- RTC-Web - alvestrand.no

Requirement F9 - Audio processing
by Kavan Seggie 15 Mar '11

15 Mar '11

Hi all Firstly congratulations on a great initiative. Video and voice chat in the web browser is incredibly powerful and will without doubt change the communications industry. A quick introduction. We are a small startup that over a two year period have created a video chat plugin. Due to our limited resources we have used open source technologies where ever possible. The one area that is very difficult to find quality open source libraries is in audio processing. As you all probably know, the overall communications experience of a video chat is greatly dependent on audio quality. The audio codec is important, but just as important are the other DSP elements like the jitter buffer, AGC, noise suppression and AEC. Without this audio processing even the best video and audio can be useless. This is what happened with the Flash plugin and AEC became the most/2nd most requested feature in Flash. They have finally implemented AEC, AGC and noise suppression in 10.3 beta and it is now usable as a communications tool. Before, without the audio processing, it wasn't in my opinion. To the best of my knowledge the only reasonable open sourced software in this field is Jean-Marc Valin's Speex DSP library. We have been working on optimising this for a few months and will continue to do so. But it is a basic implementation and to be honest our optimisations are not great yet. The infinite permutations of hardware, differing latency in the network and the system, and the different environments the user can be in, make this an incredibly difficult problem to solve. Also if we want to support different audio codecs, these libraries need to be optimised for each codec. I have read in the proposed documents that AEC is a requirement (F9) but I have not been able to find any plans on addressing the issue. After Google bought GIPS, I thought that Google may open source GIPS's rock solid audio technology. However that was almost a year ago so there may be issues with this. It is also just a guess, Google may have other plans for the technology. So perhaps, since this is an important requirement for the adoption of RTC, the group could work on bettering the Speex algorithms? We would be happy to contribute our amendments to the Speex algorithms as well as other work we have done, and so kick start a project. We could also commit our audio engineer to work on it almost full time. He has considerable experience in creating voice codecs and is learning his way through the other DSP elements. All thoughts and recommendations welcome. Regards Kavan Seggie SayMama

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Re: [RTW] [dispatch] Does RTC-WEB need to pick a signaling protocol?
by Dean Willis 15 Mar '11

15 Mar '11

Well, THAT was an interesting time-warp. Sorry about that. But I guess the sentiment still stands even if the mail was stuck in my outbox for a while... On Mar 14, 2011, at 11:37 PM, Dean Willis wrote: > > Oh hell no.... > > Really, what we need are the primitives that make a web app able to do things like slurp data from a microphone and send it someplace useful. We might also need interfaces to A/V streams (maybe even RTP), and mixers. We need those primitives exposed to things like Javascript. What we don't need is an API that turns every browser into a generic SIP UA capable of being reused in haphazard fashion, such that everybody and his chihuahua reskins their "SIP browser phone" with CSS, and every browser swells to the size of the Hindenberg just before the fatal spark. > > -- > Dean >

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Does RTC-WEB need to pick a signaling protocol?
by Jonathan Rosenberg 15 Mar '11

