- RTC-Web - alvestrand.no

НА: [dispatch] RTC-Web I-D about interworking between RTC-Web and SIP-RTP
by Slava Borilin 09 Feb '11

09 Feb '11

Also work or just work? Слава ----- Reply message ----- От: "Xavier Marjou" <xavier.marjou(a)gmail.com> Дата: ср, фев 9, 2011 22:26 Тема: [dispatch] [RTW] RTC-Web I-D about interworking between RTC-Web and SIP-RTP Кому: "Henry Sinnreich" <henry.sinnreich(a)gmail.com> Копия: "rtc-web(a)alvestrand.no" <rtc-web(a)alvestrand.no>, "DISPATCH list" <dispatch(a)ietf.org> Hi, Hi Henry, These requirements certainly do not want to cripple innovation (RTC-Web). They rather want to catch attention so that RTC-Web is designed in a way it can also work with existing SIP-RTP devices. Cheers, Xavier On Wed, Feb 9, 2011 at 6:36 PM, Henry Sinnreich <henry.sinnreich(a)gmail.com<mailto:henry.sinnreich@gmail.com>> wrote: GENERIC-REQ-1 The [RTC-Web] solution MUST be designed in a such way it allows interworking with SIP-RTP applications both at the signalling and media level. Please help me understand how this and the other requirements do not cripple the whole concept of the RTC-Web (innovation based on simplicity and flexibility) by making it just another extension of legacy telecom design? Or did the authors mean requirements for gateways only? Thanks, Henry On 2/9/11 3:06 AM, "Xavier Marjou" <xavier.marjou(a)orange-ftgroup.com<http://xavier.marjou@orange-ftgroup.com>> wrote: Hi, We have posted a draft about interworking requirements between RTC-Web and SIP-RTP. http://www.ietf.org/internet-drafts/draft-marjou-dispatch-rtcweb-sip-rtp-in… Cheers, Xavier and Jean-François ________________________________ _______________________________________________ dispatch mailing list dispatch(a)ietf.org<http://dispatch@ietf.org> https://www.ietf.org/mailman/listinfo/dispatch _______________________________________________ RTC-Web mailing list RTC-Web(a)alvestrand.no<mailto:RTC-Web@alvestrand.no> http://www.alvestrand.no/mailman/listinfo/rtc-web

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Agenda proposal, RTCWEB BOF in Prague
by Harald Alvestrand 04 Feb '11

04 Feb '11

Ted Hardie and I put this together. Please comment! Name: RTCWEB - Real Time Communication on the World Wide Web BOF Chairs: Harald Alvestrand, Ted Hardie Expected # of attendees: 150 Length of session: 2.5 hours Conflicts to avoid: RAI area, DISPATCH, CODEC, HYBI Webex NOT required. Description: A proposal for a new working group in the RAI area is presently before the RAI area chairs. The charter proposal is here: <URL coming> This BOF will give an overview of the problem space and possible action items, discuss the technology that is being proposed, and describe the relationship with other work efforts inside and outside the IETF. Relevant mailing list: rtcweb(a)alvestrand.no <mailto:rtcweb@alvestrand.no>- www.alvestrand.no/mailman/listinfo/rtcweb <http://www.alvestrand.no/mailman/listinfo/rtcweb>to subscribe. Drafts: draft-alvestrand-dispatch-rtcweb-datagram draft-alvestrand-dispatch-rtcweb-protocols <this list will grow> Agenda: 00:00 Agenda bashing, blue sheet, scribe selection, note well 00:05 Brief introduction of the problem space and scope limitations 00:10 Overview of proposed action items for the WG, should it be chartered (also give forward reference to W3C coordination description to come). 00:20 Technology Discussion 00:20 Overall model of RTW in the Web - draft: rtcweb-protocols 00:30 Requirements/scenarios work - Describe how to gather, document and triage requirements 00:55 Profiling STUN for RTCWeb - draft: rtcweb-datagrams 01:10 Negotiation mechanisms 01:25 Criteria for work done in the browser Keywords: Security models, DDoS, execution speed, resource access 01:40 Scope and methods of W3C API work - see posted W3C charter proposal 01:50 Criteria for mandatory to implement codecs, syntax <name?> 02:05 Call for volunteers for specific work items 02:20 Charter hum

