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Kungsbron 2 Stockholm 11122 Sweden harald@alvestrand.no

This document gives an overview and context of a protocol suite intended for use with real-time applications that can be deployed in browsers -- "real-time communication on the Web". It intends to serve as a starting and coordination point to make sure that (1) all the parts that are needed to achieve this goal are findable and (2) the parts that belong in the Internet protocol suite are fully specified and on the right publication track. This document is an applicability statement -- it does not itself specify any protocol, but it specifies which other specifications implementations are supposed to follow to be compliant with Web Real-Time Communication (WebRTC).

Status of This Memo This is an Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at .

Copyright Notice Copyright (c) 2021 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 () 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 described in the Simplified BSD License.

Table of Contents

• .  Introduction
• .  Principles and Terminology
  • .  Goals of This Document
  • .  Relationship between API and Protocol
  • .  On Interoperability and Innovation
  • .  Terminology
• .  Architecture and Functionality Groups
• .  Data Transport
• .  Data Framing and Securing
• .  Data Formats
• .  Connection Management
• .  Presentation and Control
• .  Local System Support Functions
• . IANA Considerations
• . Security Considerations
• . References
  • .  Normative References
  • .  Informative References
• Acknowledgements
• Author's Address

Introduction The Internet was, from very early in its lifetime, considered a possible vehicle for the deployment of real-time, interactive applications -- with the most easily imaginable being audio conversations (aka "Internet telephony") and video conferencing. The first attempts to build such applications were dependent on special networks, special hardware, and custom-built software, often at very high prices or of low quality, placing great demands on the infrastructure. As the available bandwidth has increased, and as processors and other hardware have become ever faster, the barriers to participation have decreased, and it has become possible to deliver a satisfactory experience on commonly available computing hardware. Still, there are a number of barriers to the ability to communicate universally -- one of these is that there is, as of yet, no single set of communication protocols that all agree should be made available for communication; another is the sheer lack of universal identification systems (such as is served by telephone numbers or email addresses in other communications systems). Development of "The Universal Solution" has, however, proved hard. The last few years have also seen a new platform rise for deployment of services: the browser-embedded application, or "web application". It turns out that as long as the browser platform has the necessary interfaces, it is possible to deliver almost any kind of service on it. Traditionally, these interfaces have been delivered by plugins, which had to be downloaded and installed separately from the browser; in the development of HTML5 , application developers see much promise in the possibility of making those interfaces available in a standardized way within the browser. This memo describes a set of building blocks that (1) can be made accessible and controllable through a JavaScript API in a browser and (2) together form a sufficient set of functions to allow the use of interactive audio and video in applications that communicate directly between browsers across the Internet. The resulting protocol suite is intended to enable all the applications that are described as required scenarios in the WebRTC "use cases" document . Other efforts -- for instance, the W3C Web Real-Time Communications, Web Applications Security, and Devices and Sensors Working Groups -- focus on making standardized APIs and interfaces available, within or alongside the HTML5 effort, for those functions. This memo concentrates on specifying the protocols and subprotocols that are needed to specify the interactions over the network. Operators should note that deployment of WebRTC will result in a change in the nature of signaling for real-time media on the network and may result in a shift in the kinds of devices used to create and consume such media. In the case of signaling, WebRTC session setup will typically occur over TLS-secured web technologies using application-specific protocols. Operational techniques that involve inserting network elements to interpret the Session Description Protocol (SDP) -- through either (1) the endpoint asking the network for a SIP server or (2) the transparent insertion of SIP Application Layer Gateways (ALGs) -- will not work with such signaling. In the case of networks using cooperative endpoints, the approaches defined in may serve as a suitable replacement for . The increase in browser-based communications may also lead to a shift away from dedicated real-time-communications hardware, such as SIP desk phones. This will diminish the efficacy of operational techniques that place dedicated real-time devices on their own network segment, address range, or VLAN for purposes such as applying traffic filtering and QoS. Applying the markings described in may be appropriate replacements for such techniques. While this document formally relies on , at the time of its publication, the majority of WebRTC implementations support the version of Interactive Connectivity Establishment (ICE) that is described in and use a pre-standard version of the Trickle ICE mechanism described in . The "ice2" attribute defined in can be used to detect the version in use by a remote endpoint and to provide a smooth transition from the older specification to the newer one. This memo uses the term "WebRTC" (note the case used) to refer to the overall effort consisting of both IETF and W3C efforts.

