You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
509 lines
22 KiB
509 lines
22 KiB
[](https://travis-ci.org/cisco/libsrtp)
|
|
[](https://scan.coverity.com/projects/cisco-libsrtp)
|
|
|
|
<a name="introduction-to-libsrtp"></a>
|
|
# Introduction to libSRTP
|
|
|
|
This package provides an implementation of the Secure Real-time
|
|
Transport Protocol (SRTP), the Universal Security Transform (UST), and
|
|
a supporting cryptographic kernel. The SRTP API is documented in include/srtp.h,
|
|
and the library is in libsrtp2.a (after compilation).
|
|
|
|
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](https://www.ietf.org/rfc/rfc3550.txt).
|
|
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 Standard,
|
|
defined in [RFC 3711](https://www.ietf.org/rfc/rfc3711.txt), 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 [Supported Features](#supported-features) section for more detailed information.
|
|
|
|
This document is also used to generate the documentation files in the /doc/
|
|
folder where a more detailed reference to the libSRTP API and related functions
|
|
can be created (requires installing doxygen.). The reference material is created
|
|
automatically 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.
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="contact-us"></a>
|
|
# Contact Us
|
|
|
|
- [libsrtp@lists.packetizer.com](mailto:libsrtp@lists.packetizer.com) general mailing list for news / announcements / discussions. This is an open list, see
|
|
[https://lists.packetizer.com/mailman/listinfo/libsrtp](https://lists.packetizer.com/mailman/listinfo/libsrtp) for singing up.
|
|
|
|
- [libsrtp-security@lists.packetizer.com](mailto:libsrtp-security@lists.packetizer.com) for disclosing security issues to the libsrtp maintenance team. This is a closed list but anyone can send to it.
|
|
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="contents"></a>
|
|
## Contents
|
|
|
|
- [Introduction to libSRTP](#introduction-to-libsrtp)
|
|
- [Contact Us](#contact-us)
|
|
- [Contents](#contents)
|
|
- [License and Disclaimer](#license-and-disclaimer)
|
|
- [libSRTP Overview](#libsrtp-overview)
|
|
- [Secure RTP Background](#secure-rtp-background)
|
|
- [Supported Features](#supported-features)
|
|
- [Implementation Notes](#implementation-notes)
|
|
- [Installing and Building libSRTP](#installing-and-building-libsrtp)
|
|
- [Changing Build Configuration](#changing-build-configuration)
|
|
- [Using Visual Studio](#using-visual-studio)
|
|
- [Applications](#applications)
|
|
- [Example Code](#example-code)
|
|
- [Credits](#credits)
|
|
- [References](#references)
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="license-and-disclaimer"></a>
|
|
# 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.
|
|
|
|
> Copyright (c) 2001-2017 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:
|
|
>
|
|
> - Redistributions of source code must retain the above copyright
|
|
> notice, this list of conditions and the following disclaimer.
|
|
> - 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.
|
|
> - 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.
|
|
>
|
|
> 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.
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="libsrtp-overview"></a>
|
|
# 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 *element*.
|
|
|
|
An `srtp_policy_t` structure contains two `srtp_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 `srtp_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
|
|
`srtp_crypto_policy_t` structure can be initialized by using either the
|
|
`srtp_crypto_policy_set_rtp_default()` or `srtp_crypto_policy_set_rtcp_default()`
|
|
functions, which set a crypto policy structure to the default policies
|
|
for RTP and RTCP protection, respectively.
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="secure-rtp-background"></a>
|
|
## 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 *SRTP 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 *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
|
|
*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 *key* is shared across multiple
|
|
streams requires care. When key sharing is used, the SSRC values that
|
|
identify the streams **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.
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="supported-features"></a>
|
|
## Supported Features
|
|
|
|
This library supports all of the mandatory-to-implement features of
|
|
SRTP (as defined in [RFC 3711](https://www.ietf.org/rfc/rfc3711.txt)). 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 SRTPevents
|
|
section in the generated documentation.
|
|
|
|
Some options that are described in the SRTP specification are not
|
|
supported. This includes
|
|
|
|
- key derivation rates other than zero,
|
|
- the cipher F8,
|
|
- the use of the packet index to select between master keys.
|
|
|
|
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 [RFC 3711](https://www.ietf.org/rfc/rfc3711.txt).
|
|
In addition, it is important that you read and understand the
|
|
terms outlined in the [License and Disclaimer](#license-and-disclaimer) section.
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="implementation-notes"></a>
|
|
## 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 for (S)RTCP is hardcoded to 128 bits in length.
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="installing-and-building-libsrtp"></a>
|
|
# Installing and Building libSRTP
|
|
|
|
To install libSRTP, download the latest release of the distribution
|
|
from [https://github.com/cisco/libsrtp/releases](https://github.com/cisco/libsrtp/releases).
|
|
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 `libsrtp-A-B-C` where `A` is the version number, `B` is the
|
|
major release number and `C` is the minor release number.
|
|
|
|
libSRTP uses the GNU `autoconf` and `make` utilities (BSD make will not work; if
|
|
both versions of make are on your platform, you can invoke GNU make as
|
|
`gmake`.). In the `libsrtp` directory, run the configure script and then
|
|
make:
|
|
|
|
~~~.txt
|
|
./configure [ options ]
|
|
make
|
|
~~~
|
|
|
|
The configure script accepts the following options:
|
|
|
|
Option | Description
|
|
-------------------------------|--------------------
|
|
\-\-help \-h | Display help
|
|
\-\-enable-debug-logging | Enable debug logging in all modules
|
|
\-\-enable-log-stdout | Enable logging to stdout
|
|
\-\-enable-openssl | Enable OpenSSL crypto engine
|
|
\-\-enable-openssl-kdf | Enable OpenSSL KDF algorithm
|
|
\-\-with-log-file | Use file for logging
|
|
\-\-with-openssl-dir | Location of OpenSSL installation
|
|
|
|
By default there is no log output, logging can be enabled to be output to stdout
|
|
or a given file using the configure options.
|
|
|
|
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).
