| DiffServ Applied to Real-time Transports | D. Black, Ed. |
| Internet-Draft | EMC |
| Intended status: Informational | P. Jones |
| Expires: February 1, 2015 | Cisco |
| July 31, 2014 |
Differentiated Services (DiffServ) and Real-time Communication
draft-ietf-dart-dscp-rtp-00
This document describes the interaction between Differentiated Services (DiffServ) network quality of service (QoS) functionality and real-time network communication, including communication based on the Real-time Transport Protocol (RTP). DiffServ is based on network nodes applying different forwarding treatments to packets whose IP headers are marked with different DiffServ Code Points (DSCPs). As a result, use of different DSCPs within a single traffic stream may cause transport protocol interactions (e.g., reordering). In addition, DSCP markings may be changed or removed between the traffic source and destination. This document covers the implications of these DiffServ aspects for real-time network communication, including RTCWEB.
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This document describes the interactions between Differentiated Services (DiffServ) network quality of service (QoS) functionality [RFC2475] and real-time network communication, including communication based on the Real-time Transport Protocol (RTP) [RFC3550]. DiffServ is based on network nodes applying different forwarding treatments to packets whose IP headers are marked with different DiffServ Code Points (DSCPs)[RFC2474]. As a result use of different DSCPs within a single traffic stream may cause transport protocol interactions (e.g., reordering). In addition, DSCP markings may be changed or removed between the traffic's source and destination. This document covers the implications of these DiffServ aspects for real-time network communication, including RTCWEB traffic [I-D.ietf-rtcweb-overview].
[Editor's Note: Current section structure draft skips around topics somewhat. The editor suggestsrestructuring to put real-time/RTP material first (new section 2, consisting of current sections 2, 2.1 and 3), then DiffServ Background (new section 3, consisting of current sections 2.2, 2.3, 2.6 and 2.7, followed by discussion of interactions (new section 4, consisting of current sections 2.4, 2.5 and 5) and guidelines (current section 4, renumbered to new section 5).]
Real-time communications enables communication in real-time over an IP network using voice, video, text, content sharing, etc. It is possible to use one or more of these modalities in parallel in order to provide a richer communication experience.
A simple example of real-time communications is a voice call placed over the Internet wherein an audio stream is transmitted in each direction between two users. A more complex example is an immersive videoconferencing system that has multiple video screens, multiple cameras, multiple microphones, and some means of sharing content. For such complex systems, there may be multiple media streams that may be transmitted via a single IP address and port or via multiple IP addresses and ports.
The most common protocol used for real time media is the Real-Time Transport Protocol (RTP) [RFC3550]. RTP defines a common encapsulation format and handling rules for real-time data transmitted over the Internet. Unfortunately, RTP terminology usage has been inconsistent. For example, this document on RTP grouping terminology [I-D.ietf-avtext-rtp-grouping-taxonomy] observes that:
Terminology in this document is based on that RTP grouping terminology document with the following terms being of particular importance (see that terminology document for full definitions):
Media encoding and packetization of a source stream results in a source RTP stream plus zero or more redundancy RTP streams that provide resilience against loss of packets from the source RTP stream [I-D.ietf-avtext-rtp-grouping-taxonomy]. Redundancy information may also be carried in the same RTP stream as the encoded source stream, e.g., see Section 7.2 of [RFC5109]. With most applications, a single media type (e.g., audio) is transmitted within a single RTP session. However, it is possible to transmit multiple, distinct source streams over the same RTP session as one or more individual RTP streams. This is referred to as RTP multiplexing.
The number of source streams and RTP streams in an overall real-time interaction can be surprisingly large. In addition to a voice source stream and a video source stream, there could be separate source streams for each of the cameras or microphones on a videoconferencing system. As noted above, there might also be separate redundancy RTP streams that provide protection to a source RTP stream, using techniques such as Forward Error Correction. Another example is simulcast transmission, where a video source stream can be transmitted as high resolution and low resolution RTP streams at the same time. In this case, a media processing function might choose to send one or both RTP streams onward to a receiver based on bandwidth availability or who the active speaker is in a multipoint conference. Lastly, a transmitter might send a the same media content concurrently as two RTP streams using different encodings (e.g., VP8 in parallel with H.264) to allow a media processing function to select a media encoding that best matches the capabilities of the receiver.
