Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need

On Apr 4, 2012 10:24 AM, "Roman Shpount" <roman@telurix.com> wrote:
>
> On Wed, Apr 4, 2012 at 12:43 PM, Iñaki Baz Castillo <ibc@aliax.net> wrote:
>>
>> Hi, nobody cares about the implications of option 2 ???
>>
>> Do all the people planning to interop with SIP assume that they'll
>> need the super B2BUA in the second image (without the possibility of
>> using a pure SIP proxy)?:
>>
>>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png
>>
>
> <sarcasm>I guess we should look at this as a business opportunity ;) I am
not sure why you assume that building such gateway would take 10 years. I
can sell a gateway like this to anybody who needs it right now.</sarcasm>
>
> My assumption is that IP phones will migrate to WebRTC effectively
eliminating SIP in end user devices.

That is your academic assumption, why does IT department need to spend
extra money to either desert or upgrade its existing SIP devices without
substantial gain? After all, tons of users are still using window xp.

Backward capability is a key to the success of new technology.

-Jesse

The only place where SIP will remain would be federation, which commonly
uses some sort of SBC anyway. This SBC will need to be extended to support
ICE anyway. Might as well through in support for DTLS-EKV-SRTP. I doubt it
will map DTLS key updates to re-invites. Most probably it will simply
re-encode.
>
> ______________
> Roman Shpount
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>