Principles and Terminology

Goals of This Document The goal of the WebRTC protocol specification is to specify a set of protocols that, if all are implemented, will allow an implementation to communicate with another implementation using audio, video, and data sent along the most direct possible path between the participants. This document is intended to serve as the roadmap to the WebRTC specifications. It defines terms used by other parts of the WebRTC protocol specifications, lists references to other specifications that don't need further elaboration in the WebRTC context, and gives pointers to other documents that form part of the WebRTC suite. By reading this document and the documents it refers to, it should be possible to have all information needed to implement a WebRTC-compatible implementation.

Relationship between API and Protocol The total WebRTC effort consists of two major parts, each consisting of multiple documents:

• A protocol specification, done in the IETF
• A JavaScript API specification, defined in a series of W3C documents

Together, these two specifications aim to provide an environment where JavaScript embedded in any page, when suitably authorized by its user, is able to set up communication using audio, video, and auxiliary data, as long as the browser supports these specifications. The browser environment does not constrain the types of application in which this functionality can be used. The protocol specification does not assume that all implementations implement this API; it is not intended to be necessary for interoperation to know whether the entity one is communicating with is a browser or another device implementing the protocol specification. The goal of cooperation between the protocol specification and the API specification is that for all options and features of the protocol specification, it should be clear which API calls to make to exercise that option or feature; similarly, for any sequence of API calls, it should be clear which protocol options and features will be invoked. Both are subject to constraints of the implementation, of course. The following terms are used across the documents specifying the WebRTC suite, with the specific meanings given here. Not all terms are used in this document. Other terms are used per their commonly used meanings.

Agent:

Undefined term. See "SDP Agent" and "ICE Agent".

Application Programming Interface (API):

A specification of a set of calls and events, usually tied to a programming language or an abstract formal specification such as WebIDL, with its defined semantics.

Browser:

Used synonymously with "interactive user agent" as defined in . See also the "WebRTC Browser" (aka "WebRTC User Agent") definition below.

Data Channel:

An abstraction that allows data to be sent between WebRTC endpoints in the form of messages. Two endpoints can have multiple data channels between them.

ICE Agent:

An implementation of the Interactive Connectivity Establishment (ICE) protocol . An ICE Agent may also be an SDP Agent, but there exist ICE Agents that do not use SDP (for instance, those that use Jingle ).

Interactive:

Communication between multiple parties, where the expectation is that an action from one party can cause a reaction by another party, and the reaction can be observed by the first party, where the total time required for the action/reaction/observation is on the order of no more than hundreds of milliseconds.

Media:

Audio and video content. Not to be confused with "transmission media" such as wires.

Media Path:

The path that media data follows from one WebRTC endpoint to another.

Protocol:

A specification of a set of data units, their representation, and rules for their transmission, with their defined semantics. A protocol is usually thought of as going between systems.

Real-Time Media:

Media where the generation and display of content are intended to occur closely together in time (on the order of no more than hundreds of milliseconds). Real-time media can be used to support interactive communication.

SDP Agent:

The protocol implementation involved in the Session Description Protocol (SDP) offer/answer exchange, as defined in .

Signaling:

Communication that happens in order to establish, manage, and control media paths and data paths.

Signaling Path:

The communication channels used between entities participating in signaling to transfer signaling. There may be more entities in the signaling path than in the media path.

WebRTC Browser (also called a "WebRTC User Agent" or "WebRTC UA"):

​ Something that conforms to both the protocol specification and the JavaScript API cited above.

WebRTC Non-Browser:

Something that conforms to the protocol specification but does not claim to implement the JavaScript API. This can also be called a "WebRTC device" or "WebRTC native application".

