Re: [MMUSIC] Draft new: draft-holmberg-mmusic-t140-usage-data-channel

[Gunnar Hellström <[e-mail]>]

I hope I can stop introducing new topics soon, and contribute to 
resolving them instead... But another topic to cover is multi-party 
session support.  The requirement is:

    U-C 5:  Realtime text chat during an audio and/or video call with an
            individual or with multiple people in a conference.

I hope that will be straightforward.

/Gunnar



Den 2019-08-23 kl. 12:09, skrev Christer Holmberg:
> Hi,
>
>>>> I want to add one issue for the security section: Can we specify a way to achieve end-to-end encryption of T.140
>>>> data between a WebRTC endpoint and a traditional SIP/RFC 4103 endpoint through a gateway? I know that that is a
>>>> desired feature.
>>> How would you do that? The data channel uses DTLS encryption, and SIP/RFC 4103 uses SRTP encryption, so
>>> doesn't the gateway have to decrypt/encrypt the T.140 traffic?
>>     
>>     I have just heard the requirement to have end-to-end encryption of RTT,
>>     I do not have the solution. One possibility would maybe be to have media
>>     encryption end-to-end as well as the two transport encryptions. But that
>>     complicates the possibility to insert the missing text markers by the
>>     gateway if text loss is detected.
>    
> Yes.
>
> However, keep in mind that the scope of the draft is how to use SDP O/A to negotiate a T.140 WebRTC data channel. We DO include some text regarding interworking with SIP/RFC 4103, because we know there are environments where such interworking takes place.
>
> But, extending T.140 and/or RFC 4103 (e.g., defining a new application level encryption mechanism for T.140) is outside the scope.
>
> Regards,
>
> Christer
>
>
>
>      >
>      >
>      > Den 2019-08-22 kl. 16:28, skrev Christer Holmberg:
>      >> I have created a pull request, which will be used for the changes based on Gunnar's comments:
>      >>
>      >> https://protect2.fireeye.com/url?k=f453465c-a88743e3-f45306c7-868f633d
>      >> bf25-4deb49c05b8a2375&q=1&u=https%3A%2F%2Fgithub.com%2Fcdh4u%2Fdraft-d
>      >> atachannel-t140%2Fpull%2F5
>      >>
>      >> Regards,
>      >>
>      >> Christer
>      >>
>      >> On 22/08/2019, 13.39, "mmusic on behalf of Christer Holmberg" <[e-mail] on behalf of [e-mail]> wrote:
>      >>
>      >>       Hi Gunnar,
>      >>
>      >>       Thanks you for your support (I assume :) and comments on the draft!
>      >>
>      >>       See inline.
>      >>
>      >>       >A couple of comments:
>      >>       >1) In 3.2, the attribute "cps" is misspelled "cpc" once.
>      >>
>      >>       Will fix.
>      >>
>      >>       ---
>      >>
>      >>       >2) Section 5 has some historical references to real-time text transports that may not be of much interest anymore
>      >>       >and instead confuse the reader, while some other more relevant transports may be added.
>      >>
>      >>       I took these from the schwarz draft. You probably know better
>      >> than me which ones are relevant, so feel to suggest which one(s)
>      >> should be removed, and which one(s) should be be added :)
>      >>
>      >>       >I would also like to discuss if it could be possible to have a few general recommendations on the webrtc to sip/rfc4103 case without
>      >>       >the problems you see with having a detailed gateway section.
>      >>
>      >>       The second last paragraph covers some things on the media plane (out of order and loss of RTP packets) that I think are worth mentioning.
>      >>
>      >>       As far as SDP interworking is concerned, this draft defines the m- line for T.140 data channel, and RFC 4103 defines the m- line for T.140 RTP, and the interworking should be very straight forwards. Do you have something specific in mind regarding general recommendations?
>      >>
>      >>       ---
>      >>
>      >>       > 3) Reliability. Section 3.1 implies that the channel is used in the reliable and ordered mode. We have been discussing back and forth
>      >>       > if that is the right choice for real-time text. I tend to think it is, but it might be useful to discuss it once again. The traditional user
>      >>       > requirement on real-time text is that produced characters shall be presented to the receiver within one second from their creation.
>      >>       > Modern usage in speech-to-text applications may require more rapid transmission. As I understand it, the reliable mode of the
>      >>       > data channel may imply long periods of choked transmission in case of network problems or by influence of heavy transmission
>      >>       > in another channel. As long as this happens only in case of network problems, I now tend to think that that might be acceptable.
>      >>       > The effects of being forced to use an unreliable channel are so far-going so I would like to avoid that.
>      >>       > However, the word "reliable" is misleading. A "reliable" channel is not really reliable. It can break in case of problems.
>      >>
>      >>       True, but "reliable" is the terminology used in both RFC 4960 (SCTP) and draft-ietf-rtcweb-data-channel.
>      >>
>      >>       > I think some recommendations should be inserted in section 4 about what to do when a channel breaks. The natural action
>      >>       > would be for both sides to try to figure out what was the last T.140 data that was transmitted and received, and then try to
>      >>       > reconnect and resume transmission if successful. If any T.140 data was lost during the break, that state should be marked
>      >>       > by inserting the "missing data" T.140 indicator in the received stream. There needs of course also be a recommended action
>      >>       > if it turns out to be impossible to reconnect after a low number of retries.
>      >>
>      >>       I can for sure add some text about that. Are there generic T.140 recommendations for failure that we can reference, or do you think there is something T.140 data channel specific?
>      >>
>      >>       Regards,
>      >>
>      >>       Christer
>      >>
>      >>
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>      >>
>      >>
>      > --
>      > -----------------------------------------
>      > Gunnar Hellström
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>      Gunnar Hellström
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-----------------------------------------
Gunnar Hellström
Omnitor
[e-mail]
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