15 Mar '11

I'm starting a separate thread on this, since I don't want to confound it with the charter discussion. This is a topic that should be resolved within the group itself, and here are my thoughts on it. If one asks the question on whether it is actually NECESSARY to require that a browser implement something like SIP in order to enable voip natively, the answer is definitively NO. The browser already provides a tool for exchanging messaging of arbitrary content between the browser and a server - its called HTTP (and websockets). Through client-side Javascript that comes from the server, an application can craft arbitrary protocol messaging of its own design between the client and the server. As an obvious example, in order to read mail on Gmail, the browser doesn't need to have an implementation of IMAP or POP; Gmail's Javascript implements the client side of a protocol of Google's design, and it talks to a web server which implements the server side of that protocol. The protocol is then then carried over HTTP. As such, if we take our charter here to define only what is truly REQUIRED of a browser, in order to enable voip without a plugin, then we do NOT need to pick a signaling protocol. All we need are the things which are truly impossible or grossly unsuitable for HTTP, and that is the real-time media path only. There need only be APIs for pushing in, and extracting out, the data that must be exchanged through HTTP-based signaling - and those are things like IP addresses and codec selections. That said, even if one asks the question of whether it is a good idea for us to pick something, I think the answer is no. The enormous benefit of the web model is its ability for innovation and velocity. Standardization is not needed for communications within the domain of the provider; new features can be developed and deployed as quickly as they can be conceived. This is something which, despite our best efforts here at IETF over the years, we have failed to achieve. I think it is critical that we allow web-based voip to innovate with the same kind of pace we've seen in the web overall. One of the arguments made on the list about why we should pick something, is that building their own signaling protocols and messaging is "hard" for a tiny web developer that just wants to add a bit of voice to their app. In such a case, I fully expect that within weeks or months of specification and implementation of RTC-WEB stuff in browsers, smart people will develop Javascript libraries which do all of this "hard work", along with PHP and many other server-side libraries with sit on the other side. None of it requires standardization, and we can let the open source community and the marketplace innovate on whatever solutions are needed. Thanks, Jonathan R. -- Jonathan D. Rosenberg, Ph.D. SkypeID: jdrosen Chief Technology Strategist Mobile: +1 (732) 766-2496 Skype SkypeIn: +1 (408) 465-0361 jdrosen(a)skype.net http://www.skype.com jdrosen(a)jdrosen.net http://www.jdrosen.net

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Fwd: New Version Notification for draft-alvestrand-dispatch-rtcweb-protocols-01
by Harald Alvestrand 13 Mar '11

13 Mar '11

I don't think there should be much additional contoversial stuff here, but some of the language may have improved. Biggest additional text by line count: tried to add in the reasoning behind my argument in favour of a mandatory-to-implement baseline. This is still my input, not attempting to represent a consensus position (yet). Harald -------- Original Message -------- Subject: New Version Notification for draft-alvestrand-dispatch-rtcweb-protocols-01 Date: Sun, 13 Mar 2011 01:34:00 -0800 (PST) From: IETF I-D Submission Tool <idsubmission(a)ietf.org> To: harald(a)alvestrand.no A new version of I-D, draft-alvestrand-dispatch-rtcweb-protocols-01.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository. Filename: draft-alvestrand-dispatch-rtcweb-protocols Revision: 01 Title: Overview: Real Time Protocols for Brower-based Applications Creation_date: 2011-03-13 WG ID: Independent Submission Number_of_pages: 12 Abstract: This document gives an overview and context of a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web". It intends to serve as a starting and coordination point to make sure all the parts that are needed to achieve this goal are findable, and that the parts that belong in the Internet protocol suite are fully specified and on the right publication track. This work is an attempt to synthesize the input of many people, but makes no claims to fully represent the views of any of them. All parts of the document should be regarded as open for discussion, with the intended discussion forum being the "RTCWEB" WG (in formation). Currently, discussion is on the rtc-web(a)alvestrand.no mailing list. The IETF Secretariat.

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Fwd: I-D Action:draft-perkins-rtcweb-rtp-usage-00.txt
by Magnus Westerlund 13 Mar '11

13 Mar '11

Hi, Colin Perkins, Joerg Ott and I have created this draft about RTC-WEB's usage of RTP. Feedback and discussion are most appreciated. Cheers Magnus

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Fwd: [new-work] Proposed W3C Charter: Web Real-Time Communications Working Group (until 2011-03-18)
by Cullen Jennings 09 Mar '11

