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Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-08 == Outdated reference: A later version (-17) exists of draft-ietf-rtcweb-transports-12 ** Downref: Normative reference to an Informational RFC: RFC 4594 ** Downref: Normative reference to an Informational RFC: RFC 7657 == Outdated reference: A later version (-09) exists of draft-ietf-rmcat-coupled-cc-02 -- Obsolete informational reference (is this intentional?): RFC 3662 (Obsoleted by RFC 8622) Summary: 2 errors (**), 0 flaws (~~), 4 warnings (==), 2 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group P. Jones 3 Internet-Draft S. Dhesikan 4 Intended status: Standards Track C. Jennings 5 Expires: November 21, 2016 Cisco Systems 6 D. Druta 7 AT&T 8 May 20, 2016 10 DSCP Packet Markings for WebRTC QoS 11 draft-ietf-tsvwg-rtcweb-qos-17 13 Abstract 15 Many networks, such as service provider and enterprise networks, can 16 provide different forwarding treatments for individual packets based 17 on Differentiated Services Code Point (DSCP) values on a per-hop 18 basis. This document provides the recommended DSCP values for web 19 browsers to use for various classes of WebRTC traffic. 21 Status of This Memo 23 This Internet-Draft is submitted in full conformance with the 24 provisions of BCP 78 and BCP 79. 26 Internet-Drafts are working documents of the Internet Engineering 27 Task Force (IETF). Note that other groups may also distribute 28 working documents as Internet-Drafts. The list of current Internet- 29 Drafts is at http://datatracker.ietf.org/drafts/current/. 31 Internet-Drafts are draft documents valid for a maximum of six months 32 and may be updated, replaced, or obsoleted by other documents at any 33 time. It is inappropriate to use Internet-Drafts as reference 34 material or to cite them other than as "work in progress." 36 This Internet-Draft will expire on November 21, 2016. 38 Copyright Notice 40 Copyright (c) 2016 IETF Trust and the persons identified as the 41 document authors. All rights reserved. 43 This document is subject to BCP 78 and the IETF Trust's Legal 44 Provisions Relating to IETF Documents 45 (http://trustee.ietf.org/license-info) in effect on the date of 46 publication of this document. Please review these documents 47 carefully, as they describe your rights and restrictions with respect 48 to this document. Code Components extracted from this document must 49 include Simplified BSD License text as described in Section 4.e of 50 the Trust Legal Provisions and are provided without warranty as 51 described in the Simplified BSD License. 53 Table of Contents 55 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 56 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 57 3. Relation to Other Specifications . . . . . . . . . . . . . . 3 58 4. Inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 59 5. DSCP Mappings . . . . . . . . . . . . . . . . . . . . . . . . 5 60 6. Security Considerations . . . . . . . . . . . . . . . . . . . 8 61 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 62 8. Downward References . . . . . . . . . . . . . . . . . . . . . 8 63 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 64 10. Dedication . . . . . . . . . . . . . . . . . . . . . . . . . 8 65 11. Document History . . . . . . . . . . . . . . . . . . . . . . 9 66 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 9 67 12.1. Normative References . . . . . . . . . . . . . . . . . . 9 68 12.2. Informative References . . . . . . . . . . . . . . . . . 10 69 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11 71 1. Introduction 73 Differentiated Services Code Point (DSCP) [RFC2474] packet marking 74 can help provide QoS in some environments. This specification 75 provides default packet marking for browsers that support WebRTC 76 applications, but does not change any advice or requirements in other 77 IETF RFCs. The contents of this specification are intended to be a 78 simple set of implementation recommendations based on the previous 79 RFCs. 81 Networks where these DSCP markings are beneficial (likely to improve 82 QoS for WebRTC traffic) include: 84 1. Private, wide-area networks. Network administrators have control 85 over remarking packets and treatment of packets. 87 2. Residential Networks. If the congested link is the broadband 88 uplink in a cable or DSL scenario, often residential routers/NAT 89 support preferential treatment based on DSCP. 91 3. Wireless Networks. If the congested link is a local wireless 92 network, marking may help. 94 There are cases where these DSCP markings do not help, but, aside 95 from possible priority inversion for "less than best effort traffic" 96 (see Section 5), they seldom make things worse if packets are marked 97 appropriately. 