/**
 
@mainpage Introduction to libSRTP
 
This document describes libSRTP, the Open Source Secure RTP library
from Cisco Systems, Inc.  RTP is the Real-time Transport Protocol, an
IETF standard for the transport of real-time data such as telephony,
audio, and video, defined by RFC 3550.  Secure RTP (SRTP) is an RTP
profile for providing confidentiality to RTP data and authentication
to the RTP header and payload.  SRTP is an IETF Proposed Standard,
defined in RFC 3711, and was developed in the IETF Audio/Video
Transport (AVT) Working Group.  This library supports all of the
mandatory features of SRTP, but not all of the optional features.  See
the @ref Features section for more detailed information.
 
This document is organized as follows.  The first chapter provides 
background material on SRTP and overview of libSRTP.  The following
chapters provide a detailed reference to the libSRTP API and related
functions.  The reference material is created automatically (using the
doxygen utility) from comments embedded in some of the C header
files. The documentation is organized into modules in order to improve
its clarity.  These modules do not directly correspond to files. An
underlying cryptographic kernel provides much of the basic
functionality of libSRTP, but is mostly undocumented because it does
its work behind the scenes.

@section LICENSE License and Disclaimer

libSRTP is distributed under the following license, which is included
in the source code distribution.  It is reproduced in the manual in
case you got the library from another source.
	
@latexonly
\begin{quote}
Copyright (c) 2001-2005 Cisco Systems, Inc.  All rights reserved.

Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
\begin{itemize}
\item  Redistributions of source code must retain the above copyright
  notice, this list of conditions and the following disclaimer.
\item Redistributions in binary form must reproduce the above
  copyright notice, this list of conditions and the following
  disclaimer in the documentation and/or other materials provided
  with the distribution.
\item Neither the name of the Cisco Systems, Inc. nor the names of its
  contributors may be used to endorse or promote products derived
  from this software without specific prior written permission.
\end{itemize}
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
COPYRIGHT HOLDERS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
OF THE POSSIBILITY OF SUCH DAMAGE.
\end{quote}
@endlatexonly

@section Features Supported Features

This library supports all of the mandatory-to-implement features of
SRTP (as defined by the most recent Internet Draft).  Some of these
features can be selected (or de-selected) at run time by setting an
appropriate policy; this is done using the structure srtp_policy_t.
Some other behaviors of the protocol can be adapted by defining an
approriate event handler for the exceptional events; see the @ref
SRTPevents section.  

Some options that are not included in the specification are supported.
Most notably, the TMMH authentication function is included, though it
was removed from the SRTP Internet Draft during the summer of 2002.


@latexonly
Some options that are described in the SRTP specification are not
supported.  This includes 
\begin{itemize}
\item the Master Key Index (MKI),
\item key derivation rates other than zero,
\item the cipher F8,
\item anti-replay lists with sizes other than 128,
\item the use of the packet index to select between master keys.
\end{itemize}
@endlatexonly
 
The user should be aware that it is possible to misuse this libary,
and that the result may be that the security level it provides is
inadequate.  If you are implementing a feature using this library, you
will want to read the Security Considerations section of the Internet
Draft.  In addition, it is important that you read and understand the
terms outlined in the @ref LICENSE section.


@section Installing Installing and Building libSRTP

@latexonly

To install libSRTP, download the latest release of the distribution
from \texttt{srtp.sourceforge.net}.  The format of the names of the
distributions are \texttt{srtp-A.B.C.tgz}, where \texttt{A} is the
version number, \texttt{B} is the major release number, \texttt{C} is
the minor release number, and \texttt{tgz} is the file
extension\footnote{The extension \texttt{.tgz} is identical to
\texttt{tar.gz}, and indicates a compressed tar file.}  You probably
want to get the most recent release.  Unpack the distribution and
extract the source files; the directory into which the source files
will go is named \texttt{srtp}.

libSRTP uses the GNU \texttt{autoconf} and \texttt{make}
utilities\footnote{BSD make will not work; if both versions of make
are on your platform, you can invoke GNU make as \texttt{gmake}.}.  In
the \texttt{srtp} directory, run the configure script and then make:
\begin{verbatim}
  ./configure [ options ]       
  make                          
\end{verbatim}
The configure script accepts the following options:
\begin{quote}
\begin{description}
\item[--help]              provides a usage summary.
\item[--disable-debug]     compiles libSRTP without the runtime 
			   dynamic debugging system.
\item[--enable-generic-aesicm] compile in changes for ismacryp
\item[--enable-syslog]     use syslog for error reporting.
\item[--disable-stdout]    diables stdout for error reporting.
\item[--enable-console]    use \texttt{/dev/console} for error reporting
\item[--gdoi]              use GDOI key management (disabled at present).
\end{description}
\end{quote}

By default, dynamic debugging is enabled and stdout is used for
debugging.  You can use the configure options to have the debugging
output sent to syslog or the system console.  Alternatively, you can
define ERR\_REPORTING\_FILE in \texttt{include/conf.h} to be any other
file that can be opened by libSRTP, and debug messages will be sent to
it.

