ChangeLog: Updated for 14.0.0-beta2

Release summaries: Add summaries for 14.0.0-beta2
Release summaries: Remove previous versions
2025-10-19 03:07:59 +00:00 · 2016-08-29 07:30:23 -05:00 · 2016-08-29 07:29:55 -05:00 · 2016-08-29 07:29:29 -05:00 · 2016-08-29 07:29:29 -05:00 · 2016-08-29 07:29:29 -05:00
33 changed files with 1460 additions and 15045 deletions
--- a/.version
+++ b/.version
@@ -1 +1 @@
-14.0.0-beta1
+14.0.0-beta2
--- a/11
+++ b/11
@@ -243,12 +243,11 @@ Core
   - 'media cache delete <item>' - remove an item from the cache
   - 'media cache create <uri>' - retrieve a URI and store it in the cache

- * The ability for hints to be automatically created as a result of device state
-   changes now exists in the PBX. This functionality is referred to as "autohints"
-   and is configurable in extensions.conf by placing "autohints=yes" in the
-   context. If enabled then a hint will be automatically created with the name of
-   the device.
-
+ * The ability for device state hints to be automatically created as a result of
+   device state changes now exists in the PBX. This functionality is referred to
+   as "autohints" and is configurable in extensions.conf by placing "autohints=yes"
+   in the context. If enabled a device state hint will be automatically created
+   with the name of the device.

 Functions
 ------------------
--- a/198
+++ b/198
@@ -1,3 +1,201 @@
+2016-08-29 12:30 +0000  Asterisk Development Team <asteriskteam@digium.com>
+
+	* asterisk 14.0.0-beta2 Released.
+
+2016-08-29 07:29 +0000 [9cdf44668d]  Joshua Colp <jcolp@digium.com>
+
+	* Release summaries: Add summaries for 14.0.0-beta2
+
+2016-08-29 07:29 +0000 [73d39f2029]  Joshua Colp <jcolp@digium.com>
+
+	* Release summaries: Remove previous versions
+
+2016-08-29 07:29 +0000 [e8a97775ee]  Joshua Colp <jcolp@digium.com>
+
+	* .version: Update for 14.0.0-beta2
+
+2016-08-29 07:29 +0000 [345409825a]  Joshua Colp <jcolp@digium.com>
+
+	* .lastclean: Update for 14.0.0-beta2
+
+2016-08-29 07:29 +0000 [105c1168f7]  Joshua Colp <jcolp@digium.com>
+
+	* realtime: Add database scripts for 14.0.0-beta2
+
+2016-08-29 06:31 +0000 [8927b52634]  Joshua Colp <jcolp@digium.com>
+
+	* alembic: Fix downgrade path.
+
+	  The 3772f8f828da version was referencing a previous version
+	  that did not exist in the 14.0 branch. It has been fixed to
+	  reference the correct previous version.
+
+	  Change-Id: I004d0fcfdfe1d1bb6f01c6dac2b69f6b1f40ae51
+
+2016-08-11 12:18 +0000 [9a95c6dea3]  gtjoseph <gjoseph@digium.com>
+
+	* res_pjsip:  Fail global load if debug or default_from_user are empty
+
+	  If debug was specified in the global configuration but left blank,
+	  the logger would treat it as a wildcard and log all hosts.  If
+	  default_from_user was empty, a crash would result.
+
+	  The global apply handler now checks for empty strings.
+
+	  ASTERISK-26239 #close
+	  ASTERISK-26238 #close
+
+	  Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336
+
+2016-08-11 11:24 +0000 [aaee8160bc]  gtjoseph <gjoseph@digium.com>
+
+	* res_pjsip_caller_id:  Copy header name to short header name
+
+	  When compact_headers was set, we were sending a zero-length header name
+	  for PAI and RPID because we always forced the short header name length
+	  to 0.  We did this because we cloned the header from "From" and wanted
+	  to clear "f" from the sname.  By cloning however, we bypass pjproject's
+	  automatic logic that sets sname to name if there's no compact form of
+	  the header, which there isn't for PAI and RPID.  So now we force sname
+	  to be the same as name right after we set name.
+
+	  res_pjsip_diversion needed the same treatment for the Diversion header.
+
+	  ASTERISK-26241 #close
+
+	  Change-Id: I633ec139630cd83809aae00336cee4a10077e467
+
+2016-08-11 12:01 +0000 [7af0eac02a]  gtjoseph <gjoseph@digium.com>
+
+	* autohints:  Update CHANGES and extensions.conf.sample
+
+	  Make it clear that we're talking about device state hints and add
+	  an entry to the sample config.
+
+	  Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433
+
+2016-08-11 10:50 +0000 [ef0bf47bb3]  Kevin Harwell <kharwell@digium.com>
+
+	* alembic: add auth_username to endpoint's identify_by enum
+
+	  A new identify_by option was added recently, auth_username. However, this
+	  setting was not added as an allowable choice in the database enumeration
+	  value.
+
+	  This patch updates the current enumeration, adding in the new setting.
+
+	  ASTERISK-26268 #close
+
+	  Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8
+
+2016-08-08 14:50 +0000 [a1d6b14c40]  Richard Mudgett <rmudgett@digium.com>
+
+	* res_srtp: Move SDP SRTP code from the core to res_srtp.
+
+	  A patch made to the master branch (Now the 14 branch) inadvertently made
+	  libsrtp a required dependency in order to compile Asterisk.  Rather than
+	  create dummy defines to substitute for the defines supplied by libsrtp
+	  when libsrtp is not available, most of the code in sdp_srtp.c is moved
+	  into res_srtp.c.  This gets more code out of Asterisk's core that isn't
+	  used when SRTP is not available.  This also makes another inadvertent
+	  required dependency on libsrtp by Asterisk's core unlikely.
+
+	  ASTERISK-26253 #close
+	  Reported by: Ben Merrills
+
+	  Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7
+
+2016-08-09 12:07 +0000 [a783e1e60d]  Kevin Harwell <kharwell@digium.com>
+
+	* alembic/sqlalchemy: auto increment only allowed on a single column
+
+	  The extensions table defined two columns (id and priority) as primary key
+	  autoincrement columns. However only one is allowed when defining the primary
+	  key.
+
+	  This patch removes the autoincrement attribute from the priority column since
+	  it does not need to be as such and really should not have been on there in the
+	  first place.
+
+	  This patch also removes 'context', 'exten', and 'priority' from the primary key
+	  index and creates a new combined unique contraint index on them.
+
+	  ASTERISK-26183 #close
+
+	  Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b
+
+2016-08-10 11:47 +0000 [9c56f798f6]  gtjoseph <gjoseph@digium.com>
+
+	* res_resolver_unbound:  Allow compilation with libunbound version < 1.5
+
+	  libunbound at version 1.4.20 (which CentOS still uses) declared all
+	  of their string function parameters as as 'char *'.  1.4.21 changed
+	  them all to 'const char *'.  Thankfully 1.4.21 also introduced the
+	  UNBOUND_VERSION_MAJOR define so configure now checks for that and
+	  sets HAVE_UNBOUND_CONST_PARAMS.  res_resolver_unbound then checks
+	  that and casts away the 'const' if it's not set.
+
+	  Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and
+	  Fedora24 (1.5.4).  There are a few failing tests to be addressed though.
+
+	  ASTERISK-26283 #close
+
+	  Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148
+
+2016-08-01 16:13 +0000 [1ad00c1c30]  gtjoseph <gjoseph@digium.com>
+
+	* menuselect:  Add an opaque "member_data" string to the acceptable xml
+
+	  Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe
+
+2016-07-17 18:28 +0000 [815b6f72f8]  gtjoseph <gjoseph@digium.com>
+
+	* pjproject_bundled:  Update for pjproject 2.5.5
+
+	  Add more --disable-* switches to Makefile.rules including
+	  --disable-opus which was causing bundled pjproject to fail with
+	  "undefined reference" errors in libasteriskpj.
+
+	  Changed PJ_ENABLE_EXTRA_CHECK to 1.
+
+	  Removed 2 obsolete patches and added a new one.
+	  The new one was merged by Teluu on 6/27/2016.
+
+	  ASTERISK-26148 #close
+
+	  Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063
+	  (cherry picked from commit 4cf02b5584ce33bb0a64408c27bf20c19bc4ce13)
+
+2016-07-29 13:13 +0000 [c95b611a73]  Mark Michelson <mmichelson@digium.com>
+
+	* Remove SILK payload mappings from Asterisk core.
+
+	  SILK is a bit of a hog when it comes to using up our limited number of
+	  dynamic payload types in the RTP engine. By freeing up four slots, it
+	  allows for other codecs to potentially take the place.
+
+	  Now, codec_silk.so will dynamically use the payload slots in the RTP
+	  engine when it loads.
+
+	  A better fix would be make RTP dynamic payload types actually
+	  dynamic. However, at this stage of Asterisk 14 development, this is a
+	  risky move that would be imprudent.
+
+	  Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612
+
+2016-07-27 12:36 +0000 [bc94ccbcdd]  Kevin Harwell <kharwell@digium.com>
+
+	* rtp_engine: Failed assertion and wrong name given for codec
+
+	  Fixed an assert check that would trigger when the passed in value was negative.
+	  The negative value was being cast to an unsigned value. This resulted in the
+	  check failing.
+
+	  Also fixed another problem when loading formats in the engine. When setting the
+	  mime type the format's name was being passed in instead of the codec's name.
+
+	  Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c
+
 2016-07-26 23:19 +0000  Asterisk Development Team <asteriskteam@digium.com>