WebRTC Endpoint:

Either a WebRTC browser or a WebRTC non-browser. It conforms to the protocol specification.

WebRTC-Compatible Endpoint:

An endpoint that is able to successfully communicate with a WebRTC endpoint but may fail to meet some requirements of a WebRTC endpoint. This may limit where in the network such an endpoint can be attached or may limit the security guarantees that it offers to others. It is not constrained by this specification; when it is mentioned at all, it is to note the implications on WebRTC-compatible endpoints of the requirements placed on WebRTC endpoints.

WebRTC Gateway:

A WebRTC-compatible endpoint that mediates media traffic to non-WebRTC entities.

All WebRTC browsers are WebRTC endpoints, so any requirement on a WebRTC endpoint also applies to a WebRTC browser. A WebRTC non-browser may be capable of hosting applications in a way that is similar to the way in which a browser can host JavaScript applications, typically by offering APIs in other languages. For instance, it may be implemented as a library that offers a C++ API intended to be loaded into applications. In this case, security considerations similar to those for JavaScript may be needed; however, since such APIs are not defined or referenced here, this document cannot give any specific rules for those interfaces. WebRTC gateways are described in a separate document .

On Interoperability and Innovation The "Mission statement for the IETF" states that "The benefit of a standard to the Internet is in interoperability - that multiple products implementing a standard are able to work together in order to deliver valuable functions to the Internet's users." Communication on the Internet frequently occurs in two phases:

• Two parties communicate, through some mechanism, what functionality they are both able to support.
• They use that shared communicative functionality to communicate or, failing to find anything in common, give up on communication.

There are often many choices that can be made for communicative functionality; the history of the Internet is rife with the proposal, standardization, implementation, and success or failure of many types of options, in all sorts of protocols. The goal of having a mandatory-to-implement function set is to prevent negotiation failure, not to preempt or prevent negotiation. The presence of a mandatory-to-implement function set serves as a strong changer of the marketplace of deployment in that it gives a guarantee that you can communicate successfully as long as (1) you conform to a specification and (2) the other party is willing to accept communication at the base level of that specification. The alternative (that is, not having a mandatory-to-implement function) does not mean that you cannot communicate; it merely means that in order to be part of the communications partnership, you have to implement the standard "and then some". The "and then some" is usually called a profile of some sort; in the version most antithetical to the Internet ethos, that "and then some" consists of having to use a specific vendor's product only.

Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 when, and only when, they appear in all capitals, as shown here.

Architecture and Functionality Groups For browser-based applications, the model for real-time support does not assume that the browser will contain all the functions needed for an application such as a telephone or a video conference. The vision is that the browser will have the functions needed for a web application, working in conjunction with its backend servers, to implement these functions. This means that two vital interfaces need specification: the protocols that browsers use to talk to each other, without any intervening servers; and the APIs that are offered for a JavaScript application to take advantage of the browser's functionality.

Browser Model +------------------------+ On-the-wire | | Protocols | Servers |---------> | | | | +------------------------+ ^ | | | HTTPS/ | WebSockets | | +----------------------------+ | JavaScript/HTML/CSS | +----------------------------+ Other ^ ^ RTC APIs | | APIs +---|-----------------|------+ | | | | | +---------+| | | Browser || On-the-wire | Browser | RTC || Protocols | | Function|-----------> | | || | | || | +---------+| +---------------------|------+ | V Native OS Services

Note that HTTPS and WebSockets are also offered to the JavaScript application through browser APIs. As for all protocol and API specifications, there is no restriction that the protocols can only be used to talk to another browser; since they are fully specified, any endpoint that implements the protocols faithfully should be able to interoperate with the application running in the browser. A commonly imagined model of deployment is depicted in . ("JS" stands for JavaScript.)