09 Mar '11

FYI Begin forwarded message: > From: Ian Jacobs <ij(a)w3.org> > Date: February 17, 2011 11:12:50 AM MST > To: new-work(a)ietf.org > Subject: [new-work] Proposed W3C Charter: Web Real-Time Communications Working Group (until 2011-03-18) > > Hello, > > Today W3C Advisory Committee Representatives received a Proposal to revise the Ubiquitous Web Applications Activity [0] (see the W3C Process Document description of Activity Proposals [1]). This proposal includes a draft charter for the Web Real-Time Communications Working Group: > http://www.w3.org/2010/12/webrtc-charter > > As part of ensuring that the community is aware of proposed work at W3C, this draft charter is public during the Advisory Committee review period. > > W3C invites public comments through 2011-03-18 on the proposed charter. Please send comments to public-new-work(a)w3.org, which has a public archive: > http://lists.w3.org/Archives/Public/public-new-work/ > > Other than comments sent in formal responses by W3C Advisory Committee Representatives, W3C cannot guarantee a response to comments. If you work for a W3C Member [2], please coordinate your comments with your Advisory Committee Representative. For example, you may wish to make public comments via this list and have your Advisory Committee Representative refer to it from his or her formal review comments. > > If you should have any questions or need further information, please contact François Daoust, Team Contact <fd(a)w3.org>. > > Thank you, > > Ian Jacobs, Head of W3C Communications > > [0] http://www.w3.org/2007/uwa/ > [1] > http://www.w3.org/2005/10/Process-20051014/activities#ActivityCreation > [2] http://www.w3.org/Consortium/Member/List > -- > Ian Jacobs (ij(a)w3.org) http://www.w3.org/People/Jacobs/ > Tel: +1 718 260 9447 > > _______________________________________________ > new-work mailing list > new-work(a)ietf.org > https://www.ietf.org/mailman/listinfo/new-work

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WG charter, take 4
by Harald Alvestrand 04 Mar '11

