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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: October 11, 2014 Ericsson 6 M. Tuexen 7 Muenster Univ. of Appl. Sciences 8 April 9, 2014 10 WebRTC Data Channels 11 draft-ietf-rtcweb-data-channel-08.txt 13 Abstract 15 The Real-Time Communication in WEB-browsers working group is charged 16 to provide protocol support for direct interactive rich communication 17 using audio, video, and data between two peers' web-browsers. This 18 document specifies the non-(S)RTP media data transport aspects of the 19 WebRTC framework. It provides an architectural overview of how the 20 Stream Control Transmission Protocol (SCTP) is used in the WebRTC 21 context as a generic transport service allowing WEB-browsers to 22 exchange generic data from peer to 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 October 11, 2014. 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. Association Setup . . . . . . . . . . . . . . . . . . . . 9 68 6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9 69 6.4. Channel Definition . . . . . . . . . . . . . . . . . . . 9 70 6.5. Opening a Channel . . . . . . . . . . . . . . . . . . . . 10 71 6.6. Transferring User Data on a Channel . . . . . . . . . . . 10 72 6.7. Closing a Channel . . . . . . . . . . . . . . . . . . . . 11 73 7. Security Considerations . . . . . . . . . . . . . . . . . . . 11 74 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11 75 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12 76 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12 77 10.1. Normative References . . . . . . . . . . . . . . . . . . 12 78 10.2. Informative References . . . . . . . . . . . . . . . . . 14 79 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14 81 1. Introduction 83 Non-(S)RTP media data types in the context of WebRTC are handled by 84 using SCTP [RFC4960] encapsulated in DTLS [RFC6347]. 86 +----------+ 87 | SCTP | 88 +----------+ 89 | DTLS | 90 +----------+ 91 | ICE/UDP | 92 +----------+ 94 Figure 1: Basic stack diagram 96 The encapsulation of SCTP over DTLS (see 97 [I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245]) 98 provides a NAT traversal solution together with confidentiality, 99 source authentication, and integrity protected transfers. This data 100 transport service operates in parallel to the (S)RTP media 101 transports, and all of them can eventually share a single transport- 102 layer port number. 104 SCTP as specified in [RFC4960] with the partial reliability extension 105 defined in [RFC3758] and the additional policies defined in 106 [I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively 107 with reliable, and the relevant partially-reliable delivery modes for 108 user messages. Using the reconfiguration extension defined in 109 [RFC6525] allows to increase the number of streams during the 110 lifetime of an SCTP association and to reset individual SCTP streams. 111 Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages 112 to avoid the monopolization and adds the support of prioritizing of 113 SCTP streams. 115 The remainder of this document is organized as follows: Section 3 and 116 Section 4 provide use cases and requirements for both unreliable and 117 reliable peer to peer data channels; Section 5 discusses SCTP over 118 DTLS over UDP; Section 6 provides the specification of how SCTP 119 should be used by the WebRTC protocol framework for transporting 120 non-(S)RTP media data between WEB-browsers. 122 2. Conventions 124 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 125 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 126 document are to be interpreted as described in [RFC2119]. 128 3. Use Cases 130 This section defines use cases specific to data channels. For 131 general use cases see [I-D.ietf-rtcweb-use-cases-and-requirements]. 133 3.1. Use Cases for Unreliable Data Channels 135 U-C 1: A real-time game where position and object state information 136 is sent via one or more unreliable data channels. Note that 137 at any time there may be no (S)RTP media channels, or all 138 (S)RTP media channels may be inactive, and that there may 139 also be reliable data channels in use. 141 U-C 2: Providing non-critical information to a user about the reason 142 for a state update in a video chat or conference, such as 143 mute state. 