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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(See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-rtp-usage-25 == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-08 == Outdated reference: A later version (-17) exists of draft-ietf-rtcweb-transports-11 ** Downref: Normative reference to an Informational RFC: RFC 4594 ** Downref: Normative reference to an Informational RFC: RFC 7657 == Outdated reference: A later version (-09) exists of draft-ietf-rmcat-coupled-cc-00 Summary: 2 errors (**), 0 flaws (~~), 5 warnings (==), 1 comment (--). 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: September 1, 2016 Cisco Systems 6 D. Druta 7 AT&T 8 February 29, 2016 10 DSCP and other packet markings for WebRTC QoS 11 draft-ietf-tsvwg-rtcweb-qos-13 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 September 1, 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. Relation to Other Specifications . . . . . . . . . . . . . . 3 57 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 58 4. Inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 59 5. DSCP Mappings . . . . . . . . . . . . . . . . . . . . . . . . 5 60 6. Security Considerations . . . . . . . . . . . . . . . . . . . 7 61 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 62 8. Downward References . . . . . . . . . . . . . . . . . . . . . 7 63 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 64 10. Dedication . . . . . . . . . . . . . . . . . . . . . . . . . 8 65 11. Document History . . . . . . . . . . . . . . . . . . . . . . 8 66 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 67 12.1. Normative References . . . . . . . . . . . . . . . . . . 8 68 12.2. Informative References . . . . . . . . . . . . . . . . . 9 69 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9 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 77 existing IETF RFCs. The contents of this specification are intended 78 to be a simple set of implementation recommendations based on the 79 previous RFCs. 81 There are many use cases where such marking does not help, but it 82 seldom makes things worse if packets are marked appropriately. There 83 are some environments where DSCP markings frequently help, though. 84 These include: 86 1. Private, wide-area networks. 88 2. Residential Networks. If the congested link is the broadband 89 uplink in a cable or DSL scenario, often residential routers/NAT 90 support preferential treatment based on DSCP. 92 3. Wireless Networks. If the congested link is a local wireless 93 network, marking may help. 95 DSCP values are in principle site specific, with each site selecting 96 its own code points for controlling per-hop-behavior to influence the 97 QoS for transport-layer flows. However in the WebRTC use cases, the 98 browsers need to set them to something when there is no site specific 99 information. In this document, "browsers" is used synonymously with 100 "Interactive User Agent" as defined in the HTML specification, 101 [W3C.REC-html5-20141028]. This document describes a subset of DSCP 102 code point values drawn from existing RFCs and common usage for use 103 with WebRTC applications. These code points are solely defaults. 105 This specification defines inputs that are provided by the WebRTC 106 application hosted in the browser that aid the browser in determining 107 how to set the various packet markings. The specification also 108 defines the mapping from abstract QoS policies (flow type, priority 109 level) to those packet markings. 111 2. Relation to Other Specifications 113 This document is a complement to [RFC7657], which describes the 114 interaction between DSCP and real-time communications. That RFC 115 covers the implications of using various DSCP values, particularly 116 focusing on Real-time Transport Protocol (RTP) [RFC3550] streams that 117 are multiplexed onto a single transport-layer flow. 119 There are a number of guidelines specified in [RFC7657] that apply to 120 marking traffic sent by WebRTC applications, as it is common for 121 multiple RTP streams to be multiplexed on the same transport-layer 122 flow. Generally, the RTP streams would be marked with a value as 123 appropriate from Table 1. A WebRTC application might also multiplex 124 data channel [I-D.ietf-rtcweb-data-channel] traffic over the same 125 5-tuple as RTP streams, which would also be marked as per that table. 