04 Mar '11

This is the current charter text - it's updated somewhat based on discussions on the list. In particular, there's now suggested language about the relationship to session management protocols. We'll make this an explicit discussion at the BOF, with the aim of getting one of three conclusions in the charter: - The WG will not address session management protocols - The WG will choose one or more session management protocols - The WG will discuss session management protocols, and do what makes sense. The last one is the one suggested by the current charter text; the other possible outcomes should be easier to write text for. Comments welcome! Harald (Note: The below is not going out in < 80 columns, I think. Please forgive the formatting issue.) ------------------------------------------------ Version: 4 Name: RTCWEB WG Chair(s): Cullen Jennings <fluffy(a)cisco.com> Body: Many implementations have been made that use a Web browser to support direct, interactive communications, including voice, video, collaboration, and gaming. In these implementations, the web server acts as the signaling path between these applications, using locally significant identifiers to set up the association. Up till now, such applications have typically required the installation of plugins or non-standard browser extensions. There is a desire to standardize this functionality, so that this type of application can be run in any compatible browser and allow for high-quality real-time communications experiences within the browser. Traditionally, the W3C has defined API and markup languages such as HTML that work in conjunction with with the IETF over the wire protocols such as HTTP to allow web browsers to display media that does not have real time interactive constraints with another human. The W3C and IETF plan to collaborate together in their traditional way to meet the evolving needs of browsers. Specifically the IETF will provide a set of on the wire protocols, including RTP, to meet the needs on interactive communications, and the W3C will define the API and markup to allow web application developers to control the on the wire protocols. This will allow application developers to write applications that run in a browser and facilitate interactive communications between users for voice and video communications, collaboration, and gaming. This working group will select and define a minimal set of protocols that will enable browsers to: have interactive real time voice and video pairwise between browsers or other devices using RTP have interactive real time application data for collaboration and gaming pairwise between browsers Fortunately very little development of new protocol at IETF is required for this, only selection of existing protocols and selection of minimum capabilities to ensure interoperability. The following protocols are candidates for including in the profile set: 1) RTP/ RTCP 2) a baseline audio codec for high quality interactive audio. Opus will be one of the codecs considered 3) a baseline audio codec for PSTN interoperability. G.711 and iLBC will be some of the codecs considered 4) a baseline video codec. H.264 and VP8 will be some of the codecs considered 5) Diffserv based QoS 6) NAT traversal using ICE 7) media based DTMF 8) support for identifying streams’ purpose using semantics labels mappable to the labels described in RFC 4574 9) Secure RTP and keying 10) support for IPv4, IPv6 and dual stack browsers Please note the above list is only a set of candidates that the WG may consider and is not list of things that will be in the profile the set. This group's primary aim is to enable web applications to manage real-time communication sessions with other web-context applications that implement this specification. When making choices, the working group will also consider the impact of those choices on interoperability with other devices that create or manage multimedia sessions, including the complexity imposed on any gateways which may be required. The Working group will identify information needed for session negotiation and management in web contexts, and its output documents will describe the relationship of this to information carried in session protocols used in other standardized contexts. The Working Group will consider the possibility of defining a browser component that implements an existing session negotiation and management protocol. The working group will cooperate closely with the W3C activity that specifies a semantic level API that allows the control and manipulation of all the functionality above. In addition, the API needs to communicate state information and events about what is happening in the browser to applications running in the browser. These events and state need to include information such as: receiving DTMF in the RTP, RTP and RTCP statistics, and the state of DTLS/SRTP handshakes. The output of this WG will form input to the W3C group that specifies the API, and if there are parts of the mapping between the API and the protocols that need to be done outside of the W3C, this group will do it. The working group will follow BCP 79, and adhere to the spirit of BCP 79. The working group cannot explicitly rule out the possibility of adopting encumbered technologies; however, the working group will try to avoid encumbered technologies that require royalties or other encumbrances that would prevent such technologies from being easy to use in web browsers. The following topics will be out of scope for the initial phase of the WG but could be added after a recharter: RTSP, RSVP, NSIS, LOST, Geolocation, IM & Presence, NSIS, Resource Priority. RTP Payload formats will not be done in this WG. Deliverables: An overview document describing the architecture A scenarios document describing scenarios that are expected to be supported A profile document specifying the protocols and options that must be supported by a conforming implementation (If needed) A mapping document describing the relationship between the protocols and the W3C-defined API. Milestones: May 2011 Main alternatives identified in drafts Aug 2011 WG draft with text reflecting agreement of what the profile set should be Sept 2011 Scenarios specification to IESG as Informational Nov 2011 Draft available of documentation specifying mapping of protocol functionality to W3C-specified API produced. This is an input to W3C API work. Dec 2011 Profile specification to IESG as PS Apr 2012 Mapping document submitted to the IESG as Informational (if needed)

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Re: [RTW] WG charter, take 4
by Elwell, John 25 Feb '11