145 3.2. Use Cases for Reliable Data Channels 147 U-C 3: A real-time game where critical state information needs to be 148 transferred, such as control information. Such a game may 149 have no (S)RTP media channels, or they may be inactive at any 150 given time, or may only be added due to in-game actions. 152 U-C 4: Non-realtime file transfers between people chatting. Note 153 that this may involve a large number of files to transfer 154 sequentially or in parallel, such as when sharing a folder of 155 images or a directory of files. 157 U-C 5: Realtime text chat during an audio and/or video call with an 158 individual or with multiple people in a conference. 160 U-C 6: Renegotiation of the configuration of the PeerConnection. 162 U-C 7: Proxy browsing, where a browser uses data channels of a 163 PeerConnection to send and receive HTTP/HTTPS requests and 164 data, for example to avoid local Internet filtering or 165 monitoring. 167 4. Requirements 169 This section lists the requirements for P2P data channels between two 170 browsers. 172 Req. 1: Multiple simultaneous data channels MUST be supported. 173 Note that there may be 0 or more (S)RTP media streams in 174 parallel with the data channels in the same PeerConnection, 175 and the number and state (active/inactive) of these (S)RTP 176 media streams may change at any time. 178 Req. 2: Both reliable and unreliable data channels MUST be 179 supported. 181 Req. 3: Data channels of a PeerConnection MUST be congestion 182 controlled; either individually, as a class, or in 183 conjunction with the (S)RTP media streams of the 184 PeerConnection, to ensure that data channels don't cause 185 congestion problems for these (S)RTP media streams, and 186 that the WebRTC PeerConnection as a whole is fair with 187 competing traffic such as TCP. 189 Req. 4: The application SHOULD be able to provide guidance as to 190 the relative priority of each data channel relative to each 191 other, and relative to the (S)RTP media streams. This will 192 interact with the congestion control algorithms. 194 Req. 5: Data channels MUST be secured; allowing for 195 confidentiality, integrity and source authentication. See 196 [I-D.ietf-rtcweb-security] and 197 [I-D.ietf-rtcweb-security-arch] for detailed info. 199 Req. 6: Data channels MUST provide message fragmentation support 200 such that IP-layer fragmentation can be avoided no matter 201 how large a message the JavaScript application passes to be 202 sent. It also MUST ensure that large data channel 203 transfers don't unduly delay traffic on other data 204 channels. 206 Req. 7: The data channel transport protocol MUST NOT encode local 207 IP addresses inside its protocol fields; doing so reveals 208 potentially private information, and leads to failure if 209 the address is depended upon. 211 Req. 8: The data channel transport protocol SHOULD support 212 unbounded-length "messages" (i.e., a virtual socket stream) 213 at the application layer, for such things as image-file- 214 transfer; Implementations might enforce a reasonable 215 message size limit. 217 Req. 9: The data channel transport protocol SHOULD avoid IP 218 fragmentation. It MUST support PMTU (Path MTU) discovery 219 and MUST NOT rely on ICMP or ICMPv6 being generated or 220 being passed back, especially for PMTU discovery. 222 Req. 10: It MUST be possible to implement the protocol stack in the 223 user application space. 225 5. SCTP over DTLS over UDP Considerations 227 The important features of SCTP in the WebRTC context are: 229 o Usage of a TCP-friendly congestion control. 231 o The congestion control is modifiable for integration with the 232 (S)RTP media stream congestion control. 234 o Support of multiple unidirectional streams, each providing its own 235 notion of ordered message delivery. 237 o Support of ordered and out-of-order message delivery. 239 o Supporting arbitrary large user messages by providing 240 fragmentation and reassembly. 242 o Support of PMTU-discovery. 244 o Support of reliable or partially reliable message transport. 246 SCTP multihoming will not be used in WebRTC. The SCTP layer will 247 simply act as if it were running on a single-homed host, since that 248 is the abstraction that the lower layer (a connection oriented, 249 unreliable datagram service) exposes. 