Re: Cisco's position on the WebRTC API

On 7/24/13 11:17 AM, Silvia Pfeiffer wrote:
> 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.

Only on the list so far, we (chair's) will need to assemble them somehow.

>
>
>> 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.

I'm glad you're repeating (I want the same features!), I just wanted to 
get more attention and input to that call.

>
> Thanks,
> Silvia.
>

Received on Wednesday, 24 July 2013 09:24:33 UTC