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

Gunnar Hellstrom <gunnar.hellstrom@omnitor.se> Fri, 31 May 2013 22:59 UTC

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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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On 2013-05-31 22:31, Peter Saint-Andre wrote:
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> On 5/31/13 1:31 PM, Mark Rejhon wrote:
>> 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.
> Although I have not read the legislation that mandates use of RFC 4103
> (and I'm not particularly interested in the Layer 8 issues), I don't
> quite agree that one can't use SIP and XMPP in concert. For example,
> one could use SIP to negotiate use of XMPP with XEP-0301 as the RTT
> channel, similar to draft-sparks-simple-jabber-sessions from way back
> in 2002:
>
> http://tools.ietf.org/id/draft-sparks-simple-jabber-sessions-00.txt
>
> That might not be palatable to the politicians, but it might be an
> acceptable technical solution.
Well formulated requirements do not say that you must always USE RFC 
4103 for real-time text.
The requirements rather say that you must SUPPORT RFC 4103 in SIP call 
control environment, and you must provide real-time text functionality 
with good flow, sufficiently short delay and sufficient reliability.
To support means to be possible to activate through negotiation.

So there is room for USING other protocols.

Gunnar


>
> Peter
>
> - -- 
> Peter Saint-Andre
> https://stpeter.im/
>
>
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