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SRTP Support in PJSIP (moved)

Moved to: https://docs.pjsip.org/en/latest/specific-guides/security/srtp.html

This article describes about SRTP support in PJSIP. SRTP support is included in version 0.9 (see ticket #61).

Table of Contents

Features
Compatibility Info
1. Build Systems
2. API Changes
Requirements
Building PJSIP with SRTP Support
1. Availability
2. Building
Using SRTP
Implementation Notes

Features

The SRTP functionality in PJSIP has the following features:

SRTP (RFC 3711), using the Open Source libsrtp library.
Keys exchange using Security Descriptions for Media Streams (SDESC, RFC 4568)
Supported cryptos:
- AES_CM_128_HMAC_SHA1_80
- AES_CM_128_HMAC_SHA1_32
Secure RTCP (SRTCP) is supported.

Negotiation of crypto session parameters in SDP is currently not supported.

Compatibility Info

Few notes regarding compatibility with previous PJSIP code.

Build Systems

There is a new third party library in the distribution, namely libsrtp, you will need to add this library into your application's input libraries specification.

For GNU build systems:

You will need to re-run ./configure, make dep and make to update build.mak and rebuild the project dependencies.
If your Makefile includes build.mak (as explained in Getting Started part 2), you just need to rebuild your application as the input libraries will be updated automatically (by build.mak).
If you maintain your own independent Makefile, please add libsrtp-$(TARGET) from third_party/lib directory to your input libraries.

For Visual Studio 6 and 2005:

New libsrtp project has been added into pjproject.dsw and pjproject-vs8.sln workspace.
If you maintain your own application workspace, you need to add libsrtp project into your application. The libsrtp project files are in third_party/build/srtp directory.

For Windows Mobile developments:

New libsrtp project has been added into wince_demos.vcw workspace (the workspace is located in pjsip-apps/build/wince-evc4 directory).
If you maintain your own application workspace, you need to add libsrtp.vcp project into your application. The libsrtp project files are in third_party/build/srtp directory.
If you're using Visual Studio rather than Embedded Visual C++ 4, make sure you import the Embedded Visual Studio project (libsrtp.vcp) rather than the Visual Studio (libsrtp.vcproj) project!

Symbian:

Updated 2009/01/15:

The libsrtp.mmp has been added to bld.inf on June '08 for version 0.9. It should be shown in your Carbide or CodeWarrior workspace when you import the bld.inf.

API Changes

Changes:

New callbacks have been added in media transport interface which may break your application, depending on the API level that you use. The changes will be explained in Changes in Media Transport Interface section below.
Removed const specification from media transport's RTP and RTCP callback to allow packets to be modified in place.

If you are working directly with PJMEDIA transport API, there may be changes required in your code.

If you are working on PJSUA-LIB API level, there should be no changes in your application.

Requirements

SRTP feature in PJSIP uses the Open Source libsrtp library created by David A. McGrew of Cisco Systems, Inc. Copy of libsrtp is included in PJSIP source tree in third_party/srtp directory. There is no other software to download.

libsrtp is distributed under BSD-like license, you must satisfy the license requirements if you incorporate SRTP in your application. Please see PJSIP licensing page for more information about this and other third party libraries licensing in PJSIP.

Building PJSIP with SRTP Support

Availability

SRTP feature is currently available in:

Visual C++ 6 and 2005 (for Windows targets)
GNU based build system (for Linux, including uC-Linux for embedded systems, Mingw, MacOS X, and *nix based platforms)
Windows Mobile targets
Symbian targets

Building

libsrtp is always built by default, from third_party/build/srtp directory.

Support for SRTP is enabled by default in PJMEDIA and PJSUA-LIB. To disable this feature, declare this in your config_site.h:

#define PJMEDIA_HAS_SRTP  0

Using SRTP

SRTP is implemented as media transport in PJMEDIA. In the high level PJSUA-LIB API, the use of SRTP is controlled by couple of settings as explained below.

Using SRTP in PJSUA-LIB

In PJSUA-LIB, the use of SRTP is controlled by settings in both pjsua_config and pjsua_acc_config. The settings in pjsua_config specify the default settings for all accounts, and the settings in pjsua_acc_config can be used to further set the behavior for that specific account.

In both pjsua_config and pjsua_acc_config, there are two configuration items related to SRTP:

use_srtp

This option controls whether secure media transport (SRTP) should be used for this account. Valid values are:

PJMEDIA_SRTP_DISABLED (0): SRTP is disabled, and incoming call with RTP/SAVP transport will be rejected with 488/Not Acceptable Here response.
PJMEDIA_SRTP_OPTIONAL (1): SRTP will be advertised and SRTP will be used if remote supports it, but the call may fall back to unsecure media. Incoming call with RTP/SAVP is accepted and responded with RTP/SAVP too.
PJMEDIA_SRTP_MANDATORY (2): secure media is mandatory, and the call can only proceed if secure media can be established.

