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Re: [Sip] RFC 4474 and PSTN
Hi Hannes,
what's than your recommendation how to handle SRTP (based on DTLS-SRTP)
for calls originated from PSTN and transferred to IP?
-Don't allow SRTP for such calls and fall back to RTP
-Allow SRTP, but don't protect the certificate fingerprint
Kai
> -----Original Message-----
> From: sip-bounces at ietf.org [mailto:sip-bounces at ietf.org] On
> Behalf Of Hannes Tschofenig
> Sent: Freitag, 4. April 2008 10:51
> To: Elwell, John
> Cc: sip at ietf.org
> Subject: Re: [Sip] RFC 4474 and PSTN
>
> I would instead not put the SIP Identity header there.
> No additional work needed.
>
> Ciao
> Hannes
>
> Elwell, John wrote:
> > There seems to be some interest in adding some indicator to
> the URI or
> > to the header field to warn of the limited guarantee of
> authenticity or
> > the opposite (i.e.,lack of such an indicator would mean the
> guarantee is
> > low, i.e., the identity has not been verified).
> >
> > I think on this issue it is just a matter of somebody putting things
> > together into a draft. I think Dean and/or Adam were the
> main initiators
> > of this proposal - something like a source=pstn parameter. Would
> > somebody care to author a draft? Or do we need further discussion?
> >
> > John
> > _______________________________________________
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> > Use sipping at ietf.org for new developments on the application of sip
> >
>
> _______________________________________________
> Sip mailing list https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors at cs.columbia.edu for questions on current sip
> Use sipping at ietf.org for new developments on the application of sip
>
_______________________________________________
Sip mailing list https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors at cs.columbia.edu for questions on current sip
Use sipping at ietf.org for new developments on the application of sip