For the RTCWEB protocol suite [I-D.ietf-rtcweb-transports], an individual source stream is a MediaStreamTrack, and a MediaStream contains one or more MediaStreamTracks [W3C.WD-mediacapture-streams-20130903]. A MediaStreamTrack is transmitted as a source RTP stream plus zero or more redundancy RTP streams, so a MediaStream that consists of one MediaStreamTrack is transmitted as a single source RTP stream plus zero or more redundancy RTP streams. For more information on use of RTP in RTCWEB, see [I-D.ietf-rtcweb-rtp-usage].
Other transport protocols may also be used to transmit real-time data or near-real-time data. For example, SCTP [RFC4960] can be utilized to carry application sharing or whiteboarding information as part of an overall interaction that includes real time media. These additional transport protocols can be multiplexed with an RTP session via UDP encapsulation, thereby using a single pair of UDP ports.
The RTCWEB protocol suite encompasses a number of forms of multiplexing:
The resulting unidirectional UDP packet flow is identified by a 5-tuple, i.e., a combination of two IP addresses (source and destination), two UDP ports (source and destination), and the use of the UDP protocol. SDP bundle negotiation restrictions [I-D.ietf-mmusic-sdp-bundle-negotiation] limit RTCWEB to using at most a single DTLS session per UDP 5-tuple. In contrast, multple SCTP associations can be mulitplexed over a single UDP 5-tuple [RFC6951].
For IPv6, addition of the flow label [RFC6437] to 5-tuples results in 6-tuples, but in practice, use of a flow label is unlikely to result in a finer-grain traffic subset than the corresponding 5-tuple (e.g., the flow label is likely to represent the combination of two ports with use of the UDP protocol). For that reason, discussion in this draft focuses on UDP 5-tuples.
The DiffServ architecture is intended to enable scalable service discrimination in the Internet without requiring each network node to store per-flow state and participate in per-flow signaling. The services may be end-to-end or within a network; they include both those that can satisfy quantitative performance requirements (e.g., peak bandwidth) and those based on relative performance (e.g., "class" differentiation). Services can be constructed by a combination of well-defined building blocks deployed in network nodes that:
A network node that supports DiffServ includes a classifier that selects packets based on the value of the DS field in IP headers, along with buffer management and packet scheduling mechanisms capable of delivering the specific packet forwarding treatment indicated by the DS field value. Setting of the DS field and fine-grain conditioning of marked packets need only be performed at network boundaries; internal network nodes operate on traffic aggregates that share a DS field value, or in some cases, a small set of related values.
The DiffServ architecture[RFC2475] maintains distinctions among:
This document focuses on PHBs and the usage of DSCPs to obtain those behaviors. In a network node's forwarding path, the DSCP is used to map a packet to a particular forwarding treatment, or per-hop behavior (PHB) that specifies the forwarding treatment.
A per-hop behavior (PHB) is a description of the externally observable forwarding behavior of a network node for network traffic marked with a DSCP that selects that PHB. In this context, "forwarding behavior" is a general concept - for example, if only one DSCP is used for all traffic on a link, the observable forwarding behavior (e.g., loss, delay, jitter) will often depend only on the relative loading of the link. To obtain useful behavioral differentiation, multiple traffic subsets are marked with different DSCPs for different PHBs for which node resources such as buffer space and bandwidth are allocated. PHBs provide the framework for a DiffServ network node to allocate resources to traffic subsets, with network-scope differentiated services constructed on top of this basic hop-by-hop (per-node) resource allocation mechanism.
The codepoints (DSCPs) may be chosen from a small set of fixed values (the class selector codepoints), or from a set of recommended values defined in PHB specifications, or from values that have purely local meanings to a specific network that supports DiffServ; in general, packets may be forwarded across multiple such networks between source and destination.