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Fwd: [dispatch] RTCWEB BOF was (Re: Summary: topics put forth for IETF-80)
by Harald Alvestrand 04 Feb '11

04 Feb '11

We have our approval for a BOF in Prague. -------- Original Message -------- Subject: [dispatch] RTCWEB BOF was (Re: Summary: topics put forth for IETF-80) Date: Wed, 2 Feb 2011 10:24:48 +0200 From: Gonzalo Camarillo <Gonzalo.Camarillo(a)ericsson.com> To: Mary Barnes <mary.ietf.barnes(a)gmail.com> CC: DISPATCH <dispatch(a)ietf.org>, "rai-ads(a)tools.ietf.org" <rai-ads(a)tools.ietf.org> Folks, > 1) RTCWEB: > http://www.ietf.org/mail-archive/web/dispatch/current/msg03171.html > > Excellent feedback and discussion thus far. An official Bof has been > approved for this topic: > http://trac.tools.ietf.org/bof/trac/#RAI Yes, we will have an RTCWEB BOF in Prague. The goal is to try and form a new WG. As you all know, the process of chartering a WG is a scoping process aimed to define what is in and what is out of the scope of the WG. Given that RTCWEB is a large area, there may be topics that come up during the discussions that do not end up in the charter of the WG for some reason. If that was the case, we would decide what to do with them in DISPATCH afterward. Right now, our focus should be to work on a good charter proposal. Cheers, Gonzalo _______________________________________________ dispatch mailing list dispatch(a)ietf.org https://www.ietf.org/mailman/listinfo/dispatch

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Re: [RTW] [dispatch] Does RTC-WEB need to pick a signaling protocol?
by Adam Roach 01 Feb '11

01 Feb '11

On 1/29/11 8:35 AM, Jonathan Rosenberg wrote: > 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 I have a number of thoughts on this topic also, but I don't think this (DISPATCH) is the forum for them. I'd like for these conversations to take place, though, so I want it to be clear in the charter that we aren't precluded from choosing one path over the other. Is it fair to characterize your statement that "this is a topic that should be resolved within the group itself" to support leaving the charter open enough to support any consensus conclusion that discussion may reach? /a

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The charter formerly know as RTC-WEB take 3
by Cullen Jennings 31 Jan '11

31 Jan '11

In my dispatch co-chair role, I tried to take all the comments I had seen on the list about this charter and see if I could address them in a new version of the charter. I probably messed up in some places. There were some conversation that did not seem to be converging so I did not make any changes for theses. Have a read and if you think something needs to be changed, propose text changes along with the reasons why and we will keep the evolving this charter. Thanks Cullen ---------------------------------------------------------------------------------- Version: 3 Possible Names: RTCWEB WEBRTC STORM: Standardized Transport Oriented for Realtime Media BURN: Browsers Using Realtime Media WAVE: Web And Voice/Video Enablement WAVVE: Web And Voice Video Enablement REALTIME WEBCOMM WREALTIME WEBTIME WEBFLOWS BRAVO Browser Realtime Audio and VideO COBWEB COmmuication Between WEBclients WHEELTIME 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 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. 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 that 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. 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. 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 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 W3C defied API to IETF protocols to IESG as Informational. This depends on the W3C API work.

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Re: [RTW] [dispatch] Does RTC-WEB need to pick a signaling protocol?
by Henry Sinnreich 30 Jan '11

30 Jan '11

If XMPP evangelists can commit not make any more pitches for XMPP, maybe the evangelists for SIP (count me in) could reciprocate :-) So, suppose we move on, especially, since application level scripts for both basic SIP and basic XMPP can be written and published as OS by those who really want to. And here we are getting to the key point IMO that both Jonathan and I have mentioned: Leave the field open to innovation. While SIP has been focused on telephony, XMPP has been focused on chat rooms and both, while successful have missed the major innovations on the Web that have happened since, mostly without any new standards: * REST * P2P RT A/V/IM * Web mail * Wikis and blogs * Social networks * Web conferencing * Web office apps and enterprise apps * Cloud services * Streaming HTTP and CDNs, etc, etc. Note that except for P2P RT A/V/IM, mostly basic/core HTTP and HTML standards were used. >From this perspective, it is enough to let SIP and XMPP folks publish their application level scripts and support some mechanism for endpoints to use either or both if so desired. The key however is not to shackle innovators with either SIP or XMPP or both (the worst outcome) as mandatory. Actually, as mentioned here, is not the W3C supposed to specify the API for RT-Web? Thanks, Henry

