idnits 2.17.1 draft-ietf-tsvwg-rtcweb-qos-09.txt: Checking boilerplate required by RFC 5378 and the IETF Trust (see https://trustee.ietf.org/license-info): ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/1id-guidelines.txt: ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/checklist : ---------------------------------------------------------------------------- No issues found here. Miscellaneous warnings: ---------------------------------------------------------------------------- == The copyright year in the IETF Trust and authors Copyright Line does not match the current year -- The document date (January 23, 2016) is 3014 days in the past. Is this intentional? 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(See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-rtp-usage-25 == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-08 == Outdated reference: A later version (-17) exists of draft-ietf-rtcweb-transports-10 ** Downref: Normative reference to an Informational RFC: RFC 4594 ** Downref: Normative reference to an Informational RFC: RFC 7657 Summary: 2 errors (**), 0 flaws (~~), 4 warnings (==), 1 comment (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group S. Dhesikan 3 Internet-Draft C. Jennings 4 Intended status: Standards Track Cisco Systems 5 Expires: July 26, 2016 D. Druta, Ed. 6 AT&T 7 P. Jones 8 Cisco Systems 9 January 23, 2016 11 DSCP and other packet markings for WebRTC QoS 12 draft-ietf-tsvwg-rtcweb-qos-09 14 Abstract 16 Many networks, such as service provider and enterprise networks, can 17 provide treatment for individual packets based on Differentiated 18 Services Code Point (DSCP) values on a per-hop basis. This document 19 provides the recommended DSCP values for web browsers to use for 20 various classes of WebRTC traffic. 22 Status of This Memo 24 This Internet-Draft is submitted in full conformance with the 25 provisions of BCP 78 and BCP 79. 27 Internet-Drafts are working documents of the Internet Engineering 28 Task Force (IETF). Note that other groups may also distribute 29 working documents as Internet-Drafts. The list of current Internet- 30 Drafts is at http://datatracker.ietf.org/drafts/current/. 32 Internet-Drafts are draft documents valid for a maximum of six months 33 and may be updated, replaced, or obsoleted by other documents at any 34 time. It is inappropriate to use Internet-Drafts as reference 35 material or to cite them other than as "work in progress." 37 This Internet-Draft will expire on July 26, 2016. 39 Copyright Notice 41 Copyright (c) 2016 IETF Trust and the persons identified as the 42 document authors. All rights reserved. 44 This document is subject to BCP 78 and the IETF Trust's Legal 45 Provisions Relating to IETF Documents 46 (http://trustee.ietf.org/license-info) in effect on the date of 47 publication of this document. Please review these documents 48 carefully, as they describe your rights and restrictions with respect 49 to this document. Code Components extracted from this document must 50 include Simplified BSD License text as described in Section 4.e of 51 the Trust Legal Provisions and are provided without warranty as 52 described in the Simplified BSD License. 54 Table of Contents 56 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 57 2. Relation to Other Standards . . . . . . . . . . . . . . . . . 3 58 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 59 4. Inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 60 5. DSCP Mappings . . . . . . . . . . . . . . . . . . . . . . . . 5 61 6. Security Considerations . . . . . . . . . . . . . . . . . . . 7 62 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 63 8. Downward References . . . . . . . . . . . . . . . . . . . . . 7 64 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 65 10. Dedication . . . . . . . . . . . . . . . . . . . . . . . . . 7 66 11. Document History . . . . . . . . . . . . . . . . . . . . . . 8 67 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 68 12.1. Normative References . . . . . . . . . . . . . . . . . . 8 69 12.2. Informative References . . . . . . . . . . . . . . . . . 9 70 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9 72 1. Introduction 74 Differentiated Services Code Points (DSCP) [RFC2474] packet marking 75 can help provide QoS in some environments. This specification 76 proposes how WebRTC applications can mark packets, but does not 77 contradict or redefine any advice from previous IETF RFCs. Rather, 78 it merely provides a simple set of recommendations for implementers 79 based on the 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. As 83 one example of where it does not help, if too many packets, say all 84 audio or all audio and video, are marked for a given network 85 condition then it can prevent desirable results. Either too much 86 other traffic will be starved, or there is not enough capacity for 87 the preferentially marked packets (i.e., audio and/or video). 89 There are some environments where DSCP markings frequently help. 90 These include: 92 1. Private, wide-area networks. 