Re: [rtcweb] Text communication in RTCWEB sessions -job overview

After a rapid browse, this is my view of what needs to be done to 
specify real-time text in RTCWEB and WebRTC.

1. draft-ietf-rtcweb-overview
      Make definition of Media to include real-time text
     Define the short form rtt
     Add rtt to descriptions in e.g. sections 2.1 and 2.2
     non-critical

2. draft-ietf-rtcweb-use-cases-and-requirements
     add a couple of rtt use cases
     add requirements for rtt
     add api requirements for rtt
     important, initiated

3. draft-nandakumar-rtcweb-sdp
     add codec description for rtt
     add sdp for rtt for the use cases
     important, initiated

4. draft-x-rtcweb-rtt
     rtt specifics for rtt codec and rtp usage
     ( take inspiration from draft-ietf-rtcweb-audio )
     new needed draft - not initiated

5. draft-ietf-rtcweb-jsep
     add rtt to API MediaHints  clause 6.1.1
     add rtt to examples in chapter 7.
    important

6. draft-ietf-rtcweb-rtpusage
     mention rtt
     add rtt to mixer discussion
     non-critical

7. draft-thomson-rtcweb-api-req
     mention rtt
     non-critical

8. draft-aboba-rtcweb-ecrit
     adjust rtt discussion for emergency calls to current status
     non critical

9. w3c.webrtc
     add rtt API, e.g. to Network Stream API
     important, urgent

10. w3c getusermedia
     add rtt to GetUserMedia API
     important, urgent

Any more?
Any less?

/Gunnar

On 2012-11-13 10:55, Harald Alvestrand wrote:
> On 11/13/2012 09:55 AM, Olle E. Johansson wrote:
>>
>> 12 nov 2012 kl. 13:15 skrev Harald Alvestrand <harald@alvestrand.no 
>> <mailto:harald@alvestrand.no>>:
>>
>>> I would prefer this to be added as a separate specification, rather 
>>> than done at this time.
>>> The reason being that this should be relatively easy to add on top 
>>> of the data channel functionality, but will definitely take some 
>>> time to specify, so should not be on the critical path for this 
>>> round of specifications.
>> T.140 is a codec in the  RTP flow and is implemented in Asterisk and 
>> a couple of Video SIP phones.
>> I see no reason to move it to the data channel, that would limit 
>> interoperability.
>>
>> Much easier - and faster - to implement in the RTP subsystem as it is 
>> already covered by SDP offer/answer.
>>
> Anything is possible, if someone is willing to do it.
>
> Olle and Gunnar, can you undertake to write a complete specification 
> for how to use T.140 with RTCWEB, including how it should fit in with 
> the API specification, and whether or not it supports the needs that 
> Gunnar has claimed for text services?
>
> I don't understand T.140 - so I can't do the work. Someone who wants 
> it should.
>
>
>
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