| RTCWEB Working Group | C.H. Holmberg |
| Internet-Draft | S.H. Hakansson |
| Intended status: Informational | G.E. Eriksson |
| Expires: August 16, 2014 | Ericsson |
| February 12, 2014 |
Web Real-Time Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-14.txt
This document describes web based real-time communication use-cases. Requirements on the browser functionality are derived from the use-cases.
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 August 16, 2014.
Copyright (c) 2014 IETF Trust and the persons identified as the document authors. All rights reserved.
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This document presents a few use-cases of web applications that are executed in a browser and use real-time communication capabilities. In most of the use-cases all end-user clients are web applications, but there are some use-cases where at least one of the end-user clients is of another type (e.g. a mobile phone or a SIP UA).
Based on the use-cases, the document derives requirements related to browser functionality. These requirements are named "Fn", where n is an integer, and are listed in conjunction with the use-cases. A summary is provided in Section 4.2.
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 WGs efforts so far. It is proposed to be used in a later phase to evaluate the protocols and solutions developed by the WG.
This document also lists requirements related to the API to be used by web applications as an appendix. The reason is that the W3C WebRTC WG has decided to not develop its own use-case/requirement document, but instead use this document. These requirements are named "An", where n is an integer, and are described in Appendix A-
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14, RFC 2119 [RFC2119].
This section describes web based real-time communication use-cases, from which requirements are derived.
The following considerations are applicable to all use cases:
The requirements retrived from the Simple Video Communication Service use-case (Section 3.3.1) by default apply to all other use-cases, and are considred common. For each individual use-case, only the additional requirements are listed.
Two or more users have loaded a video communication web application into their browsers, provided by the same service provider, and logged into the service it provides. The web service publishes information about user login status by pushing updates to the web application in the browsers. When one online user selects a peer online user, a 1-1 audiovisual communication session between the browsers of the two peers is initiated. The invited user might accept or reject the session.
During session establishment a self-view is displayed, and once the session has been established the video sent from the remote peer is displayed in addition to the self-view. During the session, each user can select to remove and re-insert the self-view as often as desired. Each user can also change the sizes of his/her two video displays during the session. Each user can also pause sending of media (audio, video, or both) and mute incoming media
It is essential that media and data be encrypted, authenticated and integrity protected on a per-packet basis and that media and data packets failing the integrity check not be delivered to the application.
The application gives the users the opportunity to stop it from exposing the host IP address to the application of the other user.
Any session participant can end the session at any time.
The two users may be using communication devices with different operating systems and browsers from different vendors.
The web service monitors the quality of the service (focus on quality of audio and video) the end-users experience.
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F1 The browser must be able to use microphones and
cameras as input devices to generate streams.
----------------------------------------------------------------
F2 The browser must be able to send streams and
data to a peer in the presence of NATs.
----------------------------------------------------------------
F3 Transmitted streams and data must be rate
controlled (meaning that the browser must, regardless
of application behavior, reduce send rate when
there is congestion).
----------------------------------------------------------------
F4 The browser must be able to receive, process and
render streams and data ("render" does not
apply for data) from peers.
----------------------------------------------------------------
F5 The browser should be able to render good quality
audio and video even in the presence of
reasonable levels of jitter and packet losses.
----------------------------------------------------------------
F6 The browser must detect when a stream from a
peer is not received anymore
----------------------------------------------------------------
F7 When there are both incoming and outgoing audio
streams, echo cancellation must be made
available to avoid disturbing echo during
conversation.
----------------------------------------------------------------
F8 The browser must support synchronization of
audio and video.
----------------------------------------------------------------
F9 The browser should use encoding of streams
suitable for the current rendering (e.g.
video display size) and should change parameters
if the rendering changes during the session
----------------------------------------------------------------
F10 The browser must support a baseline audio and
video codec
----------------------------------------------------------------
F11 It must be possible to protect streams and data
from wiretapping [RFC2804].
----------------------------------------------------------------
F12 The browser must enable verification, given
the right circumstances and by use of other
trusted communication, that streams and
data received have not been manipulated by
any party.
