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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group R. Jesup 3 Internet-Draft Mozilla 4 Intended status: Standards Track S. Loreto 5 Expires: April 1, 2015 Ericsson 6 M. Tuexen 7 Muenster Univ. of Appl. Sciences 8 September 28, 2014 10 WebRTC Data Channels 11 draft-ietf-rtcweb-data-channel-12.txt 13 Abstract 15 The WebRTC framework specifies protocol support for direct 16 interactive rich communication using audio, video, and data between 17 two peers' web-browsers. This document specifies the non-media data 18 transport aspects of the WebRTC framework. It provides an 19 architectural overview of how the Stream Control Transmission 20 Protocol (SCTP) is used in the WebRTC context as a generic transport 21 service allowing WEB-browsers to exchange generic data from peer to 22 peer. 24 Status of This Memo 26 This Internet-Draft is submitted in full conformance with the 27 provisions of BCP 78 and BCP 79. 29 Internet-Drafts are working documents of the Internet Engineering 30 Task Force (IETF). Note that other groups may also distribute 31 working documents as Internet-Drafts. The list of current Internet- 32 Drafts is at http://datatracker.ietf.org/drafts/current/. 34 Internet-Drafts are draft documents valid for a maximum of six months 35 and may be updated, replaced, or obsoleted by other documents at any 36 time. It is inappropriate to use Internet-Drafts as reference 37 material or to cite them other than as "work in progress." 39 This Internet-Draft will expire on April 1, 2015. 41 Copyright Notice 43 Copyright (c) 2014 IETF Trust and the persons identified as the 44 document authors. All rights reserved. 46 This document is subject to BCP 78 and the IETF Trust's Legal 47 Provisions Relating to IETF Documents 48 (http://trustee.ietf.org/license-info) in effect on the date of 49 publication of this document. Please review these documents 50 carefully, as they describe your rights and restrictions with respect 51 to this document. Code Components extracted from this document must 52 include Simplified BSD License text as described in Section 4.e of 53 the Trust Legal Provisions and are provided without warranty as 54 described in the Simplified BSD License. 56 Table of Contents 58 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 59 2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3 60 3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3 61 3.1. Use Cases for Unreliable Data Channels . . . . . . . . . 3 62 3.2. Use Cases for Reliable Data Channels . . . . . . . . . . 4 63 4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4 64 5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . 5 65 6. The Usage of SCTP for Data Channels . . . . . . . . . . . . . 8 66 6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8 67 6.2. SCTP Association Management . . . . . . . . . . . . . . . 9 68 6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9 69 6.4. Data Channel Definition . . . . . . . . . . . . . . . . . 9 70 6.5. Opening a Data Channel . . . . . . . . . . . . . . . . . 10 71 6.6. Transferring User Data on a Data Channel . . . . . . . . 11 72 6.7. Closing a Data Channel . . . . . . . . . . . . . . . . . 12 73 7. Security Considerations . . . . . . . . . . . . . . . . . . . 12 74 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13 75 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 13 76 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 14 77 10.1. Normative References . . . . . . . . . . . . . . . . . . 14 78 10.2. Informative References . . . . . . . . . . . . . . . . . 15 79 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16 81 1. Introduction 83 In the WebRTC framework, communication between the parties consists 84 of media (for example audio and video) and non-media data. Media is 85 sent using SRTP, and is not specified further here. Non-media data 86 is handled by using SCTP [RFC4960] encapsulated in DTLS [RFC4347]. 88 +----------+ 89 | SCTP | 90 +----------+ 91 | DTLS | 92 +----------+ 93 | ICE/UDP | 94 +----------+ 96 Figure 1: Basic stack diagram 98 The encapsulation of SCTP over DTLS (see 99 [I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245]) 100 provides a NAT traversal solution together with confidentiality, 101 source authentication, and integrity protected transfers. This data 102 transport service operates in parallel to the SRTP media transports, 103 and all of them can eventually share a single transport-layer port 104 number. 