25 Feb '11

+1 John > -----Original Message----- > From: rtc-web-bounces(a)alvestrand.no > [mailto:rtc-web-bounces@alvestrand.no] On Behalf Of > Rosenberg, Jonathan > Sent: 24 February 2011 16:33 > To: 'Harald Alvestrand'; RTC-Web(a)alvestrand.no > Subject: Re: [RTW] WG charter, take 4 > > A few comments: > > I'm happy with the charter language around discussion of session > management protocols being in scope for the WG. I don't > understand why, > during the BoF, we will discuss whether to bake a decision on > this into > the charter. Determining the protocols needed for browser RTC > - both the > types of protocols and specific protocol instances and > profiles - is the > work of the group and NOT the work of the charter. > > This piece of wording is also concerning: > >have interactive real time voice and video pairwise between > browsers or > >other devices using RTP > > While I think we agree this is a goal, there are real > security issues in > play here that the group needs to iron out. I do not want to bake in a > charter requirement that says we MUST deliver a solution that allows > point-to-point media (i.e., no gateways or SBCs or other > intermediaries) > between browsers and existing RTP endpoints. I'd suggest > something softer > like: > > "enable real-time voice and video communications between browsers and > non-browser endpoints, with direct media when possible based > on security > and interoperability considerations" > > Which states the intent but leaves the recommendation to the wg. > > Thanks, > Jonathan R. > > > Jonathan D. Rosenberg, Ph.D. SkypeID: jdrosen > Skype Chief Technology Strategist > jdrosen(a)skype.net http://www.skype.com > jdrosen(a)jdrosen.net http://www.jdrosen.net > > > -----Original Message----- > From: rtc-web-bounces(a)alvestrand.no > [mailto:rtc-web-bounces@alvestrand.no] > On Behalf Of Harald Alvestrand > Sent: Thursday, February 24, 2011 8:45 AM > To: RTC-Web(a)alvestrand.no > Subject: [RTW] WG charter, take 4 > > This is the current charter text - it's updated somewhat based on > discussions on the list. > > In particular, there's now suggested language about the > relationship to > session management protocols. > We'll make this an explicit discussion at the BOF, with the aim of > getting one of three conclusions in the charter: > > - The WG will not address session management protocols > - The WG will choose one or more session management protocols > - The WG will discuss session management protocols, and do what makes > sense. > > The last one is the one suggested by the current charter > text; the other > possible outcomes should be easier to write text for. > > Comments welcome! > > Harald > > (Note: The below is not going out in < 80 columns, I think. Please > forgive the formatting issue.) > > ------------------------------------------------ > > Version: 4 > > Name: RTCWEB > WG Chair(s): Cullen Jennings <fluffy(a)cisco.com> > > Body: > > Many implementations have been made that use a Web browser to support > direct, interactive communications, including voice, video, > collaboration, and gaming. In these implementations, the web > server acts > as the signaling path between these applications, using locally > significant identifiers to set up the association. Up till now, such > applications have typically required the installation of plugins or > non-standard browser extensions. There is a desire to standardize this > functionality, so that this type of application can be run in any > compatible browser and allow for high-quality real-time communications > experiences within the browser. > > Traditionally, the W3C has defined API and markup languages > such as HTML > that work in conjunction with with the IETF over the wire > protocols such > as HTTP to allow web browsers to display media that does not have real > time interactive constraints with another human. > > The W3C and IETF plan to collaborate together in their traditional way > to meet the evolving needs of browsers. Specifically the IETF will > provide a set of on the wire protocols, including RTP, to > meet the needs > on interactive communications, and the W3C will define the API and > markup to allow web application developers to control the on the wire > protocols. This will allow application developers to write > applications > that run in a browser and facilitate interactive > communications between > users for voice and video communications, collaboration, and gaming. > > This working group will select and define a minimal set of protocols > that will enable browsers to: > have interactive real time voice