Browser RTC Trapezoid +-----------+ +-----------+ | Web | | Web | | | | | | |------------------| | | Server | Signaling Path | Server | | | | | +-----------+ +-----------+ / \ / \ Application-defined / \ over / \ HTTPS/WebSockets / Application-defined over \ / HTTPS/WebSockets \ / \ +-----------+ +-----------+ |JS/HTML/CSS| |JS/HTML/CSS| +-----------+ +-----------+ +-----------+ +-----------+ | | | | | | | | | Browser |--------------------------------| Browser | | | Media Path | | | | | | +-----------+ +-----------+

In this drawing, the critical part to note is that the media path ("low path") goes directly between the browsers, so it has to be conformant to the specifications of the WebRTC protocol suite; the signaling path ("high path") goes via servers that can modify, translate, or manipulate the signals as needed. If the two web servers are operated by different entities, the inter-server signaling mechanism needs to be agreed upon, by either standardization or other means of agreement. Existing protocols (e.g., SIP or the Extensible Messaging and Presence Protocol (XMPP) ) could be used between servers, while either a standards-based or proprietary protocol could be used between the browser and the web server. For example, if both operators' servers implement SIP, SIP could be used for communication between servers, along with either a standardized signaling mechanism (e.g., SIP over WebSockets) or a proprietary signaling mechanism used between the application running in the browser and the web server. Similarly, if both operators' servers implement XMPP, XMPP could be used for communication between XMPP servers, with either a standardized signaling mechanism (e.g., XMPP over WebSockets or Bidirectional-streams Over Synchronous HTTP (BOSH) ) or a proprietary signaling mechanism used between the application running in the browser and the web server. The choice of protocols for client-server and inter-server signaling, and the definition of the translation between them, are outside the scope of the WebRTC protocol suite described in this document. The functionality groups that are needed in the browser can be specified, more or less from the bottom up, as:

Data transport:

For example, TCP and UDP, and the means to securely set up connections between entities, as well as the functions for deciding when to send data: congestion management, bandwidth estimation, and so on.

Data framing:

RTP, the Stream Control Transmission Protocol (SCTP), DTLS, and other data formats that serve as containers, and their functions for data confidentiality and integrity.

Data formats:

Codec specifications, format specifications, and functionality specifications for the data passed between systems. Audio and video codecs, as well as formats for data and document sharing, belong in this category. In order to make use of data formats, a way to describe them (e.g., a session description) is needed.

Connection management:

For example, setting up connections, agreeing on data formats, changing data formats during the duration of a call. SDP, SIP, and Jingle/XMPP belong in this category.

Presentation and control:

What needs to happen in order to ensure that interactions behave in an unsurprising manner. This can include floor control, screen layout, voice-activated image switching, and other such functions, where part of the system requires cooperation between parties. Centralized Conferencing (XCON) and Cisco⁠/Tandberg's Telepresence Interoperability Protocol (TIP) were some attempts at specifying this kind of functionality; many applications have been built without standardized interfaces to these functions.

Local system support functions:

Functions that need not be specified uniformly, because each participant may implement these functions as they choose, without affecting the bits on the wire in a way that others have to be cognizant of. Examples in this category include echo cancellation (some forms of it), local authentication and authorization mechanisms, OS access control, and the ability to do local recording of conversations.

Within each functionality group, it is important to preserve both freedom to innovate and the ability for global communication. Freedom to innovate is helped by doing the specification in terms of interfaces, not implementation; any implementation able to communicate according to the interfaces is a valid implementation. The ability to communicate globally is helped by both (1) having core specifications be unencumbered by IPR issues and (2) having the formats and protocols be fully enough specified to allow for independent implementation. One can think of the first three groups as forming a "media transport infrastructure" and of the last three groups as forming a "media service". In many contexts, it makes sense to use a common specification for the media transport infrastructure, which can be embedded in browsers and accessed using standard interfaces, and "let a thousand flowers bloom" in the "media service" layer; to achieve interoperable services, however, at least the first five of the six groups need to be specified.

Data Transport Data transport refers to the sending and receiving of data over the network interfaces, the choice of network-layer addresses at each end of the communication, and the interaction with any intermediate entities that handle the data but do not modify it (such as Traversal Using Relays around NAT (TURN) relays). It includes necessary functions for congestion control, retransmission, and in-order delivery. WebRTC endpoints MUST implement the transport protocols described in .