 	* asterisk 14.0.0-beta1 Released.
--- a/asterisk-14.0.0-beta1-summary.html
+++ b/asterisk-14.0.0-beta1-summary.html
--- a/asterisk-14.0.0-beta1-summary.txt
+++ b/asterisk-14.0.0-beta1-summary.txt
--- a/asterisk-14.0.0-beta2-summary.html
+++ b/asterisk-14.0.0-beta2-summary.html
@@ -0,0 +1,60 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-14.0.0-beta2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-14.0.0-beta2</h3><h3 align="center">Date: 2016-08-29</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
+<li><a href="#summary">Summary</a></li>
+<li><a href="#contributors">Contributors</a></li>
+<li><a href="#closed_issues">Closed Issues</a></li>
+<li><a href="#commits">Other Changes</a></li>
+<li><a href="#diffstat">Diffstat</a></li>
+</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is the first release of a major new version of Asterisk. For a list of new features that have been included with this release, please see the CHANGES file inside the source package. Since this is a new major release, users are encouraged to do extended testing before upgrading to this version in a production environment.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-14.0.0-beta1.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
+<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
+<tr valign="top"><td width="33%">6 George Joseph <gjoseph@digium.com><br/>5 Joshua Colp <jcolp@digium.com><br/>3 Kevin Harwell <kharwell@digium.com><br/>1 Richard Mudgett <rmudgett@digium.com><br/>1 Mark Michelson <mmichelson@digium.com><br/></td><td width="33%"><td width="33%">3 George Joseph <gjoseph@digium.com><br/>2 Joshua Colp <jcolp@digium.com><br/>1 Ben Merrills <ben@xdev.net><br/>1 Kevin Harwell <kharwell@digium.com><br/>1 Hans van Eijsden <info@hansvaneijsden.nl><br/></td></tr>
+</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26183">ASTERISK-26183</a>: alembic: error when using sqlalchemy version 1.1.0b2<br/>Reported by: Kevin Harwell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a783e1e60dd9594e830359ad3eb20c3ef0e761c3">[a783e1e60d]</a> Kevin Harwell -- alembic/sqlalchemy: auto increment only allowed on a single column</li>
+</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26253">ASTERISK-26253</a>: sdp_srtp: libsrtp now a required dependency, shouldn't be<br/>Reported by: Ben Merrills<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1d6b14c40498c3e6c6373fca1c2a41922ab7ebe">[a1d6b14c40]</a> Richard Mudgett -- res_srtp: Move SDP SRTP code from the core to res_srtp.</li>
+</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26238">ASTERISK-26238</a>: res_pjsip: Empty global default_from_user causes crash<br/>Reported by: Joshua Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a95c6dea39ecd2cf14b84f5d923d0a07a42138f">[9a95c6dea3]</a> gtjoseph -- res_pjsip:  Fail global load if debug or default_from_user are empty</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26241">ASTERISK-26241</a>: res_pjsip:  When using compact headers, rpid and pai are incorrectly generated<br/>Reported by: George Joseph<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aaee8160bcb875763eb8375e2b61b1c3a6a5db8a">[aaee8160bc]</a> gtjoseph -- res_pjsip_caller_id:  Copy header name to short header name</li>
+</ul><br><h4>Category: Resources/res_pjsip/Bundling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26148">ASTERISK-26148</a>: pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."<br/>Reported by: Hans van Eijsden<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=815b6f72f8796d27e17d91a6f5a14d9569f49564">[815b6f72f8]</a> gtjoseph -- pjproject_bundled:  Update for pjproject 2.5.5</li>
+</ul><br><h4>Category: Resources/res_pjsip_logger</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26239">ASTERISK-26239</a>: res_pjsip_logger:  An empty global/debug option is treated as a "match all" hostname<br/>Reported by: George Joseph<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a95c6dea39ecd2cf14b84f5d923d0a07a42138f">[9a95c6dea3]</a> gtjoseph -- res_pjsip:  Fail global load if debug or default_from_user are empty</li>
+</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
+<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73d39f20294168fbb366f9bc7fe5f0b5cf8f5cf2">73d39f2029</a></td><td>Joshua Colp</td><td>Release summaries: Remove previous versions</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e8a97775ee2ce128a9063f40b5657f1fbf5b3b07">e8a97775ee</a></td><td>Joshua Colp</td><td>.version: Update for 14.0.0-beta2</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=345409825a3e1e6ceb5b019855faf2bcedb44e98">345409825a</a></td><td>Joshua Colp</td><td>.lastclean: Update for 14.0.0-beta2</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=105c1168f7d634b2222d431a2793dec8960d8a33">105c1168f7</a></td><td>Joshua Colp</td><td>realtime: Add database scripts for 14.0.0-beta2</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8927b52634d7c7396cd31f2bcc258d26d20e0060">8927b52634</a></td><td>Joshua Colp</td><td>alembic: Fix downgrade path.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7af0eac02a57a92adf6274ae8c8e6dd8741e12ab">7af0eac02a</a></td><td>gtjoseph</td><td>autohints:  Update CHANGES and extensions.conf.sample</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ad00c1c30ce3726c0942595ac68b4d4038da10e">1ad00c1c30</a></td><td>gtjoseph</td><td>menuselect:  Add an opaque "member_data" string to the acceptable xml</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c95b611a73889cc29fdaaf037411972847660e56">c95b611a73</a></td><td>Mark Michelson</td><td>Remove SILK payload mappings from Asterisk core.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc94ccbcdddc80992d07a42986abc751d958f78d">bc94ccbcdd</a></td><td>Kevin Harwell</td><td>rtp_engine: Failed assertion and wrong name given for codec</td></tr>
+</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-14.0.0-beta1-summary.html                                                            | 4244 ----
+asterisk-14.0.0-beta1-summary.txt                                                             |10040 ----------
+b/.version                                                                                    |    2
+b/CHANGES                                                                                     |   11
+b/configs/samples/extensions.conf.sample                                                      |    5
+b/configure                                                                                   |   45
+b/configure.ac                                                                                |    3
+b/contrib/ast-db-manage/config/versions/3772f8f828da_update_identify_by.py                    |   44
+b/contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py                     |   10
+b/contrib/realtime/mssql/mssql_config.sql                                                     |   17
+b/contrib/realtime/mysql/mysql_config.sql                                                     |    9
+b/contrib/realtime/oracle/oracle_config.sql                                                   |   17
+b/contrib/realtime/postgresql/postgresql_config.sql                                           |   15
+b/include/asterisk/autoconfig.h.in                                                            |    3
+b/include/asterisk/sdp_srtp.h                                                                 |   95
+b/main/rtp_engine.c                                                                           |   24
+b/main/sdp_srtp.c                                                                             |  606
+b/menuselect/menuselect.c                                                                     |    6
+b/menuselect/menuselect.h                                                                     |    2
+b/res/res_pjsip/config_global.c                                                               |   12
+b/res/res_pjsip_caller_id.c                                                                   |    2
+b/res/res_pjsip_diversion.c                                                                   |    2
+b/res/res_resolver_unbound.c                                                                  |   53
+b/res/res_srtp.c                                                                              |  587
+b/third-party/pjproject/Makefile.rules                                                        |   31
+third-party/pjproject/patches/0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch |   73
+third-party/pjproject/patches/0001-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch |   26
+27 files changed, 963 insertions(+), 15021 deletions(-)</pre><br></html>
--- a/asterisk-14.0.0-beta2-summary.txt
+++ b/asterisk-14.0.0-beta2-summary.txt
@@ -0,0 +1,181 @@
+                                Release Summary
+
+                             asterisk-14.0.0-beta2
+
+                                Date: 2016-08-29
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Closed Issues
+    4. Other Changes
+    5. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This is the first release of a major new version of Asterisk. For a list
+   of new features that have been included with this release, please see the
+   CHANGES file inside the source package. Since this is a new major release,
+   users are encouraged to do extended testing before upgrading to this
+   version in a production environment.
+
+   The data in this summary reflects changes that have been made since the
+   previous release, asterisk-14.0.0-beta1.
+
+     ----------------------------------------------------------------------
+
+                                  Contributors
+
+                                 [Back to Top]
+
+   This table lists the people who have submitted code, those that have
+   tested patches, as well as those that reported issues on the issue tracker
+   that were resolved in this release. For coders, the number is how many of
+   their patches (of any size) were committed into this release. For testers,
+   the number is the number of times their name was listed as assisting with
+   testing a patch. Finally, for reporters, the number is the number of
+   issues that they reported that were affected by commits that went into
+   this release.
+
+   Coders                   Testers                  Reporters                
+   6 George Joseph                                   3 George Joseph          
+   5 Joshua Colp                                     2 Joshua Colp            
+   3 Kevin Harwell                                   1 Ben Merrills           
+   1 Richard Mudgett                                 1 Kevin Harwell          
+   1 Mark Michelson                                  1 Hans van Eijsden       
+
+     ----------------------------------------------------------------------
+
+                                 Closed Issues
+
+                                 [Back to Top]
+
+   This is a list of all issues from the issue tracker that were closed by
+   changes that went into this release.
+
+  Bug
+
+    Category: Contrib/General
+
+   ASTERISK-26183: alembic: error when using sqlalchemy version 1.1.0b2
+   Reported by: Kevin Harwell
+     * [a783e1e60d] Kevin Harwell -- alembic/sqlalchemy: auto increment only
+       allowed on a single column
+
+    Category: Core/General
+
+   ASTERISK-26253: sdp_srtp: libsrtp now a required dependency, shouldn't be
+   Reported by: Ben Merrills
+     * [a1d6b14c40] Richard Mudgett -- res_srtp: Move SDP SRTP code from the
+       core to res_srtp.
+
+    Category: Resources/res_pjsip
+
+   ASTERISK-26238: res_pjsip: Empty global default_from_user causes crash
+   Reported by: Joshua Colp
+     * [9a95c6dea3] gtjoseph -- res_pjsip: Fail global load if debug or
+       default_from_user are empty
+   ASTERISK-26241: res_pjsip: When using compact headers, rpid and pai are
+   incorrectly generated
+   Reported by: George Joseph
+     * [aaee8160bc] gtjoseph -- res_pjsip_caller_id: Copy header name to
+       short header name
+
+    Category: Resources/res_pjsip/Bundling
+
+   ASTERISK-26148: pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so:
+   undefined reference to..."
+   Reported by: Hans van Eijsden
+     * [815b6f72f8] gtjoseph -- pjproject_bundled: Update for pjproject 2.5.5
+
+    Category: Resources/res_pjsip_logger
+
+   ASTERISK-26239: res_pjsip_logger: An empty global/debug option is treated
+   as a "match all" hostname
+   Reported by: George Joseph
+     * [9a95c6dea3] gtjoseph -- res_pjsip: Fail global load if debug or
+       default_from_user are empty
+
+     ----------------------------------------------------------------------
+
+                      Commits Not Associated with an Issue
+
+                                 [Back to Top]
+
+   This is a list of all changes that went into this release that did not
+   reference a JIRA issue.
+
+   +------------------------------------------------------------------------+
+   | Revision   | Author         | Summary                                  |
+   |------------+----------------+------------------------------------------|
+   | 73d39f2029 | Joshua Colp    | Release summaries: Remove previous       |
+   |            |                | versions                                 |
+   |------------+----------------+------------------------------------------|
+   | e8a97775ee | Joshua Colp    | .version: Update for 14.0.0-beta2        |
+   |------------+----------------+------------------------------------------|
+   | 345409825a | Joshua Colp    | .lastclean: Update for 14.0.0-beta2      |
+   |------------+----------------+------------------------------------------|
+   | 105c1168f7 | Joshua Colp    | realtime: Add database scripts for       |
+   |            |                | 14.0.0-beta2                             |
+   |------------+----------------+------------------------------------------|
+   | 8927b52634 | Joshua Colp    | alembic: Fix downgrade path.             |
+   |------------+----------------+------------------------------------------|
+   | 7af0eac02a | gtjoseph       | autohints: Update CHANGES and            |
+   |            |                | extensions.conf.sample                   |
+   |------------+----------------+------------------------------------------|
+   | 1ad00c1c30 | gtjoseph       | menuselect: Add an opaque "member_data"  |
+   |            |                | string to the acceptable xml             |
+   |------------+----------------+------------------------------------------|
+   | c95b611a73 | Mark Michelson | Remove SILK payload mappings from        |
+   |            |                | Asterisk core.                           |
+   |------------+----------------+------------------------------------------|
+   | bc94ccbcdd | Kevin Harwell  | rtp_engine: Failed assertion and wrong   |
+   |            |                | name given for codec                     |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                Diffstat Results
+
+                                 [Back to Top]
+
+   This is a summary of the changes to the source code that went into this
+   release that was generated using the diffstat utility.
+
+ asterisk-14.0.0-beta1-summary.html                                                            | 4244 ----
+ asterisk-14.0.0-beta1-summary.txt                                                             |10040 ----------
+ b/.version                                                                                    |    2
+ b/CHANGES                                                                                     |   11
+ b/configs/samples/extensions.conf.sample                                                      |    5
+ b/configure                                                                                   |   45
+ b/configure.ac                                                                                |    3
+ b/contrib/ast-db-manage/config/versions/3772f8f828da_update_identify_by.py                    |   44
+ b/contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py                     |   10
+ b/contrib/realtime/mssql/mssql_config.sql                                                     |   17
+ b/contrib/realtime/mysql/mysql_config.sql                                                     |    9
+ b/contrib/realtime/oracle/oracle_config.sql                                                   |   17
+ b/contrib/realtime/postgresql/postgresql_config.sql                                           |   15
+ b/include/asterisk/autoconfig.h.in                                                            |    3
+ b/include/asterisk/sdp_srtp.h                                                                 |   95
+ b/main/rtp_engine.c                                                                           |   24
+ b/main/sdp_srtp.c                                                                             |  606
+ b/menuselect/menuselect.c                                                                     |    6
+ b/menuselect/menuselect.h                                                                     |    2
+ b/res/res_pjsip/config_global.c                                                               |   12
+ b/res/res_pjsip_caller_id.c                                                                   |    2
+ b/res/res_pjsip_diversion.c                                                                   |    2
+ b/res/res_resolver_unbound.c                                                                  |   53
+ b/res/res_srtp.c                                                                              |  587
+ b/third-party/pjproject/Makefile.rules                                                        |   31
+ third-party/pjproject/patches/0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch |   73
+ third-party/pjproject/patches/0001-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch |   26
+ 27 files changed, 963 insertions(+), 15021 deletions(-)
--- a/configs/samples/extensions.conf.sample
+++ b/configs/samples/extensions.conf.sample
@@ -197,6 +197,11 @@ TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
 ; before "B".
 ;
 ;[context]
+;
+;autohints = yes
+; If enabled for a context, a device state hint will be automatically created in
+; the context with the name of the device and updated with device state changes.
+;
 ;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
 ;
 ; Timing list for includes is
--- a/45
+++ b/45
@@ -23683,7 +23683,7 @@ if test "x${PBX_UNBOUND}" != "x1" -a "${USE_UNBOUND}" != "no"; then
         pbxlibdir="-L${UNBOUND_DIR}"
      fi
   fi
-   pbxfuncname="ub_ctx_add_ta_autr"
+   pbxfuncname="ub_ctx_delete"
   if test "x${pbxfuncname}" = "x" ; then   # empty lib, assume only headers
      AST_UNBOUND_FOUND=yes
   else
@@ -23777,6 +23777,49 @@ fi



