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

[Iñaki Baz Castillo <[e-mail]>]

2012/4/4 Roman Shpount <[e-mail]>:
> On Wed, Apr 4, 2012 at 12:43 PM, Iñaki Baz Castillo <[e-mail]> 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. 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... I hope you missed the </sarcasm> closing tag just HERE, am I
right? XDDD

-- 
Iñaki Baz Castillo
<[e-mail]>