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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: July 8, 2015 Ericsson 6 M. Tuexen 7 Muenster Univ. of Appl. Sciences 8 January 4, 2015 10 WebRTC Data Channels 11 draft-ietf-rtcweb-data-channel-13.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 July 8, 2015. 41 Copyright Notice 43 Copyright (c) 2015 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 . . . . . . . . . 4 62 3.2. Use Cases for Reliable Data Channels . . . . . . . . . . 4 63 4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4 64 5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . 6 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 . . . . . . . . . . . . . . . . . 10 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 . . . . . . . . . . . . . . . . . . . 13 74 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13 75 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 14 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. DTLS 1.0 is 87 defined in [RFC4347] and the present latest version, DTLS 1.2, is 88 defined in [RFC6347]. 90 +----------+ 91 | SCTP | 92 +----------+ 93 | DTLS | 94 +----------+ 95 | ICE/UDP | 96 +----------+ 98 Figure 1: Basic stack diagram 100 The encapsulation of SCTP over DTLS (see 101 [I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245]) 102 provides a NAT traversal solution together with confidentiality, 103 source authentication, and integrity protected transfers. This data 104 transport service operates in parallel to the SRTP media transports, 105 and all of them can eventually share a single UDP port number. 107 SCTP as specified in [RFC4960] with the partial reliability extension 108 defined in [RFC3758] and the additional policies defined in 109 [I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively 110 with reliable, and the relevant partially-reliable delivery modes for 111 user messages. Using the reconfiguration extension defined in 112 [RFC6525] allows to increase the number of streams during the 113 lifetime of an SCTP association and to reset individual SCTP streams. 114 Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages 115 to avoid the monopolization and adds the support of prioritizing of 116 SCTP streams. 118 The remainder of this document is organized as follows: Section 3 and 119 Section 4 provide use cases and requirements for both unreliable and 120 reliable peer to peer data channels; Section 5 discusses SCTP over 121 DTLS over UDP; Section 6 provides the specification of how SCTP 122 should be used by the WebRTC protocol framework for transporting non- 123 media data between WEB-browsers. 125 2. Conventions 127 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 128 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 129 document are to be interpreted as described in [RFC2119]. 131 3. Use Cases 133 This section defines use cases specific to data channels. Please 134 note that this section is informational only. 136 3.1. Use Cases for Unreliable Data Channels 138 U-C 1: A real-time game where position and object state information 139 is sent via one or more unreliable data channels. Note that 140 at any time there may be no SRTP media channels, or all SRTP 141 media channels may be inactive, and that there may also be 142 reliable data channels in use. 144 U-C 2: Providing non-critical information to a user about the reason 145 for a state update in a video chat or conference, such as 146 mute state. 148 3.2. Use Cases for Reliable Data Channels 150 U-C 3: A real-time game where critical state information needs to be 151 transferred, such as control information. Such a game may 152 have no SRTP media channels, or they may be inactive at any 153 given time, or may only be added due to in-game actions. 155 U-C 4: Non-realtime file transfers between people chatting. Note 156 that this may involve a large number of files to transfer 157 sequentially or in parallel, such as when sharing a folder of 158 images or a directory of files. 160 U-C 5: Realtime text chat during an audio and/or video call with an 161 individual or with multiple people in a conference. 163 U-C 6: Renegotiation of the configuration of the PeerConnection. 165 U-C 7: Proxy browsing, where a browser uses data channels of a 166 PeerConnection to send and receive HTTP/HTTPS requests and 167 data, for example to avoid local Internet filtering or 168 monitoring. 170 4. Requirements 172 This section lists the requirements for P2P data channels between two 173 browsers. Please note that this section is informational only. 175 Req. 1: Multiple simultaneous data channels must be supported. 