+    if test "x${PBX_UNBOUND_CONST_PARAMS}" != "x1" -a "${USE_UNBOUND_CONST_PARAMS}" != "no"; then
+        { $as_echo "$as_me:${as_lineno-$LINENO}: checking for UNBOUND_VERSION_MAJOR declared in unbound.h" >&5
+$as_echo_n "checking for UNBOUND_VERSION_MAJOR declared in unbound.h... " >&6; }
+        saved_cppflags="${CPPFLAGS}"
+        if test "x${UNBOUND_CONST_PARAMS_DIR}" != "x"; then
+            UNBOUND_CONST_PARAMS_INCLUDE="-I${UNBOUND_CONST_PARAMS_DIR}/include"
+        fi
+        CPPFLAGS="${CPPFLAGS} ${UNBOUND_CONST_PARAMS_INCLUDE}"
+
+        cat confdefs.h - <<_ACEOF >conftest.$ac_ext
+/* end confdefs.h.  */
+ #include <unbound.h>
+int
+main ()
+{
+#if !defined(UNBOUND_VERSION_MAJOR)
+                                    (void) UNBOUND_VERSION_MAJOR;
+                                #endif
+
+  ;
+  return 0;
+}
+_ACEOF
+if ac_fn_c_try_compile "$LINENO"; then :
+     { $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
+$as_echo "yes" >&6; }
+                PBX_UNBOUND_CONST_PARAMS=1
+
+$as_echo "#define HAVE_UNBOUND_CONST_PARAMS 1" >>confdefs.h
+
+
+
+else
+     { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5
+$as_echo "no" >&6; }
+
+fi
+rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext
+
+        CPPFLAGS="${saved_cppflags}"
+    fi
+
+

 if test "x${PBX_UNIXODBC}" != "x1" -a "${USE_UNIXODBC}" != "no"; then
   pbxlibdir=""
--- a/configure.ac
+++ b/configure.ac
@@ -2101,7 +2101,8 @@ AST_EXT_LIB_CHECK([NEWT], [newt], [newtBell], [newt.h])
 # script bug which does not find the ldns library.  The bug is fixed in
 # v1.4.22 but that version is not easily detectable.
 #
-AST_EXT_LIB_CHECK([UNBOUND], [unbound], [ub_ctx_add_ta_autr], [unbound.h], [])
+AST_EXT_LIB_CHECK([UNBOUND], [unbound], [ub_ctx_delete], [unbound.h], [])
+AST_C_DECLARE_CHECK([UNBOUND_CONST_PARAMS], [UNBOUND_VERSION_MAJOR], [unbound.h])

 AST_EXT_LIB_CHECK([UNIXODBC], [odbc], [SQLConnect], [sql.h], [])

--- a/contrib/ast-db-manage/config/versions/3772f8f828da_update_identify_by.py
+++ b/contrib/ast-db-manage/config/versions/3772f8f828da_update_identify_by.py
@@ -0,0 +1,44 @@
+"""update_identify_by
+
+Revision ID: 3772f8f828da
+Revises: c7a44a5a0851
+Create Date: 2016-08-11 10:47:29.211063
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '3772f8f828da'
+down_revision = '4a6c67fa9b7a'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def enum_update(table_name, column_name, enum_name, enum_values):
+    if op.get_context().bind.dialect.name != 'postgresql':
+        op.alter_column(table_name, column_name,
+                        type_=sa.Enum(*enum_values, name=enum_name))
+        return
+
+    # Postgres requires a few more steps
+    tmp = enum_name + '_tmp'
+
+    op.execute('ALTER TYPE ' + enum_name + ' RENAME TO ' + tmp)
+
+    updated = sa.Enum(*enum_values, name=enum_name)
+    updated.create(op.get_bind(), checkfirst=False)
+
+    op.execute('ALTER TABLE ' + table_name + ' ALTER COLUMN ' + column_name +
+               ' TYPE ' + enum_name + ' USING identify_by::text::' + enum_name)
+
+    op.execute('DROP TYPE ' + tmp)
+
+
+def upgrade():
+    enum_update('ps_endpoints', 'identify_by', 'pjsip_identify_by_values',
+                ['username', 'auth_username'])
+
+
+def downgrade():
+    enum_update('ps_endpoints', 'identify_by', 'pjsip_identify_by_values',
+                ['username'])
--- a/contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py
+++ b/contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py
@@ -31,20 +31,18 @@ down_revision = '43956d550a44'
 from alembic import op
 import sqlalchemy as sa

-
 def upgrade():
    op.create_table(
        'extensions',
        sa.Column('id', sa.BigInteger, primary_key=True, nullable=False,
                  unique=True, autoincrement=True),
-        sa.Column('context', sa.String(40), primary_key=True, nullable=False),
-        sa.Column('exten', sa.String(40), primary_key=True, nullable=False),
-        sa.Column('priority', sa.Integer, primary_key=True, nullable=False,
-                  autoincrement=True),
+        sa.Column('context', sa.String(40), nullable=False),
+        sa.Column('exten', sa.String(40), nullable=False),
+        sa.Column('priority', sa.Integer, nullable=False),
        sa.Column('app', sa.String(40), nullable=False),
        sa.Column('appdata', sa.String(256), nullable=False),
+        sa.UniqueConstraint('context', 'exten', 'priority')
    )

-
 def downgrade():
    op.drop_table('extensions')
--- a/contrib/realtime/mssql/mssql_config.sql
+++ b/contrib/realtime/mssql/mssql_config.sql
@@ -557,7 +557,8 @@ CREATE TABLE extensions (
    priority INTEGER NOT NULL, 
    app VARCHAR(40) NOT NULL, 
    appdata VARCHAR(256) NOT NULL, 
-    PRIMARY KEY (id, context, exten, priority), 
+    PRIMARY KEY (id), 
+    UNIQUE (context, exten, priority), 
    UNIQUE (id)
 );

@@ -1508,6 +1509,20 @@ UPDATE alembic_version SET version_num='4a6c67fa9b7a' WHERE alembic_version.vers

 GO

+-- Running upgrade 4a6c67fa9b7a -> 3772f8f828da
+
+ALTER TABLE ps_endpoints ALTER COLUMN identify_by VARCHAR(13);
+
+GO
+
+ALTER TABLE ps_endpoints ADD CONSTRAINT pjsip_identify_by_values CHECK (identify_by IN ('username', 'auth_username'));
+
+GO
+
+UPDATE alembic_version SET version_num='3772f8f828da' WHERE alembic_version.version_num = '4a6c67fa9b7a';
+
+GO
+
 COMMIT;

 GO
--- a/contrib/realtime/mysql/mysql_config.sql
+++ b/contrib/realtime/mysql/mysql_config.sql
@@ -398,7 +398,8 @@ CREATE TABLE extensions (
    priority INTEGER NOT NULL, 
    app VARCHAR(40) NOT NULL, 
    appdata VARCHAR(256) NOT NULL, 
-    PRIMARY KEY (id, context, exten, priority), 
+    PRIMARY KEY (id), 
+    UNIQUE (context, exten, priority), 
    UNIQUE (id)
 );

