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

[Christer Holmberg <[e-mail]>]

Hi,

>1. The cited U-C 5 requirement for multi-party support is from section 3.2 of draft-ietf-rtcweb-data-channel-13

I think the draft supports that requirement, but let me know if you think something is missing.

>2. The end-to-end encryption between technologies is hearsay from the NANC INTEROPERABLE VIDEO >CALLING group. I do not have the exact source.

I’d be happy to get more information about that, and if there are some mechanisms available we may be able to reference them. But, defining such mechanism is outside the scope of the draft.

>3. The user requirement for a maximum of one second delay from text entry to far end presentation is from ITU-T >F.700 / F.703 (the requirement may be more strict nowadays because of new applications)

That may be worth to take a look at. I am not sure there is much we can do in the draft in order guarantee such max delay, but it may be useful to point out.

>4. The requirement to insert a missing text marker in case of suspected loss is from ITU-T T.140 amendment 1

The text currently says:

“Data sending and reporting procedures conform to [T140].”

We can for sure add text if the missing text marker is not covered by that statement.

But, the idea is to only reference generic T.140 procedures, and only define the data channel specific aspects.

>5. The requirement to try to reconnect in case of channel break is just general user expectation on call behavior. If >the session is still up and the RTP media flows, then the RTT is also expected to be maintained. Only if the whole >session is also lost, then it is accepted to give up on the RTT.

That is for sure something we can add text about.

But, again, I don’t think it’s a data channel specific requirement, but we can describe how the re-connection etc takes place when using a data channel.

Thanks!

Regards,

Christer



On 8/23/19 8:51 AM, Gunnar Hellström wrote:

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]><mailto:[e-mail]@ericsson.com>  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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     >>
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     > Gunnar Hellström
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