176 Note that there may be 0 or more SRTP media streams in 177 parallel with the data channels in the same PeerConnection, 178 and the number and state (active/inactive) of these SRTP 179 media streams may change at any time. 181 Req. 2: Both reliable and unreliable data channels must be 182 supported. 184 Req. 3: Data channels of a PeerConnection must be congestion 185 controlled; either individually, as a class, or in 186 conjunction with the SRTP media streams of the 187 PeerConnection, to ensure that data channels don't cause 188 congestion problems for these SRTP media streams, and that 189 the WebRTC PeerConnection does not cause excessive problems 190 when run in parallel with TCP connections. 192 Req. 4: The application should be able to provide guidance as to 193 the relative priority of each data channel relative to each 194 other, and relative to the SRTP media streams. This will 195 interact with the congestion control algorithms. 197 Req. 5: Data channels must be secured; allowing for 198 confidentiality, integrity and source authentication. See 199 [I-D.ietf-rtcweb-security] and 200 [I-D.ietf-rtcweb-security-arch] for detailed info. 202 Req. 6: Data channels must provide message fragmentation support 203 such that IP-layer fragmentation can be avoided no matter 204 how large a message the JavaScript application passes to be 205 sent. It also must ensure that large data channel 206 transfers don't unduly delay traffic on other data 207 channels. 209 Req. 7: The data channel transport protocol must not encode local 210 IP addresses inside its protocol fields; doing so reveals 211 potentially private information, and leads to failure if 212 the address is depended upon. 214 Req. 8: The data channel transport protocol should support 215 unbounded-length "messages" (i.e., a virtual socket stream) 216 at the application layer, for such things as image-file- 217 transfer; Implementations might enforce a reasonable 218 message size limit. 220 Req. 9: The data channel transport protocol should avoid IP 221 fragmentation. It must support PMTU (Path MTU) discovery 222 and must not rely on ICMP or ICMPv6 being generated or 223 being passed back, especially for PMTU discovery. 225 Req. 10: It must be possible to implement the protocol stack in the 226 user application space. 228 5. SCTP over DTLS over UDP Considerations 230 The important features of SCTP in the WebRTC context are: 232 o Usage of a TCP-friendly congestion control. 234 o The congestion control is modifiable for integration with the SRTP 235 media stream congestion control. 237 o Support of multiple unidirectional streams, each providing its own 238 notion of ordered message delivery. 240 o Support of ordered and out-of-order message delivery. 242 o Supporting arbitrary large user messages by providing 243 fragmentation and reassembly. 245 o Support of PMTU-discovery. 247 o Support of reliable or partially reliable message transport. 249 The WebRTC Data Channel mechanism does not support SCTP multihoming. 250 The SCTP layer will simply act as if it were running on a single- 251 homed host, since that is the abstraction that the DTLS layer (a 252 connection oriented, unreliable datagram service) exposes. 254 The encapsulation of SCTP over DTLS defined in 255 [I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source 256 authenticated, and integrity protected transfers. Using DTLS over 257 UDP in combination with ICE enables middlebox traversal in IPv4 and 258 IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in 259 combination with the extension defined in [RFC3758] and provides the 260 following features for transporting non-media data between browsers: 262 o Support of multiple unidirectional streams. 264 o Ordered and unordered delivery of user messages. 266 o Reliable and partial-reliable transport of user messages. 268 Each SCTP user message contains a Payload Protocol Identifier (PPID) 269 that is passed to SCTP by its upper layer on the sending side and 270 provided to its upper layer on the receiving side. The PPID can be 271 used to multiplex/demultiplex multiple upper layers over a single 272 SCTP association. In the WebRTC context, the PPID is used to 273 distinguish between UTF-8 encoded user data, binary encoded userdata 274 and the Data Channel Establishment Protocol defined in 276 [I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not 277 accessible via the Javascript API. 279 The encapsulation of SCTP over DTLS, together with the SCTP features 280 listed above satisfies all the requirements listed in Section 4. 282 The layering of protocols for WebRTC is shown in the following 283 Figure 2. 