Re: [rtcweb] Working group last call for draft-ietf-rtcweb-audio
Ted Hardie <ted.ietf@gmail.com> Fri, 18 December 2015 22:22 UTC
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Date: Fri, 18 Dec 2015 14:22:10 -0800
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From: Ted Hardie <ted.ietf@gmail.com>
To: Roman Shpount <roman@telurix.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Working group last call for draft-ietf-rtcweb-audio
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On Fri, Dec 18, 2015 at 2:10 PM, Roman Shpount <roman@telurix.com> wrote: > On Fri, Dec 18, 2015 at 4:44 PM, Ted Hardie <ted.ietf@gmail.com> wrote: > >> On Fri, Dec 18, 2015 at 1:10 PM, Roman Shpount <roman@telurix.com> wrote: >> >>> On Fri, Dec 18, 2015 at 2:34 PM, Ted Hardie <ted.ietf@gmail.com> wrote: >>> >>>> On Fri, Dec 18, 2015 at 10:32 AM, Roman Shpount <roman@telurix.com> >>>> wrote: >>>> >>>>> Hi All, >>>>> >>>>> Since the decision not to support A-D DTMF digits was never made on >>>>> the IETF list (I have looked through the archive and cannot find it), I do >>>>> not think they should be removed. >>>>> >>>>> >>>> Can you say more about what use there would be in supporting them? >>>> >>> >>> First of all, DTMF support is there for legacy support. >>> I generally dislike when features are removed from an established >>> capability set without a good reason. >>> >> >> We have stretched very far in many directions to be reasonable in our >> support of WebRTC to legacy use cases, but I don't think we can argue that >> WebRTC has an established capability set that includes these, nor has >> anyone given flows where a legacy system would reasonably emit them toward >> a WebRTC client. >> > > There is no additional implementation burden to support DTMF tones A-D, so > there is no stretching as far as any additional work is concerned. > > I can guarantee you that legacy systems will generate DTMF tones towards > WebRTC client. > Sorry that this was not clear; I meant specifically A-D. Offer/answer telephone event negotiation is normally by directional. This > normally enabled DTMF detection in PSTN gateways. So, if you have any > system that connects WebRTC client to PSTN network, and if the person on > PSTN network will press a keypad digit, this digit will be sent to WebRTC > client. Furthermore, it is common for DTMF digit detection to be caused by > talk off, i.e. some voice periodically triggers DTMF detection. Finally, > some gateways will also send ANS tones if the call hits a fax machine, so > WebRTC end point will get not only DTMF events, but events for tone that > someone just happened to enable on the gateway. Unless we want to transcode > media coming from PSTN, WebRTC client will need to deal with any incoming > telephone-events. All of this being said, it is up to WebRTC client to > decide what to do with the received events. I am fine with these tones > being discard as long as they are being discarded in a manner which does > not disrupt normal audio processing. > This seems to result in exactly the case Cullen was concerned about; if they may be discarded their functionality remains completely unreliable. But more to the point, you are not pointing to uses of A-D, but number detection and events which would still be covered. This means no unexpected jitter buffer adjustments, packet loss correction > for the audio segment covered by telephone-event, treating these packets as > lost, or payload mismatch. > > > Can anyone name me a single commercial PSTN VoIP gateway which does not >>> support A-D DTMF tones? I cannot think of one. I believe it would generally >>> make sense to implement whole set defined in >>> https://tools.ietf.org/html/rfc4733#section-3.2 instead of just 3/4. >>> From what I know there is no extra implementation to support A-D DTMF tone. >>> It looks like these tones are removed for aesthetic reasons only. >>> >> >> No, the point Cullen made was that they were sufficiently hit-or-miss in >> terms of system blocking that they were effectively unreliable *in the >> contexts where WebRTC and legacy might interact*. >> > > All DTMF are often unreliable. There are also enough cases when they are > absolutely reliable. In places where they have been used, they can continue > to be used for legacy interop. In cases where they were unreliable they > will continue to be unreliable. It is not our job to fix or change this. > While this is no doubt not your intent, I must say your message reads as "Sure it's broken now, but we must make