----------------------------------------------------------------
F13 The browser must encrypt, authenticate and
integrity protect media and data on a
per-packet basis, and must drop incoming media
and data packets that fail the per-packet
integrity check. In addition, the browser
must support a mechanism for cryptographically
binding media and data security keys to the
user identity (see R-ID-BINDING in [RFC5479]).
----------------------------------------------------------------
F14 The browser must make it possible to set up a
call between two parties without one party
learning the other party's host IP address.
----------------------------------------------------------------
F15 The browser must be able to collect statistics,
related to the transport of audio and video
between peers, needed to estimate quality of
experience.
----------------------------------------------------------------
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A25, A26
This use-case is almost identical to the Simple Video Communication Service use-case (Section 3.3.1). The difference is that one of the users is behind a NAT/Firewall that blocks UDP traffic.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F18 The browser must be able to send streams and data to a peer in the presence of NATs and Firewalls that block UDP traffic. ----------------------------------------------------------------
This use-case is almost identical to the Simple Video Communication Service use-case (Section 3.3.1). The difference is that one of the users is behind a Firewall that only allows traffic via a HTTP Proxy.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F21 The browser must be able to send streams and data to a peer in the presence of Firewalls that only allows traffic via a HTTP Proxy, when Firewall policy allows WebRTC traffic. ----------------------------------------------------------------
This use-case is almost identical to the Simple Video Communication Service use-case (Section 3.3.1).
What is added is that the service provider is operating over large geographical areas (or even globally).
Assuming that ICE will be used, this means that the service provider would like to be able to provide several STUN and TURN servers (via the app) to the browser; selection of which one(s) to use is part of the ICE processing. Other reasons for wanting to provide several STUN and TURN servers include support for IPv4 and IPv6, load balancing and redundancy.
Note that ICE support being mandatory does not preclude a WebRTC endpoint from supporting more traversal mechanisms than ICE using STUN and TURN.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F19 The browser must be able to use several STUN and TURN servers ----------------------------------------------------------------
A22
This use-case is similar to the Simple Video Communication Service use-case (Section 3.3.1).
What is added is aspects when using the service in enterprises. ICE is assumed in the further description of this use-case.
An enterprise that uses a RTCWEB based web application for communication desires to audit all RTCWEB based application sessions used from inside the company towards any external peer. To be able to do this they deploy a TURN server that straddles the boundary between the internal and the external network.
The firewall will block all attempts to use STUN with an external destination unless they go to the enterprise auditing TURN server. In cases where employees are using RTCWEB applications provided by an external service provider they still want the traffic to stay inside their internal network and in addition not load the straddling TURN server, thus they deploy a STUN server allowing the RTCWEB client to determine its server reflexive address on the internal side. Thus enabling cases where peers are both on the internal side to connect without the traffic leaving the internal network. It must be possible to configure the browsers used in the enterprise with network specific STUN and TURN servers. This should be possible to achieve by auto-configuration methods. The RTCWEB functionality will need to utilize both network specific STUN and TURN resources and STUN and TURN servers provisioned by the web application.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F20 The browser must support the use of STUN and TURN servers that are supplied by entities other than the web application (i.e. the network provider). ----------------------------------------------------------------
This use-case is almost identical to the Simple Video Communication Service use-case (Section 3.3.1). The difference is that the user changes network access during the session.
The communication device used by one of the users has several network adapters (Ethernet, WiFi, Cellular). The communication device is accessing the Internet using Ethernet, but the user has to start a trip during the session. The communication device automatically changes to use WiFi when the Ethernet cable is removed and then moves to cellular access to the Internet when moving out of WiFi coverage. The session continues even though the access method changes.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F17 The communication session must survive across a change of the network interface used by the session ----------------------------------------------------------------
This use-case is almost identical to the Simple Video Communication Service, access change use-case (Section 3.3.6). The use of Quality of Service (QoS) capabilities is added:
The user in the previous use case that starts a trip is behind a common residential router that supports prioritization of traffic. In addition, the user's provider of cellular access has QoS support enabled. The user is able to take advantage of the QoS support both when accessing via the residential router and when using cellular.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F17 The communication session must survive across a change of the network interface used by the session ---------------------------------------------------------------- F22 The browser must be able to receive streams and data from multiple peers concurrently. ----------------------------------------------------------------
This use-case has the audio and video communication of the Simple Video Communication Service use-case (Section 3.3.1).
