RTCWEB M. Thomson Internet-Draft B. Aboba Intended status: Standards Track Microsoft Expires: January 10, 2013 July 9, 2012 Media API Requirements for Real-Time Interoperation of Web User Agents draft-thomson-rtcweb-api-reqs-00 Abstract Direct, real-time communications between web user agents (browsers) requires two points of standardization. The first describes the on- the-wire protocol that is used. The second is the API that controls and configures what gets put on the wire and how. The capabilities of the protocol determines the nature of the API. This document outlines how the constraints imposed upon the API by the protocol. Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. 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 January 10, 2013. Copyright Notice Copyright (c) 2012 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 carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as Thomson & Aboba Expires January 10, 2013 [Page 1]
Internet-Draft Tao of Web July 2012 described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 4 2. Peer-to-Peer Transport . . . . . . . . . . . . . . . . . . . . 4 2.1. Overview of ICE Operation . . . . . . . . . . . . . . . . 4 2.2. Transport ICE Requirements . . . . . . . . . . . . . . . . 5 2.3. Established Flow Requirements . . . . . . . . . . . . . . 6 3. Securing Media . . . . . . . . . . . . . . . . . . . . . . . . 6 4. Sending Peer-to-Peer Media . . . . . . . . . . . . . . . . . . 7 5. DTMF Tones . . . . . . . . . . . . . . . . . . . . . . . . . . 9 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 7. Security Considerations . . . . . . . . . . . . . . . . . . . 9 8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 9 9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9 9.1. Normative References . . . . . . . . . . . . . . . . . . . 9 9.2. Informative References . . . . . . . . . . . . . . . . . . 10 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11 Thomson & Aboba Expires January 10, 2013 [Page 2]
Internet-Draft Tao of Web July 2012 1. Introduction The exchange of media - audio and video - between peers is a basic foundation of telecommunications. The fact that the web platform has successfully managed to avoid this fundamental feature for so long is a testament to the usefulness of its other features. The WebRTC/ rtcweb effort attempts to remedy this shortcoming. Figure 1 shows the architecture for real-time communications on the web. _________ (( )) .-(( The Cloud ))-. ? / ((_________)) \ ? / \ +------------/----+ +--\--------------+ | +---------/---+ | | +-\-----------+ | | | Javascript | | | | Javascript | | | | | | | | | | | +--<W3C API>--+ | IETF | +--<W3C API>--+ | | === protocols === | | Web User Agent | | Web User Agent | +-----------------+ +-----------------+ Figure 1: Architecture for the Real-Time Web Two points of interoperation are necessary in this architecture: API and peer-to-peer protocols. Critically, the points marked with a '?' are not subject to standardization. These points are effectively internal to the application that exists both in "The Cloud" and in the Javascript virtual machine provided by the web user agent. The IETF rtcweb working group has settled on Real-Time Transport Protocol (RTP) [RFC3550], Secure RTP (SRTP) [RFC3711] and Interactive Connectivity Establishment (ICE) [RFC5245] for the peer to peer protocol components. Use of RTP is documented in [I-D.ietf-rtcweb-rtp-usage]; use of SRTP and ICE is described in the rtcweb security architecture [I-D.ietf-rtcweb-security-arch]. ICE and RTP are unable to operate without an out-of-band communications channel. For ICE this provides bootstrapping information; for RTP, a common understanding of the key protocol parameters. Though SRTP can operate without an out-of-band channel if certain assumptions are made, many uses depend on some form of out-of-band communication. The out-of-band channel is provided by the web application. The necessary information passes through the API to the browser. Thomson & Aboba Expires January 10, 2013 [Page 3]
Internet-Draft Tao of Web July 2012 This document describes what information must pass between a web application and a web user agent in order to successfully establish peer-to-peer media communications. No attempt is made to constrain what the API looks like. After all: [...] when [a] IETF documents start telling you how to build Javascript APIs, you should run far away... quickly. :) -- [I-D.kaplan-rtcweb-api-reqs] 1.1. Terminology In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as described in BCP 14, RFC 2119 [RFC2119] and indicate requirement levels for compliant implementations. 2. Peer-to-Peer Transport ICE [RFC5245] provides a mechanism for establishing peer-to-peer communications using UDP. The ICE process applies to a single UDP flow from a source address and port to a destination address and port. Where multiple flows are required, the process is applied once for each flow. 2.1. Overview of ICE Operation In ICE, for each UDP flow, each peer performs the following steps: o Candidate gathering. A candidate is an UDP port with associated session authentication parameters. Candidates are attributed to the peer, but they can be collected from the local host, by querying servers about the server perspective of the peer address, or a TURN [RFC5766] relay can allocate server-based ports that forward packets to the peer. o Candidate exchange. Each peer learns of the candidates gathered by the other peer. o Candidate pair testing. Candidates are paired, one local against one remote. A connectivity check is performed where a STUN [RFC5389] packet is sent from the local candidate to the remote. The pair is valid if a response arrives. Packets might be lost, so this process is repeated. Thomson & Aboba Expires January 10, 2013 [Page 4]