251 The encapsulation of SCTP over DTLS defined in 252 [I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source 253 authenticated, and integrity protected transfers. Using DTLS over 254 UDP in combination with ICE enables middlebox traversal in IPv4 and 255 IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in 256 combination with the extension defined in [RFC3758] and provides the 257 following features for transporting non-(S)RTP media data between 258 browsers: 260 o Support of multiple unidirectional streams. 262 o Ordered and unordered delivery of user messages. 264 o Reliable and partial-reliable transport of user messages. 266 Each SCTP user message contains a Payload Protocol Identifier (PPID) 267 that is passed to SCTP by its upper layer on the sending side and 268 provided to its upper layer on the receiving side. The PPID can be 269 used to multiplex/demultiplex multiple upper layers over a single 270 SCTP association. In the WebRTP context, the PPID is used to 271 distinguish between UTF-8 encoded user data, binary encoded userdata 272 and the Data Channel Establishment Protocol defined in 273 [I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not 274 accessible via the Javascript API. 276 The encapsulation of SCTP over DTLS, together with the SCTP features 277 listed above satisfies all the requirements listed in Section 4. 279 The layering of protocols for WebRTC is shown in the following 280 Figure 2. 282 +------+------+------+ 283 | DCEP | UTF-8|Binary| 284 | | data | data | 285 +------+------+------+ 286 | SCTP | 287 +----------------------------------+ 288 | STUN | SRTP | DTLS | 289 +----------------------------------+ 290 | ICE | 291 +----------------------------------+ 292 | UDP1 | UDP2 | ... | 293 +----------------------------------+ 295 Figure 2: WebRTC protocol layers 297 This stack (especially in contrast to DTLS over SCTP [RFC6083] in 298 combination with SCTP over UDP [RFC6951]) has been chosen because it 300 o supports the transmission of arbitrary large user messages. 302 o shares the DTLS connection with the (S)RTP media channels of the 303 PeerConnection. 305 o provides privacy for the SCTP control information. 307 Considering the protocol stack of Figure 2 the usage of DTLS over UDP 308 is specified in [RFC6347], while the usage of SCTP on top of DTLS is 309 specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the 310 demultiplexing STUN vs. (S)RTP vs. DTLS is done as described in 311 Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS. 313 Since DTLS is typically implemented in user-land, the SCTP stack also 314 needs to be a user-land stack. 316 When using DTLS as the lower layer, only single homed SCTP 317 associations are supported, since DTLS does not expose any address 318 management to its upper layer. The ICE/UDP layer can handle IP 319 address changes during a session without needing interaction with the 320 DTLS and SCTP layers. However, SCTP SHOULD be notified when an 321 address changes has happened. In this case SCTP SHOULD retest the 322 Path MTU and reset the congestion state to the initial state. In 323 case of a window based congestion control like the one specified in 324 [RFC4960], this means setting the congestion window and slow start 325 threshold to its initial values. 327 Incoming ICMP or ICMPv6 messages can't be processed by the SCTP 328 layer, since there is no way to identify the corresponding 329 association. Therefore SCTP MUST support performing Path MTU 330 discovery without relying on ICMP or ICMPv6 as specified in [RFC4821] 331 using probing messages specified in [RFC4820]. The initial Path MTU 332 at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for 333 IPv6. 335 In general, the lower layer interface of an SCTP implementation 336 SHOULD be adapted to address the differences between IPv4 and IPv6 337 (being connection-less) or DTLS (being connection-oriented). 339 When the protocol stack of Figure 2 is used, DTLS protects the 340 complete SCTP packet, so it provides confidentiality, integrity and 341 source authentication of the complete SCTP packet. 343 This SCTP stack and its upper layer MUST support the usage of 344 multiple SCTP streams. A user message can be sent ordered or 345 unordered and with partial or full reliability. The partial 346 reliability extension MUST support policies to limit 348 o the transmission and retransmission by time. 350 o the number of retransmissions. 