Secure Call

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"Privacy is ultimately about liberty while surveillance is always about control"

Contents

Introduction

The GNU Telephony Secure Calling initiative is intended to make passive voice communication intercept a thing of the past, and thereby help both national governments, and private corporations, which at present seem unable to comply with their obligations to the public that they serve and with the law of the land. While it is true that technological means for mass communication intercept has grown with incremental improvements in communication technology, the means to apply and use techniques to counter these abuses on a large scale have also grown. By making "secure by design", security capabilities simple to embed, and by enabling the largest possible participation in developing such solutions through free software, we hope to break down those remaining barriers that prevent secure voice telephony from being widely deployed.

Goals

To empower people, individually and collectively, to communicate and collaborate privately and securely in real-time worldwide

To establish secure communications as the default communication infrastructure

To enable secure anonymous communication worldwide and protect users who exercise their basic human freedom of privacy

To provide secure communication services universally on all computing platforms

Elements of Secure Calling

There are I perceive three elements to secure calling. The first and most obvious is securing the voice (bearer channel) itself. To achieve this, we initially offer GNU ccrtp 1.5.0. GNU ccrtp now offers basic voice encryption capability using the GNU gcrypt library and implements the Secure RTP profile as per RFC 3711.

In addition to SRTP capabilities starting in GNU ccrtp 1.5, we have an expansion library, libzrtpcpp. This is a library that extends ccrtp with direct support for the ZRTP protocol. The following document gives some background information about ZRTP. Have a look at Phil Zimmermann's Zfone project to get first-hand information from the originator of ZRTP. While Zfone operates as an external service (bump-in-the-wire) that needs to listen and intercept the signaling protocol (SIP) of the clients in order to apply ZRTP and SRTP, GNU ZRTP (libzrtpcpp) can be used to create softphone clients and VOIP servers that natively speak ZRTP without needing an external service. Programs that use GNU ZRTP are not required to use a signaling protocol to use ZRTP and to set up a secure RTP session. The first example is the Twinkle softphone client, starting with release 0.9.0.

One of the reason's for using ZRTP in particular is it's management and use of self generated keys, rather than requiring certificate authorities. This also means that the user is not required to obtain or manage secret keys or certificates nor has any knowledge of the underlying communications protocols. Hence, neither the user, nor an issuing authority, offers any ability to obtain backdoors into or the means to compromise ZRTP sessions.

While securing the voice channel is fundamental to privacy, it is not in itself sufficient to assure true privacy from passive observance. As we have learned, given the hunger for data mining that some national governments and private corporations have, it is equally important to secure the signalling and call detail information from external observation.

With GNU osip2 and libeXosip2, we have some capability to apply SIP over TCP with TLS. This is sufficient to secure session information from casual observance. However, it does not obscure the IP addresses of the physical endpoints which are communicating. To achieve this, we are working on forward publishing SIP contact information into anonymous proxies. However, there is no requirement that SIP must be used as the signalling protocol in conjunction with GNU ccRTP or GNU ZRTP, so we may also look at or introduce other signaling/session protocols as well in the future for this purpose.

Our intent is to apply standard compliant protocols and methods where they exist and are usable to offer secure calling. Where existing protocols need to be extended, we will offer and document such extensions. Where new and emerging protocols exist for secure calling, such as ZRTP, we will make use of those as well. Our goal is to have a broad GPL licensed free software framework for development of secure calling services that cover the full range of possibilities (hence, for example, both SRTP and ZRTP), rather than narrowly implementing a single variation.

Issues and Limitations

When we speak of securing VOIP from passive intercept through encryption, we are of course presuming that while there is the possibility that there exists undiscovered and undisclosed shortcuts in some of the core algorithms, even if true, the computational cost and time to decode a communication session would remain so great that it would remain only of value to attempt against very select targets.

Nor does encrypted voice in any way interfere with legitimate law enforcement tasks where specific individuals are targeted for lawful investigation, presumably with judicially sought warrants. However, the means to do intercept would require on-premise bugging rather than external passive intercept. The cost to use such methods for the very small number of lawful investigations is indeed very small compared both to the cost of deploying passive intercept capabilities everywhere, and especially compared to the potential loss of everyone's right to free expression and free association experienced by the deceptive deployment of police state inspired mass intercept.

However, while encrypting voice has generally been understood for a long time, securing connection information is a different problem. We can apply TLS to existing session protocols so that they do not reveal secondary information, and this helps reduce the visible communication profile even further. However, the final task of obscuring who communicated with whom by IP endpoint address is much harder to achieve.