Internet-Draft Tao of Web July 2012 o Candidate pair selection. As soon as one or more candidates are marked as valid, one of these can be selected and used. 2.2. Transport ICE Requirements The following requirements ensure that an application is able to establish a UDP flow between peers with proper consent: ICE-1 The application MUST be able to request the gathering of host candidates. Candidates are comprised of an IP address (v4 or v6), a UDP port number, an ICE username fragment and an ICE password. ICE-2 The application MUST be able to request the gathering of server reflexive address information using STUN. Input to this is the identity of a STUN server and any credentials required by the server. ICE-3 The application MUST be able to request the allocation of TURN relay candidates. Input to this is the identity of a TURN server and any credentials required by the server. ICE-4 The application MUST be able to request that a connectivity check be sent to a remote candidate and to be informed of success. Input to this is a choice of local candidate and details of the remote candidate. This establishes: whether the UDP packet is able to reach the remote peer and whether the remote peer consents to the receipt of packets from this peer. ICE-5 The application MUST be able to add STUN attributes to the STUN messages that are sent for connectivity checks. ICE-6 The application MUST be able to examine STUN attributes received on successful responses to connectivity checks. ICE-7 The application MUST be able to retrieve peer reflexive addresses that are discovered when connectivity checks are successful. ICE-8 The application MUST be able to request the creation of a UDP flow on a valid candidate pair. ICE-9 The application MUST be notified when a successful connectivity check is received from remote peer. The notification MUST include peer addressing information. Thomson & Aboba Expires January 10, 2013 [Page 5]
Internet-Draft Tao of Web July 2012 2.3. Established Flow Requirements The following requirements apply to established flows: UDP-1 The application MUST be able to send connectivity checks on an active UDP flow to test liveness of the flow. UDP-2 The application MUST be notified when consent for an active UDP flow expires. UDP-3 The application MUST be able to retrieve the bandwidth that the remote peer consents to receive for the UDP flow. UDP-4 The application MUST be able to learn the bandwidth limit that the browser has detected for the flow based on feedback from the network. UDP-5 The application MUST be notified when the browser detects changes in the available bandwidth for the UDP flow. UDP-6 The application MUST be notified when the browser detects network congestion that affects the UDP flow. UDP-7 The application MUST be notified when changes in network connectivity occur. UDP-8 The application MUST be able to pause connectivity checking on an active UDP flow. This allows an application to conserve resources for inactive flows, especially for battery-powered devices. 3. Securing Media SRTP [RFC3711] provides confidentiality, message authentication and replay protection for RTP media. SRTP requires external key negotiation. Two methods are possible: Datagram Transport Layer Security (DTLS) can be used for key negotiation (DTLS-SRTP) [RFC5764], or the SRTP master keys can be provided directly. The application chooses which of these modes the application uses. [[Ed: requirement for second option not established]] With DTLS key negotiation, the asymmetry of the DTLS handshake means that out-of-band methods must be used to determine whether peers act in the server or client roles. DTLS offers a means for peer Thomson & Aboba Expires January 10, 2013 [Page 6]
Internet-Draft Tao of Web July 2012 authentication, allowing the application to learn of the identity of the peer through the certificate they offer during the handshake. Without DTLS key negotiation, the SRTP master keys are determined out-of-band and provided to the browser through the API (e.g., [RFC4568]). SRTP does not offer any method for peer authentication. The following requirements apply: SEC-1 The application MUST be able to select how SRTP key negotiation is performed, either DTLS-SRTP or through the API. SEC-2 Where DTLS-SRTP is chosen, the application MUST be able to specify whether the browser is to take the client or server role. SEC-3 Where DTLS-SRTP is chosen, the application MUST be able to view the certificate offered by the peer. SEC-4 Where DTLS-SRTP is chosen, the application MUST be able to discover if the peer certificate is a trusted certificate for an identified domain. SEC-5 Where DTLS-SRTP is chosen, the application MUST be able to reject communication with peers based on information presented in their certificate. SEC-6 Where SRTP is chosen without DTLS for key negotiation, the application MUST be able to set SRTP parameters, including: the encryption, key generation and authentication algorithms; key and parameter size; key and salt value; key usage lifetime; key derivation interval; and an optional master key identifier. 