The mandatory DSCPs are the class selector code points as specified in [RFC2474]. The class selector codepoints (CS0-CS7) extend the deprecated concept of IP Precedence in the IPv4 header; three bits are added, so that the class selector DSCPs are of the form 'xxx000'. The all-zero DSCP ('000000' or CS0) designates a Default PHB that provides best-effort forwarding behavior and the remaining class selector code points are intended to provide relatively better per-hop-forwarding behavior in increasing numerical order, but:
Applications and traffic sources cannot rely upon different class selector codepoints providing differentiated services or upon the presence of an LE service that is selected by the CS1 DSCP. There is no effective way for a network endpoint to determine whether the CS1 DSCP selects an LE service on a specific network, let alone end-to-end. Packets marked with the CS1 DSCP may be forwarded with best effort service or another "higher" priority service, see [RFC2474].
Although Differentiated Services is a general architecture that may be used to implement a variety of services, three fundamental forwarding behaviors (PHBs) have been defined and characterized for general use. These are:
[Editor's note: Add a sentence or two on DCCP - it is not necessary to include every known transport protocol.]
Transport protocols provide data communication behaviors beyond those possible at the IP layer. An important example is that TCP [RFC0793] provides reliable in-order delivery of data with congestion control. SCTP [RFC4960] provides additional properties such as preservation of message boundaries, and the ability to avoid head-of-line blocking that may occur with TCP. In contrast, UDP [RFC0768] is a basic unreliable datagram protocol that provides port-based multiplexing and demultiplexing on top of IP.
Transport protocols that provide reliable delivery (e.g., TCP, SCTP) are sensitive to network reordering of traffic. When a protocol that provides reliable delivery receives a packet other than the next expected packet, the protocol usually assumes that the expected packet has been lost and respond with a retransmission request for that packet. In addition, congestion control functionality in transport protocols usually infers congestion when packets are lost, creating an additional sensitivity to significant reordering - such reordering may be (mis-)interpreted as indicating congestion-caused packet loss, causing a reduction in transmission rate. This remains true even when ECN [RFC3168] is in use, as ECN receivers are required to treat missing packets as potential indications of congestion. This requirement is based on two factors:
[RFC2914] for further discussion.
Congestion control is an important aspect of the Internet architecture, see
In general, marking packets with different DSCPs results in different PHBs being applied at network nodes, making reordering possible due to use of different pools of forwarding resources for each PHB. The primary exception is that reordering is prohibited within each AF class (e.g., AF1x), as the three PHBs in an AF class differ solely in drop precedence. Reordering within a PHB or AF class may occur for other transient reasons (e.g., route flap or ECMP rebalancing).
Reordering also affects other forms of congestion control, such as techniques for RTP congestion control that were under development when this document was published, see [I-D.ietf-rmcat-cc-requirements] for requirements. These techniques prefer use of a common (coupled) congestion controller for RTP streams between the same endpoints in order to reduce packet loss and delay by reducing competition for resources at any shared bottleneck.
Shared bottlenecks can be detected via correlations of measured metrics such as one-way delay. An alternative approach assumes that the set of packets on a single 5-tuple marked with DSCPs that do not allow reordering will utilize a common network path and common forwarding resources at each network node. Under that assumption, any bottleneck encountered by such packets is shared among all of them, making it safe to use a common (coupled) congestion controller, see [I-D.welzl-rmcat-coupled-cc]. This is not a safe assumption when the packets involved are marked with DSCP values that allow reordering because a bottleneck may not be shared among all such packets (e.g., if the DSCPs result in use of different queues at a network node, only one of which is a bottleneck).
UDP is the primary transport protocol that is not sensitive to reordering in the network, because it does not provide reliable delivery or congestion control. On the other hand, when UDP is used to encapsulate other protocols (e.g., as is the case for RTCWEB, see Section 2.1), the reordering considerations for the encapsulated protocols apply. For the specific usage of UDP by RTCWEB, every encapsulated protocol (i.e., RTP, SCTP and TCP) is sensitive to reordering as further discussed in this document.
Real-time communications are also sensitive to network reordering of packets. Such reordering may lead to spurious NACK generation and unneeded retransmission, as is the case for reliable delivery protocols (see Section 2.4). The degree of sensitivity depends on protocol or stream timers, in contrast to reliable delivery protocols that usually react to all reordering.
Receiver jitter buffers have important roles in the effect of reordering on real time communications:
Network packet reordering caused by use of different DSCPs has no effective upper bound, and can exceed the size of any reasonable jitter buffer - in practice, the size of jitter buffers for replay is limited by external factors such as the amount of time that a human is willing to wait for replay to start.
