RTCWEB Working Group C. Holmberg Internet-Draft S. Hakansson Intended status: Informational G. Eriksson Expires: February 29, 2012 Ericsson August 28, 2011 Web Real-Time Communication Use-cases and Requirements draft-ietf-rtcweb-use-cases-and-requirements-03.txt Abstract This document describes web based real-time communication use-cases. Based on the use-cases, the document also derives requirements related to the browser, and the API used by web applications to request and control media stream services provided by the browser. Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." This Internet-Draft will expire on February 29, 2012. Copyright Notice Copyright (c) 2011 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Holmberg, et al. Expires February 29, 2012 [Page 1]
Internet-Draft RTC-Web August 2011 Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3 4. Use-cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3 4.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 3 4.2. Browser-to-browser use-cases . . . . . . . . . . . . . . . 3 4.2.1. Simple Video Communication Service . . . . . . . . . . 3 4.2.2. Simple Video Communication Service, NAT/FW that blocks UDP . . . . . . . . . . . . . . . . . . . . . . 4 4.2.3. Simple Video Communication Service, access change . . 4 4.2.4. Simple Video Communication Service, QoS . . . . . . . 5 4.2.5. Simple video communication service with inter-operator calling . . . . . . . . . . . . . . . . 5 4.2.6. Hockey Game Viewer . . . . . . . . . . . . . . . . . . 6 4.2.7. Multiparty video communication . . . . . . . . . . . . 6 4.2.8. Multiparty on-line game with voice communication . . . 7 4.2.9. Distributed Music Band . . . . . . . . . . . . . . . . 8 4.3. Browser - GW/Server use cases . . . . . . . . . . . . . . 8 4.3.1. Telephony terminal . . . . . . . . . . . . . . . . . . 8 4.3.2. Fedex Call . . . . . . . . . . . . . . . . . . . . . . 8 4.3.3. Video conferencing system with central server . . . . 9 5. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 10 5.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 10 5.2. Browser requirements . . . . . . . . . . . . . . . . . . . 10 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12 7. Security Considerations . . . . . . . . . . . . . . . . . . . 12 7.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 12 7.2. Browser Considerations . . . . . . . . . . . . . . . . . . 13 7.3. Web Application Considerations . . . . . . . . . . . . . . 13 8. Additional use cases . . . . . . . . . . . . . . . . . . . . . 13 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14 10. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . . 14 11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15 11.1. Normative References . . . . . . . . . . . . . . . . . . . 15 11.2. Informative References . . . . . . . . . . . . . . . . . . 16 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16 Holmberg, et al. Expires February 29, 2012 [Page 2]
Internet-Draft RTC-Web August 2011 1. Introduction This document presents a few use-case of web applications that are executed in a browser and use real-time communication capabilities. Based on the use-cases, the document derives requirements related to the browser and the API used by web applications in the browser. The requirements related to the browser are named "Fn" and are described in section Section 5.2 The requirements related to the API are named "An" and are described in the external document [webrtc_reqs] The document focuses on requirements related to real-time media streams. Requirements related to privacy, signalling between the browser and web server etc are currently not considered. 2. Conventions 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]. 3. Definitions TBD 4. Use-cases 4.1. Introduction This section describes web based real-time communication use-cases, from which requirements are later derived. 4.2. Browser-to-browser use-cases 4.2.1. Simple Video Communication Service 4.2.1.1. Description In the service the users have loaded, and logged into, a video communication web application into their browsers, provided by the same service provider. The web service publishes information about user login status, by pushing updates to the web application in the Holmberg, et al. Expires February 29, 2012 [Page 3]
Internet-Draft RTC-Web August 2011 browsers. By selecting an online peer user, a 1-1 video communication session between the browsers of the peers is initiated. The invited user might accept or reject the session. When the session has been established, a self-view, as well as the video sent from the remote peer, are displayed. The users can change the sizes of the video displays during the session. The users can also pause sending of media (audio, video, or both), and mute incoming media. Any session participant can end the session at any time. The users are using communication devices of different makes, with different Operating Systems and Browsers from different vendors. One user has an unreliable internet connection. It sometimes has packet losses, and is sometimes goes down completely. One user is located behind a Network Address Translator (NAT). 4.2.1.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F22, F25 A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13 4.2.2. Simple Video Communication Service, NAT/FW that blocks UDP 4.2.2.1. Description This use case is almost identical to the previos one. The difference is that one of the users is behind a NAT that blocks UDP traffic. 4.2.2.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F22, F23, F25, F26 A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13 4.2.3. Simple Video Communication Service, access change 4.2.3.1. Description This use case is almost identical to "4.2.1 Simple Video Communication Service". The difference is that the user changes network access during the session: The communication device used byt one of the users have several Holmberg, et al. Expires February 29, 2012 [Page 4]
