--- 1/draft-ietf-rtcweb-use-cases-and-requirements-12.txt 2014-02-06 03:14:36.445217070 -0800 +++ 2/draft-ietf-rtcweb-use-cases-and-requirements-13.txt 2014-02-06 03:14:36.493218254 -0800 @@ -1,107 +1,108 @@ RTCWEB Working Group C. Holmberg Internet-Draft S. Hakansson Intended status: Informational G. Eriksson -Expires: April 17, 2014 Ericsson - October 14, 2013 +Expires: August 10, 2014 Ericsson + February 6, 2014 Web Real-Time Communication Use-cases and Requirements - draft-ietf-rtcweb-use-cases-and-requirements-12.txt + draft-ietf-rtcweb-use-cases-and-requirements-13.txt Abstract This document describes web based real-time communication use-cases. - Requirements on the browser functionality are derived from use-cases. + Requirements on the browser functionality are derived from the use- + cases. 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 April 17, 2014. + This Internet-Draft will expire on August 10, 2014. Copyright Notice - Copyright (c) 2013 IETF Trust and the persons identified as the + Copyright (c) 2014 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. Table of Contents - 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 + 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Use-cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3.1. Introduction . . . . . . . . . . . . . . . . . . . . . . 3 3.2. Common requirements . . . . . . . . . . . . . . . . . . . 4 3.3. Browser-to-browser use-cases . . . . . . . . . . . . . . 4 3.3.1. Simple Video Communication Service . . . . . . . . . 4 - 3.3.2. Simple Video Communication Service, NAT/FW that + 3.3.2. Simple Video Communication Service, NAT/Firewall that blocks UDP . . . . . . . . . . . . . . . . . . . . . 5 - 3.3.3. Simple Video Communication Service, FW that only - allows http . . . . . . . . . . . . . . . . . . . . . 5 + 3.3.3. Simple Video Communication Service, Firewall that + only allows traffic via a HTTP Proxy . . . . . . . . 5 3.3.4. Simple Video Communication Service, global service provider . . . . . . . . . . . . . . . . . . . . . . 5 3.3.5. Simple Video Communication Service, enterprise aspects . . . . . . . . . . . . . . . . . . . . . . . 6 3.3.6. Simple Video Communication Service, access change . . 7 3.3.7. Simple Video Communication Service, QoS . . . . . . . 7 - 3.3.8. Simple Video Communication Service with sharing . . . 8 + 3.3.8. Simple Video Communication Service with screen + sharing . . . . . . . . . . . . . . . . . . . . . . . 8 3.3.9. Simple Video Communication Service with file exchange 8 3.3.10. Hockey Game Viewer . . . . . . . . . . . . . . . . . 8 3.3.11. Multiparty video communication . . . . . . . . . . . 9 3.3.12. Multiparty on-line game with voice communication . . 10 - 3.4. Browser - GW/Server use cases . . . . . . . . . . . . . . 11 + 3.4. Browser - GW/Server use cases . . . . . . . . . . . . . . 10 3.4.1. Telephony terminal . . . . . . . . . . . . . . . . . 11 3.4.2. Fedex Call . . . . . . . . . . . . . . . . . . . . . 11 3.4.3. Video conferencing system with central server . . . . 11 - 4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 13 - 4.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 13 + 4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 12 + 4.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 12 4.2. Browser requirements . . . . . . . . . . . . . . . . . . 13 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16 6. Security Considerations . . . . . . . . . . . . . . . . . . . 16 6.1. Introduction . . . . . . . . . . . . . . . . . . . . . . 16 6.2. Browser Considerations . . . . . . . . . . . . . . . . . 16 6.3. Web Application Considerations . . . . . . . . . . . . . 17 - 7. Additional use-cases . . . . . . . . . . . . . . . . . . . . 17 - 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 18 - 9. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . 19 - 10. Normative References . . . . . . . . . . . . . . . . . . . . 24 - Appendix A. API requirements . . . . . . . . . . . . . . . . . . 24 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 27 + 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 17 + 8. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . 17 + 9. Normative References . . . . . . . . . . . . . . . . . . . . 23 + Appendix A. API requirements . . . . . . . . . . . . . . . . . . 23 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 26 1. Introduction 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 - client is of another type (e.g. a telephone). + 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 described 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. @@ -133,34 +134,33 @@ o Clients can be on IPv6-only o Clients can be on dual-stack o Clients can be connected to networks with different throughput capabilities o Clients can be on variable-media-quality networks (wireless) o Clients can be on congested networks - o Clients can be on firewalled networks with no UDP allowed o Clients can be on networks with a NAT using any type of Mapping and Filtering behaviors (as described in RFC4787). 3.2. Common requirements - The requirements retrived from the "Simple Video Communication - Service" by default apply to all other use-cases, and are considred - common. For each individual use-case, only the additional - requirements are listed. The following requirements can be retrieved - from, and apply to, each of the documented use-cases. For each - individual use-case, only requirements that are not part of the + 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. The following requirements can + be derived from, and apply to, each of the documented use-cases. For + each individual use-case, only requirements that are not part of the common requirements are listed. 3.3. Browser-to-browser use-cases 3.3.1. Simple Video Communication Service 3.3.1.1. Description Two or more users have loaded a video communication web application into their browsers, provided by the same service provider, and @@ -182,176 +182,184 @@ 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 of different makes, - with different operating systems and browsers from different vendors. + 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. 