Re: [rtcweb] Use of offer / answer semantics

[Paul Kyzivat <[e-mail]>]

On 9/8/11 12:05 PM, Roman Shpount wrote:
> If the goal is to create something that will interop with an existing
> SIP infrastructure using a signalling proxy only, we would need is to
> fully support RFC 3264. I think what we will need is an ability to
> generate an initial offer (as SDP), process the provisional SDP
> response, process final SDP response, process an offer and generate the
> response, and generate a new offer for the existing call (similar to the
> response to the SIP INVITE with no body) in accordance with rules in RFC
> 3264. We will need to decide which SDP related RFC would need to be
> supported, but I think RFC 4566, RFC 3551, RFC 5245, and RFC 3605 are
> the minimum.

If I understand you, then I think we are in agreement.

	Thanks,
	Paul


> _____________
> Roman Shpount
>
>
> On Thu, Sep 8, 2011 at 11:52 AM, Paul Kyzivat <[e-mail]
> <mailto:[e-mail]>> wrote:
>
>     On 9/8/11 12:45 AM, Bernard Aboba wrote:
>
>
>          > > 1) The media negotiations will be done using the same SDP
>         offer/answer semantics that are used in SIP.
>          > To be precise - you're suggesting that we use RFC 3264
>         offer/answer
>          > semantics. (That RFC is 25 pages long. RFC 3261, the core SIP
>         document,
>          > is 269 pages, and is NOT a normative reference from 3264. I
>         am anxious
>          > to avoid having a normative dependency on 3261.)
>          >
>          > I agree with this.
>
>         [BA] I do *not* agree that RTCWEB should have to support every
>         aspect of
>         SDP offer/answer. Basic offer/answer, sure. All potential corner
>         cases?
>         Not necessarily.
>
>
>     I'm not sure where you are going here?
>     Are you suggesting that all the mandatory to implement O/A semantics
>     of 3264 might not be supported? Or are you saying that O/A support
>     may not work for all defined SDP extensions?
>
>     I think that all mandatory O/A should be supported. If you have
>     something specific in mind that is problematic, then maybe we should
>     investigate why you think it needs to be excluded. My guess is that
>     maybe it is something broken in the spec that ought to be fixed there.
>
>
>          > > 2) It will be possible to gateway between legacy SIP
>         devices that
>         support ICE and appropriate RTP / SDP mechanisms and codecs without
>         using a media gateway. A signaling gateway to convert between the
>         signaling on the web side to the SIP signaling may be needed.
>
>          > I agree with this - I think the "may be needed" will turn out
>         to be
>          > "will be needed", but some portion of that gateway can be
>         implemented by
>          > Javascript running in the browser, if desirable.
>
>         [BA] This seems like a good principle, but I'm not clear that it
>         will
>         work with all use cases. For example, what happens in the E911
>         use cases
>         when an RTCWEB implementation attempts to make a call to a PSAP
>         implementing NENA i3 Stage 3? If you don't have a media gateway,
>         then
>         the browser will need to implement one of the mandated codecs on the
>         PSAP side. So in those use cases, eliminating the media gateway
>         implies
>         making G.711 and H.264 mandatory-to-implement.
>
>
>     AFAIK, not every rtcweb application will be obligated to support E911.
>     (In particular, any application that doesn't identify callees by
>     phone number is a good candidate to be exempt from E911.
>
>     Certainly a server without a media gateway will be limited in what
>     it can call based on codec compatibility. That may or may not be a
>     limitation, depending on the application. Those that find it an
>     unacceptable limitation will probably find a way to incorporate a
>     transcoder when needed.
>
>             Thanks,
>             Paul
>
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