@@ -937,3 +938,9 @@ ALTER TABLE ps_endpoints ADD COLUMN fax_detect_timeout INTEGER;

 UPDATE alembic_version SET version_num='4a6c67fa9b7a' WHERE alembic_version.version_num = '9deac0ae4717';

+-- Running upgrade 4a6c67fa9b7a -> 3772f8f828da
+
+ALTER TABLE ps_endpoints MODIFY identify_by ENUM('username','auth_username') NULL;
+
+UPDATE alembic_version SET version_num='3772f8f828da' WHERE alembic_version.version_num = '4a6c67fa9b7a';
+
--- a/contrib/realtime/oracle/oracle_config.sql
+++ b/contrib/realtime/oracle/oracle_config.sql
@@ -555,7 +555,8 @@ CREATE TABLE extensions (
    priority INTEGER NOT NULL, 
    app VARCHAR2(40 CHAR) NOT NULL, 
    appdata VARCHAR2(256 CHAR) NOT NULL, 
-    PRIMARY KEY (id, context, exten, priority), 
+    PRIMARY KEY (id), 
+    UNIQUE (context, exten, priority), 
    UNIQUE (id)
 )

@@ -1506,3 +1507,17 @@ UPDATE alembic_version SET version_num='4a6c67fa9b7a' WHERE alembic_version.vers

 /

+-- Running upgrade 4a6c67fa9b7a -> 3772f8f828da
+
+ALTER TABLE ps_endpoints MODIFY identify_by VARCHAR(13 CHAR)
+
+/
+
+ALTER TABLE ps_endpoints ADD CONSTRAINT pjsip_identify_by_values CHECK (identify_by IN ('username', 'auth_username'))
+
+/
+
+UPDATE alembic_version SET version_num='3772f8f828da' WHERE alembic_version.version_num = '4a6c67fa9b7a'
+
+/
+
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
@@ -450,7 +450,8 @@ CREATE TABLE extensions (
    priority INTEGER NOT NULL, 
    app VARCHAR(40) NOT NULL, 
    appdata VARCHAR(256) NOT NULL, 
-    PRIMARY KEY (id, context, exten, priority), 
+    PRIMARY KEY (id), 
+    UNIQUE (context, exten, priority), 
    UNIQUE (id)
 );

@@ -1007,5 +1008,17 @@ ALTER TABLE ps_endpoints ADD COLUMN fax_detect_timeout INTEGER;

 UPDATE alembic_version SET version_num='4a6c67fa9b7a' WHERE alembic_version.version_num = '9deac0ae4717';

+-- Running upgrade 4a6c67fa9b7a -> 3772f8f828da
+
+ALTER TYPE pjsip_identify_by_values RENAME TO pjsip_identify_by_values_tmp;
+
+CREATE TYPE pjsip_identify_by_values AS ENUM ('username', 'auth_username');
+
+ALTER TABLE ps_endpoints ALTER COLUMN identify_by TYPE pjsip_identify_by_values USING identify_by::text::pjsip_identify_by_values;
+
+DROP TYPE pjsip_identify_by_values_tmp;
+
+UPDATE alembic_version SET version_num='3772f8f828da' WHERE alembic_version.version_num = '4a6c67fa9b7a';
+
 COMMIT;

--- a/include/asterisk/autoconfig.h.in
+++ b/include/asterisk/autoconfig.h.in
@@ -1108,6 +1108,9 @@
 /* Define to 1 if you have the unbound library. */
 #undef HAVE_UNBOUND

+/* Define if your system has UNBOUND_VERSION_MAJOR declared. */
+#undef HAVE_UNBOUND_CONST_PARAMS
+
 /* Define to 1 if you have the <unistd.h> header file. */
 #undef HAVE_UNISTD_H

--- a/include/asterisk/sdp_srtp.h
+++ b/include/asterisk/sdp_srtp.h
@@ -63,6 +63,100 @@ struct ast_sdp_srtp *ast_sdp_srtp_alloc(void);
 */
 void ast_sdp_srtp_destroy(struct ast_sdp_srtp *srtp);

+/*! \brief Destroy a previously allocated ast_sdp_crypto struct */
+typedef void (*sdp_crypto_destroy_cb)(struct ast_sdp_crypto *crypto);
+
+/*!
+ * \brief Initialize and return an ast_sdp_crypto struct
+ *
+ * \details
+ * This function allocates a new ast_sdp_crypto struct and initializes its values
+ *
+ * \retval NULL on failure
+ * \retval a pointer to a  new ast_sdp_crypto structure
+ */
+typedef struct ast_sdp_crypto *(*sdp_crypto_alloc_cb)(void);
+
+/*!
+ * \brief Generate an SRTP a=crypto offer
+ *
+ * \details
+ * The offer is stored on the ast_sdp_crypto struct in a_crypto
+ *
+ * \param crypto A valid ast_sdp_crypto struct
+ * \param taglen Length
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+typedef int (*sdp_crypto_build_offer_cb)(struct ast_sdp_crypto *crypto, int taglen);
+
+/*!
+ * \brief Parse the a=crypto line from SDP and set appropriate values on the
+ * ast_sdp_crypto struct.
+ *
+ * The attribute line should already have "a=crypto:" removed.
+ *
+ * \param p A valid ast_sdp_crypto struct
+ * \param attr the a:crypto line from SDP
+ * \param rtp The rtp instance associated with the SDP being parsed
+ * \param srtp SRTP structure
+ *
+ * \retval 0 success
+ * \retval nonzero failure
+ */
+typedef int (*sdp_crypto_parse_offer_cb)(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *srtp, const char *attr);
+
+/*!
+ * \brief Get the crypto attribute line for the srtp structure
+ *
+ * \details
+ * The attribute line does not contain the initial "a=crypto:" and does
+ * not terminate with "\r\n".
+ *
+ * \param srtp The ast_sdp_srtp structure for which to get an attribute line
+ * \param dtls_enabled Whether this connection is encrypted with datagram TLS
+ * \param default_taglen_32 Whether to default to a tag length of 32 instead of 80
+ *
+ * \retval An attribute line containing cryptographic information
+ * \retval NULL if the srtp structure does not require an attribute line containing crypto information
+ */
+typedef const char *(*sdp_srtp_get_attr_cb)(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32);
+
+struct ast_sdp_crypto_api {
+	/*! Destroy a crypto struct */
+	sdp_crypto_destroy_cb dtor;
+	/*! Allocate a crypto struct */
+	sdp_crypto_alloc_cb alloc;
+	/*! Build a SDP a=crypto offer line parameter string */
+	sdp_crypto_build_offer_cb build_offer;
+	/*! Parse a SDP a=crypto offer line parameter string */
+	sdp_crypto_parse_offer_cb parse_offer;
+	/*! Get the SDP a=crypto offer line parameter string */
+	sdp_srtp_get_attr_cb get_attr;
+};
+
+/*!
+ * \brief Register SDP SRTP crypto processing routines.
+ * \since 14.0.0
+ *
+ * \param api Callbacks to register.
+ *
+ * \retval 0 on success.
+ * \retval -1 on error.
+ */
+int ast_sdp_crypto_register(struct ast_sdp_crypto_api *api);
+
+/*!
+ * \brief Unregister SDP SRTP crypto processing routines.
+ * \since 14.0.0
+ *
+ * \param api Callbacks to unregister.
+ *
+ * \return Nothing
+ */
+void ast_sdp_crypto_unregister(struct ast_sdp_crypto_api *api);
+
 /*! \brief Initialize an return an ast_sdp_crypto struct
 *
 * \details
@@ -104,7 +198,6 @@ int ast_sdp_crypto_process(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *sr
 */
 int ast_sdp_crypto_build_offer(struct ast_sdp_crypto *p, int taglen);

-
 /*! \brief Get the crypto attribute line for the srtp structure
 *
 * The attribute line does not contain the initial "a=crypto:" and does
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -983,6 +983,7 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
 		} else {
 			new_type->format = t->payload_type.format;
 		}
+
 		if (new_type->format) {
 			/* SDP parsing automatically increases the reference count */
 			new_type->format = ast_format_parse_sdp_fmtp(new_type->format, "");
@@ -2257,7 +2258,11 @@ static void add_static_payload(int map, struct ast_format *format, int rtp_code)
 	int x;
 	struct ast_rtp_payload_type *type;

-	ast_assert(map < ARRAY_LEN(static_RTP_PT));
+	/*
+	 * ARRAY_LEN's result is cast to an int so 'map' is not autocast to a size_t,
+	 * which if negative would cause an assertion.
+	 */
+	ast_assert(map < (int)ARRAY_LEN(static_RTP_PT));

 	ast_rwlock_wrlock(&static_RTP_PT_lock);
 	if (map < 0) {
@@ -2268,6 +2273,7 @@ static void add_static_payload(int map, struct ast_format *format, int rtp_code)
 				break;
 			}
 		}
+
 		if (map < 0) {
 			if (format) {
 				ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n",
@@ -2300,14 +2306,10 @@ static void add_static_payload(int map, struct ast_format *format, int rtp_code)

 int ast_rtp_engine_load_format(struct ast_format *format)
 {
-	char *codec_name = ast_strdupa(ast_format_get_name(format));
-
-	codec_name = ast_str_to_upper(codec_name);
-
 	set_next_mime_type(format,
 		0,
 		ast_codec_media_type2str(ast_format_get_type(format)),
-		codec_name,
+		ast_format_get_codec_name(format),
 		ast_format_get_sample_rate(format));
 	add_static_payload(-1, format, 0);

@@ -2690,11 +2692,6 @@ int ast_rtp_engine_init(void)
 	/* Opus and VP8 */
 	set_next_mime_type(ast_format_opus, 0,  "audio", "opus", 48000);
 	set_next_mime_type(ast_format_vp8, 0,  "video", "VP8", 90000);
-	/* DA SILK */
-	set_next_mime_type(ast_format_silk8, 0, "audio", "silk", 8000);
-	set_next_mime_type(ast_format_silk12, 0, "audio", "silk", 12000);
-	set_next_mime_type(ast_format_silk16, 0, "audio", "silk", 16000);
-	set_next_mime_type(ast_format_silk24, 0, "audio", "silk", 24000);

 	/* Define the static rtp payload mappings */
 	add_static_payload(0, ast_format_ulaw, 0);
@@ -2748,11 +2745,6 @@ int ast_rtp_engine_init(void)
 	add_static_payload(100, ast_format_vp8, 0);
 	add_static_payload(107, ast_format_opus, 0);

-	add_static_payload(108, ast_format_silk8, 0);
-	add_static_payload(109, ast_format_silk12, 0);
-	add_static_payload(113, ast_format_silk16, 0);
-	add_static_payload(114, ast_format_silk24, 0);
-
 	return 0;
 }

