FLUTE/ALC protocol works on top of UDP (though it could work on top of any datagram delivery transport protocol), without requiring any connectivity from receivers to the sender. Purely unidirectional networks are therefore supported by FLUTE/ALC. This guarantees scalability to an unlimited number of receivers in a session, since the sender behaves exactly the same regardless of the number of receivers.
FLUTE/ALC supports the transfer of bulk objects such as file or in- memory content, using either a push or an on-demand mode. in push mode, content is sent once to the receivers, while in on-demand mode, content is sent continuously during periods of time that can greatly exceed the average time required to download the session objects.
This enables receivers to join a session asynchronously, at their own discretion, receive the content and leave the session. In this case, data content is typically sent continuously, in loops (also known as “carousels”). FLUTE/ALC also supports the transfer of an object stream, with loose real-time constraints. This is particularly useful to carry 3GPP DASH when scalability is a requirement and unicast transmissions over HTTP cannot be used ([MBMS], section 5.6). In this case, packets are sent in sequence using push mode. FLUTE/ALC is not well adapted to byte- and message-streaming and other solutions could be preferred (e.g., FECFRAME [RFC6363] with real-time flows).
The FLUTE file delivery instantiation of ALC provides a metadata delivery service. Each object of the FLUTE/ALC session is described in a dedicated entry of a File Delivery Table (FDT), using an XML format (see [RFC6726], section 3.2). This metadata can include, but is not restricted to, a URI attribute (to identify and locate the object), a media type attribute, a size attribute, an encoding attribute, or a message digest attribute. Since the set of objects sent within a session can be dynamic, with new objects being added and old ones removed, several instances of the FDT can be sent and a mechanism is provided to identify a new FDT Instance.
To provide robustness against packet loss and improve the efficiency of the on-demand mode, FLUTE/ALC relies on packet erasure coding (AL-FEC). AL-FEC encoding is proactive (since there is no feedback and therefore no (N)ACK-based retransmission) and ALC packets containing repair data are sent along with ALC packets containing source data. Several FEC Schemes have been standardized; FLUTE/ALC does not mandate the use of any particular one. Several strategies concerning the transmission order of ALC source and repair packets are possible, in particular in on-demand mode where it can deeply impact the service provided (e.g., to favor the recovery of objects in sequence, or at the other extreme, to favor the recovery of all objects in parallel), and FLUTE/ALC does not mandate nor recommend the use of any particular one.
A FLUTE/ALC session is composed of one or more channels, associated to different destination unicast and/or multicast IP addresses. ALC packets are sent in those channels at a certain transmission rate, with a rate that often differs depending on the channel. FLUTE/ALC does not mandate nor recommend any strategy to select which ALC packet to send on which channel. FLUTE/ALC can use a multiple rate congestion control building block (e.g., WEBRC) to provide congestion control that is feedback free, where receivers adjust their reception rates individually by joining and leaving channels associated with the session. To that purpose, the ALC header provides a specific field to carry congestion control specific information. However FLUTE/ALC does not mandate the use of a particular congestion control mechanism although WEBRC is mandatory to support for the Internet ([RFC6726], section 1.1.4). FLUTE/ALC is often used over a network path with pre-provisioned capacity [I-D.ietf-tsvwg-rfc5405bis] where there are no flows competing for capacity. In this case, a sender-based rate control mechanism and a single channel is sufficient.
[RFC6584] provides per-packet authentication, integrity, and anti-replay protection in the context of the ALC and NORM protocols. Several mechanisms are proposed that seamlessly integrate into these protocols using the ALC and NORM header extension mechanisms.
The FLUTE/ALC specification does not describe a specific application programming interface (API) to control protocol operation.
Open source reference implementations of FLUTE/ALC are available at http://planete-bcast.inrialpes.fr/ (no longer maintained) and http://mad.cs.tut.fi/ (no longer maintained), and these implementations specify and document their own APIs. Commercial versions are also available, some derived from the above implementations, with their own API.
The transport features provided by FLUTE/ALC are:
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. 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 acknowledgments. The packet erasure coding can also be proactively applied for forward protection from packet loss. NORM transmissions are governed by the 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.
The NORM protocol is encapsulated in UDP datagrams and thus provides multiplexing for multiple sockets on hosts using port numbers. For 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 features provided by NORM are:
Transport Layer Security (TLS) and Datagram TLS (DTLS) are IETF protocols that provide several security-related features to applications. TLS is designed to run on top of a reliable streaming transport protocol (usually TCP), while DTLS is designed to run on top of a best-effort datagram protocol (UDP or DCCP [RFC5238]). At the time of writing, the current version of TLS is 1.2; which is defined in [RFC5246]. DTLS provides nearly identical functionality to applications; it is defined in [RFC6347] and its current version is also 1.2. The TLS protocol evolved from the Secure Sockets Layer (SSL) protocols developed in the mid 90s to support protection of HTTP traffic.
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.
Both TLS and DTLS provide the same security features and can thus be discussed together. The features they provide are:
The authentication of the peer entity can be omitted; a common web use case is where the server is authenticated and the client is not. TLS also provides a completely anonymous operation mode in which neither peer’s identity is authenticated. It is important to note that TLS itself does not specify how a peering entity’s identity should be interpreted. For 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 authorization decisions. Perfect forward secrecy, if enabled and supported by the selected algorithms, ensures that traffic encrypted and captured during a session at time t0 cannot be later decrypted at time t1 (t1 > t0), even if the long-term secrets of the communicating peers are later compromised.
