Re: [rtcweb] Draft agenda for IETF 87

Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com> Fri, 12 July 2013 10:34 UTC

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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
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Thread-Topic: [rtcweb] Draft agenda for IETF 87
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Date: Fri, 12 Jul 2013 10:34:08 +0000
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Draft agenda for IETF 87
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On 7/12/13 11:11 AM, Iñaki Baz Castillo wrote:
> 2013/7/12 Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>om>:
>> * JS APIs to:
>> ** Apply an SDP (e.g. received on the signaling channel) to the browser
>> ** Hand an SDP generated by the browser over to the application (for
>> transmission over the signaling wire presumably)
>> ** Influencing/modifying the contents of the SDP
>> * All belongs to the W3C WebRTC
>
> I've never seen an API in which the application requests some data to
> the core, and then mangles such a data before sending it to the peer.
> This is like if I ask the browser to send video, the browser gives me
> a media description (I will not assume SDP) with ptime=20, and I
> mangle the media description to set ptime=50, while at the same time I
> have no idea whether my browser can send RTP with ptime=50 or not.

I think this is not at all the way it should be done; if people think 
there is a need to define a specific ptime there should be an API that 
allows for that (and that has means of letting the app know what the 
allowed ranges are, or at least what is actually used). Let's go and do 
that in the WebRTC WG.

> It's just awesome.
>
>
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>
>