--- a/main/sdp_srtp.c
+++ b/main/sdp_srtp.c
@@ -33,18 +33,12 @@

 ASTERISK_REGISTER_FILE()

-#include <math.h>                       /* for pow */
-#include <srtp/srtp.h>                  /* for SRTP_MAX_KEY_LEN, etc */
-
 #include "asterisk/linkedlists.h"       /* for AST_LIST_NEXT, etc */
 #include "asterisk/logger.h"            /* for ast_log, LOG_ERROR, etc */
-#include "asterisk/rtp_engine.h"        /* for ast_rtp_engine_dtls, etc */
 #include "asterisk/sdp_srtp.h"          /* for ast_sdp_srtp, etc */
-#include "asterisk/strings.h"           /* for ast_strlen_zero */
-#include "asterisk/utils.h"             /* for ast_set_flag, ast_test_flag, etc */

-extern struct ast_srtp_res *res_srtp;
-extern struct ast_srtp_policy_res *res_srtp_policy;
+/*! Registered SDP crypto API */
+static struct ast_sdp_crypto_api *sdp_crypto_api;

 struct ast_sdp_srtp *ast_sdp_srtp_alloc(void)
 {
@@ -63,603 +57,49 @@ void ast_sdp_srtp_destroy(struct ast_sdp_srtp *srtp)
 	for (next = AST_LIST_NEXT(srtp, sdp_srtp_list);
 	     srtp;
 	     srtp = next, next = srtp ? AST_LIST_NEXT(srtp, sdp_srtp_list) : NULL) {
-		if (srtp->crypto) {
-			ast_sdp_crypto_destroy(srtp->crypto);
-		}
+		ast_sdp_crypto_destroy(srtp->crypto);
 		srtp->crypto = NULL;
 		ast_free(srtp);
 	}
 }

-struct ast_sdp_crypto {
-	char *a_crypto;
-	unsigned char local_key[SRTP_MAX_KEY_LEN];
-	int tag;
-	char local_key64[((SRTP_MAX_KEY_LEN) * 8 + 5) / 6 + 1];
-	unsigned char remote_key[SRTP_MAX_KEY_LEN];
-	int key_len;
-};
-
-static struct ast_sdp_crypto *sdp_crypto_alloc(const int key_len);
-static struct ast_sdp_crypto *crypto_init_keys(struct ast_sdp_crypto *p, const int key_len);
-static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, int key_len, unsigned long ssrc, int inbound);
-
 void ast_sdp_crypto_destroy(struct ast_sdp_crypto *crypto)
 {
-	ast_free(crypto->a_crypto);
-	crypto->a_crypto = NULL;
-	ast_free(crypto);
-}
-
-static struct ast_sdp_crypto *crypto_init_keys(struct ast_sdp_crypto *p, const int key_len)
-{
-	unsigned char remote_key[key_len];
-
-	if (res_srtp->get_random(p->local_key, key_len) < 0) {
-		return NULL;
+	if (sdp_crypto_api) {
+		sdp_crypto_api->dtor(crypto);
 	}
-
-	ast_base64encode(p->local_key64, p->local_key, key_len, sizeof(p->local_key64));
-
-	p->key_len = ast_base64decode(remote_key, p->local_key64, sizeof(remote_key));
-
-	if (p->key_len != key_len) {
-		ast_log(LOG_ERROR, "base64 encode/decode bad len %d != %d\n", p->key_len, key_len);
-		return NULL;
-	}
-
-	if (memcmp(remote_key, p->local_key, p->key_len)) {
-		ast_log(LOG_ERROR, "base64 encode/decode bad key\n");
-		return NULL;
-	}
-
-	ast_debug(1 , "local_key64 %s len %zu\n", p->local_key64, strlen(p->local_key64));
-
-	return p;
-}
-
-static struct ast_sdp_crypto *sdp_crypto_alloc(const int key_len)
-{
-	struct ast_sdp_crypto *p, *result;
-
-	if (!ast_rtp_engine_srtp_is_registered()) {
-		return NULL;
-	}
-
-	if (!(p = ast_calloc(1, sizeof(*p)))) {
-		return NULL;
-	}
-	p->tag = 1;
-
-	/* default is a key which uses AST_AES_CM_128_HMAC_SHA1_xx */
-	result = crypto_init_keys(p, key_len);
-	if (!result) {
-		ast_sdp_crypto_destroy(p);
-	}
-
-	return result;
 }

 struct ast_sdp_crypto *ast_sdp_crypto_alloc(void)
 {
-	return sdp_crypto_alloc(SRTP_MASTER_KEY_LEN);
-}
-
-static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, int key_len, unsigned long ssrc, int inbound)
-{
-	if (!ast_rtp_engine_srtp_is_registered()) {
-		return -1;
+	if (!sdp_crypto_api) {
+		return NULL;
 	}
-
-	if (res_srtp_policy->set_master_key(policy, master_key, key_len, NULL, 0) < 0) {
-		return -1;
-	}
-
-	if (res_srtp_policy->set_suite(policy, suite_val)) {
-		ast_log(LOG_WARNING, "Could not set remote SRTP suite\n");
-		return -1;
-	}
-
-	res_srtp_policy->set_ssrc(policy, ssrc, inbound);
-
-	return 0;
-}
-
-static int crypto_activate(struct ast_sdp_crypto *p, int suite_val, unsigned char *remote_key, int key_len, struct ast_rtp_instance *rtp)
-{
-	struct ast_srtp_policy *local_policy = NULL;
-	struct ast_srtp_policy *remote_policy = NULL;
-	struct ast_rtp_instance_stats stats = {0,};
-	int res = -1;
-
-	if (!ast_rtp_engine_srtp_is_registered()) {
-		return -1;
-	}
-
-	if (!p) {
-		return -1;
-	}
-
-	if (!(local_policy = res_srtp_policy->alloc())) {
-		return -1;
-	}
-
-	if (!(remote_policy = res_srtp_policy->alloc())) {
-		goto err;
-	}
-
-	if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_LOCAL_SSRC)) {
-		goto err;
-	}
-
-	if (set_crypto_policy(local_policy, suite_val, p->local_key, key_len, stats.local_ssrc, 0) < 0) {
-		goto err;
-	}
-
-	if (set_crypto_policy(remote_policy, suite_val, remote_key, key_len, 0, 1) < 0) {
-		goto err;
-	}
-
-	/* Add the SRTP policies */
-	if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy, 0)) {
-		ast_log(LOG_WARNING, "Could not set SRTP policies\n");
-		goto err;
-	}
-
-	ast_debug(1 , "SRTP policy activated\n");
-	res = 0;
-
-err:
-	if (local_policy) {
-		res_srtp_policy->destroy(local_policy);
-	}
-
-	if (remote_policy) {
-		res_srtp_policy->destroy(remote_policy);
-	}
-
-	return res;
+	return sdp_crypto_api->alloc();
 }

