[asterisk-dev] RFC4733 SRTP option

[Olle E. Johansson]

27 nov 2009 kl. 10.40 skrev Nick Lewis:

>> Well, you can actually exchange keys for SRTP without 
>> revealing the keys in the open in the SIP messaging 
>> channel. So it's not that crazy, Nick.
> 
> I was thinking more from an ietf marketing point of
> view than a technical one. RFC2833 has been obsoleted 
> so new implementations must use RFC4733. But RFC4733 
> mandates SRTP. Therefore in an implementation that 
> does not need security (e.g. soho pbx with pstn 
> trunks) no rtp event DTMF method can be used.
> 
I think you make the wrong assumption. Even in a SOHO pbx, users can call the bank and expose account and pin code over DTMF. Anyone with wireshark will be able to pick it up easily.

> In my view there needs to be a base level ietf 
> product that specifies functionality without any
> security 
Assuming that there is a local office zone inside the firewall that is a secure zone is extremely often wrong and a bad method. Suddenly, someone wants to place calls from home with a soft phone and the assumption is broken the second that someone messes with the NAT and creates port forwarding to the PBX that you thought was feeling good in a secure zone.

I think mandating security for DTMF is a good thing. Remember that the RFC doesn't require you to have SRTP for all media streams.

However, it will propably take more than five years until the installed world can actually accomplish this. The DTLS/SRTP standards need to settle, and the GNUtls and OpenSSL stacks needs some stable implementations of DTLS before that can happen. But here on the developers list, we should be leaders :-)

/O