Re: [rtcweb] RTT Education: Neat Demonstration of NON-peer-to-peer RTT (for future webrtc standardization purposes)

Mark Rejhon <markybox@gmail.com> Fri, 31 May 2013 19:32 UTC

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From: Mark Rejhon <markybox@gmail.com>
Date: Fri, 31 May 2013 15:31:57 -0400
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To: Adam Roach <adam@nostrum.com>
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Subject: Re: [rtcweb] RTT Education: Neat Demonstration of NON-peer-to-peer RTT (for future webrtc standardization purposes)
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Actually, it's not quite that simple.

Some countries made RFC4103 part of their government legislation.
There are two open real-time text standards that is necessary to make
provisions for, to cover world-wide usage.

Everytime SIP is used for anything, it should always use RFC4013.
If WebRTC implementations touch SIP and not XMPP, and utilize real-time
text, then it needs to use RFC4103, even if it has to use the sockets
feature.  Thus, we need to make sure that it's also usable for RFC4103.
So it's not a 100% solved problem; there is still a SIP need.

Thanks,
Mark Rejhon

On Fri, May 31, 2013 at 3:22 PM, Adam Roach <adam@nostrum.com> wrote:

> Mark:
>
> This is all very neat. To be clear about what's being shown here:
>
>  * This does not use RFC 4103 RTT-over-RTP
>  * This text is transmitted over a traditional XMPP
>    client->server->server->client architecture
>
> Correct? If so -- and if this is being held out as a preferred technology
> for RTT -- then this is great news. As is shown by numerous live
> deployments (Gmail, Facebook, LiveJournal IM, etc), the model of [JS
> Client]->[Web Server]->[XMPP Server]->[XMPP Client] works quite well, and
> has been supported by existing web technologies since at least 2006.
>
> I'm glad to have a demonstration that this is a solved problem that we
> don't need to make special provisions for. Thanks.
>
> /a
>