 int ast_sdp_crypto_process(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *srtp, const char *attr)
 {
-	char *str = NULL;
-	char *tag = NULL;
-	char *suite = NULL;
-	char *key_params = NULL;
-	char *key_param = NULL;
-	char *session_params = NULL;
-	char *key_salt = NULL;       /* The actual master key and key salt */
-	char *lifetime = NULL;       /* Key lifetime (# of RTP packets) */
-	char *mki = NULL;            /* Master Key Index */
-	int found = 0;
-	int key_len_from_sdp;
-	int key_len_expected;
-	int tag_from_sdp;
-	int suite_val = 0;
-	unsigned char remote_key[SRTP_MAX_KEY_LEN];
-	int taglen;
-	double sdes_lifetime;
-	struct ast_sdp_crypto *crypto;
-	struct ast_sdp_srtp *tmp;
-
-	if (!ast_rtp_engine_srtp_is_registered()) {
+	if (!sdp_crypto_api) {
 		return -1;
 	}
-
-	str = ast_strdupa(attr);
-
-	tag = strsep(&str, " ");
-	suite = strsep(&str, " ");
-	key_params = strsep(&str, " ");
-	session_params = strsep(&str, " ");
-
-	if (!tag || !suite) {
-		ast_log(LOG_WARNING, "Unrecognized crypto attribute a=%s\n", attr);
-		return -1;
-	}
-
-	/* RFC4568 9.1 - tag is 1-9 digits, greater than zero */
-	if (sscanf(tag, "%30d", &tag_from_sdp) != 1 || tag_from_sdp <= 0 || tag_from_sdp > 999999999) {
-		ast_log(LOG_WARNING, "Unacceptable a=crypto tag: %s\n", tag);
-		return -1;
-	}
-
-	if (!ast_strlen_zero(session_params)) {
-		ast_log(LOG_WARNING, "Unsupported crypto parameters: %s\n", session_params);
-		return -1;
-	}
-
-	/* On egress, Asterisk sent several crypto lines in the SIP/SDP offer
-	   The remote party might have choosen another line than the first */
-	for (tmp = srtp; tmp && tmp->crypto && tmp->crypto->tag != tag_from_sdp;) {
-		tmp = AST_LIST_NEXT(tmp, sdp_srtp_list);
-	}
-	if (tmp) { /* tag matched an already created crypto line */
-		unsigned int flags = tmp->flags;
-
-		/* Make that crypto line the head of the list, not by changing the
-		   list structure but by exchanging the content of the list members */
-		crypto = tmp->crypto;
-		tmp->crypto = srtp->crypto;
-		tmp->flags = srtp->flags;
-		srtp->crypto = crypto;
-		srtp->flags = flags;
-	} else {
-		crypto = srtp->crypto;
-		crypto->tag = tag_from_sdp;
-	}
-
-	if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
-		suite_val = AST_AES_CM_128_HMAC_SHA1_80;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
-		key_len_expected = 30;
-	} else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
-		suite_val = AST_AES_CM_128_HMAC_SHA1_32;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
-		key_len_expected = 30;
-#ifdef HAVE_SRTP_192
-	} else if (!strcmp(suite, "AES_192_CM_HMAC_SHA1_80")) {
-		suite_val = AST_AES_CM_192_HMAC_SHA1_80;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
-		key_len_expected = 38;
-	} else if (!strcmp(suite, "AES_192_CM_HMAC_SHA1_32")) {
-		suite_val = AST_AES_CM_192_HMAC_SHA1_32;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
-		key_len_expected = 38;
-	/* RFC used a different name while in draft, some still use that */
-	} else if (!strcmp(suite, "AES_CM_192_HMAC_SHA1_80")) {
-		suite_val = AST_AES_CM_192_HMAC_SHA1_80;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
-		key_len_expected = 38;
-	} else if (!strcmp(suite, "AES_CM_192_HMAC_SHA1_32")) {
-		suite_val = AST_AES_CM_192_HMAC_SHA1_32;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
-		key_len_expected = 38;
-#endif
-#ifdef HAVE_SRTP_256
-	} else if (!strcmp(suite, "AES_256_CM_HMAC_SHA1_80")) {
-		suite_val = AST_AES_CM_256_HMAC_SHA1_80;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
-		key_len_expected = 46;
-	} else if (!strcmp(suite, "AES_256_CM_HMAC_SHA1_32")) {
-		suite_val = AST_AES_CM_256_HMAC_SHA1_32;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
-		key_len_expected = 46;
-	/* RFC used a different name while in draft, some still use that */
-	} else if (!strcmp(suite, "AES_CM_256_HMAC_SHA1_80")) {
-		suite_val = AST_AES_CM_256_HMAC_SHA1_80;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
-		key_len_expected = 46;
-	} else if (!strcmp(suite, "AES_CM_256_HMAC_SHA1_32")) {
-		suite_val = AST_AES_CM_256_HMAC_SHA1_32;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
-		key_len_expected = 46;
-#endif
-#ifdef HAVE_SRTP_GCM
-	} else if (!strcmp(suite, "AEAD_AES_128_GCM")) {
-		suite_val = AST_AES_GCM_128;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_16);
-		key_len_expected = AES_128_GCM_KEYSIZE_WSALT;
-	} else if (!strcmp(suite, "AEAD_AES_256_GCM")) {
-		suite_val = AST_AES_GCM_256;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_16);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
-		key_len_expected = AES_256_GCM_KEYSIZE_WSALT;
-	/* RFC contained a (too) short auth tag for RTP media, some still use that */
-	} else if (!strcmp(suite, "AEAD_AES_128_GCM_8")) {
-		suite_val = AST_AES_GCM_128_8;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_8);
-		key_len_expected = AES_128_GCM_KEYSIZE_WSALT;
-	} else if (!strcmp(suite, "AEAD_AES_256_GCM_8")) {
-		suite_val = AST_AES_GCM_256_8;
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_8);
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
-		key_len_expected = AES_256_GCM_KEYSIZE_WSALT;
-#endif
-	} else {
-		ast_verb(1, "Unsupported crypto suite: %s\n", suite);
-		return -1;
-	}
-
-	while ((key_param = strsep(&key_params, ";"))) {
-		unsigned int n_lifetime;
-		char *method = NULL;
-		char *info = NULL;
-
-		method = strsep(&key_param, ":");
-		info = strsep(&key_param, ";");
-		sdes_lifetime = 0;
-
-		if (strcmp(method, "inline")) {
-			continue;
-		}
-
-		key_salt = strsep(&info, "|");
-
-		/* The next parameter can be either lifetime or MKI */
-		lifetime = strsep(&info, "|");
-		if (!lifetime) {
-			found = 1;
-			break;
-		}
-
-		mki = strchr(lifetime, ':');
-		if (mki) {
-			mki = lifetime;
-			lifetime = NULL;
-		} else {
-			mki = strsep(&info, "|");
-		}
-
-		if (mki && *mki != '1') {
-			ast_log(LOG_NOTICE, "Crypto MKI handling is not supported: ignoring attribute %s\n", attr);
-			continue;
-		}
-
-		if (lifetime) {
-			if (!strncmp(lifetime, "2^", 2)) {
-				char *lifetime_val = lifetime + 2;
-
-				/* Exponential lifetime */
-				if (sscanf(lifetime_val, "%30u", &n_lifetime) != 1) {
-					ast_log(LOG_NOTICE, "Failed to parse lifetime value in crypto attribute: %s\n", attr);
-					continue;
-				}
-
-				if (n_lifetime > 48) {
-					/* Yeah... that's a bit big. */
-					ast_log(LOG_NOTICE, "Crypto lifetime exponent of '%u' is a bit large; using 48\n", n_lifetime);
-					n_lifetime = 48;
-				}
-				sdes_lifetime = pow(2, n_lifetime);
-			} else {
-				/* Decimal lifetime */
-				if (sscanf(lifetime, "%30u", &n_lifetime) != 1) {
-					ast_log(LOG_NOTICE, "Failed to parse lifetime value in crypto attribute: %s\n", attr);
-					continue;
-				}
-				sdes_lifetime = n_lifetime;
-			}
-
-			/* Accept anything above 10 hours. Less than 10; reject. */
-			if (sdes_lifetime < 1800000) {
-				ast_log(LOG_NOTICE, "Rejecting crypto attribute '%s': lifetime '%f' too short\n", attr, sdes_lifetime);
-				continue;
-			}
-		}
-
-		ast_debug(2, "Crypto attribute '%s' accepted with lifetime '%f', MKI '%s'\n",
-			attr, sdes_lifetime, mki ? mki : "-");
-
-		found = 1;
-		break;
-	}
-
-	if (!found) {
-		ast_log(LOG_NOTICE, "SRTP crypto offer not acceptable: '%s'\n", attr);
-		return -1;
-	}
-
-	key_len_from_sdp = ast_base64decode(remote_key, key_salt, sizeof(remote_key));
-	if (key_len_from_sdp != key_len_expected) {
-		ast_log(LOG_WARNING, "SRTP descriptions key length is '%d', not '%d'\n",
-			key_len_from_sdp, key_len_expected);
-		return -1;
-	}
-
-	/* on default, the key is 30 (AES-128); throw that away (only) when the suite changed actually */
-	/* ingress: optional, but saves one expensive call to get_random(.) */
-	/*  egress: required, because the local key was communicated before the remote key is processed */
-	if (crypto->key_len != key_len_from_sdp) {
-		if (!crypto_init_keys(crypto, key_len_from_sdp)) {
-			return -1;
-		}
-	} else if (!memcmp(crypto->remote_key, remote_key, key_len_from_sdp)) {
-		ast_debug(1, "SRTP remote key unchanged; maintaining current policy\n");
-		ast_set_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
-		return 0;
-	}
-
-	if (key_len_from_sdp > sizeof(crypto->remote_key)) {
-		ast_log(LOG_ERROR,
-			"SRTP key buffer is %zu although it must be at least %d bytes\n",
-			sizeof(crypto->remote_key), key_len_from_sdp);
-		return -1;
-	}
-	memcpy(crypto->remote_key, remote_key, key_len_from_sdp);
-
-	if (crypto_activate(crypto, suite_val, remote_key, key_len_from_sdp, rtp) < 0) {
-		return -1;
-	}
-
-	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_32)) {
-		taglen = 32;
-	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_16)) {
-		taglen = 16;
-	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_8)) {
-		taglen = 8;
-	} else {
-		taglen = 80;
-	}
-	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_256)) {
-		taglen |= 0x0200;
-	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_192)) {
-		taglen |= 0x0100;
-	}
-	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME)) {
-		taglen |= 0x0080;
-	}
-
-	/* Finally, rebuild the crypto line */
-	if (ast_sdp_crypto_build_offer(crypto, taglen)) {
-		return -1;
-	}
-
-	ast_set_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
-	return 0;
+	return sdp_crypto_api->parse_offer(rtp, srtp, attr);
 }

 int ast_sdp_crypto_build_offer(struct ast_sdp_crypto *p, int taglen)
 {
-	/* Rebuild the crypto line */
-	if (p->a_crypto) {
-		ast_free(p->a_crypto);
+	if (!sdp_crypto_api) {
+		return -1;
 	}
-
-	if ((taglen & 0x007f) == 8) {
-		if (ast_asprintf(&p->a_crypto, "%d AEAD_AES_%d_GCM_%d inline:%s",
-				 p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64) == -1) {
-				 ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
-			return -1;
-		}
-	} else if ((taglen & 0x007f) == 16) {
-		if (ast_asprintf(&p->a_crypto, "%d AEAD_AES_%d_GCM inline:%s",
-				 p->tag, 128 + ((taglen & 0x0300) >> 2), p->local_key64) == -1) {
-				 ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
-			return -1;
-		}
-	} else if ((taglen & 0x0300) && !(taglen & 0x0080)) {
-		if (ast_asprintf(&p->a_crypto, "%d AES_%d_CM_HMAC_SHA1_%d inline:%s",
-				 p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64) == -1) {
-				 ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
-			return -1;
-		}
-	} else {
-		if (ast_asprintf(&p->a_crypto, "%d AES_CM_%d_HMAC_SHA1_%d inline:%s",
-				 p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64) == -1) {
-				 ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
-			return -1;
-		}
-	}
-
-	ast_debug(1, "Crypto line: a=crypto:%s\n", p->a_crypto);
-
-	return 0;
+	return sdp_crypto_api->build_offer(p, taglen);
 }