106 SCTP as specified in [RFC4960] with the partial reliability extension 107 defined in [RFC3758] and the additional policies defined in 108 [I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively 109 with reliable, and the relevant partially-reliable delivery modes for 110 user messages. Using the reconfiguration extension defined in 111 [RFC6525] allows to increase the number of streams during the 112 lifetime of an SCTP association and to reset individual SCTP streams. 113 Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages 114 to avoid the monopolization and adds the support of prioritizing of 115 SCTP streams. 117 The remainder of this document is organized as follows: Section 3 and 118 Section 4 provide use cases and requirements for both unreliable and 119 reliable peer to peer data channels; Section 5 discusses SCTP over 120 DTLS over UDP; Section 6 provides the specification of how SCTP 121 should be used by the WebRTC protocol framework for transporting non- 122 media data between WEB-browsers. 124 2. Conventions 126 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 127 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 128 document are to be interpreted as described in [RFC2119]. 130 3. Use Cases 132 This section defines use cases specific to data channels. For WebRTC 133 use cases see [I-D.ietf-rtcweb-use-cases-and-requirements]. 135 3.1. Use Cases for Unreliable Data Channels 137 U-C 1: A real-time game where position and object state information 138 is sent via one or more unreliable data channels. Note that 139 at any time there may be no SRTP media channels, or all SRTP 140 media channels may be inactive, and that there may also be 141 reliable data channels in use. 143 U-C 2: Providing non-critical information to a user about the reason 144 for a state update in a video chat or conference, such as 145 mute state. 147 3.2. Use Cases for Reliable Data Channels 149 U-C 3: A real-time game where critical state information needs to be 150 transferred, such as control information. Such a game may 151 have no SRTP media channels, or they may be inactive at any 152 given time, or may only be added due to in-game actions. 154 U-C 4: Non-realtime file transfers between people chatting. Note 155 that this may involve a large number of files to transfer 156 sequentially or in parallel, such as when sharing a folder of 157 images or a directory of files. 159 U-C 5: Realtime text chat during an audio and/or video call with an 160 individual or with multiple people in a conference. 162 U-C 6: Renegotiation of the configuration of the PeerConnection. 164 U-C 7: Proxy browsing, where a browser uses data channels of a 165 PeerConnection to send and receive HTTP/HTTPS requests and 166 data, for example to avoid local Internet filtering or 167 monitoring. 169 4. Requirements 171 This section lists the requirements for P2P data channels between two 172 browsers. 174 Req. 1: Multiple simultaneous data channels MUST be supported. 175 Note that there may be 0 or more SRTP media streams in 176 parallel with the data channels in the same PeerConnection, 177 and the number and state (active/inactive) of these SRTP 178 media streams may change at any time. 180 Req. 2: Both reliable and unreliable data channels MUST be 181 supported. 183 Req. 3: Data channels of a PeerConnection MUST be congestion 184 controlled; either individually, as a class, or in 185 conjunction with the SRTP media streams of the 186 PeerConnection, to ensure that data channels don't cause 187 congestion problems for these SRTP media streams, and that 188 the WebRTC PeerConnection does not cause excessive problems 189 when run in parallel with TCP connections. 191 Req. 4: The application SHOULD be able to provide guidance as to 192 the relative priority of each data channel relative to each 193 other, and relative to the SRTP media streams. This will 194 interact with the congestion control algorithms. 196 Req. 5: Data channels MUST be secured; allowing for 197 confidentiality, integrity and source authentication. See 198 [I-D.ietf-rtcweb-security] and 199 [I-D.ietf-rtcweb-security-arch] for detailed info. 201 Req. 6: Data channels MUST provide message fragmentation support 202 such that IP-layer fragmentation can be avoided no matter 203 how large a message the JavaScript application passes to be 204 sent. It also MUST ensure that large data channel 205 transfers don't unduly delay traffic on other data 206 channels. 