As described in the faq of pidgin-sipe (http://sourceforge.net/apps/mediawiki/sipe/index.php?title=FAQ) the lack of support of SRTP in farstream cause a problem to use audio/video. I'm opening this ticket, because there is very few information about the status and progress of this important feature.
There is now a SRTP plugin in gst-plugins-bad, just need to integrate it and add the API now
I'm interested in using SRTP/ZRTP using the kde/gnome communications tools named Telepathy. They tell me that SRTP/ZRTP would need to be supported by the Farstream libraries that they use. I was glad to see that there is an SRTP plugin in gst-plugins-bad (Comment#1). I'm sure that there are others who would be interested in making their chats more secure as well.
This branch is just waiting the GStreamer 1.4 release: http://cgit.collabora.com/git/user/tester/farstream.git/log/?h=srtp
Is it possible to test this with the version 1.3? or 1.4 is the minimum? Should we expect this code in the next version of farstream? The plan is to release the next version close to the release of GStreamer 1.4?
You need git master of gst-plugins-bad as of last night... So it will require 1.3.2 or later. But I'll probably hold off until 1.4 is release to merge and release this.
GStreamer 1.4 was releaed in July... :) FWIW when I send DTMF on an encrypted connection, the payload of the DTMF packets isn't encrypted? Should it be? Does that explain why the keypresses don't seem to get through? Is that a bug in the farstream part of the stack or elsewhere?
Was merged yesterday, will be in next release.
Oh, and works mostly with 1.2 too, though 1.4 is recommended. I've tested against a Chrome Browser WebRTC stack.
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