But in addition to this, one of the users can share what is being displayed on her/his screen with a peer. The user can choose to share the entire screen, part of the screen (part selected by the user) or what a selected application displays with the peer.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F36 The browser must be able to generate streams using the entire user display, a specific area of the user's display or the information being displayed by a specific application. ----------------------------------------------------------------
A21
This use-case has the audio and video communication of the Simple Video Communication Service use-case (Section 3.3.1).
But in addition to this, the users can send and receive files stored in the file system of the device used.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F35 The browser must be able to send reliable data traffic to a peer browser. ----------------------------------------------------------------
A21, A24
An ice-hockey club uses an application that enables talent scouts to, in real-time, show and discuss games and players with the club manager. The talent scouts use a mobile phone with two cameras, one front facing and one rear facing.
The club manager uses a desktop, equipped with one camera, for viewing the game and discussing with the talent scout.
Before the game starts, and during game breaks, the talent scout and the manager have a 1-1 audiovisual communication session. On the mobile phone, only the camera facing the talent scout is used. On the user display of the mobile phone, the video of the club manager is shown with a picture-in-picture thumbnail of the rear facing camera (self-view). On the display of the desktop, the video of the talent scout is shown with a picture-in-picture thumbnail of the desktop camera (self-view).
When the game is on-going, the talent scout activates the use of the front facing camera, and that stream is sent to the desktop (the stream from the rear facing camera continues to be sent all the time). The video stream captured by the front facing camera (that is capturing the game) of the mobile phone is shown in a big window on the desktop screen, with picture-in-picture thumbnails of the rear facing camera and the desktop camera (self-view). On the display of the mobile phone the game is shown (front facing camera) with picture-in-picture thumbnails of the rear facing camera (self-view) and the desktop camera. As the most important stream in this phase is the video showing the game, the application used in the talent scout's mobile sets higher priority for that stream.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F22 The browser should be able to take advantage of available capabilities (supplied by network nodes) to prioritize voice, video and data appropriately. ---------------------------------------------------------------- F25 The browser must be able to render several concurrent video streams ----------------------------------------------------------------
A17, A23
In this use-case, the Simple Video Communication Service use-case (Section 3.3.1) is extended by allowing multiparty sessions. No central server is involved - the browser of each participant sends and receives streams to and from all other session participants. The web application in the browser of each user is responsible for setting up streams to all receivers.
In order to enhance the user experience, the web application renders the audio coming from different particiapants so that it is experienced to come from different spatial locations. This is done automatically, but users can change how the different participants are placed in the (virtual) room. In addition the levels in the audio signals are adjusted before mixing.
Another feature intended to enhance the use experience is that the video window that displays the video of the currently speaking peer is highlighted.
Each video stream received is by default displayed in a thumbnail frame within the browser, but users can change the display size.
Note: What this use-case adds in terms of requirements is capabilities to send streams to and receive streams from several peers concurrently, as well as the capabilities to render the video from all received streams and be able to spatialize, level adjust and mix the audio from all received streams locally in the browser. It also adds the capability to measure the audio level/activity.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F23 The browser must be able to transmit streams and data to several peers concurrently. ---------------------------------------------------------------- F24 The browser must be able to receive streams and data from multiple peers concurrently. ---------------------------------------------------------------- F25 The browser must be able to render several concurrent video streams ---------------------------------------------------------------- F26 The browser must be able to mix several audio streams. ---------------------------------------------------------------- F27 The browser must be able to apply spatialization effects when playing audio streams. ---------------------------------------------------------------- F28 The browser must be able to measure the voice activity level in audio streams. ---------------------------------------------------------------- F29 The browser must be able to change the voice activity level in audio streams. ----------------------------------------------------------------
A13, A14, A15, A16
This use case is based on the previous one. In this use-case, the voice part of the multiparty video communication use case is used in the context of an on-line game. The received voice audio media is rendered together with game sound objects. For example, the sound of a tank moving from left to right over the screen must be rendered and played to the user together with the voice media.