208 Req. 7: The data channel transport protocol MUST NOT encode local 209 IP addresses inside its protocol fields; doing so reveals 210 potentially private information, and leads to failure if 211 the address is depended upon. 213 Req. 8: The data channel transport protocol SHOULD support 214 unbounded-length "messages" (i.e., a virtual socket stream) 215 at the application layer, for such things as image-file- 216 transfer; Implementations might enforce a reasonable 217 message size limit. 219 Req. 9: The data channel transport protocol SHOULD avoid IP 220 fragmentation. It MUST support PMTU (Path MTU) discovery 221 and MUST NOT rely on ICMP or ICMPv6 being generated or 222 being passed back, especially for PMTU discovery. 224 Req. 10: It MUST be possible to implement the protocol stack in the 225 user application space. 227 5. SCTP over DTLS over UDP Considerations 229 The important features of SCTP in the WebRTC context are: 231 o Usage of a TCP-friendly congestion control. 233 o The congestion control is modifiable for integration with the SRTP 234 media stream congestion control. 236 o Support of multiple unidirectional streams, each providing its own 237 notion of ordered message delivery. 239 o Support of ordered and out-of-order message delivery. 241 o Supporting arbitrary large user messages by providing 242 fragmentation and reassembly. 244 o Support of PMTU-discovery. 246 o Support of reliable or partially reliable message transport. 248 The WebRTC Data Channel mechanism does not support SCTP multihoming. 249 The SCTP layer will simply act as if it were running on a single- 250 homed host, since that is the abstraction that the DTLS layer (a 251 connection oriented, unreliable datagram service) exposes. 253 The encapsulation of SCTP over DTLS defined in 254 [I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source 255 authenticated, and integrity protected transfers. Using DTLS over 256 UDP in combination with ICE enables middlebox traversal in IPv4 and 257 IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in 258 combination with the extension defined in [RFC3758] and provides the 259 following features for transporting non-media data between browsers: 261 o Support of multiple unidirectional streams. 263 o Ordered and unordered delivery of user messages. 265 o Reliable and partial-reliable transport of user messages. 267 Each SCTP user message contains a Payload Protocol Identifier (PPID) 268 that is passed to SCTP by its upper layer on the sending side and 269 provided to its upper layer on the receiving side. The PPID can be 270 used to multiplex/demultiplex multiple upper layers over a single 271 SCTP association. In the WebRTP context, the PPID is used to 272 distinguish between UTF-8 encoded user data, binary encoded userdata 273 and the Data Channel Establishment Protocol defined in 274 [I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not 275 accessible via the Javascript API. 277 The encapsulation of SCTP over DTLS, together with the SCTP features 278 listed above satisfies all the requirements listed in Section 4. 280 The layering of protocols for WebRTC is shown in the following 281 Figure 2. 283 +------+------+------+ 284 | DCEP | UTF-8|Binary| 285 | | data | data | 286 +------+------+------+ 287 | SCTP | 288 +----------------------------------+ 289 | STUN | SRTP | DTLS | 290 +----------------------------------+ 291 | ICE | 292 +----------------------------------+ 293 | UDP1 | UDP2 | UDP3 | ... | 294 +----------------------------------+ 296 Figure 2: WebRTC protocol layers 298 This stack (especially in contrast to DTLS over SCTP [RFC6083] in 299 combination with SCTP over UDP [RFC6951]) has been chosen because it 301 o supports the transmission of arbitrary large user messages. 303 o shares the DTLS connection with the SRTP media channels of the 304 PeerConnection. 306 o provides privacy for the SCTP control information. 308 Considering the protocol stack of Figure 2 the usage of DTLS over UDP 309 is specified in [RFC4347], while the usage of SCTP on top of DTLS is 310 specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the 311 demultiplexing STUN vs. SRTP vs. DTLS is done as described in 312 Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS. 314 Since DTLS is typically implemented in user application space, the 315 SCTP stack also needs to be a user application space stack. 