This package has been tested on the following platforms: Mac OS X
(powerpc-apple-darwin1.4), Cygwin (i686-pc-cygwin), Solaris
(sparc-sun-solaris2.6), RedHat Linux 7.1 and 9 (i686-pc-linux), and
OpenBSD (sparc-unknown-openbsd2.7).


@endlatexonly

@section Applications Applications

@latexonly

Several test drivers and a simple and portable srtp application are
included in the \texttt{test/} subdirectory.

\begin{center}
\begin{tabular}{ll}
\hline
Test driver    	& Function tested	\\
\hline
kernel\_driver   & crypto kernel (ciphers, auth funcs, rng) \\
srtp\_driver	& srtp in-memory tests (does not use the network) \\
rdbx\_driver	& rdbx (extended replay database) \\
roc\_driver	& extended sequence number functions \\ 
replay\_driver	& replay database  \\
cipher\_driver	& ciphers  \\
auth\_driver	& hash functions \\
\hline
\end{tabular}
\end{center}

The app rtpw is a simple rtp application which reads words from
/usr/dict/words and then sends them out one at a time using [s]rtp.
Manual srtp keying uses the -k option; automated key management
using gdoi will be added later.

The usage for rtpw is

\texttt{rtpw [[-d $<$debug$>$]* [-k $<$key$>$ [-a][-e]] [-s | -r] dest\_ip
dest\_port] | [-l]}

Either the -s (sender) or -r (receiver) option must be chosen.  The
values dest\_ip, dest\_port are the IP address and UDP port to which
the dictionary will be sent, respectively.  The options are:
\begin{center}
\begin{tabular}{ll}
  -s		& (S)RTP sender - causes app to send words \\
  -r		& (S)RTP receive - causes app to receive words \\
  -k $<$key$>$      & use SRTP master key $<$key$>$, where the 
		key is a hexadecimal value (without the
                leading "0x") \\
  -e            & encrypt/decrypt (for data confidentiality)
                (requires use of -k option as well)\\
  -a            & message authentication 
                (requires use of -k option as well) \\
  -l            & list the available debug modules \\
  -d $<$debug$>$    & turn on debugging for module $<$debug$>$ \\
\end{tabular}
\end{center}

In order to get a random 30-byte value for use as a key/salt pair, you
can use the \texttt{rand\_gen} utility in the \texttt{test/}
subdirectory.

An example of an SRTP session using two rtpw programs follows:

\begin{verbatim}
[sh1] set k=`test/rand_gen -n 30`
[sh1] echo $k
c1eec3717da76195bb878578790af71c4ee9f859e197a414a78d5abc7451
[sh1]$ test/rtpw -s -k $k -ea 0.0.0.0 9999 
Security services: confidentiality message authentication
set master key/salt to C1EEC3717DA76195BB878578790AF71C/4EE9F859E197A414A78D5ABC7451
setting SSRC to 2078917053
sending word: A
sending word: a
sending word: aa
sending word: aal
sending word: aalii
sending word: aam
sending word: Aani
sending word: aardvark
...

[sh2] set k=c1eec3717da76195bb878578790af71c4ee9f859e197a414a78d5abc7451
[sh2]$ test/rtpw -r -k $k -ea 0.0.0.0 9999 
security services: confidentiality message authentication
set master key/salt to C1EEC3717DA76195BB878578790AF71C/4EE9F859E197A414A78D5ABC7451
19 octets received from SSRC 2078917053 word: A
19 octets received from SSRC 2078917053 word: a
20 octets received from SSRC 2078917053 word: aa
21 octets received from SSRC 2078917053 word: aal
...
\end{verbatim}


@endlatexonly


@section Review Secure RTP Background

In this section we review SRTP and introduce some terms that are used
in libSRTP.  An RTP session is defined by a pair of destination
transport addresses, that is, a network address plus a pair of UDP
ports for RTP and RTCP.  RTCP, the RTP control protocol, is used to
coordinate between the participants in an RTP session, e.g. to provide
feedback from receivers to senders.  An @e SRTP @e session is
similarly defined; it is just an RTP session for which the SRTP
profile is being used.  An SRTP session consists of the traffic sent
to the SRTP or SRTCP destination transport addresses.  Each
participant in a session is identified by a synchronization source
(SSRC) identifier.  Some participants may not send any SRTP traffic;
they are called receivers, even though they send out SRTCP traffic,
such as receiver reports.

RTP allows multiple sources to send RTP and RTCP traffic during the
same session.  The synchronization source identifier (SSRC) is used to
distinguish these sources.  In libSRTP, we call the SRTP and SRTCP
traffic from a particular source a @e stream.  Each stream has its own
SSRC, sequence number, rollover counter, and other data.  A particular
choice of options, cryptographic mechanisms, and keys is called a @e
policy.  Each stream within a session can have a distinct policy
applied to it.  A session policy is a collection of stream policies.