Internet-Draft RTC-Web August 2011 network adapters (Ethernet, WiFi, Cellular). The communication device is access 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. 4.2.3.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F22, F23, F25 A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13 4.2.4. Simple Video Communication Service, QoS 4.2.4.1. Description This use case is almost identical to the previos one. The use of 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. 4.2.4.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F21, F22, F23, F25 A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13 4.2.5. Simple video communication service with inter-operator calling 4.2.5.1. Description Two users have logged into two different web applications, provided by different service providers. The service providers are interconnected by some means, but exchange no more information about the users than what can be carried using SIP. NOTE: More profiling of what this means may be needed. Each web service publishes information about user login status for users that have a relationship with the other user; how this is Holmberg, et al. Expires February 29, 2012 [Page 5]
Internet-Draft RTC-Web August 2011 established is out of scope. The same functionality as in the "4.2.1 Simple Video Communication Service" is available. The same issues with connectivity apply. 4.2.5.2. Derived requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F22, F24, F25 A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13 4.2.6. Hockey Game Viewer 4.2.6.1. Description 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 for viewing the game and discussing with the talent scout. 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, while a thumbnail of the rear facing camera is overlaid. Most of the mobile phone screen is covered by a self view of the front facing camera. A thumbnail of the rear facing cameras view is overlaid. 4.2.6.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F14 A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15 4.2.7. Multiparty video communication 4.2.7.1. Description In this use case the simple video communication service 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. Holmberg, et al. Expires February 29, 2012 [Page 6]
Internet-Draft RTC-Web August 2011 The audio sent by each participant is a mono stream. However, in order to enhance intelligibility, the web application pans the audio from different participants differently when rendering the audio. This is done automatically, but users can change how the different participants are placed in the (virtual) room. 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 uses 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 recevied streams and be able to spatialize and mix the audio from all received streams locally in the browser. 4.2.7.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F11, F12, F13, F14, F22 A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13, A14, A15 4.2.8. Multiparty on-line game with voice communication 4.2.8.1. Description In this use-case, the voice part of the multiparty video communication application 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. 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 a file with the sound of the tank, that file could be stored locally or remotely. 4.2.8.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F11, F12, F13, F15, F20 A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A16 Holmberg, et al. Expires February 29, 2012 [Page 7]
Internet-Draft RTC-Web August 2011 4.2.9. Distributed Music Band 4.2.9.1. Description In this use-case, a music band is playing music while the members are at different physical locations. No central server is used, instead all streams are set up in a mesh fashion. Discussion: This use case was briefly discussed at the Quebec webrtc meeting and it got support. So far the only concrete requirement (A17) derived is that the application must be able to ask the browser to treat the audio signal as audio (in contrast to speech). However, the use case should be further analysed to determine other requirements (could be e.g. on delay mic->speaker, level control of audio signals, etc.). 4.2.9.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F11, F12, F13 A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A17 4.3. Browser - GW/Server use cases 4.3.1. Telephony terminal 4.3.1.1. Description 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 will be shown in the same manner as when a mobile phone used. 4.3.1.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F18 A1, A2, A3, A4, A7, A8, A9, A10, A11, A12, A13 4.3.2. Fedex Call 4.3.2.1. Description Alice uses her web browser with a service something like Skype to be able to phone PSTN numbers. Alice calls 1-800-gofedex. Alice should be able to hear the initial prompts from the fedex IVR and when the IVR says press 1, there should be a way for Alice to navigate the Holmberg, et al. Expires February 29, 2012 [Page 8]