352 Limiting the number of retransmissions to zero combined with 353 unordered delivery provides a UDP-like service where each user 354 message is sent exactly once and delivered in the order received. 356 SCTP provides congestion control on a per-association base. This 357 means that all SCTP streams within a single SCTP association share 358 the same congestion window. Traffic not being sent over SCTP is not 359 covered by the SCTP congestion control. Using a congestion control 360 different from than the standard one might improve the impact on the 361 parallel (S)RTP media streams. 363 6. The Usage of SCTP for Data Channels 365 6.1. SCTP Protocol Considerations 367 The DTLS encapsulation of SCTP packets as described in 368 [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used. 370 The following SCTP protocol extensions are required: 372 o The stream reset extension defined in [RFC6525] MUST be supported. 373 It is used for closing channels. 375 o The dynamic address reconfiguration extension defined in [RFC5061] 376 MUST be used to signal the support of the stream reset extension 377 defined in [RFC6525], other features of [RFC5061] are not REQUIRED 378 to be implemented. 380 o The partial reliability extension defined in [RFC3758] MUST be 381 supported. In addition to the timed reliability PR-SCTP policy 382 defined in [RFC3758], the limited retransmission policy defined in 383 [I-D.ietf-tsvwg-sctp-prpolicies] MUST be supported. 385 The support for message interleaving as defined in 386 [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used. 388 6.2. Association Setup 390 The SCTP association will be set up when the two endpoints of the 391 WebRTC PeerConnection agree on opening it, as negotiated by JSEP 392 (typically an exchange of SDP) [I-D.ietf-rtcweb-jsep]. It will use 393 the DTLS connection selected via ICE; typically this will be shared 394 via BUNDLE or equivalent with DTLS connections used to key the (S)RTP 395 media streams. 397 The number of streams negotiated during SCTP association setup SHOULD 398 be 65535, which is the maximum number of streams that can negotiated 399 during the association setup. 401 6.3. SCTP Streams 403 SCTP defines a stream as a unidirectional logical channel existing 404 within an SCTP association to another SCTP endpoint. The streams are 405 used to provide the notion of in-sequence delivery and for 406 multiplexing. Each user message is sent on a particular stream, 407 either ordered or unordered. Ordering is preserved only for ordered 408 messages sent on the same stream. 410 6.4. Channel Definition 412 The W3C has consensus on defining the application API for WebRTC 413 DataChannels to be bidirectional. They also consider the notions of 414 in-sequence, out-of-sequence, reliable and unreliable as properties 415 of Channels. One strong wish is for the application-level API to be 416 close to the API for WebSockets, which implies bidirectional streams 417 of data and waiting for onopen to fire before sending, a textual 418 label used to identify the meaning of the stream, among other things. 419 Each data channel also has a priority. These priorities MUST NOT be 420 strict priorities. 422 The realization of a bidirectional Data Channel is a pair of one 423 incoming stream and one outgoing SCTP stream having the same stream 424 SCTP identifier. 426 How stream values are selected is protocol and implementation 427 dependent. 429 6.5. Opening a Channel 431 Data channels can be opened by using negotiation within the SCTP 432 association, called in-band negotiation, or out-of-band negotiation. 433 Out-of-band negotiation is defined as any method which results in an 434 agreement as to the parameters of a channel and the creation thereof. 435 The details are out of scope of this document. 437 A simple protocol for in-band negotiation is specified in 438 [I-D.ietf-rtcweb-data-protocol]. 