Each DiffServ AF class consists of three PHBs that differ solely in drop precedence (e.g., AF3x consists of AF31, AF32 and AF33). Reordering is prohibited among packets on the same 5-tuple that use PHBs within a single AF class; further, these packets can be expected to draw upon the same forwarding resources on network nodes (e.g., use the same router queue) and hence use of multple drop precedences within an AF class is not expected to impact latency.
When PHBs within a single AF class are mixed for a protocol session, the resulting drop likelihood is a mix of the drop likelihoods of the PHBs involved. The primary effect of multiple drop precedences is to influence decisions on what to drop with the goal that less important packets are dropped in preference to more important packets.
There are situations in which drop precedences should not be mixed. A simple example is that there is little value in mixing drop precedences iwthin a TCP connection, because TCP's ordered delivery behavior results in any drop requiring the receiver to wait for the dropped packet to be retransmitted. Any resulting delay depends on the RTT and not the packe that was dropped. Hence a single PHB and DSCP should be used for all packets in a TCP connection.
SCTP [RFC4960] differs from TCP in a number of ways, including the ability to deliver messages in an order that differs from the order in which they were sent and support for unreliable streams. However, SCTP performs congestion control and retransmission across the entire association, and not on a per-stream basis. Although there may be advantages to using multiple drop precedence across SCTP streams or within an SCTP stream that does not use reliable ordered delivery, there is no practical operational experience in doing so (e.g., the SCTP sockets API [RFC6458] does not support use of more than one DSCP for an SCTP association). As a consequence, the impacts on SCTP protocol and implementation behavior are unknown and difficult to predict. Hence a single PHB and DSCP should be used for all packets in an SCTP association, independent of the number or nature of streams in that association.
RTCP multi-stream reporting optimizations for an RTP session [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP streams involved experience the same packet loss behavior. This mechanism is highly inappropriate if the RTP streams involved use different PHBs, even if those PHBs differ solely in drop precedence.
DSCP markings are not end-to-end in general. Each network can make its own decisions about what PHBs to use and which DSCP maps to each PHB. While every PHB specification includes a recommended DSCP, and RFC 4594 [RFC4594] recommends their end-to-end usage, there is no requirement that every network support any PHBs or use any specific DSCPs, with the exception of the class selector codepoint requirements in RFC 2474 [RFC2474]. When DiffServ is used, the edge or boundary nodes of a network are responsible for ensuring that all traffic entering that network conforms to that network's policies for DSCP and PHB usage, and such nodes remark traffic (change the DSCP marking as part of traffic conditioning) accordingly. As a result, DSCP remarking is possible at any network boundary, including the first network node that traffic sent by a host encounters. Remarking is also possible within a network, e.g., for traffic shaping.
DSCP remarking is part of traffic conditioning; the traffic conditioning functionality applied to packets at a network node is determined by a traffic classifier [RFC2475]. Edge nodes of a DiffServ network classify traffic based on selected packet header fields; typical implementations do not look beyond the traffic's 5-tuple in the IP and transport protocol headers. As a result, when multiple DSCPs are used for traffic that shares a 5-tuple, remarking at a network boundary may result in all of the traffic being forwarded with a single DSCP, thereby removing any differentiation within the 5-tuple downstream of the remarking location. Network nodes within a DiffServ network generally classify traffic based solely on DSCPs, but may perform finer grain traffic conditioning similar to that performed by edge nodes.
So, for two arbitrary network endpoints, there can be no assurance that the DSCP set at the source endpoint will be preserved and presented at the destination endpoint. Rather, it is quite likely that the DSCP will be set to zero (e.g., at the boundary of a network operator that distrusts or does not use the DSCP field) or to a value deemed suitable by an ingress classifier for whatever 5-tuple it carries. DiffServ classifiers generally ignore embedded protocol headers (e.g., for SCTP or RTP embedded in UDP, header-based classification is unlikely to look beyond the outer UDP header).
In addition, remarking may remove application-level distinctions in forwarding behavior - e.g., if multiple PHBs within an AF class are used to distinguish different types of frames within a video RTP stream, token-bucket-based remarkers operating in Color-Blind mode (see [RFC2697] and [RFC2698] for examples) may remark solely based on flow rate and burst behavior, removing the drop precedence distinctions specified by the source.