126 The guidance in [RFC7657] says that all data channel traffic would be 127 marked with a single value that is typically different than the 128 value(s) used for RTP streams multiplexed with the data channel 129 traffic over the same 5-tuple, assuming RTP streams are marked with a 130 value other than default forwarding (DF). This is expanded upon 131 further in the next section. 133 This specification does not change or override the advice in any 134 other standards about setting packet markings. Rather, it simply 135 selects a subset of DSCP values that is relevant in the WebRTC 136 context. 138 The DSCP value set by the endpoint is not trusted by the network. In 139 addition, the DSCP value may be remarked at any place in the network 140 for a variety of reasons to any other DSCP value, including default 141 forwarding (DF) value to provide basic best effort service. Even so, 142 there is benefit in marking traffic even if it only benefits the 143 first few hops. The implications are discussed in Secton 3.2 of 144 [RFC7657]. Further, a mitigation for such action is through an 145 authorization mechanism. Such an authorization mechanism is outside 146 the scope of this document. 148 3. Terminology 150 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 151 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 152 document are to be interpreted as described in [RFC2119]. 154 4. Inputs 156 WebRTC applications send and receive two types of flows of 157 significance to this document: 159 o media flows which are RTP streams [I-D.ietf-rtcweb-rtp-usage] 161 o data flows which are data channels [I-D.ietf-rtcweb-data-channel] 163 Each of the RTP streams and distinct data channels consists of all of 164 the packets associated with an independent media entity, so an RTP 165 stream or distinct data channel is not always equivalent to a 166 transport-layer flow defined by a 5-tuple (source address, 167 destination address, source port, destination port, and protocol). 168 There may be multiple RTP streams and data channels multiplexed over 169 the same 5-tuple, with each having a different level of importance to 170 the application and, therefore, potentially marked using different 171 DSCP values than another RTP stream or data channel within the same 172 transport-layer flow. (Note that there are restrictions with respect 173 to marking different data channels carried within the same SCTP 174 association as outlined in Section 5.) 176 The following are the inputs provided by the WebRTC application to 177 the browser: 179 o Flow Type: The browser provides this input as it knows if the flow 180 is audio, interactive video with or without audio, non-interactive 181 video with or without audio, or data. 183 o Application Priority: Another input is the relative importance of 184 an RTP stream or data channel. Many applications have multiple 185 flows of the same Flow Type and often some flows are more 186 important than others. For example, in a video conference where 187 there are usually audio and video flows, the audio flow may be 188 more important than the video flow. JavaScript applications can 189 tell the browser whether a particular flow is high, medium, low or 190 very low importance to the application. 192 [I-D.ietf-rtcweb-transports] defines in more detail what an 193 individual flow is within the WebRTC context and priorities for media 194 and data flows. 196 5. DSCP Mappings 198 The DSCP values for each flow type of interest to WebRTC based on 199 application priority are shown in the following table. These values 200 are based on the framework and recommended values in [RFC4594]. A 201 web browser SHOULD use these values to mark the appropriate media 202 packets. More information on EF can be found in [RFC3246]. More 203 information on AF can be found in [RFC2597]. DF is default 204 forwarding which provides the basic best effort service [RFC2474]. 206 +------------------------+-------+------+-------------+-------------+ 207 | Flow Type | Very | Low | Medium | High | 208 | | Low | | | | 209 +------------------------+-------+------+-------------+-------------+ 210 | Audio | CS1 | DF | EF (46) | EF (46) | 211 | | (8) | (0) | | | 212 | | | | | | 213 | Interactive Video with | CS1 | DF | AF42, AF43 | AF41, AF42 | 214 | or without audio | (8) | (0) | (36, 38) | (34, 36) | 215 | | | | | | 216 | Non-Interactive Video | CS1 | DF | AF32, AF33 | AF31, AF32 | 217 | with or without audio | (8) | (0) | (28, 30) | (26, 28) | 218 | | | | | | 219 | Data | CS1 | DF | AF11 | AF21 | 220 | | (8) | (0) | | | 221 +------------------------+-------+------+-------------+-------------+ 223 Table 1: Recommended DSCP Values for WebRTC Applications 225 The application priority, indicated by the columns "very low", "low", 226 "Medium", and "high", signifies the relative importance of the flow 227 within the application. It is an input that the browser receives to 228 assist in selecting the DSCP value and adjusting the network 229 transport behavior. 