README

TLS module

Peter Griffiths

   unknown

Klaus Darilion

   enum.at

Edited by

Klaus Darilion

Edited by

Bogdan-Andrei Iancu

Edited by

Cesc Santasusana

Edited by

Klaus Darilion

Edited by

Christian Lahme

Edited by

Ionut-Razvan Ionita

   Copyright © 2005 Voice Sistem SRL

   Copyright © 2005 Cesc Santasusana

   Copyright © 2006 enum.at

   Copyright © 2013 Secusmart GmbH

   Copyright © 2015 OpenSIPS Solutions
     __________________________________________________________

   Table of Contents

   1. Admin Guide

        1.1. Overview
        1.2. History
        1.3. Scenario
        1.4. Dependencies

              1.4.1. OpenSIPS Modules

        1.5. OpenSIPS Exported parameters

              1.5.1. listen=interface
              1.5.2. tls_port (integer)
              1.5.3. tls_crlf_pingpong (integer)
              1.5.4. tls_crlf_drop (integer)
              1.5.5. tls_max_msg_chunks (integer)

   2. Developer Guide

        2.1. TLS_SERVER

              2.1.1. SSL data per connection
              2.1.2. tls_print_errstack
              2.1.3. tls_tcpconn_init
              2.1.4. tls_tcpconn_clean
              2.1.5. tls_blocking_write
              2.1.6. tls_read
              2.1.7. tls_fix_read_conn

   3. Frequently Asked Questions

   List of Examples

   1.1. Set listen variable
   1.2. Set tls_port variable
   1.3. Set tls_crlf_pingpong parameter
   1.4. Set tls_crlf_drop parameter
   1.5. Set tls_max_msg_chunks parameter

Chapter 1. Admin Guide

1.1. Overview

   TLS, as defined in SIP RFC 3261, is a mandatory feature for
   proxies and can be used to secure the SIP signalling on a
   hop-by-hop basis (not end-to-end). TLS works on top of TCP.
   DTLS, or TLS over UDP is already defined by IETF and may become
   available in the future. This module also implements TLS
   related functions to use in the routing script, and exports
   pseudo variables with certificate and TLS parameters.

1.2. History

   The TLS support was originally developed by Peter Griffiths and
   posted as a patch on SER development mailing list. Thanks to
   Cesc Santasusana, several problems were fixed and some
   improvements were added.

   The TLS support was simultaneously added in both projects. In
   SER, the support was committed in a separate “experimental” CVS
   tree, as patch to the main CVS tree. In OpenSIPS, the support
   was integrated directly into the CVS tree, as a built-in
   component, and is part of stable OpenSIPS since release
   >=1.0.0.

1.3. Scenario

   By the increased number of providers the SIP world is
   continuously growing. More users means more calls and more
   calls means a high probability for a user to receive calls from
   totally unknown people or, in the worst case, to receive
   unwanted calls. To prevent this, a defense mechanism must be
   adopted by the SIP provider. Since only the called user is
   fully able to classify a call as being unwanted, the SIP
   server, based on all information regarding the call should
   notify the user about the desirability of the call. Information
   like the caller domain, the received source or the incoming
   protocol can be very useful for a SIP server to establish the
   nature of the call.

   As this information is quite limited, is very improbable for a
   server to be able detect the unwanted calls - there are many
   calls that it cannot predict anything about its status (neutral
   calls). So, instead on alerting the called user about unwanted
   calls, the server can notify the user about calls that are
   considered trusted - calls for which the server is 100% sure
   there are not unwanted.

   So, a trust concept must be defined for SIP servers. Which
   calls are trusted and which are not? A call is trusted if the
   caller can be identify as a trustable user - a user about we
   have reliable information.

   Since all the user from its domain are authenticated (or should
   be), a SIP server can consider all the calls generated by its
   user as trusted. Now we have to extend the trust concept to the
   multi-domain level. A mutual agreement, between several
   domains, can establish a trusting relationship. So, a domain
   (called A) will consider also as trusted calls all the calls
   generated by user from a different domain (called B) and
   vice-versa. But just an agreement is not enough; since the
   authentication information is strictly limited to a domain (a
   domain can authenticate only its own user, not the user from
   other domains), there is still the problem of checking the
   authenticity of the caller - he can impersonate (by a false
   FROM header) a user from a domain that is trusted.