317 The ICE/UDP layer can handle IP address changes during a session 318 without needing interaction with the DTLS and SCTP layers. However, 319 SCTP SHOULD be notified when an address changes has happened. In 320 this case SCTP SHOULD retest the Path MTU and reset the congestion 321 state to the initial state. In case of a window based congestion 322 control like the one specified in [RFC4960], this means setting the 323 congestion window and slow start threshold to its initial values. 325 Incoming ICMP or ICMPv6 messages can't be processed by the SCTP 326 layer, since there is no way to identify the corresponding 327 association. Therefore SCTP MUST support performing Path MTU 328 discovery without relying on ICMP or ICMPv6 as specified in [RFC4821] 329 using probing messages specified in [RFC4820]. The initial Path MTU 330 at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for 331 IPv6. 333 In general, the lower layer interface of an SCTP implementation 334 SHOULD be adapted to address the differences between IPv4 and IPv6 335 (being connection-less) or DTLS (being connection-oriented). 337 When the protocol stack of Figure 2 is used, DTLS protects the 338 complete SCTP packet, so it provides confidentiality, integrity and 339 source authentication of the complete SCTP packet. 341 SCTP provides congestion control on a per-association base. This 342 means that all SCTP streams within a single SCTP association share 343 the same congestion window. Traffic not being sent over SCTP is not 344 covered by the SCTP congestion control. Using a congestion control 345 different from than the standard one might improve the impact on the 346 parallel SRTP media streams. 348 SCTP uses the same port number concept as TCP and UDP do. Therefore 349 an SCTP association uses two port numbers, one at each SCTP end- 350 point. 352 6. The Usage of SCTP for Data Channels 354 6.1. SCTP Protocol Considerations 356 The DTLS encapsulation of SCTP packets as described in 357 [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used. 359 This SCTP stack and its upper layer MUST support the usage of 360 multiple SCTP streams. A user message can be sent ordered or 361 unordered and with partial or full reliability. 363 The following SCTP protocol extensions are required: 365 o The stream reconfiguration extension defined in [RFC6525] MUST be 366 supported. It is used for closing channels. 368 o The dynamic address reconfiguration extension defined in [RFC5061] 369 MUST be used to signal the support of the stream reset extension 370 defined in [RFC6525], other features of [RFC5061] are not REQUIRED 371 to be implemented. 373 o The partial reliability extension defined in [RFC3758] MUST be 374 supported. In addition to the timed reliability PR-SCTP policy 375 defined in [RFC3758], the limited retransmission policy defined in 376 [I-D.ietf-tsvwg-sctp-prpolicies] MUST be supported. Limiting the 377 number of retransmissions to zero combined with unordered delivery 378 provides a UDP-like service where each user message is sent 379 exactly once and delivered in the order received. 381 The support for message interleaving as defined in 382 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used. 384 6.2. SCTP Association Management 386 In the WebRTC context, the SCTP association will be set up when the 387 two endpoints of the WebRTC PeerConnection agree on opening it, as 388 negotiated by JSEP (typically an exchange of SDP) 389 [I-D.ietf-rtcweb-jsep]. It will use the DTLS connection selected via 390 ICE; typically this will be shared via BUNDLE or equivalent with DTLS 391 connections used to key the SRTP media streams. 393 The number of streams negotiated during SCTP association setup SHOULD 394 be 65535, which is the maximum number of streams that can be 395 negotiated during the association setup. 397 SCTP supports two ways of terminating an SCTP association. A 398 graceful one, using a procedure which ensures that no messages are 399 lost during the shutdown of the association. The second method is a 400 non-graceful one, where one side can just abort the association. 402 Each SCTP end-point supervises continuously the reachability of its 403 peer by monitoring the number of retransmissions of user messages and 404 test messages. In case of excessive retransmissions, the association 405 is terminated in a non-graceful way. 407 If an SCTP association is closed in a graceful way, all of its data 408 channels are closed. In case of a non-graceful teardown, all data 409 channels are also closed, but an error indication SHOULD be provided 410 if possible. 