99 DSCP values are in principle site specific, with each site selecting 100 its own code points for controlling per-hop-behavior to influence the 101 QoS for transport-layer flows. However in the WebRTC use cases, the 102 browsers need to set them to something when there is no site specific 103 information. This document describes a subset of DSCP code point 104 values drawn from existing RFCs and common usage for use with WebRTC 105 applications. These code points are intended to be the default 106 values used by a WebRTC application. While other values could be 107 used, using a non-default value may result in unexpected per-hop 108 behavior. It is RECOMMENDED that WebRTC applications use non-default 109 values only in private networks that are configured to use different 110 values. 112 This specification defines inputs that are provided by the WebRTC 113 application hosted in the browser that aid the browser in determining 114 how to set the various packet markings. The specification also 115 defines the mapping from abstract QoS policies (flow type, priority 116 level) to those packet markings. 118 2. Terminology 120 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 121 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 122 document are to be interpreted as described in [RFC2119]. 124 The terms "browser" and "non-browser" are defined in [RFC7742] and 125 carry the same meaning in this document. 127 3. Relation to Other Specifications 129 This document is a complement to [RFC7657], which describes the 130 interaction between DSCP and real-time communications. That RFC 131 covers the implications of using various DSCP values, particularly 132 focusing on Real-time Transport Protocol (RTP) [RFC3550] streams that 133 are multiplexed onto a single transport-layer flow. 135 There are a number of guidelines specified in [RFC7657] that apply to 136 marking traffic sent by WebRTC applications, as it is common for 137 multiple RTP streams to be multiplexed on the same transport-layer 138 flow. Generally, the RTP streams would be marked with a value as 139 appropriate from Table 1. A WebRTC application might also multiplex 140 data channel [I-D.ietf-rtcweb-data-channel] traffic over the same 141 5-tuple as RTP streams, which would also be marked as per that table. 142 The guidance in [RFC7657] says that all data channel traffic would be 143 marked with a single value that is typically different than the 144 value(s) used for RTP streams multiplexed with the data channel 145 traffic over the same 5-tuple, assuming RTP streams are marked with a 146 value other than default forwarding (DF). This is expanded upon 147 further in the next section. 149 This specification does not change or override the advice in any 150 other IETF RFCs about setting packet markings. Rather, it simply 151 selects a subset of DSCP values that is relevant in the WebRTC 152 context. 154 The DSCP value set by the endpoint is not trusted by the network. In 155 addition, the DSCP value may be remarked at any place in the network 156 for a variety of reasons to any other DSCP value, including default 157 forwarding (DF) value to provide basic best effort service. Even so, 158 there is benefit in marking traffic even if it only benefits the 159 first few hops. The implications are discussed in Secton 3.2 of 160 [RFC7657]. Further, a mitigation for such action is through an 161 authorization mechanism. Such an authorization mechanism is outside 162 the scope of this document. 164 4. Inputs 166 WebRTC applications send and receive two types of flows of 167 significance to this document: 169 o media flows which are RTP streams [I-D.ietf-rtcweb-rtp-usage] 171 o data flows which are data channels [I-D.ietf-rtcweb-data-channel] 173 Each of the RTP streams and distinct data channels consists of all of 174 the packets associated with an independent media entity, so an RTP 175 stream or distinct data channel is not always equivalent to a 176 transport-layer flow defined by a 5-tuple (source address, 177 destination address, source port, destination port, and protocol). 178 There may be multiple RTP streams and data channels multiplexed over 179 the same 5-tuple, with each having a different level of importance to 180 the application and, therefore, potentially marked using different 181 DSCP values than another RTP stream or data channel within the same 182 transport-layer flow. (Note that there are restrictions with respect 183 to marking different data channels carried within the same SCTP 184 association as outlined in Section 5.) 