285 +------+------+------+ 286 | DCEP | UTF-8|Binary| 287 | | data | data | 288 +------+------+------+ 289 | SCTP | 290 +----------------------------------+ 291 | STUN | SRTP | DTLS | 292 +----------------------------------+ 293 | ICE | 294 +----------------------------------+ 295 | UDP1 | UDP2 | UDP3 | ... | 296 +----------------------------------+ 298 Figure 2: WebRTC protocol layers 300 This stack (especially in contrast to DTLS over SCTP [RFC6083] in 301 combination with SCTP over UDP [RFC6951]) has been chosen because it 303 o supports the transmission of arbitrary large user messages. 305 o shares the DTLS connection with the SRTP media channels of the 306 PeerConnection. 308 o provides privacy for the SCTP control information. 310 Considering the protocol stack of Figure 2 the usage of DTLS 1.0 over 311 UDP is specified in [RFC4347] and the usage of DTLS 1.2 over UDP in 312 specified in [RFC6347], while the usage of SCTP on top of DTLS is 313 specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the 314 demultiplexing STUN vs. SRTP vs. DTLS is done as described in 315 Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS. 317 Since DTLS is typically implemented in user application space, the 318 SCTP stack also needs to be a user application space stack. 320 The ICE/UDP layer can handle IP address changes during a session 321 without needing interaction with the DTLS and SCTP layers. However, 322 SCTP SHOULD be notified when an address changes has happened. In 323 this case SCTP SHOULD retest the Path MTU and reset the congestion 324 state to the initial state. In case of a window based congestion 325 control like the one specified in [RFC4960], this means setting the 326 congestion window and slow start threshold to its initial values. 328 Incoming ICMP or ICMPv6 messages can't be processed by the SCTP 329 layer, since there is no way to identify the corresponding 330 association. Therefore SCTP MUST support performing Path MTU 331 discovery without relying on ICMP or ICMPv6 as specified in [RFC4821] 332 using probing messages specified in [RFC4820]. The initial Path MTU 333 at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for 334 IPv6. 336 In general, the lower layer interface of an SCTP implementation 337 should be adapted to address the differences between IPv4 and IPv6 338 (being connection-less) or DTLS (being connection-oriented). 340 When the protocol stack of Figure 2 is used, DTLS protects the 341 complete SCTP packet, so it provides confidentiality, integrity and 342 source authentication of the complete SCTP packet. 344 SCTP provides congestion control on a per-association base. This 345 means that all SCTP streams within a single SCTP association share 346 the same congestion window. Traffic not being sent over SCTP is not 347 covered by the SCTP congestion control. Using a congestion control 348 different from than the standard one might improve the impact on the 349 parallel SRTP media streams. 351 SCTP uses the same port number concept as TCP and UDP do. Therefore 352 an SCTP association uses two port numbers, one at each SCTP end- 353 point. 355 6. The Usage of SCTP for Data Channels 357 6.1. SCTP Protocol Considerations 359 The DTLS encapsulation of SCTP packets as described in 360 [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used. 362 This SCTP stack and its upper layer MUST support the usage of 363 multiple SCTP streams. A user message can be sent ordered or 364 unordered and with partial or full reliability. 366 The following SCTP protocol extensions are required: 368 o The stream reconfiguration extension defined in [RFC6525] MUST be 369 supported. It is used for closing channels. 371 o The dynamic address reconfiguration extension defined in [RFC5061] 372 MUST be used to signal the support of the stream reset extension 373 defined in [RFC6525]. Other features of [RFC5061] are OPTIONAL. 375 o The partial reliability extension defined in [RFC3758] MUST be 376 supported. In addition to the timed reliability PR-SCTP policy 377 defined in [RFC3758], the limited retransmission policy defined in 378 [I-D.ietf-tsvwg-sctp-prpolicies] MUST be supported. Limiting the 379 number of retransmissions to zero combined with unordered delivery 380 provides a UDP-like service where each user message is sent 381 exactly once and delivered in the order received. 383 The support for message interleaving as defined in 384 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used. 386 6.2. SCTP Association Management 388 In the WebRTC context, the SCTP association will be set up when the 389 two endpoints of the WebRTC PeerConnection agree on opening it, as 390 negotiated by JSEP (typically an exchange of SDP) 391 [I-D.ietf-rtcweb-jsep]. It will use the DTLS connection selected via 392 ICE; typically this will be shared via BUNDLE or equivalent with DTLS 393 connections used to key the SRTP media streams. 395 The number of streams negotiated during SCTP association setup SHOULD 396 be 65535, which is the maximum number of streams that can be 397 negotiated during the association setup. 