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Re: [RTW] Does RTC-WEB need to pick a signaling protocol?
by Bernard Aboba 30 Jan '11

30 Jan '11

Just to provide some context for those who may have only recently been exposed to the realtime web. The realtime web is not new -- it has been in operation for more than a decade, as embodied in commercial services such as Adobe Connect, WebEx, Live Meeting, etc. which serve millions of users on a daily basis. The realtime web is supported on virtually every major browser and web server platform. Javascript libraries for signalling are not a "future development" -- they have been around for a while. All that is really new here is the ability to engineer these services without plugins. That's important because it will make realtime web services considerably easier to develop and maintain. However, the "major event" occurred during the early 2000s. _____ Date: Sun, 30 Jan 2011 00:02:59 -0800 From: erik(a)hookflash.com To: jdrosen(a)jdrosen.net CC: rtc-web(a)alvestrand.no; dispatch(a)ietf.org Subject: Re: [RTW] Does RTC-WEB need to pick a signaling protocol? I understand the argument and agree that pure http might be the right course at some point in the future but the fact remains, SIP rules in communications "today", it will do so for years to come. It might be wise for us to move forward on a standard that is understood and appreciated by not only the developer community but also the business community paying the bills. Yes, we can build anew atop http, all it takes is time and money. An http effort will take a while, I think we all know that. How long did it take for SIP to displace H.323? I think the unanimous answer is "too long". We have a standard that exists today, one that we are all very familiar with, one that works! It would seem a shame not to leverage all we (and our employers) have invested, which is considerable. This is a major event in communications, we need to consider this carefully but we also owe it to "standards" to do it quickly, which has not been the norm it seems. Erik Lagerway | hookflash | m. 604.562.8647 On Sat, Jan 29, 2011 at 6:35 AM, Jonathan Rosenberg <jdrosen(a)jdrosen.net> wrote: 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 _______________________________________________ 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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НА: [dispatch] Does RTC-WEB need to pick a signaling protocol?
by Slava Borilin 29 Jan '11

29 Jan '11

+1 Slava borilin, Spirit ----- Reply message ----- От: "Jonathan Rosenberg" <jdrosen(a)jdrosen.net> Дата: сб, янв 29, 2011 17:36 Тема: [dispatch] Does RTC-WEB need to pick a signaling protocol? Кому: "'DISPATCH list'" <dispatch(a)ietf.org>, "rtc-web(a)alvestrand.no" <rtc-web(a)alvestrand.no> 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 _______________________________________________ dispatch mailing list dispatch(a)ietf.org https://www.ietf.org/mailman/listinfo/dispatch

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НА: [dispatch] Does RTC-WEB need to pick a signaling protocol?
by Slava Borilin 29 Jan '11

29 Jan '11

+1 Slava borilin, Spirit ----- Reply message ----- От: "Jonathan Rosenberg" <jdrosen(a)jdrosen.net> Дата: сб, янв 29, 2011 17:36 Тема: [dispatch] Does RTC-WEB need to pick a signaling protocol? Кому: "'DISPATCH list'" <dispatch(a)ietf.org>, "rtc-web(a)alvestrand.no" <rtc-web(a)alvestrand.no> 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 _______________________________________________ dispatch mailing list dispatch(a)ietf.org https://www.ietf.org/mailman/listinfo/dispatch

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Use SIP and let's get on with it already!
by Erik Lagerway 28 Jan '11

28 Jan '11

Come on guys, we all know SIP is bit busted but isn't it also the de-facto standard? Yes, we can debate the pros and cons of this approach until the cows come home but it would be really nice if we could move forward with just one protocol (for now) that will enable immediate usage across the vast majority of all IP networks. A protocol that everyone already knows. 'X'oIP is finally getting close to crossing the chasm, why would we drop in Jingle (or a completely new protocol) and prevent mass adoption sooner rather than later, or worse create yet another silo? Also, proposing only part of the solution will get us nowhere. Jingle might be more monolithic but it's also not the standard. Carriers, providers, vendors & users need this in 2011! Btw, I agree with Adam that this is best handled in the WG but I simply could not resist :) <bracing for impact> -Erik

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