186 The following are the inputs provided by the WebRTC application to 187 the browser: 189 o Flow Type: The application provides this input because it knows if 190 the flow is audio, interactive video [RFC4594] [G.1010] with or 191 without audio, or data. 193 o Application Priority: Another input is the relative importance of 194 an RTP stream or data channel. Many applications have multiple 195 flows of the same Flow Type and often some flows are more 196 important than others. For example, in a video conference where 197 there are usually audio and video flows, the audio flow may be 198 more important than the video flow. JavaScript applications can 199 tell the browser whether a particular flow is high, medium, low or 200 very low importance to the application. 202 [I-D.ietf-rtcweb-transports] defines in more detail what an 203 individual flow is within the WebRTC context and priorities for media 204 and data flows. 206 Currently in WebRTC, media sent over RTP is assumed to be interactive 207 [I-D.ietf-rtcweb-transports] and browser APIs do not exist to allow 208 an application to to differentiate between interactive and non- 209 interactive video. 211 5. DSCP Mappings 213 The DSCP values for each flow type of interest to WebRTC based on 214 application priority are shown in the following table. These values 215 are based on the framework and recommended values in [RFC4594]. A 216 web browser SHOULD use these values to mark the appropriate media 217 packets. More information on EF can be found in [RFC3246]. More 218 information on AF can be found in [RFC2597]. DF is default 219 forwarding which provides the basic best effort service [RFC2474]. 221 +------------------------+-------+------+-------------+-------------+ 222 | Flow Type | Very | Low | Medium | High | 223 | | Low | | | | 224 +------------------------+-------+------+-------------+-------------+ 225 | Audio | CS1 | DF | EF (46) | EF (46) | 226 | | (8) | (0) | | | 227 | | | | | | 228 | Interactive Video with | CS1 | DF | AF42, AF43 | AF41, AF42 | 229 | or without Audio | (8) | (0) | (36, 38) | (34, 36) | 230 | | | | | | 231 | Non-Interactive Video | CS1 | DF | AF32, AF33 | AF31, AF32 | 232 | with or without Audio | (8) | (0) | (28, 30) | (26, 28) | 233 | | | | | | 234 | Data | CS1 | DF | AF11 | AF21 | 235 | | (8) | (0) | | | 236 +------------------------+-------+------+-------------+-------------+ 238 Table 1: Recommended DSCP Values for WebRTC Applications 240 The application priority, indicated by the columns "very low", "low", 241 "Medium", and "high", signifies the relative importance of the flow 242 within the application. It is an input that the browser receives to 243 assist in selecting the DSCP value and adjusting the network 244 transport behavior. 246 The above table assumes that packets marked with CS1 are treated as 247 "less than best effort", such as the LE behavior described in 248 [RFC3662]. However, the treatment of CS1 is implementation 249 dependent. If an implementation treats CS1 as other than "less than 250 best effort", then the actual priority (or, more precisely, the per- 251 hop-behavior) of the packets may be changed from what is intended. 252 It is common for CS1 to be treated the same as DF, so applications 253 and browsers using CS1 cannot assume that CS1 will be treated 254 differently than DF [RFC7657]. However, it is also possible per 255 [RFC2474] for CS1 traffic to be given better treatment than DF, thus 256 caution should be exercised when electing to use CS1. This is one of 257 the cases where marking packets using these recommendations can make 258 things worse. 260 Implementers should also note that excess EF traffic is dropped. 261 This could mean that a packet marked as EF may not get through, 262 although the same packet marked with a different DSCP value would 263 have gotten through. This is not a flaw, but how excess EF traffic 264 is intended to be treated. 266 The browser SHOULD first select the flow type of the flow. Within 267 the flow type, the relative importance of the flow SHOULD be used to 268 select the appropriate DSCP value. 270 Currently, all WebRTC video is assumed to be interactive 271 [I-D.ietf-rtcweb-transports], for which the Interactive Video DSCP 272 values in Table 1 SHOULD be used. Browsers MUST NOT use the AF3x 273 DSCP values (for Non-Interactive Video in Table 1) for WebRTC 274 applications. Non-browser implementations of WebRTC MAY use the AF3x 275 DSCP values for video that is known not to be interactive, e.g., all 276 video in a WebRTC video playback application that is not implemented 277 in a browser. 