399 SCTP supports two ways of terminating an SCTP association. A 400 graceful one, using a procedure which ensures that no messages are 401 lost during the shutdown of the association. The second method is a 402 non-graceful one, where one side can just abort the association. 404 Each SCTP end-point supervises continuously the reachability of its 405 peer by monitoring the number of retransmissions of user messages and 406 test messages. In case of excessive retransmissions, the association 407 is terminated in a non-graceful way. 409 If an SCTP association is closed in a graceful way, all of its data 410 channels are closed. In case of a non-graceful teardown, all data 411 channels are also closed, but an error indication SHOULD be provided 412 if possible. 414 6.3. SCTP Streams 416 SCTP defines a stream as a unidirectional logical channel existing 417 within an SCTP association to another SCTP endpoint. The streams are 418 used to provide the notion of in-sequence delivery and for 419 multiplexing. Each user message is sent on a particular stream, 420 either ordered or unordered. Ordering is preserved only for ordered 421 messages sent on the same stream. 423 6.4. Data Channel Definition 425 Data channels are defined such that their accompanying application- 426 level API can closely mirror the API for WebSockets, which implies 427 bidirectional streams of data and a textual field called 'label' used 428 to identify the meaning of the data channel. 430 The realization of a data channel is a pair of one incoming stream 431 and one outgoing SCTP stream having the same SCTP stream identifier. 432 How these SCTP stream identifiers are selected is protocol and 433 implementation dependent. This allows a bidirectional communication. 435 Additionally, each data channel has the following properties in each 436 direction: 438 o reliable or unreliable message transmission. In case of 439 unreliable transmissions, the same level of unreliability is used. 440 Please note that in SCTP this is a property of an SCTP user 441 message and not of an SCTP stream. 443 o in-order or out-of-order message delivery for message sent. 444 Please note that in SCTP this is a property of an SCTP user 445 message and not of an SCTP stream. 447 o A priority, which is a 2 byte unsigned integer. These priorities 448 MUST be interpreted as weighted-fair-queuing scheduling priorities 449 per the definition of the corresponding stream scheduler 450 supporting interleaving in [I-D.ietf-tsvwg-sctp-ndata]. For use 451 in WebRTC, the values used SHOULD be one of 128 ("below normal"), 452 256 ("normal"), 512 ("high") or 1024 ("extra high"). 454 o an optional label. 456 o an optional protocol. 458 Please note that for a data channel being negotiated with the 459 protocol specified in [I-D.ietf-rtcweb-data-protocol] all of the 460 above properties are the same in both directions. 462 6.5. Opening a Data Channel 464 Data channels can be opened by using negotiation within the SCTP 465 association, called in-band negotiation, or out-of-band negotiation. 466 Out-of-band negotiation is defined as any method which results in an 467 agreement as to the parameters of a channel and the creation thereof. 468 The details are out of scope of this document. Applications using 469 data channels need to use the negotiation methods consistently on 470 both end-points. 472 A simple protocol for in-band negotiation is specified in 473 [I-D.ietf-rtcweb-data-protocol]. 475 When one side wants to open a channel using out-of-band negotiation, 476 it picks a stream. Unless otherwise defined or negotiated, the 477 streams are picked based on the DTLS role (the client picks even 478 stream identifiers, the server odd stream identifiers). However, the 479 application is responsible for avoiding collisions with existing 480 streams. If it attempts to re-use a stream which is part of an 481 existing data channel, the addition MUST fail. In addition to 482 choosing a stream, the application SHOULD also determine the options 483 to use for sending messages. The application MUST ensure in an 484 application-specific manner that the application at the peer will 485 also know the selected stream to be used, and the options for sending 486 data from that side. 488 6.6. Transferring User Data on a Data Channel 490 All data sent on a data channel in both directions MUST be sent over 491 the underlying stream using the reliability defined when the data 492 channel was opened unless the options are changed, or per-message 493 options are specified by a higher level. 