Ticket: #5709

Update: This will be resolved with the release of Tails Server (#5688), which will include Mumble.

We need to find out how VoIP can be usable in the context of Tails.

Preliminary testing showed OnionCat + Mumble to be a working and relatively easy to setup Tor-enabled VoIP solution; the 1/2s - 1s delay is only slightly annoying.

As it was pointed out in the "Adding voip to torchat" thread on or-talk, OnionCat before r555 provides no bidirectional authentication: the caller has (limited) certainty to be talking to the call receiver, but the reverse is not true. So this shall be used in combination with zRTP or similar, unless the new unidirectional mode is good enough.


It looks like Linphone 3.5.1 or newer has everything Tails need, so it would be good to test it (probably with OnionCat).

The new OnionCat unidirectional mode (default since r555) should be tested for VoIP.


Encryption and authentication

Note: these are relatively old notes that should be updated and further researched.

On the UI side, something similar to Pidgin's OTR would be perfect.


  • IETF chose DTLS+SRTP over zRTP
  • a PKI is needed to authenticate peers :/






User-friendly peer authentication with a voice-based "short authentication string". How strong is this?


RFC 4353

RFC 4353: three-peers SIP conferencing, using one of them as a central mixer.

RFC 4575

RFC 4575: N-peers SIP conference rooms, using one of the peers as a central mixer. One can see who is saying what.

Mixer-to-client Audio Level Indication

A mechanism for RTP-level mixers in audio conferences to deliver information about the audio level of the individual participants => helps detecting where bad noise comes from.

VoIP software

Last updated: 20121122


  • in Debian Squeeze and Wheezy
  • supposed to support zRTP... some day:
  • their TODO item
  • last update as of 200904
  • supports IPv6 in 3.3.x (Debian experimental only, as of 20121129) but not before (Debian bug #375056, [[!gnomebug 331041 desc="upstream bug]])


  • homepage
  • SIP account => insists to connect to SIP server => impossible to setup a p2p voice call between onioncat IPv6 addresses, at least without registering SIP accounts.
  • cannot connect to a XMPP server running behind a hidden service (2.30.3-3)
  • Link-local XMPP connection manager (telepathy-salut 0.5.0-3) does not support voice calls



  • (previously known as SIP Communicator)
  • homepage, wikipedia page
  • LGPL, written in Java
  • in Debian Jessie
  • supports IPv6, SIP, XMPP
  • supports zRTP for key negotiation, SRTP for voice encryption, and TLS for signaling encryption
  • supports audio SIP and XMPP conference calls; what conferencing protocol?
  • supports OTR for text IM
  • reported to work over Tor
  • we're told it supports Jingle


  • homepage, wikipedia page
  • in Debian Wheezy
  • supports SIP over TCP and TLS
  • supports IPv6
  • supports zRTP since version 3.5.1, but it's not enabled in the Wheezy package (Debian bug #671815)
  • test results: 5-10s lag but one of us was using a really bad Internet connection
  • successfully tested over OnionCat by Whonix folks; see the "Why OnionCat + Mumble - why not just Mumble?" thread on tails-dev@ (August, 2014) for details.
  • audio conferencing since 3.5.0


  • homepage, wikipedia page
  • in Debian Squeeze
  • primary engineering effort targeted at low-latency
  • successfully tested in combination with OnionCat
  • TLS and OCB-AES128; seems to depend on a PKI for peer authentication
  • supports IPv6
  • Tor project's (mttp and Phoul) guide on using Mumble with Tor
  • Plumble is a Mumble client for Android



  • homepage
  • in Debian Squeeze but it has been removed from Wheezy: ROM; dead upstream, obsolete components (KDE3/ QT3/ libccrtp1).
  • was included in Incognito
  • supports SIP, zRTP and SRTP
  • IPv6 is on the roadmap
  • Qt application, but does not depend on KDE libs
  • no release between 20090225 and 20110429 => asked on 20110510 for their plans; no answer so far
  • it's the client advised by GNU Telephony


  • homepage
  • allows to use zRTP on other VoIP software
  • supposed to work with Ekiga
  • some packages in Debian: libzrtpcpp-1.6-0, is that enough?
  • last release was a public beta, out in March 2009
  • license seems inadequate: according to Zfone, "only the libZRTP SDK libraries are provided under the AGPL. The parts of Zfone that are not part of the libZRTP SDK libraries are not licensed under the AGPL or any other open source license. Although the source code of those components is published for peer review, they remain proprietary. The Zfone proprietary license also contains a time bomb provision."