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

[Gunnar Hellström <[e-mail]>]

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.

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"<[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
>>>      > Omnitor
>>>      >[e-mail]
>>>      > [phone]
>>>      >
>>>      > _______________________________________________
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>>>           --
>>>      -----------------------------------------
>>>      Gunnar Hellström
>>>      Omnitor
>>>      [e-mail]
>>>      [phone]
>>>
>> -- 
>> -----------------------------------------
>> Gunnar Hellström
>> Omnitor
>> [e-mail]
>> [phone]
>>
>>
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Gunnar Hellström
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