   The answer to this problem is TLS (Transport Layer Security).
   All calls via domain A and domain B will be done via TLS.
   Authentication in origin domain plus TLS transport between
   domains will make the call 100% trusted for the target domain.

   For such a mechanism to work, the following requirements must
   be met:
     * all UA must have set as outbound proxy their home server.
     * all SIP servers must authenticated all the calls generated
       by their own users.
     * all SIP servers must relay the calls generated be their
       user to a trusted domain via TLS.

   Based on this, a server can classify as trusted a call for one
   of its user only if the call is also generated by one of its
   users or is the call is received from a trusted domain ( which
   is equivalent with a call received via TLS). Untrusted call
   will be calls received from users belonging to untrusted
   domains or from users from trusted domains, but whose calls are
   not routed via their home server (so, they are not
   authenticated by there home servers).

   Once the server is able to tell if the call is trusted or not,
   the still open issue is about the mechanism used by server to
   notify the called user about the nature of the incoming call.

   One way to do it is by remotely changing the ringing type of
   the called user's phone. This can be done by inserting special
   header into the INVITE request. Such feature is supported by
   now by several hardphones like CISCO ATA, CISCO 7960 and SNOM.
   This phones can change their ringing tone based on the present
   or content of the "Alert-Info" SIP header as follows:
     * CISCO ATA - it has 4 pre-defined ringing types. The
       Alert-Info header must look like “Alert-info: Bellcore-drX
       EOH"” where X can be between 1 and 4. Note that 1 is the
       phone default ringing tone.
     * CISCO 7960 - it has 2 pre-defined ringing types and the
       possibility of uploading new ones. The “Alert-Info” header
       must look like “Alert-info: X EOH” where X can be whatever
       number. When this header is present, the phones will not
       change the ringing tone, but the ringing pattern. Normally,
       the phone rings like [ring.........ring..........ring]
       where [ring] is the ringing tone; if the header is present,
       the ringing pattern will be
       [ring.ring.........ring.ring........]. So, to be able to
       hear some difference between the two patterns (and not only
       as length), its strongly recommended to have a highly
       asymmetric ringing type (as the pre-defined are not!!).
     * SNOM - The “Alert-Info” header must look like “Alert-info:
       URL EOH"” where URL can be a HTTP URL (for example) from
       where the phone can retrieve a ringing tone.

1.4. Dependencies

1.4. Dependencies

1.4.1. OpenSIPS Modules

   The following modules must be loaded before this module:
     * tls_mgm.

   OpenSIPS TLS v1.0 support requires the following packages:
     * openssl or libssl >= 0.9.6
     * openssl-dev or libssl-dev

   OpenSIPS TLS v1.1/1.2 support requires the following packages:
     * openssl or libssl >= 1.0.1e
     * openssl-dev or libssl-dev

1.5. OpenSIPS Exported parameters

   All these parameters can be used from the opensips.cfg file, to
   configure the behavior of OpenSIPS-TLS.

1.5.1. listen=interface

   Not specific to TLS. Allows to specify the protocol (udp, tcp,
   tls), the IP address and the port where the listening server
   will be.

   Example 1.1. Set listen variable
...
listen = tls:1.2.3.4:5061
...

1.5.2. tls_port (integer)

   Sets the default TLS listening port.

   Default value is 5061.

   Example 1.2. Set tls_port variable
...
modparam("proto_tls", "tls_port", 5062)
...

1.5.3. tls_crlf_pingpong (integer)

   Send CRLF pong (\r\n) to incoming CRLFCRLF ping messages over
   TLS. By default it is enabled (1).

   Default value is 1 (enabled).

   Example 1.3. Set tls_crlf_pingpong parameter
...
modparam("proto_tls", "tls_crlf_pingpong", 0)
...

1.5.4. tls_crlf_drop (integer)

   Drop CRLF (\r\n) ping messages. When this parameter is enabled,
   the TLS layer drops packets that contains a single CRLF
   message. If a CRLFCRLF message is received, it is handled
   according to the tls_crlf_pingpong parameter.

   Default value is 0 (disabled).

