- From: cowwoc <cowwoc@bbs.darktech.org>
- Date: Tue, 02 Jul 2013 13:59:46 -0400
- To: Eric Rescorla <ekr@rtfm.com>
- CC: Adrian Georgescu <ag@ag-projects.com>, Jesús Leganés Combarro <piranna@gmail.com>, Robin Raymond <robin@hookflash.com>, Ted Hardie <ted.ietf@gmail.com>, Roman Shpount <roman@telurix.com>, Adam Bergkvist <adam.bergkvist@ericsson.com>, "public-webrtc@w3.org" <public-webrtc@w3.org>
- Message-ID: <51D31512.5010209@bbs.darktech.org>
Eric, Thanks for the clarification. Those are good points. Gili On 02/07/2013 10:38 AM, Eric Rescorla wrote: > > > > On Tue, Jul 2, 2013 at 7:14 AM, cowwoc <cowwoc@bbs.darktech.org > <mailto:cowwoc@bbs.darktech.org>> wrote: > > Adrian, > > Help us understand where this is coming from. I understand > that equipment vendors already need to parse SDP (which is > essentially just a bunch of key-value pairs) but clearly WebRTC > requires them to modify their parsing engine as we're introducing > new keys and values. So two points here: > > > > This seems incorrect in at least two respects: > > 1. The SDP mechanisms are enhancements motivated by WebRTC, so > they are designed to be backward compatible (though obviously older > peers won't get any benefit). So, equipment vendors aren't required > to modify their endpoints in order to get any interop and the new > feature adds can be done incrementally. > > 2. Most of the work (BUNDLE, trickle, etc.) is generally applicable > improvements, so the vendors are going to want to make these > improvements anyway, even for non-WebRTC systems. > > > Conversely, even if we were to discard SDP entirely at the interface to > WebRTC, we would still need to specify SDP representations of these > concepts in order to allow sensible interop of these features with > SDP-using > systems on the other end. For instance, if WebRTC implementations are > to use trickle ICE and get that when they interoperate with SIP systems, > then trickle ICE needs to be added to SDP. > > -Ekr > > > > > > > > 1. One option is to replace SDP with an equally-simple mechanism. > 2. Another option is to replace SDP inside WebRTC and ask > equipment vendors to use a gateway which will convert to SDP. > This kind of conversion would be very cheap (as compared to a > media gateway which is very expensive). > > If you could explain (or point to documents that do) the drive > behind using SDP then it would help us moving forward. We should > focus on *what* we're trying to accomplish, not *how*. > > Thanks, > Gili > > On 02/07/2013 9:06 AM, Adrian Georgescu wrote: >> SDP is here to stay for the benefits of network equipment >> vendors, end-users be damned. Which part of this you do not get? >> >> Stop complaining about SDP usage in WebRTC, it is a lost cause. >> >> Regards, >> Adrian >> >> >> On Jun 29, 2013, at 9:15 AM, piranna@gmail.com >> <mailto:piranna@gmail.com> wrote: >> >>> It has been said always that SDP was choosed so WebRTC would be >>> compatible with current VoIP devices and software, so you could >>> make calls from your browser without needing a proxy. Problem >>> is, that WebRTC has evolve a lot adding a lot of features >>> (panswer SDP message, new codecs...) that make them incompatible >>> with legacy VoIP systems so they should upgrade their firmware >>> or build a proxy anyway, so the main reason to use SDP has >>> dissapeared some time ago. >>> >>> El 29/06/2013 07:19, "Robin Raymond" <robin@hookflash.com >>> <mailto:robin@hookflash.com>> escribió: >>> >>> >>> Thanks - and in some ways that's absolutely true about >>> security requirements. If we don't have a document that >>> explains each choice along the way, things can be overlooked >>> if you try some new approach. >>> >>> Even though security is obviously really important, the >>> larger issue I'm finding is knowing what exactly must be >>> supported use case wise, especially in the context of being >>> compatible with legacy/existing systems, including all the >>> on-the-wire oddities that can happen, e.g. the mapping of >>> streams, tracks, ssrcs, dtls muxing, multitrack payload >>> based demuxing, CNAME correlation, video rules, incoming >>> unmapped RTP tracks/streams, CNAME/RTP correlation race, >>> SSRC collision, pre-mapping requirements, forking, etc. >>> >>> Sure makes it fun! We'll undoubtedly get some stuff wrong. >>> But hopefully the model is right so it's a matter of >>> tweaking the parameters and rules to follow. >>> >>> I can fully appreciate the reasons why the original design >>> went down the SDP path. I think most people knew SDP was >>> wonky. Having everything described in details in some master >>> blob made the wire easier to understand. But that blob makes >>> a horrible programmers interface with bad control and with >>> lots of short/long term problems/consequence. I think >>> there's a way to fix this API to make it semi-palatable for >>> JS developers, make it on-the-wire compatible/understandable >>> and achieve the requirement goals specified for WebRTC (even >>> if they are a bit vague in some areas still). >>> >>> We are giving it our best to fix this situation... >>> >>> -Robin >>> >>> >>>> cowwoc <mailto:cowwoc@bbs.darktech.org> >>>> 29 June, 2013 12:26 AM >>>> >>>> I wish you good luck (honestly). I did have one >>>> question, however: how do you know what the "established >>>> RTC principles and security considerations" are? Part of >>>> the problem, as I see it, is that we don't have a design >>>> document explaining the use-cases that were considered and >>>> why each design decision was made. We're missing the >>>> "design blueprint". I believe you're going to find it very >>>> hard to design an equivalent API without this information. >>>> >>>> Gili >>>> >>>> On 28/06/2013 8:17 PM, Robin Raymond wrote: >>>> >> > >
Received on Tuesday, 2 July 2013 18:00:22 UTC