3.3.1.2. Common Requirements F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F35, F36, F38, F39 A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A25, A26 -3.3.2. Simple Video Communication Service, NAT/FW that blocks UDP +3.3.2. Simple Video Communication Service, NAT/Firewall that blocks UDP 3.3.2.1. Description - 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 that blocks 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 NAT/Firewall that + blocks UDP traffic. 3.3.2.2. Additional Requirements F29 -3.3.3. Simple Video Communication Service, FW that only allows http +3.3.3. Simple Video Communication Service, Firewall that only allows + traffic via a HTTP Proxy 3.3.3.1. Description - 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 FW that only allows traffic via a HTTP Proxy. + 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. 3.3.3.2. Additional Requirements F37 3.3.4. Simple Video Communication Service, global service provider + 3.3.4.1. Description - This use-case is almost identical to the Simple Video Communication - Service use-case (Section 3.3.1). + 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 additional traversal mechanisms. + 3.3.4.2. Additional Requirements F31 A22 3.3.5. Simple Video Communication Service, enterprise aspects 3.3.5.1. Description 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 session + 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 straddle the boundary - between the internal network and the external. + 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 to have 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 + 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. 3.3.5.2. Additional Requirements F32 3.3.6. Simple Video Communication Service, access change 3.3.6.1. Description - 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: + 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 have 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. + 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. 3.3.6.2. Additional Requirements F26 3.3.7. Simple Video Communication Service, QoS 3.3.7.1. Description - 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: + 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. 3.3.7.2. Additional Requirements F24, F26 -3.3.8. Simple Video Communication Service with sharing +3.3.8. Simple Video Communication Service with screen sharing 3.3.8.1. Description - This use-case has the audio and video communication of the Simple - Video Communication Service use-case (Section 3.3.1). + 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. 3.3.8.2. Additional Requirements F30 A21 3.3.9. Simple Video Communication Service with file exchange 3.3.9.1. Description - This use-case has the audio and video communication of the Simple - Video Communication Service use-case (Section 3.3.1). + 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. 3.3.9.2. Additional Requirements F30, F33 A21, A24 @@ -384,40 +392,41 @@ 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. 3.3.10.2. Additional Requirements - F17, F34 + F17, F24 A17, A23 3.3.11. Multiparty video communication 3.3.11.1. Description - In this use-case is the Simple Video Communication Service use-case + 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 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. In addition - the levels in the audio signals are adjusted before mixing. + 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 @@ -448,26 +457,25 @@ 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. 3.3.12.2. Additional Requirements - F12, F13, F14, F15, F16, F18 + F12, F13, F14, F15, F16, F18, F23, F24 A13, A14, A15, A16, A17, A18, A23 3.4. Browser - GW/Server use cases - 3.4.1. Telephony terminal 3.4.1.1. 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 is shown in the same manner as when a mobile phone is used. @@ -477,21 +485,21 @@ 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. 3.4.1.2. Additional Requirements - F21 + F21, F27 3.4.2. Fedex Call 3.4.2.1. Description 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. @@ -494,27 +502,28 @@ 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. 3.4.2.2. Additional Requirements F21, F22 3.4.3. Video conferencing system with central server + 3.4.3.1. Description 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 send an audio stream (type in terms + 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 @@ -525,21 +534,21 @@ 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 F7, on encryption of media and on ability to traverse very - restrictive FWs. There exist several solutions that enable the + restrictive Firewalls. 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. 3.4.3.2. Additional Requirements @@ -636,79 +645,76 @@ based Interactive voice response (IVR) System ---------------------------------------------------------------- F23 The browser must be able to send short latency unreliable datagram traffic to a peer browser [RFC5405]. ---------------------------------------------------------------- F24 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 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 ---------------------------------------------------------------- - F26 It must be possible to move from one network - interface to another one + F26 The communication session must survive across a + change of the network interface used by the + session ---------------------------------------------------------------- F27 The browser must be able to initiate and accept a media session where the data needed for establishment can be carried in SIP. ---------------------------------------------------------------- F28 The browser must support a baseline audio and video codec ---------------------------------------------------------------- F29 The browser must be able to send streams and - data to a peer in the presence of NATs that - block UDP traffic. + data to a peer in the presence of NATs and + Firewalls that block UDP traffic. ---------------------------------------------------------------- - F30 The browser must be able to use the screen (or - a specific area of the screen) or what a certain - application displays on the screen to generate - streams. + F30 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. ---------------------------------------------------------------- F31 The browser must be able