231 The above table assumes that packets marked with CS1 are treated as 232 "less than best effort". However, the treatment of CS1 is 233 implementation dependent. If an implementation treats CS1 as other 234 than "less than best effort", then the actual priority (or, more 235 precisely, the per-hop-behavior) of the packets may be changed from 236 what is intended. It is common for CS1 to be treated the same as DF, 237 so applications and browsers using CS1 cannot assume that CS1 will be 238 treated differently than DF [RFC7657]. However, it is also possible 239 per [RFC2474] for CS1 traffic to be given better treatment than DF, 240 thus caution should be exercised when electing to use CS1. 242 Implementers should also note that excess EF traffic is dropped. 243 This could mean that a packet marked as EF may not get through as 244 opposed to a packet marked with a different DSCP value. This is not 245 a flaw, but how excess EF traffic is intended to be treated. 247 The browser SHOULD first select the flow type of the flow. Within 248 the flow type, the relative importance of the flow SHOULD be used to 249 select the appropriate DSCP value. 251 The combination of flow type and application priority provides 252 specificity and helps in selecting the right DSCP value for the flow. 253 All packets within a flow SHOULD have the same application priority. 254 In some cases, the selected application priority cell may have 255 multiple DSCP values, such as AF41 and AF42. These offer different 256 drop precedences. The different drop precedence values provides 257 additional granularity in classifying packets within a flow. For 258 example, in a video conference, the video flow may have medium 259 application priority. If so, either AF42 or AF43 may be selected. 260 If the I-frames in the stream are more important than the P-frames, 261 then the I-frames can be marked with AF42 and the P-frames marked 262 with AF43. 264 It is worth noting that the application priority is utilized by the 265 coupled congestion control mechanism for media flows per 266 [I-D.ietf-rmcat-coupled-cc] and the SCTP scheduler for data channel 267 traffic per [I-D.ietf-rtcweb-data-channel]. 269 For reasons discussed in Section 6 of [RFC7657], if multiple flows 270 are multiplexed using a reliable transport (e.g., TCP) then all of 271 the packets for all flows multiplexed over that transport-layer flow 272 MUST be marked using the same DSCP value. Likewise, all WebRTC data 273 channel packets transmitted over an SCTP association MUST be marked 274 using the same DSCP value, regardless of how many data channels 275 (streams) exist or what kind of traffic is carried over the various 276 SCTP streams. In the event that the browser wishes to change the 277 DSCP value in use for an SCTP association, it MUST reset the SCTP 278 congestion controller after changing values. Frequent changes in the 279 DSCP value used for an SCTP association are discouraged, though, as 280 this would defeat any attempts at effectively managing congestion. 281 It should also be noted that any change in DSCP value that results in 282 a reset of the congestion controller puts the SCTP association back 283 into slow start, which may have undesirable effects on application 284 performance. 286 For the data channel traffic multiplexed over an SCTP association, it 287 is RECOMMENDED that the DSCP value selected be the one associated 288 with the highest priority requested for all data channels multiplexed 289 over the SCTP association. Likewise, when multiplexing multiple 290 flows over a TCP connection, the DCSP value selected should be the 291 one associated with the highest priority requested for all 292 multiplexed flows. 294 If a packet enters a network that has no support for a flow type- 295 application priority combination specified in Table 1 (above), then 296 the network node at the edge will remark the DSCP value based on 297 policies. This could result in the flow not getting the network 298 treatment it expects based on the original DSCP value in the packet. 299 Subsequently, if the packet enters a network that supports a larger 300 number of these combinations, there may not be sufficient information 301 in the packet to restore the original markings. Mechanisms for 302 restoring such original DSCP is outside the scope of this document. 