94 2. Residential Networks. If the congested link is the broadband 95 uplink in a cable or DSL scenario, often residential routers/NAT 96 support preferential treatment based on DSCP. 98 3. Wireless Networks. If the congested link is a local wireless 99 network, marking may help. 101 Traditionally DSCP values have been thought of as being site 102 specific, with each site selecting its own code points for 103 controlling per-hop-behavior to influence the QoS for transport-layer 104 flows. However in the WebRTC use cases, the browsers need to set 105 them to something when there is no site specific information. In 106 this document, "browsers" is used synonymously with "Interactive User 107 Agent" as defined in the HTML specification, 108 [W3C.REC-html5-20141028]. This document describes a subset of DSCP 109 code point values drawn from existing RFCs and common usage for use 110 with WebRTC applications. These code points are solely defaults. 112 This specification defines some inputs that the browser in a WebRTC 113 application can consider to aid in determining how to set the various 114 packet markings and defines the mapping from abstract QoS policies 115 (flow type, priority level) to those packet markings. 117 2. Relation to Other Standards 119 This document exists as a complement to [RFC7657], which describes 120 the interaction between DSCP and real-time communications. It covers 121 the implications of using various DSCP values, particularly focusing 122 on Real-time Transport Protocol (RTP) [RFC3550] streams that are 123 multiplexed onto a single transport-layer flow. 125 There are a number of guidelines specified in [RFC7657] that should 126 be followed when marking traffic sent by WebRTC applications, as it 127 is common for multiple RTP streams to be multiplexed on the same 128 transport-layer flow. Generally, the RTP streams would be marked 129 with a value as appropriate from Table 1. A WebRTC application might 130 also multiplex data channel [I-D.ietf-rtcweb-data-channel] traffic 131 over the same 5-tuple as RTP streams, which would also be marked as 132 per that table. The guidance in [RFC7657] says that all data channel 133 traffic would be marked with a single value that is typically 134 different than the value(s) used for RTP streams multiplexed with the 135 data channel traffic over the same 5-tuple, assuming RTP streams are 136 marked with a value other than default forwarding (DF). This is 137 expanded upon further in the next section. 139 This specification does not change or override the advice in any 140 other standards about setting packet markings. It simply selects a 141 subset of DSCP values that is relevant in the WebRTC context. This 142 document also specifies the inputs that are needed by the browser to 143 provide to the media engine. 145 The DSCP value set by the endpoint is not always trusted by the 146 network. Therefore, the DSCP value may be remarked at any place in 147 the network for a variety of reasons to any other DSCP value, 148 including default forwarding (DF) value to provide basic best effort 149 service. The mitigation for such action is through an authorization 150 mechanism. Such authorization mechanism is outside the scope of this 151 document. There is benefit in marking traffic even if it only 152 benefits the first few hops. 154 3. Terminology 156 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 157 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 158 document are to be interpreted as described in [RFC2119]. 160 4. Inputs 162 WebRTC entities transmit and receive two types of media of 163 significance to this document: 165 o media flows which are RTP streams [I-D.ietf-rtcweb-rtp-usage] 166 o data flows which are data channels [I-D.ietf-rtcweb-data-channel] 168 Each of the RTP streams and distinct data channels consists of all of 169 the packets associated with an independent media entity and are not 170 always equivalent to a transport-layer flow defined by a 5-tuple 171 (source address, destination address, source port, destination port, 172 and protocol). There may be multiple RTP streams and data channels 173 multiplexed over the same 5-tuple, with each having a different level 174 of importance to the application and, therefore, potentially marked 175 using different DSCP values than another RTP stream or data channel 176 within the same transport-layer flow. (Note that there are 177 restrictions with respect to marking different data channels carried 178 within the same SCTP association as outlined in Section 5.) 180 The following are the inputs that the browser provides to the media 181 engine: 183 o Flow Type: The browser provides this input as it knows if the flow 184 is audio, interactive video with or without audio, non-interactive 185 video with or without audio, or data. 186 o Application Priority: Another input is the relative importance of 187 an RTP stream or data channel. Many applications have multiple 188 flows of the same Flow Type and often some flows are more 189 important than others. For example, in a video conference where 190 there are usually audio and video flows, the audio flow may be 191 more important than the video flow. JavaScript applications can 192 tell the browser whether a particular flow is high, medium, low or 193 very low importance to the application. 