Andrey Volk 0275ea1eb4 [Build-System] Use $(DefaultPlatformToolset) so it's easier to build using newer Microsoft Visual Studio versions. 4 年 前
..
build d0ead01aad remove generated files from tree, this will require re-bootstrap on updated source 15 年 前
crypto f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
doc f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
include 9b05fabd53 FS-11966: [core] Fix regression after libsrtp upgrade to 2.2.0 resulting in no audio 5 年 前
srtp 148e02bc67 [srtp] scan-build: Fix dead assignments. 4 年 前
test f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
.clang-format f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
.gitignore f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
.travis.yml f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
.update f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
AUTHORS c4b5df5315 solairs port, add new files, still WIP. 18 年 前
CHANGES f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
COPYING bd96911f4f FS-11816 [Build-System] srtp build issue 5 年 前
ChangeLog 965f8b50d5 finish solaris porting for srtp. This should work now, but still needs some more work 18 年 前
INSTALL bd96911f4f FS-11816 [Build-System] srtp build issue 5 年 前
LICENSE 3bcf5b7ff8 FS-9785: upgrade libsrtp to 2.1 7 年 前
Makefile.am f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
NEWS 965f8b50d5 finish solaris porting for srtp. This should work now, but still needs some more work 18 年 前
README a99f06dfc6 sync changes from srtp upstream 10 年 前
README.md f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
config.h_win32vc7 3bcf5b7ff8 FS-9785: upgrade libsrtp to 2.1 7 年 前
config.hw f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
configure.ac f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
configure.gnu 80c7eb85e6 update libsrtp to use openssl 10 年 前
format.sh f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
install-win.bat 3bcf5b7ff8 FS-9785: upgrade libsrtp to 2.1 7 年 前
libsrtp.2017.vcxproj 0275ea1eb4 [Build-System] Use $(DefaultPlatformToolset) so it's easier to build using newer Microsoft Visual Studio versions. 4 年 前
libsrtp2.pc.in 3bcf5b7ff8 FS-9785: upgrade libsrtp to 2.1 7 年 前
srtp.def 3bcf5b7ff8 FS-9785: upgrade libsrtp to 2.1 7 年 前
srtp.sln 3bcf5b7ff8 FS-9785: upgrade libsrtp to 2.1 7 年 前
srtp2.vcxproj f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
srtp2.vcxproj.filters f82321ee0c FS-11816: [Build-System] Update libsrtp to 2.2.0 5 年 前
timing b3d890ef25 add srtp 18 年 前
undos.sh b3d890ef25 add srtp 18 年 前
update.sh 72e2d183c1 update to cvs head srtp 11 年 前

README

Secure RTP (SRTP) Reference Implementation
David A. McGrew
Cisco Systems, Inc.
mcgrew@cisco.com


This package provides an implementation of the Secure Real-time
Transport Protocol (SRTP), the Universal Security Transform (UST), and
a supporting cryptographic kernel. These mechanisms are documented in
the Internet Drafts in the doc/ subdirectory. The SRTP API is
documented in include/srtp.h, and the library is in libsrtp.a (after
compilation). An overview and reference manual is available in
doc/libsrtp.pdf. The PDF documentation is more up to date than this
file.


Installation:

./configure [ options ] # GNU autoconf script
make # or gmake if needed; use GNU make

The configure script accepts the following options:

--help provides a usage summary
--disable-debug compile without the runtime debugging system
--enable-syslog use syslog for error reporting
--disable-stdout use stdout for error reporting
--enable-console use /dev/console for error reporting
--enable-openssl use OpenSSL crypto primitives
--gdoi use GDOI key management (disabled at present)

By default, debugging is enabled and stdout is used for debugging.
You can use the above configure options to have the debugging output
sent to syslog or the system console. Alternatively, you can define
ERR_REPORTING_FILE in 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 Mac OS X (powerpc-apple-darwin1.4),
Cygwin (i686-pc-cygwin), and Sparc (sparc-sun-solaris2.6). Previous
versions have been tested on Linux and OpenBSD on both x86 and sparc
platforms.

A quick tour of this package:

Makefile targets: all, clean, ...
README this file
CHANGES change log
VERSION version number of this package
LICENSE legal details (it's a BSD-like license)
crypto/ciphers/ ciphers (null, aes_icm, ...)
crypto/math/ crypto math routines
crypto/hash/ crypto hashing (hmac, tmmhv2, ...)
crypto/replay/ replay protection
doc/ documentation: rfcs, apis, and suchlike
include/ include files for all code in distribution
srtp/ secure real-time transport protocol implementation
tables/ apps for generating tables (useful in porting)
test/ test drivers


Applications

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

test driver function tested
-------------------------------------------------------------
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 (n.b. not used in libsrtp)
cipher_driver ciphers
auth_driver hash functions

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.

usage: rtpw [-d ]* [-k [-a][-e ][-g]] [-s | -r] dest_ip dest_port
or rtpw -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.

options:

-s (s)rtp sender - causes app to send words

-r (s)rtp receive - causes app to receive words

-k use srtp master 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)
(use 128, 192, or 256 for keysize)

-g use AES-GCM mode (must be used with -e)

-a message authentication
(requires use of -k option as well)

-l list debug modules

-d turn on debugging for module
-i specify input/output file
(instead of using dictionary file)


In order to get random 30-byte values for use as key/salt pairs , you
can use the following bash function to format the output of
/dev/random (where that device is available).

function randhex() {
cat /dev/random | od --read-bytes=32 --width=32 -x | awk '{ print $2 $3 $4 $5 $6 $7 $8 $9 $10 $11 $12 $13 $14 $15 $16 }'
}


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

set k=c1eec3717da76195bb878578790af71c4ee9f859e197a414a78d5abc7451

[sh1]$ test/rtpw -s -k $k -e 128 -a 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
...

[sh2]$ test/rtpw -r -k $k -e 128 -a 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
...

Implementation Notes

* The srtp_protect() function assumes that the buffer holding the
rtp packet has enough storage allocated that the authentication
tag can be written to the end of that packet. If this assumption
is not valid, memory corruption will ensue.

* Automated tests for the crypto functions are provided through
the cipher_type_self_test() and auth_type_self_test() functions.
These functions should be used to test each port of this code
to a new platform.

* Replay protection is contained in the crypto engine, and
tests for it are provided.

* This implementation provides calls to initialize, protect, and
unprotect RTP packets, and makes as few as possible assumptions
about how these functions will be called. For example, the
caller is not expected to provide packets in order (though if
they're called more than 65k out of sequence, synchronization
will be lost).

* The sequence number in the rtp packet is used as the low 16 bits
of the sender's local packet index. Note that RTP will start its
sequence number in a random place, and the SRTP layer just jumps
forward to that number at its first invocation. An earlier
version of this library used initial sequence numbers that are
less than 32,768; this trick is no longer required as the
rdbx_estimate_index(...) function has been made smarter.

* The replay window is 128 bits in length, and is hard-coded to this
value for now.