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Re: Cisco's position on the WebRTC API

From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Wed, 24 Jul 2013 19:16:45 +1000
Message-ID: <CAHp8n2nBtxROFjDOXup2PX2dH6DY0Y7JOs_8J5yXWLeqCZsYMQ@mail.gmail.com>
To: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
Cc: Eric Rescorla <ekr@rtfm.com>, cowwoc <cowwoc@bbs.darktech.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
On Wed, Jul 24, 2013 at 5:45 PM, Stefan Håkansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
> On 7/24/13 3:49 AM, Silvia Pfeiffer wrote:
>> On Wed, Jul 24, 2013 at 11:14 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>>> On Tue, Jul 23, 2013 at 5:03 PM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>>>> On 23/07/2013 7:45 PM, Cullen Jennings (fluffy) wrote:
>> Do take my suggestion for a .mute() function on MediaStream as such a
>> concrete proposal.
>> Another concrete proposal is to add a bandwidth limitation to the
>> constraints handling.
>> And a third is to introduce receive-only peer connections instead of
>> having to do:
>> desc.sdp = desc.sdp.replace(/a=sendrecv/g, "a=recvonly");
>> Maybe this part can go into the RTCConfiguration?
> Note that we sent out a separate call for input like this:
> http://lists.w3.org/Archives/Public/public-webrtc/2013Jul/0176.html

Excellent - were the replies captured somewhere? If mine aren't there,
please regard my input as a reply to those. I'm only just learning
what features require me to mangle SDP.

> Regarding the "mute" (stop sending temporarily) and bandwidth parts I
> did an input:
> http://lists.w3.org/Archives/Public/public-webrtc/2013Jul/0260.html

Sorry if I was just repeating things. I'm sure if I had seen
consequences of those submissions, I wouldn't have needed to repeat
it. So, hopefully we can make these happen this time around.

Received on Wednesday, 24 July 2013 09:17:32 UTC

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