 const char *ast_sdp_srtp_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
 {
-	int taglen;
-
-	if (!srtp) {
+	if (!sdp_crypto_api) {
 		return NULL;
 	}
-
-	/* Set encryption properties */
-	if (!srtp->crypto) {
-
-		if (AST_LIST_NEXT(srtp, sdp_srtp_list)) {
-			srtp->crypto = ast_sdp_crypto_alloc();
-			ast_log(LOG_ERROR, "SRTP SDP list was not empty\n");
-		} else {
-			const int len = default_taglen_32 ? AST_SRTP_CRYPTO_TAG_32 : AST_SRTP_CRYPTO_TAG_80;
-			const int attr[][3] = {
-			/* This array creates the following list:
-			 * a=crypto:1 AES_CM_128_HMAC_SHA1_ ...
-			 * a=crypto:2 AEAD_AES_128_GCM ...
-			 * a=crypto:3 AES_256_CM_HMAC_SHA1_ ...
-			 * a=crypto:4 AEAD_AES_256_GCM ...
-			 * a=crypto:5 AES_192_CM_HMAC_SHA1_ ...
-			 * something like 'AEAD_AES_192_GCM' is not specified by the RFCs
-			 *
-			 * If you want to prefer another crypto suite or you want to
-			 * exclude a suite, change this array and recompile Asterisk.
-			 * This list cannot be changed from rtp.conf because you should
-			 * know what you are doing. Especially AES-192 and AES-GCM are
-			 * broken in many VoIP clients, see
-			 * https://github.com/cisco/libsrtp/pull/170
-			 * https://github.com/cisco/libsrtp/pull/184
-			 * Furthermore, AES-GCM uses a shorter crypto-suite string which
-			 * causes Nokia phones based on Symbian/S60 to reject the whole
-			 * INVITE with status 500, even if a matching suite was offered.
-			 * AES-256 might just waste your processor cycles, especially if
-			 * your TLS transport is not secured with equivalent grade, see
-			 * https://security.stackexchange.com/q/61361
-			 * Therefore, AES-128 was preferred here.
-			 *
-			 * If you want to enable one of those defines, please, go for
-			 * CFLAGS='-DENABLE_SRTP_AES_GCM' ./configure && sudo make install
-			 */
-				{ len, 0, 30 },
-#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM)
-				{ AST_SRTP_CRYPTO_TAG_16, 0, AES_128_GCM_KEYSIZE_WSALT },
-#endif
-#if defined(HAVE_SRTP_256) && defined(ENABLE_SRTP_AES_256)
-				{ len, AST_SRTP_CRYPTO_AES_256, 46 },
-#endif
-#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM) && defined(ENABLE_SRTP_AES_256)
-				{ AST_SRTP_CRYPTO_TAG_16, AST_SRTP_CRYPTO_AES_256, AES_256_GCM_KEYSIZE_WSALT },
-#endif
-#if defined(HAVE_SRTP_192) && defined(ENABLE_SRTP_AES_192)
-				{ len, AST_SRTP_CRYPTO_AES_192, 38 },
-#endif
-			};
-			struct ast_sdp_srtp *tmp = srtp;
-			int i;
-
-			for (i = 0; i < ARRAY_LEN(attr); i++) {
-				if (attr[i][0]) {
-					ast_set_flag(tmp, attr[i][0]);
-				}
-				if (attr[i][1]) {
-					ast_set_flag(tmp, attr[i][1]);
-				}
-				tmp->crypto = sdp_crypto_alloc(attr[i][2]); /* key_len */
-				tmp->crypto->tag = (i + 1); /* tag starts at 1 */
-
-				if (i < ARRAY_LEN(attr) - 1) {
-					AST_LIST_NEXT(tmp, sdp_srtp_list) = ast_sdp_srtp_alloc();
-					tmp = AST_LIST_NEXT(tmp, sdp_srtp_list);
-				}
-			}
-		}
-	}
-
-	if (dtls_enabled) {
-		/* If DTLS-SRTP is enabled the key details will be pulled from TLS */
-		return NULL;
-	}
-
-	/* set the key length based on INVITE or settings */
-	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_80)) {
-		taglen = 80;
-	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_32)) {
-		taglen = 32;
-	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_16)) {
-		taglen = 16;
-	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_8)) {
-		taglen = 8;
-	} else {
-		taglen = default_taglen_32 ? 32 : 80;
-	}
-	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_256)) {
-		taglen |= 0x0200;
-	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_192)) {
-		taglen |= 0x0100;
-	}
-	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME)) {
-		taglen |= 0x0080;
-	}
-
-	if (srtp->crypto && (ast_sdp_crypto_build_offer(srtp->crypto, taglen) >= 0)) {
-		return srtp->crypto->a_crypto;
-	}
-
-	ast_log(LOG_WARNING, "No SRTP key management enabled\n");
-	return NULL;
+	return sdp_crypto_api->get_attr(srtp, dtls_enabled, default_taglen_32);
 }

 char *ast_sdp_get_rtp_profile(unsigned int sdes_active, struct ast_rtp_instance *instance, unsigned int using_avpf,
@@ -682,3 +122,19 @@ char *ast_sdp_get_rtp_profile(unsigned int sdes_active, struct ast_rtp_instance
 	}
 }

+int ast_sdp_crypto_register(struct ast_sdp_crypto_api *api)
+{
+	if (sdp_crypto_api) {
+		return -1;
+	}
+	sdp_crypto_api = api;
+	return 0;
+}
+
+void ast_sdp_crypto_unregister(struct ast_sdp_crypto_api *api)
+{
+	if (sdp_crypto_api == api) {
+		sdp_crypto_api = NULL;
+	}
+}
+
--- a/menuselect/menuselect.c
+++ b/menuselect/menuselect.c
@@ -398,6 +398,11 @@ static int process_xml_use_node(xmlNode *node, struct member *mem)
 	return process_xml_ref_node(node, mem, &mem->uses);
 }

+static int process_xml_member_data_node(xmlNode *node, struct member *mem)
+{
+	return 0;
+}
+
 static int process_xml_unknown_node(xmlNode *node, struct member *mem)
 {
 	fprintf(stderr, "Encountered unknown node: %s\n", node->name);
@@ -416,6 +421,7 @@ static const struct {
 	{ "depend",         process_xml_depend_node         },
 	{ "conflict",       process_xml_conflict_node       },
 	{ "use",            process_xml_use_node            },
+	{ "member_data",    process_xml_member_data_node    },
 };

 static node_handler lookup_node_handler(xmlNode *node)
--- a/menuselect/menuselect.h
+++ b/menuselect/menuselect.h
@@ -70,6 +70,8 @@ struct member {
 	const char *touch_on_change;
 	const char *support_level;
 	const char *replacement;
+	/*! member_data is just an opaque, member-specific string */
+	const char *member_data;
 	/*! This module is currently selected */
 	unsigned int enabled:1;
 	/*! This module was enabled when the config was loaded */
--- a/res/res_pjsip/config_global.c
+++ b/res/res_pjsip/config_global.c
@@ -106,6 +106,18 @@ static int global_apply(const struct ast_sorcery *sorcery, void *obj)
 	struct global_config *cfg = obj;
 	char max_forwards[10];

+	if (ast_strlen_zero(cfg->debug)) {
+		ast_log(LOG_ERROR,
+			"Global option 'debug' can't be empty.  Set it to a valid value or remove the entry to accept 'no' as the default\n");
+		return -1;
+	}
+
+	if (ast_strlen_zero(cfg->default_from_user)) {
+		ast_log(LOG_ERROR,
+			"Global option 'default_from_user' can't be empty.  Set it to a valid value or remove the entry to accept 'asterisk' as the default\n");
+		return -1;
+	}
+
 	snprintf(max_forwards, sizeof(max_forwards), "%u", cfg->max_forwards);

 	ast_sip_add_global_request_header("Max-Forwards", max_forwards, 1);
--- a/res/res_pjsip_caller_id.c
+++ b/res/res_pjsip_caller_id.c
@@ -413,7 +413,7 @@ static pjsip_fromto_hdr *create_new_id_hdr(const pj_str_t *hdr_name, pjsip_fromt
 	id_hdr = pjsip_from_hdr_create(tdata->pool);
 	id_hdr->type = PJSIP_H_OTHER;
 	pj_strdup(tdata->pool, &id_hdr->name, hdr_name);
-	id_hdr->sname.slen = 0;
+	id_hdr->sname = id_hdr->name;

 	id_name_addr = pjsip_uri_clone(tdata->pool, base->uri);
 	id_uri = pjsip_uri_get_uri(id_name_addr->uri);
--- a/res/res_pjsip_diversion.c
+++ b/res/res_pjsip_diversion.c
@@ -305,7 +305,7 @@ static void add_diversion_header(pjsip_tx_data *tdata, struct ast_party_redirect
 	hdr = pjsip_from_hdr_create(tdata->pool);
 	hdr->type = PJSIP_H_OTHER;
 	pj_strdup(tdata->pool, &hdr->name, &diversion_name);
-	hdr->sname.slen = 0;
+	hdr->sname = hdr->name;

 	name_addr = pjsip_uri_clone(tdata->pool, base);
 	uri = pjsip_uri_get_uri(name_addr->uri);
--- a/res/res_resolver_unbound.c
+++ b/res/res_resolver_unbound.c
@@ -79,6 +79,17 @@ ASTERISK_REGISTER_FILE()
 	</configInfo>
 ***/

+/*!
+ * Unbound versions <= 1.4.20 declare string function parameters as 'char *'
+ * but versions >= 1.4.21 declare them as 'const char *'.  Since CentOS6 is still
+ * at 1.4.20, we need to cast away the 'const' if we detect the earlier version.
+ */
+#ifdef HAVE_UNBOUND_CONST_PARAMS
+#define UNBOUND_CHAR const char
+#else
+#define UNBOUND_CHAR char
+#endif
+
 /*! \brief Structure for an unbound resolver */
 struct unbound_resolver {
 	/*! \brief Resolver context itself */
@@ -292,7 +303,7 @@ static int unbound_resolver_resolve(struct ast_dns_query *query)
 	data->resolver = ao2_bump(cfg->global->state->resolver);
 	ast_dns_resolver_set_data(query, data);

-	res = ub_resolve_async(data->resolver->context, ast_dns_query_get_name(query),
+	res = ub_resolve_async(data->resolver->context, (UNBOUND_CHAR *)ast_dns_query_get_name(query),
 		ast_dns_query_get_rr_type(query), ast_dns_query_get_rr_class(query),
 		ao2_bump(query), unbound_resolver_callback, &data->id);

@@ -404,7 +415,7 @@ static int unbound_config_preapply(struct unbound_config *cfg)
 	if (!strcmp(cfg->global->hosts, "system")) {
 		res = ub_ctx_hosts(cfg->global->state->resolver->context, NULL);
 	} else if (!ast_strlen_zero(cfg->global->hosts)) {
-		res = ub_ctx_hosts(cfg->global->state->resolver->context, cfg->global->hosts);
+		res = ub_ctx_hosts(cfg->global->state->resolver->context, (UNBOUND_CHAR *)cfg->global->hosts);
 	}

 	if (res) {
@@ -419,7 +430,7 @@ static int unbound_config_preapply(struct unbound_config *cfg)

 		it_nameservers = ao2_iterator_init(cfg->global->nameservers, 0);
 		while ((nameserver = ao2_iterator_next(&it_nameservers))) {
-			res = ub_ctx_set_fwd(cfg->global->state->resolver->context, nameserver);
+			res = ub_ctx_set_fwd(cfg->global->state->resolver->context, (UNBOUND_CHAR *)nameserver);

 			if (res) {
 				ast_log(LOG_ERROR, "Failed to add nameserver '%s' to unbound resolver: %s\n",
@@ -434,7 +445,7 @@ static int unbound_config_preapply(struct unbound_config *cfg)
 	if (!strcmp(cfg->global->resolv, "system")) {
 		res = ub_ctx_resolvconf(cfg->global->state->resolver->context, NULL);
 	} else if (!ast_strlen_zero(cfg->global->resolv)) {
-		res = ub_ctx_resolvconf(cfg->global->state->resolver->context, cfg->global->resolv);
+		res = ub_ctx_resolvconf(cfg->global->state->resolver->context, (UNBOUND_CHAR *)cfg->global->resolv);
 	}

 	if (res) {
@@ -444,7 +455,7 @@ static int unbound_config_preapply(struct unbound_config *cfg)
 	}

 	if (!ast_strlen_zero(cfg->global->ta_file)) {
-		res = ub_ctx_add_ta_file(cfg->global->state->resolver->context, cfg->global->ta_file);
+		res = ub_ctx_add_ta_file(cfg->global->state->resolver->context, (UNBOUND_CHAR *)cfg->global->ta_file);

 		if (res) {
 			ast_log(LOG_ERROR, "Failed to set trusted anchor file to '%s' in unbound resolver: %s\n",
@@ -740,13 +751,13 @@ static enum ast_test_result_state nominal_test(struct ast_test *test, resolve_fn
 	static const size_t V4_SIZE = sizeof(struct in_addr);
 	static const size_t V6_SIZE = sizeof(struct in6_addr);