A single policy can be used for all of the streams in a given session,
though the case in which a single @e key is shared across multiple
streams requires care.  When key sharing is used, the SSRC values that
identify the streams @b must be distinct.  This requirement can be
enforced by using the convention that each SRTP and SRTCP key is used
for encryption by only a single sender.  In other words, the key is
shared only across streams that originate from a particular device (of
course, other SRTP participants will need to use the key for
decryption).  libSRTP supports this enforcement by detecting the case
in which a key is used for both inbound and outbound data.


@section Overview libSRTP Overview

libSRTP provides functions for protecting RTP and RTCP.  RTP packets
can be encrypted and authenticated (using the srtp_protect()
function), turning them into SRTP packets.  Similarly, SRTP packets
can be decrypted and have their authentication verified (using the
srtp_unprotect() function), turning them into RTP packets.  Similar
functions apply security to RTCP packets.

The typedef srtp_stream_t points to a structure holding all of the
state associated with an SRTP stream, including the keys and
parameters for cipher and message authentication functions and the
anti-replay data.  A particular srtp_stream_t holds the information
needed to protect a particular RTP and RTCP stream.  This datatype
is intentionally opaque in order to better seperate the libSRTP
API from its implementation.

Within an SRTP session, there can be multiple streams, each
originating from a particular sender.  Each source uses a distinct
stream context to protect the RTP and RTCP stream that it is
originating.  The typedef srtp_t points to a structure holding all of
the state associated with an SRTP session.  There can be multiple
stream contexts associated with a single srtp_t.  A stream context
cannot exist indepent from an srtp_t, though of course an srtp_t can
be created that contains only a single stream context.  A device
participating in an SRTP session must have a stream context for each
source in that session, so that it can process the data that it
receives from each sender.


In libSRTP, a session is created using the function srtp_create().
The policy to be implemented in the session is passed into this
function as an srtp_policy_t structure.  A single one of these
structures describes the policy of a single stream.  These structures
can also be linked together to form an entire session policy.  A linked
list of srtp_policy_t structures is equivalent to a session policy.  
In such a policy, we refer to a single srtp_policy_t as an @e element.

An srtp_policy_t strucutre contains two crypto_policy_t structures
that describe the cryptograhic policies for RTP and RTCP, as well as
the SRTP master key and the SSRC value.  The SSRC describes what to
protect (e.g. which stream), and the crypto_policy_t structures
describe how to protect it.  The key is contained in a policy element
because it simplifies the interface to the library.  In many cases, it
is desirable to use the same cryptographic policies across all of the
streams in a session, but to use a distinct key for each stream.  A
crypto_policy_t structure can be initialized by using either the
crypto_policy_set_rtp_default() or crypto_policy_set_rtcp_default()
functions, which set a crypto policy structure to the default policies
for RTP and RTCP protection, respectively.
				   
@section Example Example Code

This section provides a simple example of how to use libSRTP.  The
example code lacks error checking, but is functional.  Here we assume
that the value ssrc is already set to describe the SSRC of the stream
that we are sending, and that the functions get_rtp_packet() and
send_srtp_packet() are available to us.  The former puts an RTP packet
into the buffer and returns the number of octets written to that
buffer.  The latter sends the RTP packet in the buffer, given the
length as its second argument.

@verbatim
   srtp_t session;   
   srtp_policy_t policy;
   uint8_t key[30];

   // initialize libSRTP 
   srtp_init();                                  

   // set policy to describe a policy for an SRTP stream 
   crypto_policy_set_rtp_default(&policy.rtp);   
   crypto_policy_set_rtcp_default(&policy.rtcp); 
   policy.ssrc = ssrc;                            
   policy.key  = key;
   policy.next = NULL;

   // set key to random value 
   crypto_get_random(key, 30);          

   // allocate and initialize the SRTP session 
   srtp_create(&session, &policy);  
   
   // main loop: get rtp packets, send srtp packets
   while (1) {
      char rtp_buffer[2048];
      unsigned len;

      len = get_rtp_packet(rtp_buffer);
      srtp_protect(session, rtp_buffer, &len);
      send_srtp_packet(rtp_buffer, len);
   }
@endverbatim

@section ISMAcryp ISMA Encryption Support

The Internet Streaming Media Alliance (ISMA) specifies a way 
to pre-encrypt a media file prior to streaming.  This method
is an alternative to SRTP encryption, which is potentially
useful when a particular media file will be streamed
multiple times.  The specification is available online 
at  http://www.isma.tv/specreq.nsf/SpecRequest.

libSRTP provides the encryption and decryption functions needed for ISMAcryp
in the library @t libaesicm.a, which is included in the default
Makefile target.  This library is used by the MPEG4IP project; see 
http://mpeg4ip.sourceforge.net/.

Note that ISMAcryp does not provide authentication for 
RTP nor RTCP, nor confidentiality for RTCP.  
ISMAcryp RECOMMENDS the use of SRTP message authentication for ISMAcryp
streams while using ISMAcryp encryption to protect the media itself.


 */