279 The combination of flow type and application priority provides 280 specificity and helps in selecting the right DSCP value for the flow. 281 All packets within a flow SHOULD have the same application priority. 282 In some cases, the selected application priority cell may have 283 multiple DSCP values, such as AF41 and AF42. These offer different 284 drop precedences. The different drop precedence values provides 285 additional granularity in classifying packets within a flow. For 286 example, in a video conference the video flow may have medium 287 application priority, thus either AF42 or AF43 may be selected. More 288 important video packets (e.g., a video picture or frame encoded 289 without any dependency on any prior pictures or frames) might be 290 marked with AF42 and less important packets (e.g., a video picture or 291 frame encoded based on the content of one or more prior pictures or 292 frames) might be marked with AF43 (e.g., receipt of the more 293 important packets enables a video renderer to continue after one or 294 more packets are lost). 296 It is worth noting that the application priority is utilized by the 297 coupled congestion control mechanism for media flows per 298 [I-D.ietf-rmcat-coupled-cc] and the SCTP scheduler for data channel 299 traffic per [I-D.ietf-rtcweb-data-channel]. 301 For reasons discussed in Section 6 of [RFC7657], if multiple flows 302 are multiplexed using a reliable transport (e.g., TCP) then all of 303 the packets for all flows multiplexed over that transport-layer flow 304 MUST be marked using the same DSCP value. Likewise, all WebRTC data 305 channel packets transmitted over an SCTP association MUST be marked 306 using the same DSCP value, regardless of how many data channels 307 (streams) exist or what kind of traffic is carried over the various 308 SCTP streams. In the event that the browser wishes to change the 309 DSCP value in use for an SCTP association, it MUST reset the SCTP 310 congestion controller after changing values. Frequent changes in the 311 DSCP value used for an SCTP association are discouraged, though, as 312 this would defeat any attempts at effectively managing congestion. 313 It should also be noted that any change in DSCP value that results in 314 a reset of the congestion controller puts the SCTP association back 315 into slow start, which may have undesirable effects on application 316 performance. 318 For the data channel traffic multiplexed over an SCTP association, it 319 is RECOMMENDED that the DSCP value selected be the one associated 320 with the highest priority requested for all data channels multiplexed 321 over the SCTP association. Likewise, when multiplexing multiple 322 flows over a TCP connection, the DCSP value selected should be the 323 one associated with the highest priority requested for all 324 multiplexed flows. 326 If a packet enters a network that has no support for a flow type- 327 application priority combination specified in Table 1 (above), then 328 the network node at the edge will remark the DSCP value based on 329 policies. This could result in the flow not getting the network 330 treatment it expects based on the original DSCP value in the packet. 331 Subsequently, if the packet enters a network that supports a larger 332 number of these combinations, there may not be sufficient information 333 in the packet to restore the original markings. Mechanisms for 334 restoring such original DSCP is outside the scope of this document. 336 In summary, DSCP marking provides neither guarantees nor promised 337 levels of service. However, DSCP marking is expected to provide a 338 statistical improvement in real-time service as a whole. The service 339 provided to a packet is dependent upon the network design along the 340 path, as well as the network conditions at every hop. 342 6. Security Considerations 344 Since the JavaScript application specifies the flow type and 345 application priority that determine the media flow DSCP values used 346 by the browser, the browser could consider application use of a large 347 number of higher priority flows to be suspicious. If the server 348 hosting the JavaScript application is compromised, many browsers 349 within the network might simultaneously transmit flows with the same 350 DSCP marking. The DiffServ architecture requires ingress traffic 351 conditioning for reasons that include protecting the network from 352 this sort of attack. 