Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt

Hi,

In BUNDLE we use "RTP-based media" terminology, eventhough BUNDLE will be used in WebRTC environments where SRTP is mandated.

Why can't we use the same here?

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
Sent: 10. huhtikuuta 2014 14:55
To: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt

On 04/09/2014 04:39 PM, Colin Perkins wrote:
> On 9 Apr 2014, at 15:33, Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>> On 9 Apr 2014, at 15:20, Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>>>>> This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>>>>>>
>>>>>>     Title           : WebRTC Data Channels
>>>>>>     Authors         : Randell Jesup
>>>>>>                       Salvatore Loreto
>>>>>>                       Michael Tuexen
>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>> 	Pages           : 15
>>>>>> 	Date            : 2014-04-09
>>>>>>
>>>>>> Abstract:
>>>>>> The Real-Time Communication in WEB-browsers working group is 
>>>>>> charged to provide protocol support for direct interactive rich 
>>>>>> communication using audio, video, and data between two peers' 
>>>>>> web-browsers.  This document specifies the non-(S)RTP media data 
>>>>>> transport aspects of the WebRTC framework.  It provides an 
>>>>>> architectural overview of how the Stream Control Transmission 
>>>>>> Protocol (SCTP) is used in the WebRTC context as a generic 
>>>>>> transport service allowing WEB-browsers to exchange generic data from peer to peer.
>>>>> This talks about "(S)RTP" throughout, but the rtp-usage draft requires that SRTP be used for WebRTC, and disallows plain RTP. I think this draft could be simplified by changing "(S)RTP" to "SRTP" throughout.
>>>> Hi Colin,
>>>>
>>>> The (S)RTP notion goes back to a comment from Magnus. If I remember it correctly he considers SRTP a profile of RTP. Since I don't wanted to just use RTP, I ended up with (S)RTP based on a discussion with Magnus.
>>>>
>>>> However, I'm fine with changing it to SRTP...
>>> SRTP is an RTP profile. My comment was that if this is for WebRTC only, then  only SRTP can be used, and not plain RTP. Using "(S)RTP" rather than "SRTP" in this draft suggests that the secure profile is optional, which isn't the case in WebRTC. If this is for more general use than WebRTC, then "(S)RTP" is fine.
>> It is clear that in WebRTC only SRTP is used...
> I don't believe your draft is clear on that, due to the use of "(S)RTP" terminology. That's why I commented.
>
Enforcing the SRTP-only rule for WebRTC is not the business of this draft, so it's not a normative statement in any way, shape or form.

If it's possible to use these channels outside of WebRTC, and some of these places claim to be able to use pure RTP with appropriate security, then making use of "SRTP" only in this draft is confusing.

I see arguments both ways, and want to leave it at editors' discretion.

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