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

[Iñaki Baz Castillo <[e-mail]>]

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]>