Our default build probably shouldn't include non-free software. With
mod_ilbc, the licensing situation is merely ambiguous. With
mod_siren, the user can't use this code without getting explicit
permission from Polycom (though it is apparently easily given).
These codecs are non-free which creates issues for distributions, so
let's not require these by default to run our example configs. We can
add back in iLBC once we resolve the licensing situation with our
in-tree implementation.
ZRTP passthrough mode allows two ZRTP-capable clients to negotiate an
end-to-end security association through FreeSWITCH. The clients are
therefore able to be certain that the FreeSWITCH instance mediating
the call cannot eavesdrop on their conversation.
Importantly, this capability is maintained across multiple FreeSWITCH
hops. If widely deployed, this enables a global network architecture
where two people can speak securely with strong cryptographically
protected authentication and confidentiality.
With this commit we implement a zrtp-passthru mode that handles all
the details of the negotiation intelligently. This mode can be
selected by setting the boolean parameter inbound-zrtp-passthru in the
sofia profile. This will also force late-negotiation as it is
essential for correctly negotiating an end-to-end ZRTP security
association.
When an incoming call with a zrtp-hash is received and this mode is
enabled, we find the first audio and the first video zrtp-hash in the
SDP and store them as remote values on this channel. Once a b-leg is
available, we set the local zrtp-hash values on that channel to the
remote zrtp-hash values collected from the a-leg.
Because zrtp-passthru absolutely requires that the channels negotiate
the same codec, we offer to the b-leg only codecs that the a-leg can
speak. Once the b-leg accepts a codec, we will force that choice onto
the a-leg.
If the b-leg sends us zrtp-hash values in the signaling, we store
those as remote values on the b-leg and copy them to the local values
on the a-leg.
At this point, each leg has the zrtp-hash values from the other, and
we know we can do ZRTP passthrough mode on the call. We send the
b-leg's zrtp-hash back to the a-leg in the 200 OK.
We then enable UDPTL mode on the rtp streams for both the audio and
the video so that we don't interfere in the ZRTP negotiation.
If the b-leg fails to return a zrtp-hash in the signaling, we set up a
ZRTP security association with the a-leg ourselves, if we are so
equipped. Likewise, if the a-leg fails to send a zrtp-hash in the
signaling, we attempt to set up a ZRTP security association ourselves
with the b-leg.
The zrtp-passthru mode can also be enabled in the dialplan by setting
the boolean channel variable zrtp_passthru. If enabled in this
manner, we can't force late-negotiation, so the user would need to be
sure this is configured.
If ZRTP passthrough mode is not enabled in either manner, this change
should have no effect.
Channel variables for each of the various zrtp-hash values are set,
though it is anticipated that there is no good reason to use them, so
they may be removed without warning. For checking whether zrtp
passthrough mode was successful, we provide the channel variable
zrtp_passthru_active which is set on both legs.
Though not implemented by this commit, the changes here should make it
more straightforward to add correct zrtp-hash values to the signaling
and verify that correct hello hash values are received when FreeSWITCH
is acting as a terminating leg of the ZRTP security association.
A historical note...
This commit replaces the recently-added sdp_zrtp_hash_string method,
commit 2ab1605a88.
This prior method sets a channel variable from the a-leg's zrtp-hash,
then relies on the dialplan to export this channel variable to the
b-leg, where it is put into the SDP.
While it was a great start and wonderful for testing, this approach
has some drawbacks that motivated the present work:
* There's no good way to pass the zrtp-hash from the b-leg back to
the a-leg. In fact, the implementation seems to send the a-leg's
zrtp-hash back to the originating client in the 200 OK. This is
not correct.
* To support video, we'd need to have a separate dialplan variable,
and the dialplan author would need to deal with that explicitly.
* The API is problematic as it requires the dialplan author to
understand intricate details of how ZRTP works to implement a
correct dialplan. Further, by providing too fine-grained control
(but at the same time, not enough control) it would limit our
ability to make the behavior smarter once people started relying on
this.