354 Otherwise, this specification does not add any additional security 355 implications beyond those addressed in the following DSCP-related 356 specifications. For security implications on use of DSCP, please 357 refer to Section 7 of [RFC7657] and Section 6 of [RFC4594]. Please 358 also see [I-D.ietf-rtcweb-security] as an additional reference. 360 7. IANA Considerations 362 This specification does not require any actions from IANA. 364 8. Downward References 366 This specification contains a downwards reference to [RFC4594] and 367 [RFC7657]. However, the parts of the former RFC used by this 368 specification are sufficiently stable for this downward reference. 369 The guidance in the latter RFC is necessary to understand the 370 Diffserv technology used in this document and the motivation for the 371 recommended DSCP values and procedures. 373 9. Acknowledgements 375 Thanks to David Black, Magnus Westerlund, Paolo Severini, Jim 376 Hasselbrook, Joe Marcus, Erik Nordmark, Michael Tuexen, and Brian 377 Carpenter for their invaluable input. 379 10. Dedication 381 This document is dedicated to the memory of James Polk, a long-time 382 friend and colleague. James made important contributions to this 383 specification, including serving initially as one of the primary 384 authors. The IETF global community mourns his loss and he will be 385 missed dearly. 387 11. Document History 389 Note to RFC Editor: Please remove this section. 391 This document was originally an individual submission in RTCWeb WG. 392 The RTCWeb working group selected it to be become a WG document. 393 Later the transport ADs requested that this be moved to the TSVWG WG 394 as that seemed to be a better match. 396 12. References 398 12.1. Normative References 400 [I-D.ietf-rtcweb-data-channel] 401 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 402 Channels", draft-ietf-rtcweb-data-channel-13 (work in 403 progress), January 2015. 405 [I-D.ietf-rtcweb-rtp-usage] 406 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 407 Communication (WebRTC): Media Transport and Use of RTP", 408 draft-ietf-rtcweb-rtp-usage-26 (work in progress), March 409 2016. 411 [I-D.ietf-rtcweb-security] 412 Rescorla, E., "Security Considerations for WebRTC", draft- 413 ietf-rtcweb-security-08 (work in progress), February 2015. 415 [I-D.ietf-rtcweb-transports] 416 Alvestrand, H., "Transports for WebRTC", draft-ietf- 417 rtcweb-transports-12 (work in progress), March 2016. 419 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 420 Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/ 421 RFC2119, March 1997, 422 . 424 [RFC4594] Babiarz, J., Chan, K., and F. Baker, "Configuration 425 Guidelines for DiffServ Service Classes", RFC 4594, DOI 426 10.17487/RFC4594, August 2006, 427 . 429 [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services 430 (Diffserv) and Real-Time Communication", RFC 7657, DOI 431 10.17487/RFC7657, November 2015, 432 . 434 [RFC7742] Roach, A., "WebRTC Video Processing and Codec 435 Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016, 436 . 438 12.2. Informative References 440 [G.1010] International Telecommunications Union, "End-user 441 multimedia QoS categories", Recommendation ITU-T G.1010, 442 November 2001. 444 [I-D.ietf-rmcat-coupled-cc] 445 Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion 446 control for RTP media", draft-ietf-rmcat-coupled-cc-02 447 (work in progress), April 2016. 449 [RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black, 450 "Definition of the Differentiated Services Field (DS 451 Field) in the IPv4 and IPv6 Headers", RFC 2474, DOI 452 10.17487/RFC2474, December 1998, 453 . 455 [RFC2597] Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski, 456 "Assured Forwarding PHB Group", RFC 2597, DOI 10.17487/ 457 RFC2597, June 1999, 458 . 460 [RFC3246] Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec, 461 J., Courtney, W., Davari, S., Firoiu, V., and D. 462 Stiliadis, "An Expedited Forwarding PHB (Per-Hop 463 Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002, 464 . 466 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 467 Jacobson, "RTP: A Transport Protocol for Real-Time 468 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 469 July 2003, . 471 [RFC3662] Bless, R., Nichols, K., and K. Wehrle, "A Lower Effort 472 Per-Domain Behavior (PDB) for Differentiated Services", 473 RFC 3662, DOI 10.17487/RFC3662, December 2003, 474 . 476 Authors' Addresses 478 Paul E. Jones 479 Cisco Systems 481 Email: paulej@packetizer.com 483 Subha Dhesikan 484 Cisco Systems 486 Email: sdhesika@cisco.com 488 Cullen Jennings 489 Cisco Systems 491 Email: fluffy@cisco.com 493 Dan Druta 494 AT&T 496 Email: dd5826@att.com