and video pairwise between > browsers or > other devices using RTP > have interactive real time application data for collaboration > and gaming > pairwise between browsers > > > Fortunately very little development of new protocol at IETF > is required > for this, only selection of existing protocols and selection > of minimum > capabilities to ensure interoperability. The following protocols are > candidates for including in the profile set: > > 1) RTP/ RTCP > > 2) a baseline audio codec for high quality interactive audio. > Opus will > be one of the codecs considered > > 3) a baseline audio codec for PSTN interoperability. G.711 > and iLBC will > be some of the codecs considered > > 4) a baseline video codec. H.264 and VP8 will be some of the codecs > considered > > 5) Diffserv based QoS > > 6) NAT traversal using ICE > > 7) media based DTMF > > 8) support for identifying streams' purpose using semantics labels > mappable to the labels described in RFC 4574 > > 9) Secure RTP and keying > > 10) support for IPv4, IPv6 and dual stack browsers > > Please note the above list is only a set of candidates that the WG may > consider and is not list of things that will be in the > profile the set. > > This group's primary aim is to enable web applications to manage > real-time communication sessions with other web-context applications > that implement this specification. When making choices, the working > group will also consider the impact of those choices on > interoperability > with other devices that create or manage multimedia sessions, > including > the complexity imposed > on any gateways which may be required. > > The Working group will identify information needed for session > negotiation and management in web contexts, and its output documents > will describe the relationship of this to information carried > in session > protocols used in other standardized contexts. > > The Working Group will consider the possibility of defining a browser > component that implements an existing session negotiation and > management > protocol. > > The working group will cooperate closely with the W3C activity that > specifies a semantic level API that allows the control and > manipulation > of all the functionality above. In addition, the API needs to > communicate state information and events about what is > happening in the > browser to applications running in the browser. These events and state > need to include information such as: receiving DTMF in the > RTP, RTP and > RTCP statistics, and the state of DTLS/SRTP handshakes. The output of > this WG will form input to the W3C group that specifies the > API, and if > there are parts of the mapping between the API and the protocols that > need to be done outside of the W3C, this group will do it. > > The working group will follow BCP 79, and adhere to the spirit of BCP > 79. The working group cannot explicitly rule out the possibility of > adopting encumbered technologies; however, the working group > will try to > avoid encumbered technologies that require royalties or other > encumbrances that would prevent such technologies from being > easy to use > in web browsers. > > The following topics will be out of scope for the initial phase of the > WG but could be added after a recharter: RTSP, RSVP, NSIS, LOST, > Geolocation, IM & Presence, NSIS, Resource Priority. RTP > Payload formats > will not be done in this WG. > > Deliverables: > An overview document describing the architecture > A scenarios document describing scenarios that are expected to be > supported > A profile document specifying the protocols and options that must be > supported by a conforming implementation > (If needed) A mapping document describing the relationship between the > protocols and the W3C-defined API. > > > Milestones: > > May 2011 Main alternatives identified in drafts > > Aug 2011 WG draft with text reflecting agreement of what the > profile set > should be > > Sept 2011 Scenarios specification to IESG as Informational > > Nov 2011 Draft available of documentation specifying mapping > of protocol > functionality to W3C-specified API produced. This is an input > to W3C API > work. > > Dec 2011 Profile specification to IESG as PS > > Apr 2012 Mapping document submitted to the IESG as Informational (if > needed) > > > _______________________________________________ > RTC-Web mailing list > RTC-Web(a)alvestrand.no > http://www.alvestrand.no/mailman/listinfo/rtc-web > _______________________________________________ > RTC-Web mailing list > RTC-Web(a)alvestrand.no > http://www.alvestrand.no/mailman/listinfo/rtc-web >