to use several STUN and TURN servers ---------------------------------------------------------------- - F32 There browser must support that STUN and TURN - servers to use are supplied by other entities - than via the web application (i.e. the network - provider). + F32 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). ---------------------------------------------------------------- F33 The browser must be able to send reliable data traffic to a peer browser. ---------------------------------------------------------------- - F34 The browser must support prioritization of - streams and data. - ---------------------------------------------------------------- F35 The browser must enable verification, given the right circumstances and by use of other - trusted communication, of that streams and + trusted communication, that streams and data received have not been manipulated by any party. ---------------------------------------------------------------- F36 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]). ---------------------------------------------------------------- F37 The browser must be able to send streams and - data to a peer in the presence of FWs that only - allows traffic via a HTTP Proxy, when FW policy + data to a peer in the presence of Firewalls that only + allows traffic via a HTTP Proxy, when Firewall policy allows WebRTC traffic. + ---------------------------------------------------------------- F38 The browser must be able to collect statistics, related to the transport of audio and video between peers, needed to estimate quality of experience. ---------------------------------------------------------------- F39 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. ---------------------------------------------------------------- @@ -766,98 +772,48 @@ finished. The browser needs to ensure that the stream negotiation procedures are not seen as Denial Of Service (DOS) by other entities. 6.3. Web Application Considerations The web application is expected to ensure user consent in sending and receiving media streams. -7. Additional use-cases - - Several additional use-cases have been discussed. 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. E911 (Paul Beaumont) http://www.ietf.org/mail-archive/web/rtcweb - /current/msg00525.html, followed up by Stephan Wenger - - 3. Local Recording and Remote recording (John): Discussed a _lot_ - on the mail lists (rtcweb as well as public-webrtc) August and - September 2011. Concrete proposal: http://www.ietf.org/mail- - archive/web/rtcweb/current/msg01006.html (remote) and http:// - www.ietf.org/mail-archive/web/rtcweb/current/msg00734.html - (local) - - 4. Emergency access for disabled (Bernard Aboba) http:// - www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html - - 5. Clue use-cases (Roni Even) http://tools.ietf.org/html/draft- - ietf-clue-telepresence-use-cases-01 - - 6. Rohan red cross (Cullen Jennings); http://www.ietf.org/mail- - archive/web/rtcweb/current/msg00323.html - - 7. Security camera/baby monitor usage http://www.ietf.org/mail- - archive/web/rtcweb/current/msg00543.html - - 8. Large multiparty session http://www.ietf.org/mail-archive/web/ - rtcweb/current/msg00530.html - - 9. Call center http://www.ietf.org/mail-archive/web/rtcweb/current/ - msg04203.html - - 10. Enterprise policies http://www.ietf.org/mail-archive/web/rtcweb/ - current/msg04271.html - - 11. Low-complex multiparty central node http://www.ietf.org/mail- - archive/web/rtcweb/current/msg04430.html - - 12. Multiparty central node that is not allowed to decipher http:// - www.ietf.org/mail-archive/web/rtcweb/current/msg04457.html - -8. Acknowledgements +7. Acknowledgements 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. -9. Change Log +8. Change Log [RFC EDITOR NOTE: Please remove this section when publishing] Changes from draft-ietf-rtcweb-use-cases-and-requirements-10 o Described that the API requirements are really from a W3C perspective and are supplied as an appendix in the introduction. Moved API requirements to an Appendix. o Removed the "Conventions" section with the key-words and reference to RFC2119. Also changed uppercase MUST's/SHOULD's to lowercase. o Added a note on the proposed use of the document to the introduction. - o Removed the note talking about WS from the "FW that only allows - http" use-case. + o Removed the note talking about WS from the "Firewall that only + allows http" use-case. o Removed the word "Skype" that was used as example in one of the use-cases. o Clarified F3 (the req saying the everything the browser sends must be rate controlled). o Removed the TBD saying we need to define reasonable levels from the requirement saying that quality must be good even in presence of packet losses (F5), and changed "must" to "should" (Based on a @@ -902,22 +858,22 @@ o Changed "video communication session" to "audiovisual communication session. Changes from draft-ietf-rtcweb-use-cases-and-requirements-08 o Changed "eavesdropping" to "wiretapping" and referenced RFC2804. o Removed informal ref webrtc_req; that document has been abandoned by the W3C webrtc WG. - o Added use-case where one user is behind a FW that only allows - http; derived req. F37. + o Added use-case where one user is behind a Firewall that only + allows http; derived req. F37. o Changed F24 slightly; MUST-> SHOULD, inserted "available". o Added a clause to "Simple video communication service" saying that the service provider monitors the quality of service, and derived reqs F38 and A26. Changes from draft-ietf-rtcweb-use-cases-and-requirements-07 o Added "and data exchange" to 1. Introduction. @@ -1029,23 +984,22 @@ o Removed description/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 "Simple video communication service, NAT/Firewall + 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" o Added section "Additional use cases" o Added text about ID handling to multiparty with central server use case @@ -1090,26 +1043,26 @@ 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 -10. Normative References +9. Normative References [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. - [RFC2804] IAB IESG, "IETF Policy on Wiretapping", RFC 2804, May + [RFC2804] IAB and IESG, "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.