The default value for this option is PJSUA_DEFAULT_USE_SRTP, which is zero (disabled).

srtp_secure_signaling

This option controls whether SRTP requires secure signaling to be used. This option is only used when use_srtp option above is non-zero. Valid values are:

0: SRTP does not require secure signaling (not recommended)
1: SRTP requires secure transport such as TLS to be used.
2: SRTP requires secure end-to-end transport (sips: URI scheme) to be used.

The default value for this option is PJSUA_DEFAULT_SRTP_SECURE_SIGNALING, which is 1 (require TLS transport).

pjsua

Two new options were added to pjsua:

--use-srtp=N

This corresponds to use_srtp setting above. Valid values are 0, 1, or 2. Default value is 0.

--srtp-secure=N

This corresponds to srtp_secure_signaling setting above. Valid values are 0, 1, or 2. Default value is 1.

Sample usage:

 $ ./pjsua --use-tls --use-srtp=1 sip:alice@example.com;transport=tls

Using SRTP Transport Directly

The SRTP transport may also be used directly without having to involve SDP negotiations (for example, to use SRTP without SIP). Please see streamutil from the samples collection for a sample application. For this to work, you will need to have a different mechanism to exchange keys between endpoints.

To use SRTP transport directly:

Call pjmedia_transport_srtp_create() to create the SRTP adapter, giving it the actual media transport instance (such as UDP transport).
Call pjmedia_transport_srtp_start() to active SRTP session, giving it both local and remote crypto settings and keys.
Call pjmedia_transport_attach() to configure the remote RTP/RTCP addresses and attach your RTP and RTCP callbacks.
Call pjmedia_transport_send_rtp() and pjmedia_transport_send_rtcp() to send RTP/RTCP packets.
Once you done with your session, call pjmedia_transport_close() to destroy the SRTP adapter (and optionally the actual transport which is attached to the SRTP adapter, depending on whether close_member_tp flag is set in the pjmedia_srtp_setting when creating the SRTP adapter).

Implementation Notes

Changes in Media Transport Interface

Since the availability of SRTP changes SDP (Session Description Protocol) contents and the SDP negotiation, we needed to modify/add new interfaces in PJMEDIA transport API to allow media transport to modify and negotiate SDP. Incidently this would work well with ICE too (previously we treat ICE as a special kind of media transport so it is treated differently, but with this new interfaces, all media transports will behave uniformly (anyway that's what API abstraction is for!)).

New interfaces in media transport are as follows (please consult the PJMEDIA transport documentation for more info):

media_create(): This callback is called by application (or PJSUA-LIB) to allow the media transport to add more information in the SDP offer, before the offer is sent to remote. Additionally, for answerer side, this callback allows the media transport to reject the offer before this offer is processed by the SDP negotiator.

media_start(): This callback is called after offer and answer are negotiated, and both SDPs are available, and before the media is started. For answerer side, this callback will be called before the answer is sent to remote, to allow media transport to put additional info in the SDP. For offerer side, this callback will be called after SDP answer is received. In this callback, the media transport has the final chance to negotiate/validate the offer and answer before media is really started (and answer is sent, for answerer side).

media_stop(): This callback is called when the media is stopped, to allow the media transport to release its resources.

simulate_lost(): This has nothing to do with SRTP, but since all media transports support this feature (packet loss simulation), we added this as a new interface.

pjmedia_transport_srtp Implementation

As we know, media transport is separated from the stream object (which does the encoding/decoding of PCM frames, (de)packetization of RTP/RTCP packets, and de-jitter buffering). The connection between stream and media transport is established when the stream is created (we need to specify media transport during stream creation), and the interconnection can be depicted from the diagram below:

I think the diagram above is self-explanatory.

SRTP functionality is implemented as some kind of "adapter", which is plugged between the stream and the actual media transport that does sending/receiving RTP/RTCP packets. When SRTP is used, the interconnection between stream and transport is like the diagram below:

So to stream, the SRTP transport behaves as if it is a media transport (because it is a media transport), and to the media transport it behaves as if it is a stream. The SRTP object then forwards RTP packets back and forth between stream and the actual transport, encrypting/decrypting the RTP/RTCP packets as necessary.

The neat thing about this design is the SRTP "adapter" then can be used to encrypt any kind of media transports. We currently have UDP and ICE media transports that can benefit SRTP, and we could add SRTP to any media transports that will be added in the future.

AES-GCM support

Pjsip 2.6 enabled the support for AES-GCM (#1943), however the bundled libSRTP (1.5.4) at that time has compatibility issue with OpenSSL 1.1.0. Updating the libSRTP was done in #1993, included in 2.7.

As an alternative to the bundled libSRTP, users are also allowed to use external libSRTP by specifying --with-external-srtp. Using #2050, it's been tested to work with external libSRTP 1.5.4 and 2.1.0. Note about this option, using libSRTP with AES-GCM would also require the user to enable building pjsip with ssl.