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

Paul Kyzivat <pkyzivat@alum.mit.edu> Fri, 23 August 2019 15:10 UTC

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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Fri, 23 Aug 2019 11:10:18 -0400
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Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-t140-usage-data-channel
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Gunnar,

On 8/23/19 10:56 AM, Gunnar Hellström wrote:
> 
> Den 2019-08-23 kl. 16:26, skrev Paul Kyzivat:
>> Gunnar,
>>
>> What is the source for these requirements?
> Paul, assuming that you ask for the source for all requirements I have 
> mentioned:
> 
> 1. The cited U-C 5 requirement for multi-party support is from section 
> 3.2 of draft-ietf-rtcweb-data-channel-13
> 
> 2. The end-to-end encryption between technologies is hearsay from the 
> NANC INTEROPERABLE VIDEO CALLING group. I do not have the exact source.
> 
> 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)
> 
> 4. The requirement to insert a missing text marker in case of suspected 
> loss is from ITU-T T.140 amendment 1
> 
> 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.

Thanks! That helps me with context.

ISTM that some of these are *system* requirements that can't be resolved 
with link-level requirements, though there may be link-level features 
that can contribute to a solution.

	Thanks,
	Paul

> Regards
> 
> Gunnar
> 
>>
>>     Thanks,
>>     Paul
>>
>> 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"<mmusic-bounces@ietf.org on behalf of 
>>>> christer.holmberg@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
>>>>      >>
>>>>      >>
>>>>      >> _______________________________________________
>>>>      >>       mmusic mailing list
>>>>      >>mmusic@ietf.org
>>>>      >>https://www.ietf.org/mailman/listinfo/mmusic
>>>>      >>
>>>>      >>
>>>>      > --
>>>>      > -----------------------------------------
>>>>      > Gunnar Hellström
>>>>      > Omnitor
>>>>      >gunnar.hellstrom@omnitor.se
>>>>      > +46 708 204 288
>>>>      >
>>>>      > _______________________________________________
>>>>      > mmusic mailing list
>>>>      >mmusic@ietf.org
>>>>      >https://www.ietf.org/mailman/listinfo/mmusic
>>>>           --
>>>>      -----------------------------------------
>>>>      Gunnar Hellström
>>>>      Omnitor
>>>> gunnar.hellstrom@omnitor.se
>>>>      +46 708 204 288
>>>>
>>> -- 
>>> -----------------------------------------
>>> Gunnar Hellström
>>> Omnitor
>>> gunnar.hellstrom@omnitor.se
>>> +46 708 204 288
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
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>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> 
> -- 
> -----------------------------------------
> Gunnar Hellström
> Omnitor
> gunnar.hellstrom@omnitor.se
> +46 708 204 288
> 
> 
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