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

Christer Holmberg <christer.holmberg@ericsson.com> Thu, 05 April 2012 11:14 UTC

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Iñaki Baz Castillo <ibc@aliax.net>, Roman Shpount <roman@telurix.com>
Date: Thu, 05 Apr 2012 13:14:17 +0200
Thread-Topic: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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> 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:

...and, in general we should not specify procedures which require an intermediary to trigger and send re-INVITEs in the first place, because that itself can then cause lots of issues.

Regards,

Christer




> 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
<ibc@aliax.net>
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