- 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