--- 1/draft-ietf-rtcweb-transports-08.txt 2015-07-06 15:15:05.350447869 -0700 +++ 2/draft-ietf-rtcweb-transports-09.txt 2015-07-06 15:15:05.390448870 -0700 @@ -1,18 +1,18 @@ Network Working Group H. Alvestrand Internet-Draft Google -Intended status: Standards Track February 27, 2015 -Expires: August 31, 2015 +Intended status: Standards Track July 6, 2015 +Expires: January 7, 2016 Transports for WebRTC - draft-ietf-rtcweb-transports-08 + draft-ietf-rtcweb-transports-09 Abstract This document describes the data transport protocols used by WebRTC, including the protocols used for interaction with intermediate boxes such as firewalls, relays and NAT boxes. Status of This Memo This Internet-Draft is submitted in full conformance with the @@ -21,21 +21,21 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on August 31, 2015. + This Internet-Draft will expire on January 7, 2016. Copyright Notice Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents @@ -44,85 +44,87 @@ include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Requirements language . . . . . . . . . . . . . . . . . . . . 3 3. Transport and Middlebox specification . . . . . . . . . . . . 3 3.1. System-provided interfaces . . . . . . . . . . . . . . . 3 - 3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . 3 + 3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . 4 3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4 3.4. Middle box related functions . . . . . . . . . . . . . . 4 3.5. Transport protocols implemented . . . . . . . . . . . . . 5 4. Media Prioritization . . . . . . . . . . . . . . . . . . . . 6 4.1. Usage of Quality of Service - DSCP and Multiplexing . . . 6 4.2. Local prioritization . . . . . . . . . . . . . . . . . . 8 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 6. Security Considerations . . . . . . . . . . . . . . . . . . . 9 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9 8.1. Normative References . . . . . . . . . . . . . . . . . . 9 - 8.2. Informative References . . . . . . . . . . . . . . . . . 11 + 8.2. Informative References . . . . . . . . . . . . . . . . . 12 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 12 A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 12 - A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 12 + A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 13 A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 13 - A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 13 - A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 13 + A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 14 + A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 14 A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 14 A.7. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 14 - A.8. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 14 - Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 14 + A.8. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 15 + A.9. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 15 + Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 15 1. Introduction WebRTC is a protocol suite aimed at real time multimedia exchange between browsers, and between browsers and other entities. WebRTC is described in the WebRTC overview document, [I-D.ietf-rtcweb-overview], which also defines terminology used in - this document. + this document, including the terms "WebRTC device" and "WebRTC + browser". This document focuses on the data transport protocols that are used by conforming implementations, including the protocols used for interaction with intermediate boxes such as firewalls, relays and NAT boxes. This protocol suite intends to satisfy the security considerations described in the WebRTC security documents, [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch]. This document describes requirements that apply to all WebRTC devices. When there are requirements that apply only to WebRTC - browsers, this is called out by using the word "browser". + browsers, this is called out explicitly. 2. Requirements language The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119]. 3. Transport and Middlebox specification 3.1. System-provided interfaces The protocol specifications used here assume that the following protocols are available to the implementations of the WebRTC protocols: - o UDP. This is the protocol assumed by most protocol elements - described. + o UDP [RFC0768]. This is the protocol assumed by most protocol + elements described. - o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL - and ICE-TCP. + o TCP [RFC0793]. This is used for HTTP/WebSockets, as well as for + TURN/SSL and ICE-TCP. For both protocols, IPv4 and IPv6 support is assumed. For UDP, this specification assumes the ability to set the DSCP code point of the sockets opened on a per-packet basis, in order to achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] (see Section 4.1) when multiple media types are multiplexed. It does not assume that the DSCP codepoints will be honored, and does assume that they may be zeroed or changed, since this is a local configuration issue. @@ -152,22 +154,22 @@ that temporary addresses [RFC4941] are to be preferred over permanent addresses. This is a change from the rules specified by [RFC3484]. For applications that select a single address, this is usually done by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014]. However, this rule is not completely obvious in the ICE scope. This is therefore clarified as follows: When a client gathers all IPv6 addresses on a host, and both temporary addresses and permanent addresses of the same scope are present, the client SHOULD discard the permanent addresses before - forming pairs. This is consistent with the default policy described - in [RFC6724]. + exposing addresses to the application or using them in ICE. This is + consistent with the default policy described in [RFC6724]. 