Quick updates of the game state is required, and have higher priority than the voice.
Note: the difference regarding local audio processing compared to the "Multiparty video communication" use-case is that other sound objects than the streams must be possible to be included in the spatialization and mixing. "Other sound objects" could for example be a file with the sound of the tank; that file could be stored locally or remotely.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F22 The browser should be able to take advantage of available capabilities (supplied by network nodes) to prioritize voice, video and data appropriately. ---------------------------------------------------------------- F23 The browser must be able to transmit streams and data to several peers concurrently. ---------------------------------------------------------------- F24 The browser must be able to receive streams and data from multiple peers concurrently. ---------------------------------------------------------------- F25 The browser must be able to render several concurrent video streams ---------------------------------------------------------------- F26 The browser must be able to mix several audio streams. ---------------------------------------------------------------- F27 The browser must be able to apply spatialization effects when playing audio streams. ---------------------------------------------------------------- F28 The browser must be able to measure the voice activity level in audio streams. ---------------------------------------------------------------- F29 The browser must be able to change the voice activity level in audio streams. ---------------------------------------------------------------- F30 The browser must be able to process and mix sound objects (media that is retrieved from another source than the established media stream(s) with the peer(s) with audio streams. ---------------------------------------------------------------- F34 The browser must be able to send short latency unreliable datagram traffic to a peer browser [RFC5405]. ----------------------------------------------------------------
A13, A14, A15, A16, A17, A18, A23
A mobile telephony operator allows its customers to use a web browser to access their services. After a simple log in the user can place and receive calls in the same way as when using a normal mobile phone. When a call is received or placed, the identity is shown in the same manner as when a mobile phone is used.
Note: With "place and receive calls in the same way as when using a normal mobile phone" it is meant that you can dial a number, and that your mobile telephony operator has made available your phone contacts on line, so they are available and can be clicked to call, and be used to present the identity of an incoming call. If the callee is not in your phone contacts the number is displayed. Furthermore, your call logs are available, and updated with the calls made/received from the browser. And for people receiving calls made from the web browser the usual identity (i.e. the phone number of the mobile phone) will be presented.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F31 The browser must support an audio media format (codec) that is commonly supported by existing telephony services. ---------------------------------------------------------------- F33 The browser must be able to initiate and accept a media session where the data needed for establishment can be carried in SIP. ----------------------------------------------------------------
Alice uses her web browser with a service that allows her to call PSTN numbers. Alice calls 1-800-gofedex. Alice should be able to hear the initial prompts from the fedex Interactive Voice Responder (IVR) and when the IVR says press 1, there should be a way for Alice to navigate the IVR.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F31 The browser must support an audio media format (codec) that is commonly supported by existing telephony services. ---------------------------------------------------------------- F32 There should be a way to navigate a Dual-tone multi-frequency signaling (DTMF) based Interactive voice response (IVR) System ----------------------------------------------------------------
An organization uses a video communication system that supports the establishment of multiparty video sessions using a central conference server.
The browser of each participant sends an audio stream (type in terms of mono, stereo, 5.1, ... depending on the equipment of the participant) to the central server. The central server mixes the audio streams (and can in the mixing process naturally add effects such as spatialization) and sends towards each participant a mixed audio stream which is played to the user.
The browser of each participant sends video towards the server. For each participant one high resolution video is displayed in a large window, while a number of low resolution videos are displayed in smaller windows. The server selects what video streams to be forwarded as main- and thumbnail videos respectively, based on speech activity. As the video streams to display can change quite frequently (as the conversation flows) it is important that the delay from when a video stream is selected for display until the video can be displayed is short.
All participants are authenticated by the central server, and authorized to connect to the central server. The participants are identified to each other by the central server, and the participants do not have access to each others' credentials such as e-mail addresses or login IDs.
Note: This use-case adds requirements on support for fast stream switches F16. There exist several solutions that enable the server to forward one high resolution and several low resolution video streams: a) each browser could send a high resolution, but scalable stream, and the server could send just the base layer for the low resolution streams, b) each browser could in a simulcast fashion send one high resolution and one low resolution stream, and the server just selects or c) each browser sends just a high resolution stream, the server transcodes into low resolution streams as required.