195 [I-D.ietf-rtcweb-transports] defines in more detail what an 196 individual flow is within the WebRTC context. 198 5. DSCP Mappings 200 The DSCP markings for each flow type of interest to WebRTC given the 201 application priority is shown in the following table. The DSCP 202 values for each flow type listed are a reasonable subset of code 203 point values taken from [RFC4594]. A web browser SHOULD use these 204 values to mark the appropriate media packets. More information on EF 205 can be found in [RFC3246]. More information on AF can be found in 206 [RFC2597]. DF is default forwarding which provides the basic best 207 effort service. 209 +------------------------+-------+------+-------------+-------------+ 210 | Flow Type | Very | Low | Medium | High | 211 | | Low | | | | 212 +------------------------+-------+------+-------------+-------------+ 213 | Audio | CS1 | DF | EF (46) | EF (46) | 214 | | (8) | (0) | | | 215 | | | | | | 216 | Interactive Video with | CS1 | DF | AF42, AF43 | AF41, AF42 | 217 | or without audio | (8) | (0) | (36, 38) | (34, 36) | 218 | | | | | | 219 | Non-Interactive Video | CS1 | DF | AF32, AF33 | AF31, AF32 | 220 | with or without audio | (8) | (0) | (28, 30) | (26, 28) | 221 | | | | | | 222 | Data | CS1 | DF | AF11 | AF21 | 223 | | (8) | (0) | | | 224 +------------------------+-------+------+-------------+-------------+ 226 Table 1: Recommended DSCP Values for WebRTC Applications 228 The application priority, indicated by the columns "very low", "low", 229 "Medium", and "high", signifies the relative importance of the flow 230 within the application. It is an input that the browser receives to 231 assist it in selecting the DSCP value. Application priority does not 232 refer to priority in the network transport. 234 The above table assumes that packets marked with CS1 are treated as 235 "less than best effort". However, the treatment of CS1 is 236 implementation dependent. If an implementation treats CS1 as other 237 than "less than best effort", then the actual priority (or, more 238 precisely, the per-hop-behavior) of the packets may be changed from 239 what is intended. It is common for CS1 to be treated the same as DF, 240 so anyone using CS1 cannot assume that CS1 will be treated 241 differently than DF. Implementers should also note that excess EF 242 traffic is dropped. This could mean that a packet marked as EF may 243 not get through as opposed to a packet marked with a different DSCP 244 value. 246 The browser SHOULD first select the flow type of the flow. Within 247 the flow type, the relative importance of the flow SHOULD be used to 248 select the appropriate DSCP value. 250 The combination of flow type and application priority provides 251 specificity and helps in selecting the right DSCP value for the flow. 252 All packets within a flow SHOULD have the same application priority. 253 In some cases, the selected application priority cell may have 254 multiple DSCP values, such as AF41 and AF42. These offer different 255 drop precedences. The different drop precedence values provides 256 additional granularity in classifying packets within a flow. For 257 example, in a video conference, the video flow may have medium 258 application priority. If so, either AF42 or AF43 may be selected. 259 If the I-frames in the stream are more important than the P-frames, 260 then the I-frames can be marked with AF42 and the P-frames marked 261 with AF43. 263 For reasons discussed in Section 6 of [RFC7657], if multiple flows 264 are multiplexed using a reliable transport (e.g., TCP) then all of 265 the packets for all flows multiplexed over that transport-layer flow 266 MUST be marked using the same DSCP value. Likewise, all WebRTC data 267 channel packets transmitted over an SCTP association MUST be marked 268 using the same DSCP value, regardless of how many data channels 269 (streams) exist or what kind of traffic is carried over the various 270 SCTP streams. In the event that the browser wishes to change the 271 DSCP value in use for an SCTP association, it MUST reset the SCTP 272 congestion controller after changing values. Frequent changes in the 273 DSCP value used for an SCTP association are discouraged, though, as 274 this would defeat any attempts at effectively managing congestion. 275 It should also be noted that any change in DSCP value that results in 276 a reset of the congestion controller puts the SCTP association back 277 into slow start, which may have undesirable effects on application 278 performance. 280 For the data channel traffic multiplexed over an SCTP association, it 281 is RECOMMENDED that the DSCP value selected be the one associated 282 with the highest priority requested for all data channels multiplexed 283 over the SCTP association. Likewise, when multiplexing multiple 284 flows over a TCP connection, the DCSP value selected should be the 285 one associated with the highest priority requested for all 286 multiplexed flows. 