Data Framing and Securing The format for media transport is RTP . Implementation of the Secure Real-time Transport Protocol (SRTP) is REQUIRED for all implementations. The detailed considerations for usage of functions from RTP and SRTP are given in . The security considerations for the WebRTC use case are provided in , and the resulting security functions are described in . Considerations for the transfer of data that is not in RTP format are described in , and a supporting protocol for establishing individual data channels is described in . WebRTC endpoints MUST implement these two specifications. WebRTC endpoints MUST implement , , , and the requirements they include.

Data Formats The intent of this specification is to allow each communications event to use the data formats that are best suited for that particular instance, where a format is supported by both sides of the connection. However, a minimum standard is greatly helpful in order to ensure that communication can be achieved. This document specifies a minimum baseline that will be supported by all implementations of this specification and leaves further codecs to be included at the will of the implementer. WebRTC endpoints that support audio and/or video MUST implement the codecs and profiles required in and .

Connection Management The methods, mechanisms, and requirements for setting up, negotiating, and tearing down connections comprise a large subject, and one where it is desirable to have both interoperability and freedom to innovate. The following principles apply:

1. The WebRTC media negotiations will be capable of representing the same SDP offer/answer semantics that are used in SIP, in such a way that it is possible to build a signaling gateway between SIP and the WebRTC media negotiation.
2. It will be possible to gateway between legacy SIP devices that support ICE and appropriate RTP/SDP mechanisms, codecs, and security mechanisms without using a media gateway. A signaling gateway to convert between the signaling on the web side and the SIP signaling may be needed.
3. When an SDP for a new codec is specified, no other standardization should be required for it to be possible to use that codec in the web browsers. Adding new codecs that might have new SDP parameters should not change the APIs between the browser and the JavaScript application. As soon as the browsers support the new codecs, old applications written before the codecs were specified should automatically be able to use the new codecs where appropriate, with no changes to the JavaScript applications.

The particular choices made for WebRTC, and their implications for the API offered by a browser implementing WebRTC, are described in . WebRTC browsers MUST implement . WebRTC endpoints MUST implement those functions described in that relate to the network layer (e.g., BUNDLE , "rtcp-mux" , and Trickle ICE ), but these endpoints do not need to support the API functionality described in .

Presentation and Control The most important part of control is the users' control over the browser's interaction with input/output devices and communications channels. It is important that the users have some way of figuring out where their audio, video, or texting is being sent; for what purported reason; and what guarantees are made by the parties that form part of this control channel. This is largely a local function between the browser, the underlying operating system, and the user interface; this is specified in the peer connection API and the media capture API . WebRTC browsers MUST implement these two specifications.

Local System Support Functions These functions are characterized by the fact that the quality of an implementation strongly influences the user experience, but the exact algorithm does not need coordination. In some cases (for instance, echo cancellation, as described below), the overall system definition may need to specify that the overall system needs to have some characteristics for which these facilities are useful, without requiring them to be implemented a certain way. Local functions include echo cancellation; volume control; camera management, including focus, zoom, and pan/tilt controls (if available); and more. One would want to see certain parts of the system conform to certain properties; for instance:

• Echo cancellation should be good enough to achieve the suppression of acoustical feedback loops below a perceptually noticeable level.
• Privacy concerns MUST be satisfied; for instance, if remote control of a camera is offered, the APIs should be available to let the local participant figure out who's controlling the camera and possibly decide to revoke the permission for camera usage.
• Automatic Gain Control (AGC), if present, should normalize a speaking voice into a reasonable dB range.

The requirements on WebRTC systems with regard to audio processing are found in , and that document includes more guidance about echo cancellation and AGC; the APIs for control of local devices are found in . WebRTC endpoints MUST implement the processing functions in . (Together with the requirement in , this means that WebRTC endpoints MUST implement the whole document.)

IANA Considerations This document has no IANA actions.