495 The message-orientation of SCTP is used to preserve the message 496 boundaries of user messages. Therefore, senders MUST NOT put more 497 than one application message into an SCTP user message. Unless the 498 deprecated PPID-based fragmentation and reassembly is used, the 499 sender MUST include exactly one application message in each SCTP user 500 message. 502 The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the 503 interpretation of the "Payload data". The following PPIDs MUST be 504 used (see Section 8): 506 WebRTC String: to identify a non-empty JavaScript string encoded in 507 UTF-8. 509 WebRTC String Empty: to identify an empty JavaScript string encoded 510 in UTF-8. 512 WebRTC Binary: to identify a non-empty JavaScript binary data 513 (ArrayBuffer, ArrayBufferView or Blob). 515 WebRTC Binary Empty: to identify an empty JavaScript binary data 516 (ArrayBuffer, ArrayBufferView or Blob). 518 SCTP does not support the sending of empty user messages. Therefore, 519 if an empty message has to be sent, the appropriate PPID (WebRTC 520 String Empty or WebRTC Binary Empty) is used and the SCTP user 521 message of one zero byte is sent. When receiving an SCTP user 522 message with one of these PPIDs, the receiver MUST ignore the SCTP 523 user message and process it as an empty message. 525 The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary 526 Partial" is deprecated. They were used for a PPID-based 527 fragmentation and reassembly of user messages belonging to reliable 528 and ordered data channels. 530 If a message with an unsupported PPID is received or some error 531 condition related to the received message is detected by the receiver 532 (for example, illegal ordering), the receiver SHOULD close the 533 corresponding data channel. This implies in particular that 534 extensions using additional PPIDs can't be used without prior 535 negotiation. 537 The SCTP base protocol specified in [RFC4960] does not support the 538 interleaving of user messages. Therefore sending a large user 539 message can monopolize the SCTP association. To overcome this 540 limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to 541 support message interleaving, which SHOULD be used. As long as 542 message interleaving is not supported, the sender SHOULD limit the 543 maximum message size to 16 KB to avoid monopolization. 545 It is recommended that the message size be kept within certain size 546 bounds as applications will not be able to support arbitrarily-large 547 single messages. This limit has to be negotiated, for example by 548 using [I-D.ietf-mmusic-sctp-sdp]. 550 The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to 551 minimize the latency. 553 6.7. Closing a Data Channel 555 Closing of a data channel MUST be signaled by resetting the 556 corresponding outgoing streams [RFC6525]. This means that if one 557 side decides to close the data channel, it resets the corresponding 558 outgoing stream. When the peer sees that an incoming stream was 559 reset, it also resets its corresponding outgoing stream. Once this 560 is completed, the data channel is closed. Resetting a stream sets 561 the Stream Sequence Numbers (SSNs) of the stream back to 'zero' with 562 a corresponding notification to the application layer that the reset 563 has been performed. Streams are available for reuse after a reset 564 has been performed. 566 [RFC6525] also guarantees that all the messages are delivered (or 567 abandoned) before the stream is reset. 569 7. Security Considerations 571 This document does not add any additional considerations to the ones 572 given in [I-D.ietf-rtcweb-security] and 573 [I-D.ietf-rtcweb-security-arch]. 575 It should be noted that a receiver must be prepared that the sender 576 tries to send arbitrary large messages. 578 8. IANA Considerations 580 [NOTE to RFC-Editor: 582 "RFCXXXX" is to be replaced by the RFC number you assign this 583 document. 585 ] 587 This document uses six already registered SCTP Payload Protocol 588 Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary 589 Data Last", "DOMString Partial", "WebRTC String Empty", and "WebRTC 590 Binary Empty". [RFC4960] creates the registry "SCTP Payload Protocol 591 Identifiers" from which these identifiers were assigned. IANA is 592 requested to update the reference of these six assignments to point 593 to this document and change the names of the first four PPIDs. The 594 corresponding dates should be kept. 596 Therefore these six assignments should be updated to read: 598 +-------------------------------+----------+-----------+------------+ 599 | Value | SCTP | Reference | Date | 600 | | PPID | | | 601 +-------------------------------+----------+-----------+------------+ 602 | WebRTC String | 51 | [RFCXXXX] | 2013-09-20 | 603 | WebRTC Binary Partial | 52 | [RFCXXXX] | 2013-09-20 | 604 | (Deprecated) | | | | 605 | WebRTC Binary | 53 | [RFCXXXX] | 2013-09-20 | 606 | WebRTC String Partial | 54 | [RFCXXXX] | 2013-09-20 | 607 | (Deprecated) | | | | 608 | WebRTC String Empty | 56 | [RFCXXXX] | 2014-08-22 | 609 | WebRTC Binary Empty | 57 | [RFCXXXX] | 2014-08-22 | 610 +-------------------------------+----------+-----------+------------+ 612 9. Acknowledgments 614 Many thanks for comments, ideas, and text from Harald Alvestrand, 615 Richard Barnes, Adam Bergkvist, Alissa Cooper, Benoit Claise, Spencer 616 Dawkins, Gunnar Hellstrom, Christer Holmberg, Cullen Jennings, Paul 617 Kyzivat, Eric Rescorla, Adam Roach, Irene Ruengeler, Randall Stewart, 618 Martin Stiemerling, Justin Uberti, and Magnus Westerlund. 