Re: [rtcweb] Use Case draft - legacy interop

"Dan Wing" <dwing@cisco.com> Sat, 05 May 2012 17:24 UTC

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From: Dan Wing <dwing@cisco.com>
To: 'Iñaki Baz Castillo' <ibc@aliax.net>
References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com> <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com> <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com> <4F9EC0B2.10903@alcatel-lucent.com> <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com> <4FA1575C.4050508@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F4BF@NAHALD.us.int.genesyslab.com> <4FA37A1E.4080806@alvestrand.no> <CALiegf=H5QH_YY-cJ4z29wChWZ-VoQpHvsZCeaJPjTgVp+km3Q@mai l.gmail.com> <0db701cd 2adb$98082f10$c8188d30$@com> <CALiegf=C7a6zeUDHn-Wuku9eFWmADG5N+D8oXSQbwJKSYYjcQg@mail.gmail.com>
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Date: Sat, 05 May 2012 10:24:48 -0700
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft - legacy interop
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> -----Original Message-----
> From: Iñaki Baz Castillo [mailto:ibc@aliax.net]
> Sent: Saturday, May 05, 2012 9:32 AM
> To: Dan Wing
> Cc: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft - legacy interop
> 
> 2012/5/5 Dan Wing <dwing@cisco.com>:
> >> If I have a SIP phone implementing ICE and DTLS-SRTP (which is
> >> standarized for SIP regardless it has null impementation), will my
> SIP
> >> phone be able to *directly* talk in the media plane with a WebRTC
> >> browser? or not?
> >
> > That would work.
> 
> Ok, so then, taking into account that SIP defines the usage of
> DTLS-SRTP in RFC 5763 (regardless no one device implements it), why
> does this WG assumes that "interop with non WebRTC endpoints will be
> made via *gateways*"?

Perhaps a more accurate statement would be:

  directly, if the remote endpoint supports ICE and supports
  DTLS-SRTP; otherwise, via a gateway.

> Let me please repeat my question: if my SIP phone implements RFC 5763,
> can my SIP phone directly interop at media plane with a WebRTC
> browser? (I know you already replied this, but I want to be very very
> sure) :)

Sure seems it should work.  Only nuance that I am aware of at the moment
is WEBRTC seems to want to require Consent Freshness (periodic ICE
connectivity checks) for the duration of the call.  This is okay, and
would work with an existing SIP phone -- so long as the SIP phone
continues to respond to ICE messages.  It should.  But I doubt that
is exercised in SIPIT testing.  The other nuance is that, because
doing the SHA1 for MESSAGE-INTEGRITY isn't needed for consent freshness,
there is desire to allow those periodic ICE connectivity checks to
omit the MESSAGE-INTEGRITY, which is a change to ICE.  See 
draft-muthu-behave-consent-freshness.  Of course, that doesn't require
a gateway, but would require softening the implementation on the
phone.  draft-muthu-behave-consent-freshness might also add a new
SDP attribute, I dunno yet.

-d

> Thanks a lot.
> 
> 
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>