Internet-Draft RTC-Web August 2011 IVR. 4.3.2.2. Derived Requirements F1, F2, F3, F4, F5, F6, F8, F9, F10, F18, F19 A1, A2, A3, A4, A7, A8, A9, A10, A11, A12, A13 4.3.3. Video conferencing system with central server 4.3.3.1. Description An organization uses a video communication system that supports the establishment of multiparty video sessions using a central conference server. The browsers of all participants send an audio stream (mono or stereo depending on the equipment of a participant) to the central server. The central server mixes the audio streams and sends towards the participants a mixed stereo audio stream. Each participant sends two video streams in a simulcast fashion towards the server, one low resolution and one high resolution. At 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, 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. The organization has an internal network set up with an aggressive firewall handling access to the internet. If users can not physically access the internal network, they can establish a Virtual Private Network (VPN). It is essential that the communication can not be eavesdropped. All participant 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 F7, on encryption of media and on ability to traverse very restrictive FWs. It also introduces simulcast, but no concrete Holmberg, et al. Expires February 29, 2012 [Page 9]
Internet-Draft RTC-Web August 2011 requirement is put for this. 4.3.3.2. Derived Requirements F1, F2, F3, F4, F5, F6, F7, F8, F9, F10, F14, F16, F17 A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15 5. Requirements 5.1. General This section contains the browser requirements, derived from the use- cases in section 4. For the API requirements refer to [webrtc_reqs]. 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 (OS), is outside the scope of this document. 5.2. Browser 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 to a peer in presence of NATs. ---------------------------------------------------------------- F3 Transmitted streams MUST be rate controlled. ---------------------------------------------------------------- F4 The browser MUST be able to receive, process and render streams from peers. ---------------------------------------------------------------- F5 The browser MUST be able to render good quality audio and video even in presence of reasonable levels of jitter and packet losses. TBD: What is a reasonable level? ---------------------------------------------------------------- F6 The browser MUST be able to handle high loss and jitter levels in a graceful way. ---------------------------------------------------------------- F7 The browser MUST support fast stream switches. Holmberg, et al. Expires February 29, 2012 [Page 10]
Internet-Draft RTC-Web August 2011 ---------------------------------------------------------------- F8 The browser MUST detect when a stream from a peer is not received any more ---------------------------------------------------------------- F9 When there are both incoming and outgoing audio streams, echo cancellation MUST be made available to avoid disturbing echo during conversation. QUESTION: How much control should be left to the web application? ---------------------------------------------------------------- F10 The browser MUST support synchronization of audio and video. QUESTION: How much control should be left to the web application? ---------------------------------------------------------------- F11 The browser MUST be able to transmit streams to several peers concurrently. ---------------------------------------------------------------- F12 The browser MUST be able to receive streams from multiple peers concurrently. ---------------------------------------------------------------- F13 The browser MUST be able to pan, mix and render several concurrent audio streams. ---------------------------------------------------------------- F14 The browser MUST be able to render several concurrent video streams ---------------------------------------------------------------- F15 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). ---------------------------------------------------------------- F16 Streams MUST be able to pass through restrictive firewalls. ---------------------------------------------------------------- F17 It MUST be possible to protect streams from eavesdropping. ---------------------------------------------------------------- F18 The browser MUST support an audio media format (codec) that is commonly supported by existing telephony services. QUESTION: G.711? ---------------------------------------------------------------- F19 there should be a way to navigate Holmberg, et al. Expires February 29, 2012 [Page 11]
Internet-Draft RTC-Web August 2011 the IVR ---------------------------------------------------------------- F20 The browser must be able to send short latency datagram traffic to a peer browser ---------------------------------------------------------------- F21 The browser MUST be able to take advantage of capabilities to prioritize voice and video appropriately. ---------------------------------------------------------------- F22 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 ---------------------------------------------------------------- F23 It MUST be possible to move from one network interface to another one ---------------------------------------------------------------- F24 The browser MUST be able to initiate and accept a media session where the data needed for establishment can be carried in SIP. ---------------------------------------------------------------- F25 The browser MUST support a baseline audio and video codec ---------------------------------------------------------------- F26 The browser MUST be able to send streams to a peer in presence of NATs that block UDP traffic. ---------------------------------------------------------------- 6. IANA Considerations TBD 7. Security Considerations 7.1. Introduction 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. Holmberg, et al. Expires February 29, 2012 [Page 12]
