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

Gunnar Hellström <> Fri, 23 August 2019 10:01 UTC

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Date: Fri, 23 Aug 2019 12:01:41 +0200
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Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-t140-usage-data-channel
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Den 2019-08-22 kl. 22:55, 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.


> 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:
>> bf25-4deb49c05b8a2375&q=1&
>> atachannel-t140%2Fpull%2F5
>> Regards,
>> Christer
>> On 22/08/2019, 13.39, "mmusic on behalf of Christer Holmberg" < on behalf of>; 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2) Section 5 has some historical references to real-time text transports that may not be of much interest anymore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
> --
> -----------------------------------------
> Gunnar Hellström
> Omnitor
> +46 708 204 288
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Gunnar Hellström
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