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

["Ravindran, Parthasarathi" <[e-mail]>]

IMHO, there is no need to tweak WebRTC recommendation for the sake of SIP proxy. I'm fine as long as there is a way in WebRTC to interop with SIP. For example, SIP proxy is not suitable for IETF SIPREC recording implementation itself!!!

But I'm also surprised to see that there is no response for Fabio Pietrosanti mail on DTLS-SRTP trust model mail thread.

Thanks
Partha 

>-----Original Message-----
>From: [e-mail] [mailto:[e-mail]] On Behalf
>Of Iñaki Baz Castillo
>Sent: Thursday, April 05, 2012 2:34 PM
>To: Roman Shpount
>Cc: [e-mail]
>Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
>
>2012/4/5 Roman Shpount <[e-mail]>:
>> On the more serious note, very few SIP end points offer working ICE
>support.
>> So, in a large sense, interop with them is not an option.
>
>For me there is a BIG difference when a kind of B2BUA is required (which
>involves signaling "transaction"). That's the barrier IMHO.
>
>ICE support can be implemented in a ICE-Lite RTP/SRTP proxy, see:
>
>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.png
>
>The problem arises when media encrypt/decrypt is required, and evenr
>more when a key update in RTP (like the DTLS EKT update) must be
>converted into a signaling re-INVITE by a super Signaling+Media B2BUA:
>
>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png
>
>
>
>> Out of the ones
>> that do support ICE and SRTP, very few are actually connected directly
>> to a public internet. Most of them are connected to some sort of PBX
>> or an IP PBX type service. So, in reality you do not need to bridge
>> every IP phone with WebRTC.
>
>That is a *very* limited scope of what WebRTC can provide. An IT
>department should be able to deploy its own WebRTC infrastructure (a
>Web+WebSocket server) within its "local" network, so browsers
>accessing to such a local website share the network with SIP/XMPP
>phones/devices/softphones.
>
>Please don't imagine WebRTC and SIP interop as the communication between
>two islands ;)
>
>
>
>> If a few PBX and hosted centrex vendors will add support for WebRTC
>> required features, we will get compatibility with existing end points.
>
>Pure SIP proxies do exist. A PBX is not always needed, so again, SIP
>world does not require to be "an island".
>
>
>> To support the rest you will need to deploy some sort of gateway.
>
>Not in the case SDES+SRTP is allowed in WebRTC (see Fabius's recent
>mails about SDES and DTLS).
>
>
>
>> hurdle (I am trying to be politically correct here). WebRTC enabled
>> end points, on the other hand, will offer significant benefits to
>> traditional SIP phones, since they will allow development of higher
>> quality integrated real time communications services. I hope this will
>> drive a much quicker standard adoption.
>
>The problem is that there is a *NEW* SRTP related spec for WebRTC:
>
>  http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-03
>
>Maybe it could also become a standard for SIP? I hope, but currently
>it's not, so by *mandating* DTLS-EKT-SRTP in WebRTC we are creating a
>island.
>
>
>Regards.
>
>
>--
>Iñaki Baz Castillo
><[e-mail]>
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