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Re: Cisco's position on the WebRTC API

From: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
Date: Wed, 24 Jul 2013 07:45:02 +0000
To: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
CC: Eric Rescorla <ekr@rtfm.com>, cowwoc <cowwoc@bbs.darktech.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Message-ID: <1447FA0C20ED5147A1AA0EF02890A64B1C335DBB@ESESSMB209.ericsson.se>
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

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

Br,
Stefan

>
>
> Thanks for listening,
> Silvia.
>
>
Received on Wednesday, 24 July 2013 07:45:27 UTC

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