440 When one side wants to open a channel using out-of-band negotiation, 441 it picks a stream. Unless otherwise defined or negotiated, the 442 streams are picked based on the DTLS role (the client picks even 443 stream identifiers, the server odd stream identifiers). However, the 444 application is responsible for avoiding collisions with existing 445 streams. If it attempts to re-use a stream which is part of an 446 existing Channel, the addition SHOULD fail. In addition to choosing 447 a stream, the application SHOULD also determine the options to use 448 for sending messages. The application MUST ensure in an application- 449 specific manner that the application at the peer will also know the 450 selected stream to be used, and the options for sending data from 451 that side. 453 6.6. Transferring User Data on a Channel 455 All data sent on a Channel in both directions MUST be sent over the 456 underlying stream using the reliability defined when the Channel was 457 opened unless the options are changed, or per-message options are 458 specified by a higher level. 460 No more than one message should be put into an SCTP user message. 462 The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the 463 interpretation of the "Payload data". For identifying a JavaScript 464 string encoded in UTF-8 the PPID "WebRTC String" MUST be used, for 465 JavaScript binary data (ArrayBuffer or Blob) the PPID "WebRTC Binary" 466 MUST be used (see Section 8). 468 The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary 469 Partial" is deprecated. They were used for a PPID-based 470 fragmentation and reassembly of user messages belonging to reliable 471 and ordered data channels. 473 If a message with an unsupported PPID is received or some error is 474 detected by the receiver (for example, illegal ordering), the 475 receiver SHOULD close the corresponding channel. 477 The SCTP base protocol specified in [RFC4960] does not support the 478 interleaving of user messages. Therefore sending a large user 479 message can monopolize the SCTP association. To overcome this 480 limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to 481 support message interleaving, which SHOULD be used. As long as 482 message interleaving is not supported, the sender SHOULD limit the 483 maximum message size to 16 KB to avoid monopolization. 485 It is recommended that the message size be kept within certain size 486 bounds as applications will not be able to support arbitrarily-large 487 single messages. This limit has to be negotiated, for example by 488 using [I-D.ietf-mmusic-sctp-sdp]. 490 The sender SHOULD disable the Nagle algorithm to minimize the 491 latency. 493 6.7. Closing a Channel 495 Closing of a Data Channel MUST be signaled by resetting the 496 corresponding outgoing streams [RFC6525]. This means that if one 497 side decides to close the channel, it resets the corresponding 498 outgoing stream. When the peer sees that an incoming stream was 499 reset, it also resets its corresponding outgoing stream. Once this 500 is completed, the channel is closed. Resetting a stream sets the 501 Stream Sequence Numbers (SSNs) of the stream back to 'zero' with a 502 corresponding notification to the application layer that the reset 503 has been performed. Streams are available to reuse after a reset has 504 been performed. 506 [RFC6525] also guarantees that all the messages are delivered (or 507 abandoned) before resetting the stream. 509 7. Security Considerations 511 This document does not add any additional considerations to the ones 512 given in [I-D.ietf-rtcweb-security] and 513 [I-D.ietf-rtcweb-security-arch]. 515 8. IANA Considerations 517 [NOTE to RFC-Editor: 519 "RFCXXXX" is to be replaced by the RFC number you assign this 520 document. 522 ] 524 This document uses four already registered SCTP Payload Protocol 525 Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary 526 Data Last", and "DOMString Partial". [RFC4960] creates the registry 527 "SCTP Payload Protocol Identifiers" from which these identifiers were 528 assigned. IANA is requested to update the reference of these four 529 assignments to point to this document and change the names of the 530 PPIDs. Therefore these four assignments should be updated to read: 532 +------------------------------------+-----------+-----------+ 533 | Value | SCTP PPID | Reference | 534 +------------------------------------+-----------+-----------+ 535 | WebRTC String | 51 | [RFCXXXX] | 536 | WebRTC Binary Partial (Deprecated) | 52 | [RFCXXXX] | 537 | WebRTC Binary | 53 | [RFCXXXX] | 538 | WebRTC String Partial (Deprecated) | 54 | [RFCXXXX] | 539 +------------------------------------+-----------+-----------+ 541 9. Acknowledgments 543 Many thanks for comments, ideas, and text from Harald Alvestrand, 544 Adam Bergkvist, Christer Holmberg, Cullen Jennings, Paul Kyzivat, 545 Eric Rescorla, Irene Ruengeler, Randall Stewart, Justin Uberti, and 546 Magnus Westerlund. 