---------------------------------------------------------------- REQ-ID DESCRIPTION ---------------------------------------------------------------- F16 The browser must support insertion of reference frames in outgoing media streams when requested by a peer. ---------------------------------------------------------------- F25 The browser must be able to render several concurrent video streams ----------------------------------------------------------------
This section contains the requirements on the browser derived from the use-cases in Section 3.
NOTE: It is assumed that the user applications are executed on a browser. Whether the capabilities to implement specific browser requirements are implemented by the browser application, or are provided to the browser application by the underlying operating system, is outside the scope of this document.
----------------------------------------------------------------
Common, basic requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F1 The browser must be able to use microphones and
cameras as input devices to generate streams.
----------------------------------------------------------------
F2 The browser must be able to send streams and
data to a peer in the presence of NATs.
----------------------------------------------------------------
F3 Transmitted streams and data must be rate
controlled (meaning that the browser must, regardless
of application behavior, reduce send rate when
there is congestion).
----------------------------------------------------------------
F4 The browser must be able to receive, process and
render streams and data ("render" does not
apply for data) from peers.
----------------------------------------------------------------
F5 The browser should be able to render good quality
audio and video even in the presence of
reasonable levels of jitter and packet losses.
----------------------------------------------------------------
F6 The browser must detect when a stream from a
peer is not received anymore
----------------------------------------------------------------
F7 When there are both incoming and outgoing audio
streams, echo cancellation must be made
available to avoid disturbing echo during
conversation.
----------------------------------------------------------------
F8 The browser must support synchronization of
audio and video.
----------------------------------------------------------------
F9 The browser should use encoding of streams
suitable for the current rendering (e.g.
video display size) and should change parameters
if the rendering changes during the session
----------------------------------------------------------------
F10 The browser must support a baseline audio and
video codec
----------------------------------------------------------------
F11 It must be possible to protect streams and data
from wiretapping [RFC2804].
----------------------------------------------------------------
F12 The browser must enable verification, given
the right circumstances and by use of other
trusted communication, that streams and
data received have not been manipulated by
any party.
----------------------------------------------------------------
F13 The browser must encrypt, authenticate and
integrity protect media and data on a
per-packet basis, and must drop incoming media
and data packets that fail the per-packet
integrity check. In addition, the browser
must support a mechanism for cryptographically
binding media and data security keys to the
user identity (see R-ID-BINDING in [RFC5479]).
----------------------------------------------------------------
F14 The browser must make it possible to set up a
call between two parties without one party
learning the other party's host IP address.
----------------------------------------------------------------
F15 The browser must be able to collect statistics,
related to the transport of audio and video
between peers, needed to estimate quality of
experience.
----------------------------------------------------------------
Requirements related to network and topology
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F16 The browser must support insertion of reference frames
in outgoing media streams when requested by a peer.
----------------------------------------------------------------
F17 The communication session must survive across a
change of the network interface used by the
session
----------------------------------------------------------------
F18 The browser must be able to send streams and
data to a peer in the presence of NATs and
Firewalls that block UDP traffic.
----------------------------------------------------------------
F19 The browser must be able to use several STUN
and TURN servers
----------------------------------------------------------------
F20 The browser must support the use of STUN and TURN
servers that are supplied by entities other than
the web application (i.e. the network provider).
----------------------------------------------------------------
F21 The browser must be able to send streams and
data to a peer in the presence of Firewalls that only
allows traffic via a HTTP Proxy, when Firewall policy
allows WebRTC traffic.
----------------------------------------------------------------
F22 The browser should be able to take advantage
of available capabilities (supplied by network
nodes) to prioritize voice, video and data
appropriately.
----------------------------------------------------------------
Requirements related to multiple peers and streams
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F23 The browser must be able to transmit streams and
data to several peers concurrently.
----------------------------------------------------------------
F24 The browser must be able to receive streams and
data from multiple peers concurrently.
----------------------------------------------------------------
F25 The browser must be able to render several
concurrent video streams
----------------------------------------------------------------
F26 The browser must be able to mix several
audio streams.