4. Sending Peer-to-Peer Media RTP [RFC3550] sends real-time data - media - from a source to a sink. An RTP stream is the manifestation of media as packets. Most media is a single stream, though layered codecs (such as H.264 SVC [RFC6190]) can be split into multiple streams. Browser need to handle RTP streams in two directions: outbound from a local source, and inbound toward a local sink. The following requirements apply: Thomson & Aboba Expires January 10, 2013 [Page 7]
Internet-Draft Tao of Web July 2012 MED-1 The application MUST be able to select the UDP flow that an RTP stream uses. MED-2 The application MUST be able select the UDP flow that RTCP for a given RTP stream uses. MED-3 The application MUST be able to move RTP streams (or associated RTCP) between UDP flows without losing packets. MED-4 The application MUST be able to specify the SSRC for RTP streams, both outbound and inbound. MED-5 The application MUST be able to discover the CNAME that will be used for streams that it generates. MED-6 The application MUST be able to discover the SRCNAME [I-D.westerlund-avtext-rtcp-sdes-srcname] that will be used for streams that it generates. [[Ed: conditional on acceptance of SRCNAME]] MED-7 The application MUST be able to specify the RTP packet type that is used to identify codecs in RTP streams, both inbound and outbound. MED-8 The application MUST be able to select the codecs that are used for outbound RTP streams and to specify the codecs that are present in inbound RTP streams. MED-9 The application MUST be able to limit the bandwidth allocated to a single outbound RTP stream. MED-10 The application MUST be able to specify the number of audio channels used for an audio stream. MED-11 The application MUST be able to specify codec parameters (such as appear in SDP [RFC4566] "a=fmtp" lines). MED-12 The application MUST be able to mark the priority of an RTP stream. This lets the browser choose appropriate packet marking using DiffServ and the relative sensitivity of streams to congestion. MED-13 The application MUST be able to update the configuration for an active RTP stream. Thomson & Aboba Expires January 10, 2013 [Page 8]
Internet-Draft Tao of Web July 2012 5. DTMF Tones Dual-Tone Multifrequency (DTMF) is commonly used by legacy telephony applications. DTMF tones are typically interleaved/mixed with audio streams. The following requirements apply: DTMF-1 The application MUST be able to insert DTMF tones into an audio stream. DTMF-2 The application MUST be able to be informed of the existence of DTMF tone events on an audio stream. 6. IANA Considerations This document has no IANA actions. [RFC Editor: please remove this section prior to publication.] 7. Security Considerations The security of your precious bodily fluids can only be achieved through purity of essence. 8. Acknowledgments This document takes ideas from [I-D.kaplan-rtcweb-api-reqs], including the short title you see on each page. 9. References 9.1. Normative References [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-03 (work in progress), June 2012. [I-D.ietf-rtcweb-security-arch] Rescorla, E., "RTCWEB Security Architecture", draft-ietf-rtcweb-security-arch-02 (work in progress), June 2012. Thomson & Aboba Expires January 10, 2013 [Page 9]
Internet-Draft Tao of Web July 2012 [I-D.westerlund-avtext-rtcp-sdes-srcname] Westerlund, M., Burman, B., and P. Sandgren, "RTCP SDES Item SRCNAME to Label Individual Sources", draft-westerlund-avtext-rtcp-sdes-srcname-00 (work in progress), October 2011. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004. [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, April 2010. [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 9.2. Informative References [I-D.kaplan-rtcweb-api-reqs] Kaplan, H., Stratford, N., and T. Panton, "API Requirements for WebRTC-enabled Browsers", draft-kaplan-rtcweb-api-reqs-01 (work in progress), October 2011. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, July 2006. [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for NAT (STUN)", RFC 5389, October 2008. [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. Thomson & Aboba Expires January 10, 2013 [Page 10]
Internet-Draft Tao of Web July 2012 [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, "RTP Payload Format for Scalable Video Coding", RFC 6190, May 2011. [RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC 6455, December 2011. Authors' Addresses Martin Thomson Microsoft 3210 Porter Drive Palo Alto, CA 94304 US Phone: +1 650-353-1925 Email: martin.thomson@skype.net Bernard Aboba Microsoft One Microsoft Way Redmond, WA 98052 US Email: bernard_aboba@hotmail.com Thomson & Aboba Expires January 10, 2013 [Page 11]