412 6.3. SCTP Streams 414 SCTP defines a stream as a unidirectional logical channel existing 415 within an SCTP association to another SCTP endpoint. The streams are 416 used to provide the notion of in-sequence delivery and for 417 multiplexing. Each user message is sent on a particular stream, 418 either ordered or unordered. Ordering is preserved only for ordered 419 messages sent on the same stream. 421 6.4. Data Channel Definition 423 One strong wish is for the application-level API to be close to the 424 API for WebSockets, which implies bidirectional streams of data and a 425 textual field called 'label' used to identify the meaning of the data 426 channel. 428 The realization of a data channel is a pair of one incoming stream 429 and one outgoing SCTP stream having the same SCTP stream identifier. 430 How these SCTP stream identifiers are selected is protocol and 431 implementation dependent. This allows a bidirectional communication. 433 Additionally, each data channel has the following properties in each 434 direction: 436 o reliable or unreliable message transmission. In case of 437 unreliable transmissions, the same level of unreliability is used. 438 Please note that in SCTP this is a property of an SCTP user 439 message and not of an SCTP stream. 441 o in-order or out-of-order message delivery for message sent. 442 Please note that in SCTP this is a property of an SCTP user 443 message and not of an SCTP stream. 445 o A priority, which is a 2 byte unsigned integer. These priorities 446 MUST be interpreted as weighted-fair-queuing scheduling priorities 447 per the definition of the corresponding stream scheduler 448 supporting interleaving in [I-D.ietf-tsvwg-sctp-ndata]. For use 449 in WebRTC, the values used SHOULD be one of 128 ("below normal"), 450 256 ("normal"), 512 ("high") or 1024 ("extra high"). 452 o an optional label. 454 o an optional protocol. 456 Please note that for a data channel being negotiated with the 457 protocol specified in [I-D.ietf-rtcweb-data-protocol] all of the 458 above properties are the same in both directions. 460 6.5. Opening a Data Channel 462 Data channels can be opened by using negotiation within the SCTP 463 association, called in-band negotiation, or out-of-band negotiation. 464 Out-of-band negotiation is defined as any method which results in an 465 agreement as to the parameters of a channel and the creation thereof. 466 The details are out of scope of this document. Applications using 467 data channels need to use the negotiation methods consistently on 468 both end-points. 470 A simple protocol for in-band negotiation is specified in 471 [I-D.ietf-rtcweb-data-protocol]. 473 When one side wants to open a channel using out-of-band negotiation, 474 it picks a stream. Unless otherwise defined or negotiated, the 475 streams are picked based on the DTLS role (the client picks even 476 stream identifiers, the server odd stream identifiers). However, the 477 application is responsible for avoiding collisions with existing 478 streams. If it attempts to re-use a stream which is part of an 479 existing data channel, the addition SHOULD fail. In addition to 480 choosing a stream, the application SHOULD also determine the options 481 to use for sending messages. The application MUST ensure in an 482 application-specific manner that the application at the peer will 483 also know the selected stream to be used, and the options for sending 484 data from that side. 486 6.6. Transferring User Data on a Data Channel 488 All data sent on a data channel in both directions MUST be sent over 489 the underlying stream using the reliability defined when the data 490 channel was opened unless the options are changed, or per-message 491 options are specified by a higher level. 493 No more than one message should be put into an SCTP user message. 495 The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the 496 interpretation of the "Payload data". The following PPIDs MUST be 497 used (see Section 8): 499 WebRTC String: to identify a non-empty JavaScript string encoded in 500 UTF-8. 502 WebRTC String Empty: to identify an empty JavaScript string encoded 503 in UTF-8. 505 WebRTC Binary: to identify a non-empty JavaScript binary data 506 (ArrayBuffer, ArrayBufferView or Blob). 