Re: [rtcweb] WebRTC service between SPs

I had assumed that a DNS NAPTR record could be used to provide all the
alternate "user id's". Am I right?

Cheers,
/Barry

Barry Dingle
Fixed - +61(0)3-9725-3937    Mob - +61(0)41-911-7578
Fellow of University of Melbourne, Electrical and Electronic Eng.,
Australia
> Linux + Android + Apple + Open Source software

On Sat, Jun 29, 2013 at 7:29 PM, Wangyahui (Yahui) <yahui.wang@huawei.com>wrote:

>  Thanks for your comments.****
>
> Yes, we can implement the federation between SPs through using the same
> signaling protocol (e.g. SIP) or deploying a gateway.****
>
> ** **
>
> But what I concern is how to compatible with the existing various user id
> from different SPs. For example, if Google provides WebRTC client, then the
> users should be able to login using their Gmail address. In the same way,
> Facebook support the users using FacebookId. So the format of
> identification may be number, string or email and so on.****
>
> ** **
>
> The problem is how to handle addressing users of different SPs. Should it
> be standardized to a unified WebRTC URI?****
>
> ** **
>
> Yahui****
>
> ** **
>
> *发件人:* Hrishikesh Kulkarni [mailto:rishi@turtleyogi.com]
> *发送时间:* 2013年6月29日 14:18
> *收件人:* Moises Silva
> *抄送:* Jim Barnett; Wangyahui (Yahui); rtcweb@ietf.org
> *主题:* Re: [rtcweb] WebRTC service between SPs****
>
> ** **
>
> SIP is an established standard to interoperate domains. We at
> OneKlikStreet.com developed a video/audio bridging service for WebRTC.
> Although it uses JS/JSON signaling for web based clients. Our server could
> very well federate with any other service using SIP. What does need to be
> discussed on app to app basis is what kind of federation you are looking
> for? ****
>
> In case of bridging service we could merge calls from both servers or
> redirect all the calls to the host service.****
>
> ** **
>
> regards,****
>
> Rishi****
>
> Founder, OneKlikStreet.com****
>
> ** **
>
> ** **
>
> ** **
>
> On Fri, Jun 28, 2013 at 11:55 PM, Moises Silva <moises.silva@gmail.com>
> wrote:****
>
> ** **
>
> On Fri, Jun 28, 2013 at 12:03 PM, Jim Barnett <Jim.Barnett@genesyslab.com>
> wrote:****
>
>  As I understand it, it’s not just a problem of identities.  WebRTC does
> not define the signaling protocol, but leaves it  up to the application.
> If two users download their applications/JavaScript from the same site, it
> won’t be a problem, because the same application is handling both ends of
> the call.  But if one user is on site A while the other is on site B, there
> is no guarantee that either site’s application will understand the
> signaling from the other.****
>
> ** **
>
> Unless websites agree to use something standard such as SIP/Jingle for
> federation (inter website/domain communication).
>
> -****
>
> Moy****
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb****
>
> ** **
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>