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="changing-build-configuration"></a>
|
|
## Changing Build Configuration
|
|
|
|
To build the `./configure` script mentioned above, libSRTP relies on the
|
|
[automake](https://www.gnu.org/software/automake/) toolchain. Since
|
|
`./configure` is built from `configure.in` by automake, if you make changes in
|
|
how `./configure` works (e.g., to add a new library dependency), you will need
|
|
to rebuild `./configure` and commit the updated version. In addition to
|
|
automake itself, you will need to have the `pkgconfig` tools installed as well.
|
|
|
|
For example, on macOS:
|
|
|
|
```
|
|
brew install automake pkgconfig
|
|
# Edit configure.in
|
|
autoremake -ivf
|
|
```
|
|
|
|
--------------------------------------------------------------------------------
|
|
<a name="using-visual-studio"></a>
|
|
## Using Visual Studio
|
|
|
|
On Windows one can use Visual Studio via CMake. CMake can be downloaded here:
|
|
https://cmake.org/ . To create Visual Studio build files, for example run the
|
|
following commands:
|
|
|
|
```
|
|
# Create build subdirectory
|
|
mkdir build
|
|
cd build
|
|
|
|
# Make project files
|
|
cmake .. -G "Visual Studio 15 2017"
|
|
|
|
# Or for 64 bit project files
|
|
cmake .. -G "Visual Studio 15 2017 Win64"
|
|
```
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="applications"></a>
|
|
# 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
|
|
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:
|
|
~~~.txt
|
|
rtpw [[-d <debug>]* [-k|b <key> [-a][-e <key size>][-g]] [-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:
|
|
|
|
Option | Description
|
|
--------- | -------
|
|
-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 (without the leading "0x")
|
|
-b <key> | same as -k but with base64 encoded key
|
|
-e <keysize> | 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 the available debug modules
|
|
-d <debug> | turn on debugging for module <debug>
|
|
|
|
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).
|
|
|
|
~~~.txt
|
|
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 $16 }'
|
|
}
|
|
~~~
|
|
|
|
An example of an SRTP session using two rtpw programs follows:
|
|
|
|
~~~.txt
|
|
set k=c1eec3717da76195bb878578790af71c4ee9f859e197a414a78d5abc7451
|
|
|
|
[sh1]$ 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
|
|
...
|
|
~~~
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="example-code"></a>
|
|
## 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.
|
|
|
|
~~~.c
|
|
srtp_t session;
|
|
srtp_policy_t policy;
|
|
|
|
// Set key to predetermined value
|
|
uint8_t key[30] = {0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07,
|
|
0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F,
|
|
0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17,
|
|
0x18, 0x19, 0x1A, 0x1B, 0x1C, 0x1D};
|
|
|
|
// initialize libSRTP
|
|
srtp_init();
|
|
|
|
// default policy values
|
|
memset(&policy, 0x0, sizeof(srtp_policy_t));
|
|
|
|
// set policy to describe a policy for an SRTP stream
|
|
srtp_crypto_policy_set_rtp_default(&policy.rtp);
|
|
srtp_crypto_policy_set_rtcp_default(&policy.rtcp);
|
|
policy.ssrc = ssrc;
|
|
policy.key = key;
|
|
policy.next = NULL;
|
|
|
|
// 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);
|
|
}
|
|
~~~
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="credits"></a>
|
|
# Credits
|
|
|
|
The original implementation and documentation of libSRTP was written
|
|
by David McGrew of Cisco Systems, Inc. in order to promote the use,
|
|
understanding, and interoperability of Secure RTP. Michael Jerris
|
|
contributed support for building under MSVC. Andris Pavenis
|
|
contributed many important fixes. Brian West contributed changes to
|
|
enable dynamic linking. Yves Shumann reported documentation bugs.
|
|
Randell Jesup contributed a working SRTCP implementation and other
|
|
fixes. Steve Underwood contributed x86_64 portability changes. We also give
|
|
thanks to Fredrik Thulin, Brian Weis, Mark Baugher, Jeff Chan, Bill
|
|
Simon, Douglas Smith, Bill May, Richard Preistley, Joe Tardo and
|
|
others for contributions, comments, and corrections.
|
|
|
|
This reference material, when applicable, in this documenation was generated
|
|
using the doxygen utility for automatic documentation of source code.
|
|
|
|
Copyright 2001-2005 by David A. McGrew, Cisco Systems, Inc.
|
|
|
|
--------------------------------------------------------------------------------
|
|
|
|
<a name="references"></a>
|
|
# References
|
|
|
|
SRTP and ICM References
|
|
September, 2005
|
|
|
|
Secure RTP is defined in [RFC 3711](https://www.ietf.org/rfc/rfc3711.txt).
|
|
The counter mode definition is in Section 4.1.1.
|
|
|
|
SHA-1 is defined in [FIPS PUB 180-4](http://nvlpubs.nist.gov/nistpubs/FIPS/NIST.FIPS.180-4.pdf).
|
|
|
|
HMAC is defined in [RFC 2104](https://www.ietf.org/rfc/rfc2104.txt)
|
|
and HMAC-SHA1 test vectors are available
|
|
in [RFC 2202](https://www.ietf.org/rfc/rfc2202.txt).
|
|
|
|
AES-GCM usage in SRTP is defined in [RFC 7714](https://www.ietf.org/html/rfc7714)
|