   Example 1.4. Set tls_crlf_drop parameter
...
modparam("proto_tls", "tls_crlf_drop", 1)
...

1.5.5. tls_max_msg_chunks (integer)

   The maximum number of chunks that a SIP message is expected to
   arrive via TLS. If a packet is received more fragmented than
   this, the connection is dropped (either the connection is very
   overloaded and this leads to high fragmentation - or we are the
   victim of an ongoing attack where the attacker is sending the
   traffic very fragmented in order to decrease our performance).

   Default value is 4.

   Example 1.5. Set tls_max_msg_chunks parameter
...
modparam("proto_tls", "tls_max_msg_chunks", 8)
...

Chapter 2. Developer Guide

2.1. TLS_SERVER

2.1.1. SSL data per connection

   Each TLS connection, incoming or outgoing, creates an SSL *
   object, where configuration inherited from the SSL_CTX * and
   particular info on that socket are stored. This SSL * structure
   is kept in OpenSIPS as long as the connection is alive, as part
   of the “struct tcp_connection *” object:
...
struct tcp_connection *c;
SSL *ssl;

/*create somehow SSL object*/
c->extra_data = (void *) ssl;
ssl = (SSL *) c->extra_data;
...

2.1.2. tls_print_errstack

   void tls_print_errstack(void);

   Dumps ssl error stack.

2.1.3. tls_tcpconn_init

   int tls_tcpconn_init( struct tcp_connection *c, int fd);

   Called when new tcp connection is accepted

2.1.4. tls_tcpconn_clean

   void tls_tcpconn_clean( struct tcp_connection *c);

   Shuts down the TLS connection.

2.1.5. tls_blocking_write

   size_t tls_blocking_write( struct tcp_connection *c, int fd,
   const char *buf, size_t len);

   Writes a memory chunk in blocking mode (syncron).

2.1.6. tls_read

   size_t tls_read( struct tcp_connection *c);

   Reads from a TLS connection. Return the number of bytes read.

2.1.7. tls_fix_read_conn

   void tls_tcpconn_clean( struct tcp_connection *c);

   Shuts down the TLS connection.

Chapter 3. Frequently Asked Questions

   3.1.

       Where can I post a question about TLS?

       Use one (the most appropriate) of the OpenSIPS mailing lists:
         * User Mailing List -
           http://lists.opensips.org/cgi-bin/mailman/listinfo/users
         * Developer Mailing List -
           http://lists.opensips.org/cgi-bin/mailman/listinfo/devel

       Remember: first at all, check if your question wasn't already
       answered.

   3.2.

       How can I report a bug?

       Accumulate as much as possible information (OpenSIPS version,
       opensips -V output, your OS (uname -a), OpenSIPS logs, network
       dumps, core dump files, configuration file) and send a mail to
       http://lists.opensips.org/cgi-bin/mailman/listinfo/devel

       Also you may try OpenSIPS's bug report web page:
       http://www.opensips.org/pmwiki.php?n=Development.Tracker

   3.3.

       How can I debug ssl/tls problems?

       Increase the log level in opensips.cfg (debug=4) and watch the
       log statements in syslog.

       Install the ssldump utility and start it. This will give you a
       trace of the ssl/tls connections.

   3.4.

       What is the difference between the TLS directory and the TLSOPS
       module directory?

       The code in the TLS directory implements the TLS transport
       layer. The TLSOPS module implements TLS related functions which
       can be used in the routing script.

   3.5.

       Where can I find more about OpenSIPS?

       Take a look at http://www.opensips.org/.

   3.6.

       Where can I post a question about this module?

       First at all check if your question was already answered on one
       of our mailing lists:
         * User Mailing List -
           http://lists.opensips.org/cgi-bin/mailman/listinfo/users
         * Developer Mailing List -
           http://lists.opensips.org/cgi-bin/mailman/listinfo/devel

       E-mails regarding any stable OpenSIPS release should be sent to
       <users@lists.opensips.org> and e-mails regarding development
       versions should be sent to <devel@lists.opensips.org>.

       If you want to keep the mail private, send it to
       <users@lists.opensips.org>.

   3.7.

       How can I report a bug?

       Please follow the guidelines provided at:
       https://github.com/OpenSIPS/opensips/issues.