----------------------------------------------------------------
Requirements related to audio processing
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F27 The browser must be able to apply spatialization
effects when playing audio streams.
----------------------------------------------------------------
F28 The browser must be able to measure the
voice activity level in audio streams.
----------------------------------------------------------------
F29 The browser must be able to change the
voice activity level in audio streams.
----------------------------------------------------------------
F30 The browser must be able to process and mix
sound objects (media that is retrieved from
another source than the established media
stream(s) with the peer(s) with audio streams.
----------------------------------------------------------------
Requirements related to legacy interop
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F31 The browser must support an audio media format
(codec) that is commonly supported by existing
telephony services.
----------------------------------------------------------------
F32 There should be a way to navigate
a Dual-tone multi-frequency signaling (DTMF)
based Interactive voice response (IVR) System
----------------------------------------------------------------
F33 The browser must be able to initiate and
accept a media session where the data needed
for establishment can be carried in SIP.
----------------------------------------------------------------
Other requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F34 The browser must be able to send short
latency unreliable datagram traffic to a
peer browser [RFC5405].
----------------------------------------------------------------
F35 The browser must be able to send reliable
data traffic to a peer browser.
----------------------------------------------------------------
F36 The browser must be able to generate streams
using the entire user display, a specific area
of the user's display or the information being
displayed by a specific application.
----------------------------------------------------------------
There are no IANA actions in this document.
A malicious web application might use the browser to perform Denial Of Service (DOS) attacks on NAT infrastructure, or on peer devices. Also, a malicious web application might silently establish outgoing, and accept incoming, streams on an already established connection.
Based on the identified security risks, this section will describe security considerations for the browser and web application.
The browser is expected to provide mechanisms for getting user consent to use device resources such as camera and microphone.
The browser is expected to provide mechanisms for informing the user that device resources such as camera and microphone are in use ("hot").
The browser is expected to provide mechanisms for users to revise and even completely revoke consent to use device resources such as camera and microphone.
The browser is expected to provide mechanisms for getting user consent to use the screen (or a certain part of it) or what a certain application displays on the screen as source for streams.
The browser is expected to provide mechanisms for informing the user that the screen, part thereof or an application is serving as a stream source ("hot").
The browser is expected to provide mechanisms for users to revise and even completely revoke consent to use the screen, part thereof or an application is serving as a stream source.
The browser is expected to provide mechanisms in order to assure that streams are the ones the recipient intended to receive.
The browser is expected to provide mechanisms that allows the users to verify that the streams received have not be manipulated (F12).
The browser needs to ensure that media is not sent, and that received media is not rendered, until the associated stream establishment and handshake procedures with the remote peer have been successfully finished.
The browser needs to ensure that the stream negotiation procedures are not seen as Denial Of Service (DOS) by other entities.
The web application is expected to ensure user consent in sending and receiving media streams.
The authors wish to thank Bernard Aboba, Gunnar Hellstrom, Martin Thomson, Lars Eggert, Matthew Kaufman, Emil Ivov, Eric Rescorla, Eric Burger, John Leslie, Dan Wing, Richard Barnes, Barry Dingle, Dale Worley, Ted hardie, Mary Barnes, Dan Burnett, Stephan Wenger, Harald Alvestrand, Cullen Jennings, Andrew Hutton and everyone else in the RTCWEB community that have provided comments, feedback, text and improvement proposals on the document.
[RFC EDITOR NOTE: Please remove this section when publishing]
Changes from draft-ietf-rtcweb-use-cases-and-requirements-10
Changes from draft-ietf-rtcweb-use-cases-and-requirements-09
Changes from draft-ietf-rtcweb-use-cases-and-requirements-08
Changes from draft-ietf-rtcweb-use-cases-and-requirements-07
Changes from draft-ietf-rtcweb-use-cases-and-requirements-06
Changes from draft-ietf-rtcweb-use-cases-and-requirements-05
Changes from draft-ietf-rtcweb-use-cases-and-requirements-04
Changes from draft-ietf-rtcweb-use-cases-and-requirements-03
Changes from draft-ietf-rtcweb-use-cases-and-requirements-02
Changes from draft-ietf-rtcweb-ucreqs-01
Changes from draft-ietf-rtcweb-ucreqs-00
Changes from draft-holmberg-rtcweb-ucreqs-01
Changes from draft-holmberg-rtcweb-ucreqs-00
| [RFC2119] | Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. |
| [RFC2804] | IABIESG, "IETF Policy on Wiretapping", RFC 2804, May 2000. |
| [RFC5405] | Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines for Application Designers", BCP 145, RFC 5405, November 2008. |
| [RFC5479] | Wing, D., Fries, S., Tschofenig, H. and F. Audet, "Requirements and Analysis of Media Security Management Protocols", RFC 5479, April 2009. |
This section contains the requirements on the API derived from the use-cases in Section 3.