508 WebRTC Binary Empty: to identify an empty JavaScript binary data 509 (ArrayBuffer, ArrayBufferView or Blob). 511 SCTP does not support the sending of empty user messages. Therefore, 512 if an empty message has to be sent, the appropriate PPID (WebRTC 513 String Empty or WebRTC Binary Empty) is used and the SCTP user 514 message of one zero byte is sent. When receiving an SCTP user 515 message with one of these PPIDs, the receiver MUST ignore the SCTP 516 user message and process it as an empty message. 518 The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary 519 Partial" is deprecated. They were used for a PPID-based 520 fragmentation and reassembly of user messages belonging to reliable 521 and ordered data channels. 523 If a message with an unsupported PPID is received or some error 524 condition related to the received message is detected by the receiver 525 (for example, illegal ordering), the receiver SHOULD close the 526 corresponding data channel. This implies in particular that 527 extensions using additional PPIDs can't be used without prior 528 negotiation. 530 The SCTP base protocol specified in [RFC4960] does not support the 531 interleaving of user messages. Therefore sending a large user 532 message can monopolize the SCTP association. To overcome this 533 limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to 534 support message interleaving, which SHOULD be used. As long as 535 message interleaving is not supported, the sender SHOULD limit the 536 maximum message size to 16 KB to avoid monopolization. 538 It is recommended that the message size be kept within certain size 539 bounds as applications will not be able to support arbitrarily-large 540 single messages. This limit has to be negotiated, for example by 541 using [I-D.ietf-mmusic-sctp-sdp]. 543 The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to 544 minimize the latency. 546 6.7. Closing a Data Channel 548 Closing of a data channel MUST be signaled by resetting the 549 corresponding outgoing streams [RFC6525]. This means that if one 550 side decides to close the data channel, it resets the corresponding 551 outgoing stream. When the peer sees that an incoming stream was 552 reset, it also resets its corresponding outgoing stream. Once this 553 is completed, the data channel is closed. Resetting a stream sets 554 the Stream Sequence Numbers (SSNs) of the stream back to 'zero' with 555 a corresponding notification to the application layer that the reset 556 has been performed. Streams are available for reuse after a reset 557 has been performed. 559 [RFC6525] also guarantees that all the messages are delivered (or 560 abandoned) before the stream is reset. 562 7. Security Considerations 564 This document does not add any additional considerations to the ones 565 given in [I-D.ietf-rtcweb-security] and 566 [I-D.ietf-rtcweb-security-arch]. 568 I should be noted that a receiver must be prepared that the sender 569 tries to send arbitrary large messages. 571 8. IANA Considerations 573 [NOTE to RFC-Editor: 575 "RFCXXXX" is to be replaced by the RFC number you assign this 576 document. 578 ] 580 This document uses six already registered SCTP Payload Protocol 581 Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary 582 Data Last", "DOMString Partial", "WebRTC String Empty", and "WebRTC 583 Binary Empty". [RFC4960] creates the registry "SCTP Payload Protocol 584 Identifiers" from which these identifiers were assigned. IANA is 585 requested to update the reference of these six assignments to point 586 to this document and change the names of the first four PPIDs. The 587 corresponding dates should be kept. 589 Therefore these six assignments should be updated to read: 591 +-------------------------------+----------+-----------+------------+ 592 | Value | SCTP | Reference | Date | 593 | | PPID | | | 594 +-------------------------------+----------+-----------+------------+ 595 | WebRTC String | 51 | [RFCXXXX] | 2013-09-20 | 596 | WebRTC Binary Partial | 52 | [RFCXXXX] | 2013-09-20 | 597 | (Deprecated) | | | | 598 | WebRTC Binary | 53 | [RFCXXXX] | 2013-09-20 | 599 | WebRTC String Partial | 54 | [RFCXXXX] | 2013-09-20 | 600 | (Deprecated) | | | | 601 | WebRTC String Empty | 56 | [RFCXXXX] | 2014-08-22 | 602 | WebRTC Binary Empty | 57 | [RFCXXXX] | 2014-08-22 | 603 +-------------------------------+----------+-----------+------------+ 605 9. Acknowledgments 607 Many thanks for comments, ideas, and text from Harald Alvestrand, 608 Richard Barnes, Adam Bergkvist, Gunnar Hellstrom, Christer Holmberg, 609 Cullen Jennings, Paul Kyzivat, Eric Rescorla, Irene Ruengeler, 610 Randall Stewart, Justin Uberti, and Magnus Westerlund. 