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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. 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.

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.

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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.


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). 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. In addition, it is important that you read and understand the terms outlined in the 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 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:

./configure [ options ]

The configure script accepts the following options:

--help -hDisplay help
--enable-debug-loggingEnable debug logging in all modules
--enable-log-stdoutEnable logging to stdout
--enable-opensslEnable OpenSSL crypto engine
--enable-openssl-kdfEnable OpenSSL KDF algorithm
--with-log-fileUse file for logging
--with-openssl-dirLocation 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).


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

Test driverFunction tested
kernel_drivercrypto kernel (ciphers, auth funcs, rng)
srtp_driversrtp in-memory tests (does not use the network)
rdbx_driverrdbx (extended replay database)
roc_driverextended sequence number functions
replay_driverreplay database
auth_driverhash 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.


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:

-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)
-guse AES-GCM mode (must be used with -e)
-amessage authentication (requires use of -k option as well)
-llist 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).

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 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 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.

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

// set policy to describe a policy for an SRTP stream
policy.ssrc = ssrc;
policy.key  = key; = 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);


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.


SRTP and ICM References September, 2005

Secure RTP is defined in RFC 3711. The counter mode definition is in Section 4.1.1.

SHA-1 is defined in FIPS PUB 180-4.

HMAC is defined in RFC 2104 and HMAC-SHA1 test vectors are available in RFC 2202.

AES-GCM usage in SRTP is defined in RFC 7714