620 10. References 622 10.1. Normative References 624 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 625 Requirement Levels", BCP 14, RFC 2119, March 1997. 627 [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. 628 Conrad, "Stream Control Transmission Protocol (SCTP) 629 Partial Reliability Extension", RFC 3758, May 2004. 631 [RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer 632 Security", RFC 4347, April 2006. 634 [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and 635 Parameter for the Stream Control Transmission Protocol 636 (SCTP)", RFC 4820, March 2007. 638 [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU 639 Discovery", RFC 4821, March 2007. 641 [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 642 4960, September 2007. 644 [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. 645 Kozuka, "Stream Control Transmission Protocol (SCTP) 646 Dynamic Address Reconfiguration", RFC 5061, September 647 2007. 649 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 650 (ICE): A Protocol for Network Address Translator (NAT) 651 Traversal for Offer/Answer Protocols", RFC 5245, April 652 2010. 654 [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer 655 Security Version 1.2", RFC 6347, January 2012. 657 [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control 658 Transmission Protocol (SCTP) Stream Reconfiguration", RFC 659 6525, February 2012. 661 [I-D.ietf-tsvwg-sctp-ndata] 662 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, 663 "Stream Schedulers and a New Data Chunk for the Stream 664 Control Transmission Protocol", draft-ietf-tsvwg-sctp- 665 ndata-01 (work in progress), July 2014. 667 [I-D.ietf-rtcweb-data-protocol] 668 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 669 Establishment Protocol", draft-ietf-rtcweb-data- 670 protocol-08 (work in progress), September 2014. 672 [I-D.ietf-tsvwg-sctp-dtls-encaps] 673 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 674 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 675 dtls-encaps-07 (work in progress), December 2014. 677 [I-D.ietf-rtcweb-security] 678 Rescorla, E., "Security Considerations for WebRTC", draft- 679 ietf-rtcweb-security-07 (work in progress), July 2014. 681 [I-D.ietf-rtcweb-security-arch] 682 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 683 rtcweb-security-arch-10 (work in progress), July 2014. 685 [I-D.ietf-rtcweb-jsep] 686 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 687 Session Establishment Protocol", draft-ietf-rtcweb-jsep-08 688 (work in progress), October 2014. 690 [I-D.ietf-tsvwg-sctp-prpolicies] 691 Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, 692 "Additional Policies for the Partial Reliability Extension 693 of the Stream Control Transmission Protocol", draft-ietf- 694 tsvwg-sctp-prpolicies-06 (work in progress), December 695 2014. 697 [I-D.ietf-mmusic-sctp-sdp] 698 Holmberg, C., Loreto, S., and G. Camarillo, "Stream 699 Control Transmission Protocol (SCTP)-Based Media Transport 700 in the Session Description Protocol (SDP)", draft-ietf- 701 mmusic-sctp-sdp-11 (work in progress), December 2014. 703 10.2. Informative References 705 [RFC1122] Braden, R., "Requirements for Internet Hosts - 706 Communication Layers", STD 3, RFC 1122, October 1989. 708 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 709 Security (DTLS) Extension to Establish Keys for the Secure 710 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 712 [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram 713 Transport Layer Security (DTLS) for Stream Control 714 Transmission Protocol (SCTP)", RFC 6083, January 2011. 716 [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream 717 Control Transmission Protocol (SCTP) Packets for End-Host 718 to End-Host Communication", RFC 6951, May 2013. 720 Authors' Addresses 722 Randell Jesup 723 Mozilla 724 US 726 Email: randell-ietf@jesup.org 728 Salvatore Loreto 729 Ericsson 730 Hirsalantie 11 731 Jorvas 02420 732 FI 734 Email: salvatore.loreto@ericsson.com 736 Michael Tuexen 737 Muenster University of Applied Sciences 738 Stegerwaldstrasse 39 739 Steinfurt 48565 740 DE 742 Email: tuexen@fh-muenster.de