As DTLS is generally used over an unreliable datagram transport such as UDP, applications will need to tolerate lost, re-ordered, or duplicated datagrams. Like TLS, DTLS conveys application data in a sequence of independent records. However, because records are mapped to unreliable datagrams, there are several features unique to DTLS that are not applicable to TLS:
Generally, DTLS follows the TLS design as closely as possible. To operate over datagrams, DTLS includes a sequence number and limited forms of retransmission and fragmentation for its internal operations. The sequence number may be used for detecting replayed information, according to the windowing procedure described in Section 4.1.2.6 of [RFC6347]. DTLS forbids the use of stream ciphers, which are essentially incompatible when operating on independent encrypted records.
TLS is commonly invoked using an API provided by packages such as OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the manipulation of several important abstractions, which fall into the following categories: long-term keys and algorithms, session state, and communications/connections. There may also be special APIs required to deal with time and/or random numbers, both of which are needed by a variety of encryption algorithms and protocols.
Considerable care is required in the use of TLS APIs to ensure creation of a secure application. The programmer should have at least a basic understanding of encryption and digital signature algorithms and their strengths, public key infrastructure (including X.509 certificates and certificate revocation), and the sockets API. See [RFC7525] and [RFC7457], as mentioned above.
As an example, in the case of OpenSSL, the primary abstractions are the library itself and method (protocol), session, context, cipher and connection. After initializing the library and setting the method, a cipher suite is chosen and used to configure a context object. Session objects may then be minted according to the parameters present in a context object and associated with individual connections. Depending on how precisely the programmer wishes to select different algorithmic or protocol options, various levels of details may be required.
Both TLS and DTLS employ a layered architecture. The lower layer is commonly called the record protocol. It is responsible for:
DTLS augments the TLS record protocol with:
Several protocols are layered on top of the record protocol. These include the handshake, alert, and change cipher spec protocols. There is also the data protocol, used to carry application traffic. The handshake protocol is used to establish cryptographic and compression parameters when a connection is first set up. In DTLS, this protocol also has a basic fragmentation and retransmission capability and a cookie-like mechanism to resist DoS attacks. (TLS compression is not recommended at present). The alert protocol is used to inform the peer of various conditions, most of which are terminal for the connection. The change cipher spec protocol is used to synchronize changes in cryptographic parameters for each peer.
The data protocol, when used with an appropriate cipher, provides:
The 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]. HTTP is usually transported over TCP using port 80 and 443, although it can be used with other transports. When used over TCP it inherits its properties.
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 makes it work over many types of infrastructure, and in cases where a application server often needs to support HTTP anyway.
Depending on application need, 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] addresses this issue 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.
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 using 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 transport connection. This reduces connection establishment overhead and the effect of the transport layer slow-start on each transaction, important in reducing latency 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). The TLS Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used for HTTP version negotiation within the TLS handshake, which eliminates the latency of addition round-trips. 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 an 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 works 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 features provided by HTTP, when used as a pseudotransport, are:
HTTPS (HTTP over TLS) additionally provides the following components:
The tables below summarize some key features to illustrate the range of functions provided across the IETF-specified transports. Figure 1 considers transports that may be directly layered over the network, and Figure 2 considers transports layered over another transport service.
+---------------+------+------+------+------+------+------+------+ | Feature | TCP | MPTCP| SCTP | UDP | UDP-L|DCCP |ICMP | +---------------+------+------+------+------+------+------+------+ | Datagram | No | No | Yes | Yes | Yes | Yes | Yes | +---------------+------+------+------+------+------+------+------+ | Conn. Oriented| Yes | Yes | Yes | No | No | Yes | No | +---------------+------+------+------+------+------+------+------+ | Reliability | Yes | Yes | Yes | No | No | No | No | +---------------+------+------+------+------+------+------+------+ | Partial Rel. | No | No | Pos | N/A | N/A | Yes | N/A | +---------------+------+------+------+------+------+------+------+ | Corupt. Tol | No | No | No | No | Yes | Yes | No | +---------------+------+------+------+------+------+------+------+ | Cong.Control | Yes | Yes | Yes | No | No | Yes | No | +---------------+------+------+------+------+------+------+------+ | Endpoint | 1 | >=1 | >=1 | 1 | 1 | 1 | 1 | +---------------+------+------+------+------+------+------+------+ | Multicast Cap.| No | No | No | Yes | Yes | No | No | +---------------+------+------+------+------+------+------+------+
Figure 1: Summary comparison: Transport protocols
+---------------+------+------+------+------+------+ | Feature | RTP | FLUTE| NORM |(D)TLS| HTTP | +---------------+------+------+------+------+------+ | Datagram | Yes | No | Both | Both | No | +---------------+------+------+------+------+------+ | Conn. Oriented| No | Yes | Yes | Yes | Yes | +---------------+------+------+------+------+------+ | Reliability | No | Yes | Pos | Pos | Yes | +---------------+------+------+------+------+------+ | Partial R | Pos | No | Pos | No | No | +---------------+------+------+------+------+------+ | Corupt. Tol | Poss | No | No | No | No | +---------------+------+------+------+------+------+ | Cong.Control | Poss | Poss | Poss | N/A | N/A | +---------------+------+------+------+------+------+ | Endpoint | >=1 | >=1 | >=1 | 1 | 1 | +---------------+------+------+------+------+------+ | Multicast Cap.| Yes | Yes | Yes | No | No | +---------------+------+------+------+------+------+
Figure 2: Upper layer transports and frameworks
The transport protocol components analyzed in this document that can be used as a basis for defining common transport service features, normalized and separated into categories, are as follows:
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.
In addition to the editors, this document is the work of Brian Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera, Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent Roca, and Michael Tuexen.
Thanks to Joe Touch, Michael Welzl, and the TAPS Working Group for the comments, feedback, and discussion. This work is partially supported by the European Commission under grant agreements FP7-ICT-318627 mPlane and from the Horizon 2020 research and innovation program under grant agreement No. 644334 (NEAT); support does not imply endorsement.