as proposed in one of my last e-mails I've changed the way how the
ZRTP GUI is implemented. Quite some modifications were necessary. It's
ready in my sandbox here. Before I commit I thought it is a good idea
to give you some information in advance. Everybody is invited to test
All modifications were tested and work with Phil Zimmermann's original
Zfone, this includes SAS verification.
As proposed by Emanuel we can extend the implementation to support or
use other key sharing / key negotiation mechanisms (if this will ever
happen ). Currently only ZRTP is implemented.
We now have a separate ZRTP panel that handles the display of events
and necessary data and also handles user action to set SAS
verification state. Also this action is separated from the ZRTP
The ZRTP panel is _very_ _simple_ and I kindly ask our GUI gurus to
have a look and propose a much better look-and-feel. I'm not a good
GUI designer at all .
Attached to this mail is a more detailed list of modifications I've
done to make this happen.
Due to various reasons ZRTP is enabled by default for all SIP calls
(more detailed explanation see attachment). This doe not do any harm
to clients that do not support ZRTP - they just ignore it and the user
does not notice any problems. Emanuel provided a patch about 6 weeks
ago to enable/disable ZRTP on a per account basis. After these
modifications are done we can have a look at that patch and how to
Video is not yet supported, it's somehow prepared but needs some
additional modifications and enhancements in ZRTP handling inside
SipSessionImpl and the ZRTP SCCallback class. The GUI
and other mechanisms are already well prepared to handle video.
Regarding video support: The ZRTP specification requires one "Master"
session that must negotiate the keys before additional sessions, for
example Video, can be attached. IMHO we should define the "Audio"
session as the "Master" for this case. Is this ok?
Also because of the key negotiation and the SRTP crypto context only
1-1 calls are possible if security is enabled (one call can have
several media streams but only to the same client), that is only one
participant per call (no secure 1-n calls).
zrtpGuiChanges.txt (2.12 KB)