REQ-ID DESCRIPTION ---------------------------------------------------------------- A1 The Web API must provide means for the application to ask the browser for permission to use cameras and microphones as input devices. ---------------------------------------------------------------- A2 The Web API must provide means for the web application to control how streams generated by input devices are used. ---------------------------------------------------------------- A3 The Web API must provide means for the web application to control the local rendering of streams (locally generated streams and streams received from a peer). ---------------------------------------------------------------- A4 The Web API must provide means for the web application to initiate sending of stream/stream components to a peer. ---------------------------------------------------------------- A5 The Web API must provide means for the web application to control the media format (codec) to be used for the streams sent to a peer. NOTE: The level of control depends on whether the codec negotiation is handled by the browser or the web application. ---------------------------------------------------------------- A6 The Web API must provide means for the web application to modify the media format for streams sent to a peer after a media stream has been established. ---------------------------------------------------------------- A7 The Web API must provide means for informing the web application of whether the establishment of a stream with a peer was successful or not. ---------------------------------------------------------------- A8 The Web API must provide means for the web application to mute/unmute a stream or stream component(s). When a stream is sent to a peer mute status must be preserved in the stream received by the peer. ---------------------------------------------------------------- A9 The Web API must provide means for the web application to cease the sending of a stream to a peer. ---------------------------------------------------------------- A10 The Web API must provide means for the web application to cease processing and rendering of a stream received from a peer. ---------------------------------------------------------------- A11 The Web API must provide means for informing the web application when a stream from a peer is no longer received. ---------------------------------------------------------------- A12 The Web API must provide means for informing the web application when high loss rates occur. ---------------------------------------------------------------- A13 The Web API must provide means for the web application to apply spatialization effects to audio streams. ---------------------------------------------------------------- A14 The Web API must provide means for the web application to detect the level in audio streams. ---------------------------------------------------------------- A15 The Web API must provide means for the web application to adjust the level in audio streams. ---------------------------------------------------------------- A16 The Web API must provide means for the web application to mix audio streams. ---------------------------------------------------------------- A17 The Web API must provide a way to identify streams such that an application is able to match streams on a sending peer with the same stream on all receiving peers. ---------------------------------------------------------------- A18 The Web API must provide a mechanism for sending and receiving isolated discrete chunks of data. ---------------------------------------------------------------- A19 The Web API must provide means for the web application to indicate the type of audio signal (speech, audio) for audio stream(s)/stream component(s). ---------------------------------------------------------------- A20 It must be possible for an initiator or a responder web application to indicate the types of media it is willing to accept incoming streams for when setting up a connection (audio, video, other). The types of media to be accepted can be a subset of the types of media the browser is able to accept. ---------------------------------------------------------------- A21 The Web API must provide means for the application to ask the browser for permission to the screen, a certain area on the screen or what a certain application displays on the screen as input to streams. ---------------------------------------------------------------- A22 The Web API must provide means for the application to specify several STUN and/or TURN servers to use. ---------------------------------------------------------------- A23 The Web API must provide means for the application to specify the priority to apply for outgoing streams and data. ---------------------------------------------------------------- A24 The Web API must provide a mechanism for sending and receiving files. ---------------------------------------------------------------- A25 It must be possible for the application to instruct the browser to refrain from exposing the host IP address to the application ---------------------------------------------------------------- A26 The Web API must provide means for the application to obtain the statistics (related to transport, and collected by the browser) needed to estimate quality of service. ----------------------------------------------------------------