Security Considerations Security of the web-enabled real-time communications comes in several pieces:

Security of the components:

The browsers, and other servers involved. The most target-rich environment here is probably the browser; the aim here should be that the introduction of these components introduces no additional vulnerability.

Security of the communication channels:

It should be easy for participants to reassure themselves of the security of their communication -- by verifying the crypto parameters of the links that they participate in, and to get reassurances from the other parties to the communication that those parties promise that appropriate measures are taken.

Security of the partners' identities:

Verifying that the participants are who they say they are (when positive identification is appropriate) or that their identities cannot be uncovered (when anonymity is a goal of the application).

The security analysis, and the requirements derived from that analysis, are contained in . It is also important to read the security sections of and .

References Normative References In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them. In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP). [STANDARDS-TRACK] This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK] This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK] This specification provides the requirements and considerations for WebRTC applications to send and receive video across a network. It specifies the video processing that is required as well as video codecs and their parameters. This document outlines the audio codec and processing requirements for WebRTC endpoints. RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings. This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based communication. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). This document obsoletes RFC 5245. W3C Candidate Recommendation W3C Proposed Recommendation Informative References WHATWG This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. [STANDARDS-TRACK] This memo gives a mission statement for the IETF, tries to define the terms used in the statement sufficiently to make the mission statement understandable and useful, argues why the IETF needs a mission statement, and tries to capture some of the debate that led to this point. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). ICE can be used by any protocol utilizing the offer/answer model, such as the Session Initiation Protocol (SIP). [STANDARDS-TRACK] This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions. [STANDARDS-TRACK] The Extensible Messaging and Presence Protocol (XMPP) is an application profile of the Extensible Markup Language (XML) that enables the near-real-time exchange of structured yet extensible data between any two or more network entities. This document defines XMPP's core protocol methods: setup and teardown of XML streams, channel encryption, authentication, error handling, and communication primitives for messaging, network availability ("presence"), and request-response interactions. This document obsoletes RFC 3920. [STANDARDS-TRACK] RFC 5239 defines centralized conferencing (XCON) as an association of participants with a central focus. The state of a conference is represented by a conference object. This document defines an XML- based conference information data model to be used for conference objects. A conference information data model is designed to convey information about the conference and about participation in the conference. The conference information data model defined in this document constitutes an extension of the data format specified in the Session Initiation Protocol (SIP) event package for conference State. [STANDARDS-TRACK] This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases. This document was developed in an initial phase of the work with rather minor updates at later stages. It has not really served as a tool in deciding features or scope for the WG's efforts so far. It is being published to record the early conclusions of the WG. It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future. Current Traversal Using Relays around NAT (TURN) server discovery mechanisms are relatively static and limited to explicit configuration. These are usually under the administrative control of the application or TURN service provider, and not the enterprise, ISP, or the network in which the client is located. Enterprises and ISPs wishing to provide their own TURN servers need auto-discovery mechanisms that a TURN client could use with minimal or no configuration. This document describes three such mechanisms for TURN server discovery. This document updates RFC 5766 to relax the requirement for mutual authentication in certain cases. This document describes interoperability considerations for a class of WebRTC-compatible endpoints called "WebRTC gateways", which interconnect between WebRTC endpoints and devices that are not WebRTC endpoints. Work in Progress

ian.paterson@clientside.co.uk

dizzyd@jabber.org

stpeter@jabber.org

jack@chesspark.com

lance@andyet.com

winfried@tilanus.com

scottlu@google.com

jbeda@google.com

stpeter@jabber.org

robert.mcqueen@collabora.co.uk

seanegan@google.com

jhildebr@cisco.com

Acknowledgements The number of people who have taken part in the discussions surrounding this document are too numerous to list, or even to identify. The people listed below have made special, identifiable contributions; this does not mean that others' contributions are less important. Thanks to , , , , and , who offered technical contributions to various draft versions of this document. Thanks to , , and others at Skype for the ASCII drawings in . Thanks to , , , , , , , , , , , , and for document review.

Author's Address Google

Kungsbron 2 Stockholm 11122 Sweden harald@alvestrand.no