One answer is the use of anonymous proxies. This helps in that neither party directly communicates, and each session is through a different proxy. Anonymous proxies have been used successfully in projects like Tor because they have multiple stages. This cannot be done, at least in the same way, in voice communication because of latency. Hence we are reduced to single step proxies.

In the worst case in a single step anonymous proxy, an external party with sufficient resources to observe broadly may well successfully observe that person at IP address A communicated with proxy at IP address X starting at a fixed and known time, and that proxy X formed a connection to someone at IP address B at roughly the same time. Having very many users on each proxy will obscure this, especially in conjunction with doing things with false sessions, randomly delayed startup, and separate shutdown with false traffic. However, again this requires proxies each capable of hosting a very large number of users, and there being sufficient usage to assure many of these proxies are filled to near capacity.

Project Status

Phase 1 - The Stack

Securing voice channels with Secure RTP profile and ZRTP support. Introduced with GNU ccRTP 1.5 and improved since then. Complete, initially released October 1, 2006, along with the release by Michel de Boer of a Secure Call capable Twinkle softphone client. Current releases are maintained compatible with the latest release of zfone and the latest IETF "Media Path Key Agreement for Secure RTP" drafts. Recently, a new Java implementation of the ZRTP stack has been created, and successfully integrated into the Java based SIP Communicator client. Recent downloads of SIP Communicator come with ZRTP4J support.

Phase 2 - The Switch

Many IP-PBX systems (such as Asterisk) establish media sessions with the endpoints (operating as a b2bua), and thereby require both media to pass through the server itself, and to do so in decrypted form. This is essentially a "forced" man-in-the-middle, as well as adding latency and in my opinion dis-advantaging the existing peer media/network scalable capabilities of SIP/RTP softphone devices.

GNU SIP Witch, initially released in 2008, is offered as a model for a secure VOIP telephone switch that supports TLS over SIP and that can interconnect secure endpoints without intercepting media, thereby assuring privacy without compromising security. Furthermore, GNU SIP Witch offers a "forward plugin" and calling model whereby one can support secure media interconnection directly between ZRTP capable endpoints while also operating in conjunction with and access to an insecure calling domain managed by a conventional B2BUA based IP-PBX. This is accomplished by cross-registering secure user agents with the insecure IP-PBX and routing insecure destinations to it.

Phase 3 - Domain Calling

Moving GNU SIP Witch to the desktop, and consolidating integration with a local SIP User Agent, it has become possible to offer a bottom-up participatory network that uses published protocols and allows people to directly communicate anywhere using only DNS lookup of SIP uri's. This offers a free software alternative to Skype, which depends both on source secret binary protocol libraries and a central service provider, either of which can be compromised. This is called "Domain Calling", and is being introduced as a feature in Ubuntu Lucid and Fedora F13 GNU/Linux.

Phase 4 - Worldwide Anonymous Calling

Anonymous calling proxy; Bayonne Voice Relay, with forward publishing of SIP contact information, and the ability to search and cache contact points. The goal is to allow truly anonymous and untracable calling without revealing even basic signalling meta-data. In Design.

Current Downloads

commoncpp2-1.7.3 Core platform portable threading and Sockets
ccrtp-1.7.1 GNU RTP Stack with SRTP support
libzrtcpp-1.4.5 ZRTP addon to GNU RTP stack
twinke-1.4.2 Softphone supporting GNU RTP stack SRTP/ZRTP
ucommon-4.0.1 Next generation core platform
libosip2-3.1.0 GNU oSIP messaging and parsing stack
libeXosip2-3.1.0 Transactions, event management, and transport for GNU oSIP
sipwitch-0.9.1 GNU SIP Witch call server

Todo and User Comments

Additional GUI elements for selection of alternate encryption algorithms?

Maybe more GUI elements for SAS key management, fingerprints for social key exchange?

  • Twinkle already supports GUI elements and actions to display and verify the Short Authentication String (SAS). ZRTP uses SAS to to enable users to detect and thus avoid Man-in-the-Middle (MitM) attacks. SAS is not a key but an authentication string that users read aloud and compare over the SRTP audio session.

Twinkle support for tls secured SIP (tcp SIP with tls), and certificates for traditional SRTP related operations?

Please add additional ones.

Related Resources

Some background of secure VoIP and the ZRTP specification
Zfone Project Home and ZRTP specification
Twinkle Home
SIP Communicator Java based multi-platform / multi-protocol ZRTP capable client
GNU SIP Witch
ZRTP FAQ
GNU ZRTP How To
Personal tools