288 If a packet enters a QoS domain that has no support for the above 289 defined flow types/application priority (service class), then the 290 network node at the edge will remark the DSCP value based on 291 policies. This could result in the flow not getting the network 292 treatment it expects based on the original DSCP value in the packet. 293 Subsequently, if the packet enters a QoS domain that supports a 294 larger number of service classes, there may not be sufficient 295 information in the packet to restore the original markings. 296 Mechanisms for restoring such original DSCP is outside the scope of 297 this document. 299 In summary, there are no guarantees or promised level of service with 300 the use of DSCP. The service provided to a packet is dependent upon 301 the network design along the path, as well as the congestion levels 302 at every hop. 304 6. Security Considerations 306 This specification does not add any additional security implication 307 other than the normal application use of DSCP. For security 308 implications on use of DSCP, please refer to Section 6 of [RFC4594]. 309 Please also see [I-D.ietf-rtcweb-security] as an additional 310 reference. 312 7. IANA Considerations 314 This specification does not require any actions from IANA. 316 8. Downward References 318 This specification contains a downwards reference to [RFC4594]. 319 However, the parts of that RFC used by this specification are 320 sufficiently stable for this downward reference. 322 9. Acknowledgements 324 Thanks To David Black, Magnus Westerland, Paolo Severini, Jim 325 Hasselbrook, Joe Marcus, Erik Nordmark, and Michael Tuexen for their 326 invaluable input. 328 10. Dedication 330 This document is dedicated to the memory of James Polk, a long-time 331 friend and colleague. James made important contributions to this 332 specification, including being one of its primary authors. The IETF 333 global community mourns his loss and he will be missed dearly. 335 11. Document History 337 Note to RFC Editor: Please remove this section. 339 This document was originally an individual submission in RTCWeb WG. 340 The RTCWeb working group selected it to be become a WG document. 341 Later the transport ADs requested that this be moved to the TSVWG WG 342 as that seemed to be a better match. 344 12. References 346 12.1. Normative References 348 [I-D.ietf-rtcweb-data-channel] 349 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 350 Channels", draft-ietf-rtcweb-data-channel-13 (work in 351 progress), January 2015. 353 [I-D.ietf-rtcweb-rtp-usage] 354 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 355 Communication (WebRTC): Media Transport and Use of RTP", 356 draft-ietf-rtcweb-rtp-usage-25 (work in progress), June 357 2015. 359 [I-D.ietf-rtcweb-security] 360 Rescorla, E., "Security Considerations for WebRTC", draft- 361 ietf-rtcweb-security-08 (work in progress), February 2015. 363 [I-D.ietf-rtcweb-transports] 364 Alvestrand, H., "Transports for WebRTC", draft-ietf- 365 rtcweb-transports-10 (work in progress), October 2015. 367 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 368 Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/ 369 RFC2119, March 1997, 370 . 372 [RFC4594] Babiarz, J., Chan, K., and F. Baker, "Configuration 373 Guidelines for DiffServ Service Classes", RFC 4594, DOI 374 10.17487/RFC4594, August 2006, 375 . 377 [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services 378 (Diffserv) and Real-Time Communication", RFC 7657, DOI 379 10.17487/RFC7657, November 2015, 380 . 382 12.2. Informative References 384 [RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black, 385 "Definition of the Differentiated Services Field (DS 386 Field) in the IPv4 and IPv6 Headers", RFC 2474, DOI 387 10.17487/RFC2474, December 1998, 388 . 390 [RFC2597] Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski, 391 "Assured Forwarding PHB Group", RFC 2597, DOI 10.17487/ 392 RFC2597, June 1999, 393 . 395 [RFC3246] Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec, 396 J., Courtney, W., Davari, S., Firoiu, V., and D. 397 Stiliadis, "An Expedited Forwarding PHB (Per-Hop 398 Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002, 399 . 401 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 402 Jacobson, "RTP: A Transport Protocol for Real-Time 403 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 404 July 2003, . 406 [W3C.REC-html5-20141028] 407 Hickson, I., Berjon, R., Faulkner, S., Leithead, T., 408 Navara, E., O'Connor, E., and S. Pfeiffer, "HTML5", 409 World Wide Web Consortium Recommendation REC- 410 html5-20141028, October 2014, 411 . 413 Authors' Addresses 415 Subha Dhesikan 416 Cisco Systems 418 Email: sdhesika@cisco.com 420 Cullen Jennings 421 Cisco Systems 423 Email: fluffy@cisco.com 424 Dan Druta (editor) 425 AT&T 427 Email: dd5826@att.com 429 Paul E. Jones 430 Cisco Systems 432 Email: paulej@packetizer.com