304 In summary, DSCP marking provides neither guarantees nor promised 305 levels of service. However, DSCP marking is expected to provide a 306 statistical improvement in real-time service as a whole. The service 307 provided to a packet is dependent upon the network design along the 308 path, as well as the network conditions at every hop. 310 6. Security Considerations 312 This specification does not add any additional security implication 313 other than the normal application use of DSCP not already addressed 314 by the following specifications. For security implications on use of 315 DSCP, please refer to Section 7 of [RFC7657] and Section 6 of 316 [RFC4594]. Please also see [I-D.ietf-rtcweb-security] as an 317 additional reference. 319 7. IANA Considerations 321 This specification does not require any actions from IANA. 323 8. Downward References 325 This specification contains a downwards reference to [RFC4594]. 326 However, the parts of that RFC used by this specification are 327 sufficiently stable for this downward reference. 329 9. Acknowledgements 331 Thanks to David Black, Magnus Westerland, Paolo Severini, Jim 332 Hasselbrook, Joe Marcus, Erik Nordmark, Michael Tuexen, and Brian 333 Carpenter for their invaluable input. 335 10. Dedication 337 This document is dedicated to the memory of James Polk, a long-time 338 friend and colleague. James made important contributions to this 339 specification, including being one of its primary authors. The IETF 340 global community mourns his loss and he will be missed dearly. 342 11. Document History 344 Note to RFC Editor: Please remove this section. 346 This document was originally an individual submission in RTCWeb WG. 347 The RTCWeb working group selected it to be become a WG document. 348 Later the transport ADs requested that this be moved to the TSVWG WG 349 as that seemed to be a better match. 351 12. References 353 12.1. Normative References 355 [I-D.ietf-rtcweb-data-channel] 356 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 357 Channels", draft-ietf-rtcweb-data-channel-13 (work in 358 progress), January 2015. 360 [I-D.ietf-rtcweb-rtp-usage] 361 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 362 Communication (WebRTC): Media Transport and Use of RTP", 363 draft-ietf-rtcweb-rtp-usage-25 (work in progress), June 364 2015. 366 [I-D.ietf-rtcweb-security] 367 Rescorla, E., "Security Considerations for WebRTC", draft- 368 ietf-rtcweb-security-08 (work in progress), February 2015. 370 [I-D.ietf-rtcweb-transports] 371 Alvestrand, H., "Transports for WebRTC", draft-ietf- 372 rtcweb-transports-11 (work in progress), January 2016. 374 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 375 Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/ 376 RFC2119, March 1997, 377 . 379 [RFC4594] Babiarz, J., Chan, K., and F. Baker, "Configuration 380 Guidelines for DiffServ Service Classes", RFC 4594, DOI 381 10.17487/RFC4594, August 2006, 382 . 384 [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services 385 (Diffserv) and Real-Time Communication", RFC 7657, DOI 386 10.17487/RFC7657, November 2015, 387 . 389 12.2. Informative References 391 [I-D.ietf-rmcat-coupled-cc] 392 Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion 393 control for RTP media", draft-ietf-rmcat-coupled-cc-00 394 (work in progress), September 2015. 396 [RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black, 397 "Definition of the Differentiated Services Field (DS 398 Field) in the IPv4 and IPv6 Headers", RFC 2474, DOI 399 10.17487/RFC2474, December 1998, 400 . 402 [RFC2597] Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski, 403 "Assured Forwarding PHB Group", RFC 2597, DOI 10.17487/ 404 RFC2597, June 1999, 405 . 407 [RFC3246] Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec, 408 J., Courtney, W., Davari, S., Firoiu, V., and D. 409 Stiliadis, "An Expedited Forwarding PHB (Per-Hop 410 Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002, 411 . 413 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 414 Jacobson, "RTP: A Transport Protocol for Real-Time 415 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 416 July 2003, . 418 [W3C.REC-html5-20141028] 419 Hickson, I., Berjon, R., Faulkner, S., Leithead, T., 420 Navara, E., O'Connor, E., and S. Pfeiffer, "HTML5", 421 World Wide Web Consortium Recommendation REC- 422 html5-20141028, October 2014, 423 . 425 Authors' Addresses 427 Paul E. Jones 428 Cisco Systems 430 Email: paulej@packetizer.com 431 Subha Dhesikan 432 Cisco Systems 434 Email: sdhesika@cisco.com 436 Cullen Jennings 437 Cisco Systems 439 Email: fluffy@cisco.com 441 Dan Druta 442 AT&T 444 Email: dd5826@att.com