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Re: [RTW] WG charter, take 4
by Bernard Aboba 25 Feb '11

25 Feb '11

Agree with Jonathan on the security issues below. Personally, I'd like to see an analysis of the issues added to the charter -- along with some opportunity for the security community to review the document. Allowing P2P media is a non-trivial change to the security threat model of the browser. -----Original Message----- From: rtc-web-bounces(a)alvestrand.no [mailto:rtc-web-bounces@alvestrand.no] On Behalf Of Rosenberg, Jonathan Sent: Thursday, February 24, 2011 8:33 AM To: 'Harald Alvestrand'; RTC-Web(a)alvestrand.no Subject: Re: [RTW] WG charter, take 4 A few comments: I'm happy with the charter language around discussion of session management protocols being in scope for the WG. I don't understand why, during the BoF, we will discuss whether to bake a decision on this into the charter. Determining the protocols needed for browser RTC - both the types of protocols and specific protocol instances and profiles - is the work of the group and NOT the work of the charter. This piece of wording is also concerning: >have interactive real time voice and video pairwise between browsers or >other devices using RTP While I think we agree this is a goal, there are real security issues in play here that the group needs to iron out. I do not want to bake in a charter requirement that says we MUST deliver a solution that allows point-to-point media (i.e., no gateways or SBCs or other intermediaries) between browsers and existing RTP endpoints. I'd suggest something softer like: "enable real-time voice and video communications between browsers and non-browser endpoints, with direct media when possible based on security and interoperability considerations" Which states the intent but leaves the recommendation to the wg. Thanks, Jonathan R. Jonathan D. Rosenberg, Ph.D. SkypeID: jdrosen Skype Chief Technology Strategist jdrosen(a)skype.net http://www.skype.com jdrosen(a)jdrosen.net http://www.jdrosen.net -----Original Message----- From: rtc-web-bounces(a)alvestrand.no [mailto:rtc-web-bounces@alvestrand.no] On Behalf Of Harald Alvestrand Sent: Thursday, February 24, 2011 8:45 AM To: RTC-Web(a)alvestrand.no Subject: [RTW] WG charter, take 4 This is the current charter text - it's updated somewhat based on discussions on the list. In particular, there's now suggested language about the relationship to session management protocols. We'll make this an explicit discussion at the BOF, with the aim of getting one of three conclusions in the charter: - The WG will not address session management protocols - The WG will choose one or more session management protocols - The WG will discuss session management protocols, and do what makes sense. The last one is the one suggested by the current charter text; the other possible outcomes should be easier to write text for. Comments welcome! Harald (Note: The below is not going out in < 80 columns, I think. Please forgive the formatting issue.) ------------------------------------------------ Version: 4 Name: RTCWEB WG Chair(s): Cullen Jennings <fluffy(a)cisco.com> Body: Many implementations have been made that use a Web browser to support direct, interactive communications, including voice, video, collaboration, and gaming. In these implementations, the web server acts as the signaling path between these applications, using locally significant identifiers to set up the association. Up till now, such applications have typically required the installation of plugins or non-standard browser extensions. There is a desire to standardize this functionality, so that this type of application can be run in any compatible browser and allow for high-quality real-time communications experiences within the browser. Traditionally, the W3C has defined API and markup languages such as HTML that work in conjunction with with the IETF over the wire protocols such as HTTP to allow web browsers to display media that does not have real time interactive constraints with another human. The W3C and IETF plan to collaborate together in their traditional way to meet the evolving needs of browsers. Specifically the IETF will provide a set of on the wire protocols, including RTP, to meet the needs on interactive communications, and the W3C will define the API and markup to allow web application developers to control the on the wire protocols. This will allow application developers to write applications that run in a browser and facilitate interactive communications between users for voice and video communications, collaboration, and gaming. This working group will select and define a minimal set of protocols that will enable browsers to: have interactive real time voice and video pairwise between browsers or other devices using RTP have interactive real time application data for collaboration and gaming pairwise between browsers Fortunately very little development of new protocol at IETF is required for this, only selection of existing protocols and selection of minimum capabilities to ensure interoperability. The following protocols are candidates for including in the profile set: 1) RTP/ RTCP 2) a baseline audio codec for high quality interactive audio. Opus will be one of the codecs considered 3) a baseline audio codec for PSTN interoperability. G.711 and iLBC will be some of the codecs considered 4) a baseline video codec. H.264 and VP8 will be some of the codecs considered 5) Diffserv based QoS 6) NAT traversal using ICE 7) media based DTMF 8) support for identifying streams purpose using semantics labels mappable to the labels described in RFC 4574 9) Secure RTP and keying 10) support for IPv4, IPv6 and dual stack browsers Please note the above list is only a set of candidates that the WG may consider and is not list of things that will be in the profile the set. This group's primary aim is to enable web applications to manage real-time communication sessions with other web-context applications that implement this specification. When making choices, the working group will also consider the impact of those choices on interoperability with other devices that create or manage multimedia sessions, including the complexity imposed on any gateways which may be required. The Working group will identify information needed for session negotiation and management in web contexts, and its output documents will describe the relationship of this to information carried in session protocols used in other standardized contexts. The Working Group will consider the possibility of defining a browser component that implements an existing session negotiation and management protocol. The working group will cooperate closely with the W3C activity that specifies a semantic level API that allows the control and manipulation of all the functionality above. In addition, the API needs to communicate state information and events about what is happening in the browser to applications running in the browser. These events and state need to include information such as: receiving DTMF in the RTP, RTP and RTCP statistics, and the state of DTLS/SRTP handshakes. The output of this WG will form input to the W3C group that specifies the API, and if there are parts of the mapping between the API and the protocols that need to be done outside of the W3C, this group will do it. The working group will follow BCP 79, and adhere to the spirit of BCP 79. The working group cannot explicitly rule out the possibility of adopting encumbered technologies; however, the working group will try to avoid encumbered technologies that require royalties or other encumbrances that would prevent such technologies from being easy to use in web browsers. The following topics will be out of scope for the initial phase of the WG but could be added after a recharter: RTSP, RSVP, NSIS, LOST, Geolocation, IM & Presence, NSIS, Resource Priority. RTP Payload formats will not be done in this WG. Deliverables: An overview document describing the architecture A scenarios document describing scenarios that are expected to be supported A profile document specifying the protocols and options that must be supported by a conforming implementation (If needed) A mapping document describing the relationship between the protocols and the W3C-defined API. Milestones: May 2011 Main alternatives identified in drafts Aug 2011 WG draft with text reflecting agreement of what the profile set should be Sept 2011 Scenarios specification to IESG as Informational Nov 2011 Draft available of documentation specifying mapping of protocol functionality to W3C-specified API produced. This is an input to W3C API work. Dec 2011 Profile specification to IESG as PS Apr 2012 Mapping document submitted to the IESG as Informational (if needed) _______________________________________________ RTC-Web mailing list RTC-Web(a)alvestrand.no http://www.alvestrand.no/mailman/listinfo/rtc-web _______________________________________________ RTC-Web mailing list RTC-Web(a)alvestrand.no http://www.alvestrand.no/mailman/listinfo/rtc-web

