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

Christer Holmberg <christer.holmberg@ericsson.com> Thu, 22 August 2019 20:55 UTC

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Gunnar Hellström <gunnar.hellstrom@omnitor.se>, Bernard Aboba <bernard.aboba@gmail.com>
CC: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new: draft-holmberg-mmusic-t140-usage-data-channel
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Date: Thu, 22 Aug 2019 20:55:08 +0000
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Archived-At: <https://mailarchive.ietf.org/arch/msg/mmusic/kUlHmovfQacWbOv0fgfXdmT1oP8>
Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-t140-usage-data-channel
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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?

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2) Section 5 has some historical references to real-time text transports that may not be of much interest anymore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