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- Introduction to libSRTP
- Contact Us
- Contents
- License and Disclaimer
- libSRTP Overview
- Secure RTP Background
- Supported Features
- Implementation Notes
- Installing and Building libSRTP
- Changing Build Configuration
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. 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, 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 section for more detailed information.
•libsrtp@lists.packetizer.com general mailing list for news / announcements / discussions. This is an open list, see https://lists.packetizer.com/mailman/listinfo/libsrtp for singing up.
•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.
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:
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 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.
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.
This library supports all of the mandatory-to-implement features of SRTP (as defined in RFC 3711). 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 library, 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. In addition, it is important that you read and understand the terms outlined in the License and Disclaimer section.
•It is possible to configure which 3rd party (ie openssl/nss/etc) crypto backend libSRTP will be built with. If no 3rd party backend is set then libSRTP provides an internal implementation of AES and Sha1. The internal implementation only supports AES-128 & AES-256, so to use AES-192 or the AES-GCM group of ciphers a 3rd party crypto backend must be configured. For this and performance reasons it is highly recommended to use a 3rd party crypto backend.
•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.
To install libSRTP, download the latest release of the distribution from 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:
The configure script accepts the following options:
By default there is no log output, logging can be enabled to be output to stdout or a given file using the configure options.
To build the ./configure script mentioned above, libSRTP relies on the 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:
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