612 10. References 614 10.1. Normative References 616 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 617 Requirement Levels", BCP 14, RFC 2119, March 1997. 619 [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. 620 Conrad, "Stream Control Transmission Protocol (SCTP) 621 Partial Reliability Extension", RFC 3758, May 2004. 623 [RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer 624 Security", RFC 4347, April 2006. 626 [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and 627 Parameter for the Stream Control Transmission Protocol 628 (SCTP)", RFC 4820, March 2007. 630 [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU 631 Discovery", RFC 4821, March 2007. 633 [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 634 4960, September 2007. 636 [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. 637 Kozuka, "Stream Control Transmission Protocol (SCTP) 638 Dynamic Address Reconfiguration", RFC 5061, September 639 2007. 641 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 642 (ICE): A Protocol for Network Address Translator (NAT) 643 Traversal for Offer/Answer Protocols", RFC 5245, April 644 2010. 646 [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control 647 Transmission Protocol (SCTP) Stream Reconfiguration", RFC 648 6525, February 2012. 650 [I-D.ietf-tsvwg-sctp-ndata] 651 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, 652 "Stream Schedulers and a New Data Chunk for the Stream 653 Control Transmission Protocol", draft-ietf-tsvwg-sctp- 654 ndata-01 (work in progress), July 2014. 656 [I-D.ietf-rtcweb-data-protocol] 657 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 658 Establishment Protocol", draft-ietf-rtcweb-data- 659 protocol-07 (work in progress), July 2014. 661 [I-D.ietf-tsvwg-sctp-dtls-encaps] 662 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 663 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 664 dtls-encaps-05 (work in progress), July 2014. 666 [I-D.ietf-rtcweb-security] 667 Rescorla, E., "Security Considerations for WebRTC", draft- 668 ietf-rtcweb-security-07 (work in progress), July 2014. 670 [I-D.ietf-rtcweb-security-arch] 671 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 672 rtcweb-security-arch-10 (work in progress), July 2014. 674 [I-D.ietf-rtcweb-jsep] 675 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 676 Session Establishment Protocol", draft-ietf-rtcweb-jsep-07 677 (work in progress), July 2014. 679 [I-D.ietf-tsvwg-sctp-prpolicies] 680 Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, 681 "Additional Policies for the Partial Reliability Extension 682 of the Stream Control Transmission Protocol", draft-ietf- 683 tsvwg-sctp-prpolicies-03 (work in progress), May 2014. 685 [I-D.ietf-mmusic-sctp-sdp] 686 Loreto, S. and G. Camarillo, "Stream Control Transmission 687 Protocol (SCTP)-Based Media Transport in the Session 688 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-07 689 (work in progress), July 2014. 691 10.2. Informative References 693 [RFC1122] Braden, R., "Requirements for Internet Hosts - 694 Communication Layers", STD 3, RFC 1122, October 1989. 696 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 697 Security (DTLS) Extension to Establish Keys for the Secure 698 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 700 [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram 701 Transport Layer Security (DTLS) for Stream Control 702 Transmission Protocol (SCTP)", RFC 6083, January 2011. 704 [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream 705 Control Transmission Protocol (SCTP) Packets for End-Host 706 to End-Host Communication", RFC 6951, May 2013. 708 [I-D.ietf-rtcweb-use-cases-and-requirements] 709 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 710 Time Communication Use-cases and Requirements", draft- 711 ietf-rtcweb-use-cases-and-requirements-14 (work in 712 progress), February 2014. 714 Authors' Addresses 716 Randell Jesup 717 Mozilla 718 US 720 Email: randell-ietf@jesup.org 722 Salvatore Loreto 723 Ericsson 724 Hirsalantie 11 725 Jorvas 02420 726 FI 728 Email: salvatore.loreto@ericsson.com 730 Michael Tuexen 731 Muenster University of Applied Sciences 732 Stegerwaldstrasse 39 733 Steinfurt 48565 734 DE 736 Email: tuexen@fh-muenster.de