Backbone and other carrier networks may employ a small number of DSCPs (e.g., less than half a dozen) in order to manage a small number of traffic aggregates; hosts that use a larger number of DSCPs can expect to find that much of their intended differentiation is removed by such networks. Better results may be achieved when DSCPs are used to spread traffic among a smaller number of DiffServ-based traffic subsets or aggregates, see [I-D.geib-tsvwg-diffserv-intercon] for one proposal. This is of particular importance for MPLS-based networks due to the limited size of the Traffic Class (TC) field in an MPLS label [RFC5462] that is used to carry DiffServ information and the use of that TC field for other purposes, e.g., ECN [RFC5129]. For further discussion on use of DiffServ with MPLS, see [RFC3270] and [RFC5127].
Section 2 explains how source streams can be multiplexed over RTP sessions which can in turn be multiplexed over UDP with packets generated by other transport protocols. This section provides background on why this level of multiplexing is desirable. The rationale in this section applies both to multiplexing of source streams in RTP sessions and multiplexing of an RTP session with traffic from other transport protocols via UDP encapsulation.
Multiplexing reduces the number of ports utilized for real-time and related communication in an overall interaction. While a single endpoint might have plenty of ports available for communication, this traffic often traverses points in the network that are constrained on the number of available ports. A good example is a Network Address Translator and Firewall (NAT/FW) device sitting at the network edge. As the number of simultaneous protocol sessions increases, so does the burden placed on these devices in order to provide port mapping.
The STUN [RFC5389]/ICE [RFC5245]/TURN [RFC5766]protocol family provides NAT/FW traversal and port mapping for protocols (e.g., those in the RTCWEB protocol suite) via communication with a relay server. These protocols were originally designed for use of UDP, however, they have been extended to use TCP as a transport for situations in which UDP does not work [RFC6062].
When TCP is selected for NAT/FW traversal, a single PHB and DSCP should be used for all traffic on that TCP connection for the reasons discussed in Section 2.4 and Section 2.6 above. An additional reason for this recommendation is that packetization for STUN/ICE/TURN occurs before passing the resulting packets to TCP; TCP resegmentation may result in a different packetization on the wire, breaking any association between DSCPs and specific data to which they are intended to apply.
Another reason for multiplexing is to help reduce the time required to establish bi-directional communication. Since any two communicating users might be situated behind different NAT/FW devices, it is necessary to employ techniques like STUN/ICE/TURN in order to get traffic to flow between the two devices [I-D.ietf-rtcweb-transports]. Performing the tasks required of STUN/ICE/TURN take time, especially when multiple protocol sessions are involved. While tasks for different sessions can be performed in parallel, it is nonetheless necessary for applications to wait for all sessions to be opened before communication between to users can begin. Reducing the number of STUN/ICE/TURN steps reduces the likelihood of loss of a packet for one of these protocols; any such loss adds delay to setting up a communication session. Further, reducing the number of STUN/ICE/TURN tasks places a lower burden on the STUN and TURN servers.
Multiplexing may reduce the complexity and resulting load on an endpoint. A single instance of STUN/ICE/TURN is simpler to execute and manage than multiple instances STUN/ICE/TURN operations happening in parallel, as the latter require synchronization and create more complex failure situations that have to be cleaned up by additional code.
The only use of multiple standardized PHBs and DSCPs that prevents network reordering among packets marked with different DSCPs is use of PHBs within a single AF class. All other uses of multiple PHBs and/or the class selector DSCPs allow network reordering of packets that are marked with different DSCPs. Based on this and the foregoing discussion, the following requirements apply to use of DiffServ with real-time communications - applications and other traffic sources:
There is no requirement in this document for network operators to differentiate traffic in any fashion. Networks may support all of the PHBs discussed herein, classify EF and AFxx traffic identically, or even remark all traffic to best effort at some ingress points. Nonetheless, it is useful for network endpoints to provide finer granularity DSCP marking on packets for the benefit of networks that offer QoS service differentiation. A specific example is that traffic originating from a browser may benefit from QoS service differentiation in within-building and residential access networks, even if the DSCP marking is subsequently removed or simplified. This is because such networks and the boundaries between them are likely traffic bottleneck locations (e.g., due to customer aggregation onto common links and/or speed differences among links used by the same traffic).