sure it remains broken to be backwards compatible." > > >> As far as more specific examples are concerned: >>> >>> DTMF tones A-D are used to provide a way to access hidden flows in IVR >>> systems. For instance, when inbound call center is integrated with an >>> external auto-dialer, auto-dialer places a call to call center IVR and uses >>> a hidden flow accessible via one of those tones to connect directly to a >>> waiting agent. >>> >>> Given that they likely would block the receipt of those tones from >> external clients in order to avoid this access from outside, this seems >> like one of the cases that Cullen alluded to; it is permitted only part of >> the time and the access it provides is not standard. >> > > This is absolutely incorrect. In this particular example auto-dialer is > running on a hosted platform and accesses the inbound call center over > public PSTN. > > >> DTMF tones A-D are also commonly used by ham radio operators to integrate >>> with PSTN, such as http://www.echolink.org/Help/dtmf_functions.htm >>> >>> >> This seems far outside our remit. >> > > Why? I have already seen a few virtual ham radio stations with WebRTC > interface > > The only A-D tone in the one example you gave (echolink) was to pull up a different stored configuration. > DTMF tone A-D are used to provide prioritized call handling service in ITU >>> Q.955.3 specification which is still used by some PBX and phone systems. >>> >>> >> Again, not at all clear that a WebRTC client would ever expect to use >> these features of a PBX. >> > > There are intranet WebRTC applications that access in-company legacy PBX > via a private gateway. > And they are using the A-D set? > > Finally, some calling card like systems allow remote control of functions >>> such as flash by mapping them to extended DTMF tones. >>> >>> >> If there is a calling card system using WebRTC, I have not heard of it. >> Can you provide a pointer? >> > > Once again, this would be a private WebRTC application used to place calls > through a gateway which is under control of the person who developer the > app. > > So this is theoretical. > The main point here is that we need a consistent set. I take it you do >> agree with that point, and disagree only on whether the set should include >> A-D. Do I have that right? >> > > I agree that we need a consistent set. I would also like that this set > would be defined once and will have no optional components. I do not want > to spent any more time discussion they API to discover which DTMF tones are > supported by a particular implementation. Including tones A-D removes any > excuses for such options. > > Removing them from the set does as well, at least as far as I can tell. No one has suggested that they be the subject of discovery or negotiation. Ted > Regards, > _____________ > Roman Shpount > >
- [rtcweb] Working group last call for draft-ietf-r… Ted Hardie
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Ted Hardie
- Re: [rtcweb] Working group last call for draft-ie… Bernard Aboba
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Adam Roach
- Re: [rtcweb] Working group last call for draft-ie… Adam Roach
- Re: [rtcweb] Working group last call for draft-ie… Ted Hardie
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Ted Hardie
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Bernard Aboba
- Re: [rtcweb] Working group last call for draft-ie… Cullen Jennings (fluffy)
- Re: [rtcweb] Working group last call for draft-ie… Bernard Aboba
- Re: [rtcweb] Working group last call for draft-ie… Cullen Jennings (fluffy)
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Adam Roach
- Re: [rtcweb] Working group last call for draft-ie… Barry Dingle
- Re: [rtcweb] Working group last call for draft-ie… Bernard Aboba
- Re: [rtcweb] Working group last call for draft-ie… Matthew Kaufman
- Re: [rtcweb] Working group last call for draft-ie… Gunnar Hellström
- Re: [rtcweb] Working group last call for draft-ie… DRAGE, Keith (Keith)
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… DRAGE, Keith (Keith)
- Re: [rtcweb] Working group last call for draft-ie… Martin Thomson
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… DRAGE, Keith (Keith)
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Tim Panton
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Jean-Marc Valin
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount
- Re: [rtcweb] Working group last call for draft-ie… Jean-Marc Valin
- Re: [rtcweb] Working group last call for draft-ie… Roman Shpount