3.4. Middle box related functions The primary mechanism to deal with middle boxes is ICE, which is an appropriate way to deal with NAT boxes and firewalls that accept traffic from the inside, but only from the outside if it is in response to inside traffic (simple stateful firewalls). ICE [RFC5245] MUST be supported. The implementation MUST be a full ICE implementation, not ICE-Lite. A full ICE implementation allows @@ -231,25 +234,32 @@ [I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS- SRTP, as described in [I-D.ietf-rtcweb-security-arch]. For data transport over the WebRTC data channel [I-D.ietf-rtcweb-data-channel], WebRTC implementations MUST support SCTP over DTLS over ICE. This encapsulation is specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported. - The setup protocol for WebRTC data channels is described in - [I-D.ietf-rtcweb-data-protocol]. + The setup protocol for WebRTC data channels described in + [I-D.ietf-rtcweb-data-protocol] MUST be supported. + + Note: DTLS-SRTP as defined in [RFC5764] section 6.7.1 defines the + interaction between DTLS and ICE ( [RFC5245]). The effect of this + specification is that all ICE candidate pairs associated with a + single component are part of the same DTLS association. Thus, there + will only be one DTLS handshake even if there are multiple valid + candidate pairs. WebRTC implementations MUST support multiplexing of DTLS and RTP over - the same port pair, as described in the DTLS_SRTP specification + the same port pair, as described in the DTLS-SRTP specification [RFC5764], section 5.1.2. All application layer protocol payloads over this DTLS connection are SCTP packets. Protocol identification MUST be supplied as part of the DTLS handshake, as specified in [I-D.ietf-rtcweb-alpn]. 4. Media Prioritization The WebRTC prioritization model is that the application tells the WebRTC implementation about the priority of media and data flows @@ -267,23 +277,24 @@ Implementations SHOULD attempt to set QoS on the packets sent, according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is appropriate to depart from this recommendation when running on platforms where QoS marking is not implemented. The implementation MAY turn off use of DSCP markings if it detects symptoms of unexpected behaviour like priority inversion or blocking of packets with certain DSCP markings. The detection of these conditions is implementation dependent. (Question: Does there need - to be an API knob to turn off DSCP markings?) + to be an API knob to turn off DSCP markings? If nobody argues for + it, this parenthesis will be removed.) - All packets arrying data from the SCTP association supporting the + All packets carrying data from the SCTP association supporting the data channels MUST use a single DSCP code point. All packets on one TCP connection, no matter what it carries, MUST use a single DSCP code point. More advice on the use of DSCP code points with RTP is given in [I-D.ietf-dart-dscp-rtp]. There exist a number of schemes for achieving quality of service that do not depend solely on DSCP code points. Some of these schemes @@ -419,20 +429,25 @@ [I-D.ietf-rtcweb-alpn] Thomson, M., "Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)", draft-ietf-rtcweb- alpn-00 (work in progress), July 2014. [I-D.ietf-rtcweb-data-channel] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channels", draft-ietf-rtcweb-data-channel-13 (work in progress), January 2015. + [I-D.ietf-rtcweb-data-protocol] + Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel + Establishment Protocol", draft-ietf-rtcweb-data- + protocol-09 (work in progress), January 2015. + [I-D.ietf-rtcweb-rtp-usage] Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP", draft-ietf-rtcweb-rtp-usage-22 (work in progress), February 2015. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for WebRTC", draft- ietf-rtcweb-security-07 (work in progress), July 2014. @@ -455,20 +470,26 @@ Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "Stream Schedulers and a New Data Chunk for the Stream Control Transmission Protocol", draft-ietf-tsvwg-sctp- ndata-02 (work in progress), January 2015. [I-D.martinsen-mmusic-ice-dualstack-fairness] Martinsen, P., Reddy, T., and P. Patil, "ICE IPv4/IPv6 Dual Stack Fairness", draft-martinsen-mmusic-ice- dualstack-fairness-02 (work in progress), February 2015. + [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, + August 1980. + + [RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC + 793, September 1981. + [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection- Oriented Transport", RFC 4571, July 2006. [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy Extensions for Stateless Address Autoconfiguration in IPv6", RFC 4941, September 2007. @@ -506,25 +527,20 @@ "Default Address Selection for Internet Protocol Version 6 (IPv6)", RFC 6724, September 2012. 8.2. Informative References [I-D.ietf-dart-dscp-rtp] Black, D. and P. Jones, "Differentiated Services (DiffServ) and Real-time Communication", draft-ietf-dart- dscp-rtp-10 (work in progress), November 2014. - [I-D.ietf-rtcweb-data-protocol] - Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel - Establishment Protocol", draft-ietf-rtcweb-data- - protocol-09 (work in progress), January 2015. - [I-D.ietf-rtcweb-overview] Alvestrand, H., "Overview: Real Time Protocols for Browser-based Applications", draft-ietf-rtcweb-overview-13 (work in progress), November 2014. [RFC3484] Draves, R., "Default Address Selection for Internet Protocol version 6 (IPv6)", RFC 3484, February 2003. [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 Socket API for Source Address Selection", RFC 5014, @@ -643,16 +658,20 @@ A.8. Changes from -07 to -08 o Updated references o Deleted "bundle each media type (audio, video or data) into its own 5-tuple (bundling by media type)" from MUST support configuration, since JSEP does not have a means to negotiate this configuration +A.9. Changes from -08 to -09 + + o Added a clarifying note about DTLS-SRTP and ICE interaction. + Author's Address Harald Alvestrand Google Email: harald@alvestrand.no