Overview: Real-Time Protocols for Browser-Based ApplicationsGoogleKungsbron 2Stockholm11122Swedenharald@alvestrand.noThis 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
.
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Copyright (c) 2021 IETF Trust and the persons identified as the
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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
IntroductionThe 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 TerminologyGoals of This DocumentThe 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 ProtocolThe 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 InnovationThe "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.TerminologyThe 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 GroupsFor 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.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.)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 TransportData 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 SecuringThe 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 FormatsThe 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 ManagementThe 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:
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.
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.
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 ControlThe 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 FunctionsThese 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 ConsiderationsThis document has no IANA actions.Security ConsiderationsSecurity 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 .ReferencesNormative ReferencesKey words for use in RFCs to Indicate Requirement LevelsIn 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.An Offer/Answer Model with Session Description Protocol (SDP)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]RTP: A Transport Protocol for Real-Time ApplicationsThis 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]The Secure Real-time Transport Protocol (SRTP)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]WebRTC Video Processing and Codec RequirementsThis 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.WebRTC Audio Codec and Processing RequirementsThis document outlines the audio codec and processing requirements for WebRTC endpoints.Ambiguity of Uppercase vs Lowercase in RFC 2119 Key WordsRFC 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.Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) TraversalThis 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.Security Considerations for WebRTCWebRTC Security ArchitectureJavaScript Session Establishment Protocol (JSEP)WebRTC Data ChannelsWebRTC Data Channel Establishment ProtocolMedia Transport and Use of RTP in WebRTCTransports for WebRTCMedia Capture and StreamsW3C Candidate RecommendationWebRTC 1.0: Real-time Communication Between BrowsersW3C Proposed RecommendationInformative ReferencesHTML - Living StandardWHATWGSIP: Session Initiation ProtocolThis 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]Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) ServersA Mission Statement for the IETFThis 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.Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer ProtocolsThis 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]Multiplexing RTP Data and Control Packets on a Single PortThis 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]Extensible Messaging and Presence Protocol (XMPP): CoreThe 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]Conference Information Data Model for Centralized Conferencing (XCON)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]Web Real-Time Communication Use Cases and RequirementsThis 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.Traversal Using Relays around NAT (TURN) Server Auto DiscoveryCurrent 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.Differentiated Services Code Point (DSCP) Packet Markings for WebRTC QoSTrickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) ProtocolNegotiating Media Multiplexing Using the Session Description Protocol (SDP)WebRTC GatewaysThis 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 ProgressBidirectional-streams Over Synchronous HTTP (BOSH)ian.paterson@clientside.co.ukdizzyd@jabber.orgstpeter@jabber.orgjack@chesspark.comlance@andyet.comwinfried@tilanus.comJinglescottlu@google.comjbeda@google.comstpeter@jabber.orgrobert.mcqueen@collabora.co.ukseanegan@google.comjhildebr@cisco.comAcknowledgementsThe 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 AddressGoogleKungsbron 2Stockholm11122Swedenharald@alvestrand.no