Re: [rtcweb] Low Level Javascript API Proposal avail on the webrtc list

Few questions that I have in regard to this:

1. How can you get local codec capabilities to implement something like SIP
OPTIONS?

2. Can you specify multiple stun/turn/turns servers? If you do what is the
candidate generation process? On the same note why do we provide both stun
and turn relays? Isn't turn server is normally stun server as well?

3. How do you process multiple answers for the same dialog?

4. What do we do with multiple answers from a forked dialog? Is it even
supported?

5. What is the process for synchronizing audio and video streams?
_____________
Roman Shpount

On Wed, Sep 28, 2011 at 11:59 AM, Tim Panton <tim@phonefromhere.com> wrote:

> I just want to ask folks to take a look at a proposal Neil Stratford wrote
> for the
> Low Level Javascript API and sent to the webrtc list.
>
> It  is based on his and my experience of web based audio plugins and on a
> proposal of Cullen's a while back.
>
> http://lists.w3.org/Archives/Public/public-webrtc/2011Sep/0106.html
>
> We'd appreciate any feedback either here or on the webrtc list as
> appropriate.
>
> Tim.
>
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