548 10. References 550 10.1. Normative References 552 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 553 Requirement Levels", BCP 14, RFC 2119, March 1997. 555 [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. 556 Conrad, "Stream Control Transmission Protocol (SCTP) 557 Partial Reliability Extension", RFC 3758, May 2004. 559 [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and 560 Parameter for the Stream Control Transmission Protocol 561 (SCTP)", RFC 4820, March 2007. 563 [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU 564 Discovery", RFC 4821, March 2007. 566 [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 567 4960, September 2007. 569 [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. 570 Kozuka, "Stream Control Transmission Protocol (SCTP) 571 Dynamic Address Reconfiguration", RFC 5061, September 572 2007. 574 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 575 (ICE): A Protocol for Network Address Translator (NAT) 576 Traversal for Offer/Answer Protocols", RFC 5245, April 577 2010. 579 [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer 580 Security Version 1.2", RFC 6347, January 2012. 582 [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control 583 Transmission Protocol (SCTP) Stream Reconfiguration", RFC 584 6525, February 2012. 586 [I-D.ietf-tsvwg-sctp-ndata] 587 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A 588 New Data Chunk for Stream Control Transmission Protocol", 589 draft-ietf-tsvwg-sctp-ndata-00 (work in progress), 590 February 2014. 592 [I-D.ietf-rtcweb-data-protocol] 593 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 594 Establishment Protocol", draft-ietf-rtcweb-data- 595 protocol-03 (work in progress), February 2014. 597 [I-D.ietf-tsvwg-sctp-dtls-encaps] 598 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 599 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 600 dtls-encaps-03 (work in progress), February 2014. 602 [I-D.ietf-rtcweb-security] 603 Rescorla, E., "Security Considerations for WebRTC", draft- 604 ietf-rtcweb-security-06 (work in progress), January 2014. 606 [I-D.ietf-rtcweb-security-arch] 607 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 608 rtcweb-security-arch-09 (work in progress), February 2014. 610 [I-D.ietf-rtcweb-jsep] 611 Uberti, J. and C. Jennings, "Javascript Session 612 Establishment Protocol", draft-ietf-rtcweb-jsep-06 (work 613 in progress), February 2014. 615 [I-D.ietf-tsvwg-sctp-prpolicies] 616 Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, 617 "Additional Policies for the Partial Reliability Extension 618 of the Stream Control Transmission Protocol", draft-ietf- 619 tsvwg-sctp-prpolicies-02 (work in progress), April 2014. 621 [I-D.ietf-mmusic-sctp-sdp] 622 Loreto, S. and G. Camarillo, "Stream Control Transmission 623 Protocol (SCTP)-Based Media Transport in the Session 624 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-06 625 (work in progress), February 2014. 627 10.2. Informative References 629 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 630 Security (DTLS) Extension to Establish Keys for the Secure 631 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 633 [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram 634 Transport Layer Security (DTLS) for Stream Control 635 Transmission Protocol (SCTP)", RFC 6083, January 2011. 637 [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream 638 Control Transmission Protocol (SCTP) Packets for End-Host 639 to End-Host Communication", RFC 6951, May 2013. 641 [I-D.ietf-rtcweb-use-cases-and-requirements] 642 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 643 Time Communication Use-cases and Requirements", draft- 644 ietf-rtcweb-use-cases-and-requirements-14 (work in 645 progress), February 2014. 647 Authors' Addresses 649 Randell Jesup 650 Mozilla 651 US 653 Email: randell-ietf@jesup.org 655 Salvatore Loreto 656 Ericsson 657 Hirsalantie 11 658 Jorvas 02420 659 FI 661 Email: salvatore.loreto@ericsson.com 662 Michael Tuexen 663 Muenster University of Applied Sciences 664 Stegerwaldstrasse 39 665 Steinfurt 48565 666 DE 668 Email: tuexen@fh-muenster.de