[![Build Status](https://travis-ci.org/cisco/libsrtp.svg?branch=master)](https://travis-ci.org/cisco/libsrtp) [![Coverity Scan Build Status](https://scan.coverity.com/projects/14274/badge.svg)](https://scan.coverity.com/projects/cisco-libsrtp) # 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. -------------------------------------------------------------------------------- # 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. -------------------------------------------------------------------------------- ## Contents - [Introduction to libSRTP](#introduction-to-libsrtp) - [Contact Us](#contact) - [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) - [Applications](#applications) - [Example Code](#example-code) - [Credits](#credits) - [References](#references) -------------------------------------------------------------------------------- # 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. -------------------------------------------------------------------------------- # 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` 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. -------------------------------------------------------------------------------- ## 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. -------------------------------------------------------------------------------- ## 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. -------------------------------------------------------------------------------- ## 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. -------------------------------------------------------------------------------- # 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). -------------------------------------------------------------------------------- ## 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 ``` -------------------------------------------------------------------------------- # 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 ]* [-k|b [-a][-e ][-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 | use SRTP master key , where the key is a hexadecimal (without the leading "0x") -b | same as -k but with base64 encoded key -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 the available debug modules -d | turn on debugging for module 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 ... ~~~ -------------------------------------------------------------------------------- ## 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 crypto_policy_set_rtp_default(&policy.rtp); 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); } ~~~ -------------------------------------------------------------------------------- # 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. -------------------------------------------------------------------------------- # 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)