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Re: [RTW] [dispatch] RTCWEB I-D with thoughts on the framework
by Stefan Håkansson LK 23 Feb '11

23 Feb '11

I agree to large parts of the document, and I think that the section on interoperability with existing VoIP gear brings up important aspects. However, I am not completely comfortable with a couple of things: * The application would be responsible for rate adaptation, but what is the incentive for the application developer to do something that is fair to other applications or users? I would prefer that this is part of the browser (and with the auto update feature most browsers use the update cycles could be short anyway) * Are you suggesting that the <video> (and consequently also the <audio>) tag should not be used? I don't think this is a good idea. Since they are accepted and used by the web community we should use them also for RTC-Web rather than trying to invent something new. --Stefan > -----Original Message----- > From: dispatch-bounces(a)ietf.org > [mailto:dispatch-bounces@ietf.org] On Behalf Of Jonathan Rosenberg > Sent: den 8 februari 2011 15:49 > To: 'DISPATCH list' > Subject: [dispatch] RTCWEB I-D with thoughts on the framework > > Some of my colleagues from Skype and I have put together a > draft which gives our thoughts on the scope of the protocols > and functionality for enabling browser RTC. I've just > submitted the draft, which you can find > here: > > http://www.ietf.org/id/draft-rosenberg-rtcweb-framework-00.txt > > Some of the key points: > > * keep it minimal > * APIs for controlling behaviors of the various media > components, with defaults for those that dont care > * no SIP or Jingle; leave that to proprietary over websockets/HTTP > * no ICE - just STUN. ICE can be a javascript library. > > Thanks, > Jonathan R. > -- > Jonathan D. Rosenberg, Ph.D. SkypeID: jdrosen > Skype Chief Technology Strategist > jdrosen(a)skype.net http://www.skype.com > jdrosen(a)jdrosen.net http://www.jdrosen.net > > > _______________________________________________ > dispatch mailing list > dispatch(a)ietf.org > https://www.ietf.org/mailman/listinfo/dispatch >

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