Internet-Draft RTC-Web August 2011 7.2. Browser Considerations 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 consent to use device resources such as camera and microphone. The browser is expected to provide mechanisms in order to assure that streams are the ones the recipient intended to receive. The browser is 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. 7.3. Web Application Considerations The web application is expected to ensure user consent in sending and receiving media streams. 8. Additional use cases Several additional use cases have been discusses. At this point these use cases are not included as requirement deriving use cases for different reasons (lack of documentation, overlap with existing use cases, lack of consensus). For completeness these additional use cases are listed below: 1. Use cases regarding different situations when being invited to a "session", e.g. browser open, browser open but another tab active, browser open but active in session, browser closed, .... (Matthew Kaufman); discussed at webrtc meeting 2. Different TURN provider scenarios (Cullen Jennings); discussed at the webrtc meeting 3. E911 (Paul Beaumont) http://www.ietf.org/mail-archive/web/ rtcweb/current/msg00525.html 4. Local Recording and Remote recording (John): Discussed a _lot_ on the mail lists (rtcweb as well as public-webrtc late August 2011. Not concluded at time of writing. Holmberg, et al. Expires February 29, 2012 [Page 13]
Internet-Draft RTC-Web August 2011 5. Emergency access for disabled (Bernard Aboba) http:// www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html 6. Clue use cases (Roni Even) http://tools.ietf.org/html/ draft-ietf-clue-telepresence-use-cases-01 7. Rohan red cross (Cullen Jennings); http://www.ietf.org/ mail-archive/web/rtcweb/current/msg00323.html 8. Remote assistance (ala VNC or RDP) - User is helping another user on their computer with either view-only or view-with- control, either of just the browser of the the entire screen. ht tp://www.ietf.org/mail-archive/web/rtcweb/current/msg00543.html 9. Security camera/baby monitor usage http://www.ietf.org/ mail-archive/web/rtcweb/current/msg00543.html 10. Large multiparty session http://www.ietf.org/mail-archive/web/ rtcweb/current/msg00530.html 9. Acknowledgements Harald Alvestrand and Ted Hardie have provided comments and feedback on the draft. Harald Alvestrand and Cullen Jennings have provided additional use- cases. Thank You to everyone in the RTCWEB community that have provided comments, feedback and improvement proposals on the draft content. 10. Change Log [RFC EDITOR NOTE: Please remove this section when publishing] Changes from draft-ietf-rtcweb-use-cases-and-requirements-02 o Removed desrciption/list of API requirements, instead o Reference to W3C webrtc_reqs document for API requirements Changes from draft-ietf-rtcweb-ucreqs-01 o Changed Intended status to Information o Changed "Ipr" to "trust200902" o Added use case "Simple video communication service, NAT/FW that blocks UDP", and derived new req F26 o Added use case "Distributed Music Band" and derived new req A17 o Added F24 as requirement derived from use case "Simple video communication service with inter-operator calling" Holmberg, et al. Expires February 29, 2012 [Page 14]
Internet-Draft RTC-Web August 2011 o Added section "Additional use cases" o Added text about ID handling to multiparty with central server use case o Re-phrased A1 slightly Changes from draft-ietf-rtcweb-ucreqs-00 o - Reshuffled: Just two main groups of use cases (b2b and b2GW/ Server); removed some specific use cases and added them instead as flavors to the base use case (Simple video communciation) o - Changed the fromulation of F19 o - Removed the requirement on an API for DTMF o - Removed "FX3: There SHOULD be a mapping of the minimum needed data for setting up connections into SIP, so that the restriction to SIP-carriable data can be verified. Not a rew on the browser but rather on a document" o - (see http://www.ietf.org/mail-archive/web/rtcweb/current/msg00227.html for more details) o -Added text on informing user of that mic/cam is being used and that it must be possible to revoce permission to use them in section 7. Changes from draft-holmberg-rtcweb-ucreqs-01 o - Draft name changed to draft-ietf-rtcweb-ucreqs o - Use-case grouping introduced o - Additional use-cases added o - Additional reqs added (derived from use cases): F19-F25, A16-A17 Changes from draft-holmberg-rtcweb-ucreqs-00 o - Mapping between use-cases and requirements added (Harald Alvestrand, 090311) o - Additional security considerations text (Harald Alvestrand, 090311) o - Clarification that user applications are assumed to be executed by a browser (Ted Hardie, 080311) o - Editorial corrections and clarifications 11. References 11.1. Normative References [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. Holmberg, et al. Expires February 29, 2012 [Page 15]
Internet-Draft RTC-Web August 2011 11.2. Informative References [webrtc_reqs] "Webrt requirements, http://dev.w3.org/2011/webrtc/editor/webrtc_reqs.html". Authors' Addresses Christer Holmberg Ericsson Hirsalantie 11 Jorvas 02420 Finland Email: christer.holmberg@ericsson.com Stefan Hakansson Ericsson Laboratoriegrand 11 Lulea 97128 Sweden Email: stefan.lk.hakansson@ericsson.com Goran AP Eriksson Ericsson Farogatan 6 Stockholm 16480 Sweden Email: goran.ap.eriksson@ericsson.com Holmberg, et al. Expires February 29, 2012 [Page 16]