[Editor's note: rtcweb-transports draft is not aligned with the above. The rtcweb WG and the draft author will bring it into line.]
For real-time communications, one might want to mark the audio packets using EF and the video packets as AF41. However, in a video conference receiving the audio packets ahead of the video is not useful because lip sync is necessary between audio and video. It may still be desirable to send audio with a PHB that provides better service, because early arrival of audio helps assure smooth audio rendering, which is often more important than fully faithful video rendering. There are also limits, as some devices have difficulties in synchronizing voice and video when packets that need to be rendered together arrive at significantly different times. It makes more sense to use different PHBs when the audio and video source streams do not share a strict timing relationship. For example, video content may be shared within a video conference via playback, perhaps of an unedited video clip that is intended to become part of a television advertisement. Such content sharing video does not need precise synchronization with video conference audio, and could use a different PHB, as content sharing video is more tolerant to jitter, loss, and delay.
Within a layered video RTP stream, ordering of frame communication is preferred, but importance of frame types varies, making use of PHBs with different drop precedences appropriate. For example, I-frames that contain an entire image are usually more important than P-frames that contain only changes from the previous image because loss of a P-frame (or part thereof) can be recovered (at the latest) via the next I-frame, whereas loss of an I-frame (or part thereof) may cause rendering problems for all of the P-frames that depend on the missing I-frame. For this reason, it is appropriate to mark I-frame packets with a PHB that has lower drop precedence than the PHB used for P-frames, as long as the PHBs preserve ordering among frames (e.g., are in an AF class) - AF41 for I-frames and AF43 for P-frames is one possibility. Additional spatial and temporal layers beyond the base video layer could also be marked with higher drop precedence than the base video layer, as their loss reduces video quality, but does not disrupt video rendering.
Additional RTP streams in a real-time communication interaction could be marked with CS0 and carried as best effort traffic. One example is real-time text transmitted as specified in RFC 4103 [RFC4103]. Best effort forwarding suffices because such real-time text has loose timing requirements; RFC 4103 recommends sending text in chunks every 300ms. Such text is technically real-time, but does not need a PHB promising better service than best effort, in contrast to audio or video.
This document includes no request to IANA.
The security considerations for all of the technologies discussed in this document apply; in particular see the security considerations for RTP in [RFC3550] and DiffServ in [RFC2474] and [RFC2475].
Multiplexing of multiple protocols onto a single UDP 5-tuple via encapsulation has implications for network functionality that monitors or inspects individual protocol flows, e.g., firewalls and traffic monitoring systems. When implementations of such functionality lack visibility into encapsulated traffic (likely for many current implementations), it may be difficult or impossible to apply network security policy and associated controls at a finer granularity than the overall UDP 5-tuple.
Use of multiple DSCPs to provide differentiated QoS service may reveal information about the encrypted traffic to which different service levels are provided. For example, DSCP-based identification of RTP streams combined with packet frequency and packet size could reveal the type or nature of the encrypted source streams. The IP header used for forwarding has to be unencrypted for obvious reasons, and the DSCP likewise has to be unencrypted in order to enable different IP forwarding behaviors to be applied to different packets. The nature of encrypted traffic components can be disguised via encrypted dummy data padding and encrypted dummy packets, e.g., see the discussion of traffic flow confidentiality in [RFC4303]. Encrypted dummy packets could even be added in a fashion that an observer of the overall encrypted traffic might mistake for another encrypted RTP stream.
This document is the result of many conversations that have occurred within the dart working group and multiple other working groups in the RAI and Transport areas. Many thanks to Harald Alvestrand, Erin Bournival, Brian Carpenter, Keith Drage, Ruediger Geib, Cullen Jennings, Jonathan Lennox, Karen Nielsen, Colin Perkins, James Polk, Michael Welzl, Dan York and DART WG participants for their reviews and comments.
[Editor's Note: Check which references should be Normative.]
[To be removed before RFC publication.]
Changes from draft-york-dart-dscp-rtp-00 to -01
Changes from draft-york-dart-dscp-rtp-01 to -02
Changes from draft-york-dart-dscp-rtp-02 to draft-ietf-dart-dscp-rtp-00