-	static const char *DOMAIN1 = "goose.feathers";
-	static const char *DOMAIN2 = "duck.feathers";
+	static UNBOUND_CHAR *DOMAIN1 = "goose.feathers";
+	static UNBOUND_CHAR *DOMAIN2 = "duck.feathers";

-	static const char *ADDR1 = "127.0.0.2";
-	static const char *ADDR2 = "127.0.0.3";
-	static const char *ADDR3 = "::1";
-	static const char *ADDR4 = "127.0.0.4";
+	static UNBOUND_CHAR *ADDR1 = "127.0.0.2";
+	static UNBOUND_CHAR *ADDR2 = "127.0.0.3";
+	static UNBOUND_CHAR *ADDR3 = "::1";
+	static UNBOUND_CHAR *ADDR4 = "127.0.0.4";

 	char addr1_buf[V4_SIZE];
 	char addr2_buf[V4_SIZE];
@@ -786,7 +797,7 @@ static enum ast_test_result_state nominal_test(struct ast_test *test, resolve_fn
 	ub_ctx_zone_add(resolver->context, DOMAIN2, "static");

 	for (i = 0; i < ARRAY_LEN(records); ++i) {
-		ub_ctx_data_add(resolver->context, records[i].as_string);
+		ub_ctx_data_add(resolver->context, (UNBOUND_CHAR *)records[i].as_string);
 	}

 	for (i = 0; i < ARRAY_LEN(runs); ++i) {
@@ -808,7 +819,7 @@ static enum ast_test_result_state nominal_test(struct ast_test *test, resolve_fn

 cleanup:
 	for (i = 0; i < ARRAY_LEN(records); ++i) {
-		ub_ctx_data_remove(resolver->context, records[i].as_string);
+		ub_ctx_data_remove(resolver->context, (UNBOUND_CHAR *)records[i].as_string);
 	}
 	ub_ctx_zone_remove(resolver->context, DOMAIN1);
 	ub_ctx_zone_remove(resolver->context, DOMAIN2);
@@ -1012,10 +1023,10 @@ static enum ast_test_result_state off_nominal_test(struct ast_test *test,

 	static const size_t V4_SIZE = sizeof(struct in_addr);

-	static const char *DOMAIN1 = "goose.feathers";
-	static const char *DOMAIN2 = "duck.feathers";
+	static UNBOUND_CHAR *DOMAIN1 = "goose.feathers";
+	static UNBOUND_CHAR *DOMAIN2 = "duck.feathers";

-	static const char *ADDR1 = "127.0.0.2";
+	static UNBOUND_CHAR *ADDR1 = "127.0.0.2";

 	char addr1_buf[V4_SIZE];

@@ -1046,7 +1057,7 @@ static enum ast_test_result_state off_nominal_test(struct ast_test *test,
 	ub_ctx_zone_add(resolver->context, DOMAIN2, "static");

 	for (i = 0; i < ARRAY_LEN(records); ++i) {
-		ub_ctx_data_add(resolver->context, records[i].as_string);
+		ub_ctx_data_add(resolver->context, (UNBOUND_CHAR *)records[i].as_string);
 	}

 	for (i = 0; i < ARRAY_LEN(runs); ++i) {
@@ -1196,7 +1207,7 @@ AST_TEST_DEFINE(resolve_naptr)

 	const struct ast_dns_record *record;

-	static const char * DOMAIN1 = "goose.feathers";
+	static char * DOMAIN1 = "goose.feathers";
 	int i;
 	enum ast_test_result_state res = AST_TEST_PASS;

@@ -1234,7 +1245,7 @@ AST_TEST_DEFINE(resolve_naptr)
 	ub_ctx_zone_add(resolver->context, DOMAIN1, "static");

 	for (i = 0; i < ARRAY_LEN(records); ++i) {
-		ub_ctx_data_add(resolver->context, records[i].zone_entry);
+		ub_ctx_data_add(resolver->context, (UNBOUND_CHAR *)records[i].zone_entry);
 	}

 	if (ast_dns_resolve(DOMAIN1, ns_t_naptr, ns_c_in, &result)) {
@@ -1311,8 +1322,8 @@ AST_TEST_DEFINE(resolve_srv)
 	RAII_VAR(struct unbound_config *, cfg, NULL, ao2_cleanup);
 	RAII_VAR(struct ast_dns_result *, result, NULL, ast_dns_result_free);
 	const struct ast_dns_record *record;
-	static const char *DOMAIN1 = "taco.bananas";
-	static const char *DOMAIN1_SRV = "taco.bananas 12345 IN SRV 10 20 5060 sip.taco.bananas";
+	static UNBOUND_CHAR *DOMAIN1 = "taco.bananas";
+	static UNBOUND_CHAR *DOMAIN1_SRV = "taco.bananas 12345 IN SRV 10 20 5060 sip.taco.bananas";
 	enum ast_test_result_state res = AST_TEST_PASS;

 	switch (cmd) {
--- a/res/res_srtp.c
+++ b/res/res_srtp.c
@@ -39,6 +39,7 @@

 ASTERISK_REGISTER_FILE()

+#include <math.h>                       /* for pow */
 #include <srtp/srtp.h>
 #ifdef HAVE_OPENSSL
 #include <openssl/rand.h>
@@ -50,6 +51,7 @@ ASTERISK_REGISTER_FILE()
 #include "asterisk/frame.h"             /* for AST_FRIENDLY_OFFSET */
 #include "asterisk/logger.h"            /* for ast_log, ast_debug, etc */
 #include "asterisk/module.h"            /* for ast_module_info, etc */
+#include "asterisk/sdp_srtp.h"
 #include "asterisk/res_srtp.h"          /* for ast_srtp_cb, ast_srtp_suite, etc */
 #include "asterisk/rtp_engine.h"        /* for ast_rtp_engine_register_srtp, etc */
 #include "asterisk/utils.h"             /* for ast_free, ast_calloc */
@@ -578,10 +580,587 @@ static int ast_srtp_change_source(struct ast_srtp *srtp, unsigned int from_ssrc,
 	return 0;
 }

+struct ast_sdp_crypto {
+	char *a_crypto;
+	unsigned char local_key[SRTP_MAX_KEY_LEN];
+	int tag;
+	char local_key64[((SRTP_MAX_KEY_LEN) * 8 + 5) / 6 + 1];
+	unsigned char remote_key[SRTP_MAX_KEY_LEN];
+	int key_len;
+};
+
+static void res_sdp_crypto_dtor(struct ast_sdp_crypto *crypto)
+{
+	if (crypto) {
+		ast_free(crypto->a_crypto);
+		crypto->a_crypto = NULL;
+		ast_free(crypto);
+
+		ast_module_unref(ast_module_info->self);
+	}
+}
+
+static struct ast_sdp_crypto *crypto_init_keys(struct ast_sdp_crypto *p, const int key_len)
+{
+	unsigned char remote_key[key_len];
+
+	if (srtp_res.get_random(p->local_key, key_len) < 0) {
+		return NULL;
+	}
+
+	ast_base64encode(p->local_key64, p->local_key, key_len, sizeof(p->local_key64));
+
+	p->key_len = ast_base64decode(remote_key, p->local_key64, sizeof(remote_key));
+
+	if (p->key_len != key_len) {
+		ast_log(LOG_ERROR, "base64 encode/decode bad len %d != %d\n", p->key_len, key_len);
+		return NULL;
+	}
+
+	if (memcmp(remote_key, p->local_key, p->key_len)) {
+		ast_log(LOG_ERROR, "base64 encode/decode bad key\n");
+		return NULL;
+	}
+
+	ast_debug(1 , "local_key64 %s len %zu\n", p->local_key64, strlen(p->local_key64));
+
+	return p;
+}
+
+static struct ast_sdp_crypto *sdp_crypto_alloc(const int key_len)
+{
+	struct ast_sdp_crypto *p, *result;
+
+	if (!(p = ast_calloc(1, sizeof(*p)))) {
+		return NULL;
+	}
+	p->tag = 1;
+	ast_module_ref(ast_module_info->self);
+
+	/* default is a key which uses AST_AES_CM_128_HMAC_SHA1_xx */
+	result = crypto_init_keys(p, key_len);
+	if (!result) {
+		res_sdp_crypto_dtor(p);
+	}
+
+	return result;
+}
+
+static struct ast_sdp_crypto *res_sdp_crypto_alloc(void)
+{
+	return sdp_crypto_alloc(SRTP_MASTER_KEY_LEN);
+}
+
+static int res_sdp_crypto_build_offer(struct ast_sdp_crypto *p, int taglen)
+{
+	int res;
+
+	/* Rebuild the crypto line */
+	ast_free(p->a_crypto);
+	p->a_crypto = NULL;
+
+	if ((taglen & 0x007f) == 8) {
+		res = ast_asprintf(&p->a_crypto, "%d AEAD_AES_%d_GCM_%d inline:%s",
+			p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64);
+	} else if ((taglen & 0x007f) == 16) {
+		res = ast_asprintf(&p->a_crypto, "%d AEAD_AES_%d_GCM inline:%s",
+			p->tag, 128 + ((taglen & 0x0300) >> 2), p->local_key64);
+	} else if ((taglen & 0x0300) && !(taglen & 0x0080)) {
+		res = ast_asprintf(&p->a_crypto, "%d AES_%d_CM_HMAC_SHA1_%d inline:%s",
+			p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64);
+	} else {
+		res = ast_asprintf(&p->a_crypto, "%d AES_CM_%d_HMAC_SHA1_%d inline:%s",
+			p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64);
+	}
+	if (res == -1 || !p->a_crypto) {
+		ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
+		return -1;
+	}
+
+	ast_debug(1, "Crypto line: a=crypto:%s\n", p->a_crypto);
+
+	return 0;
+}
+
+static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, int key_len, unsigned long ssrc, int inbound)
+{
+	if (policy_res.set_master_key(policy, master_key, key_len, NULL, 0) < 0) {
+		return -1;
+	}
+
+	if (policy_res.set_suite(policy, suite_val)) {
+		ast_log(LOG_WARNING, "Could not set remote SRTP suite\n");
+		return -1;
+	}
+
+	policy_res.set_ssrc(policy, ssrc, inbound);
+
+	return 0;
+}
+
+static int crypto_activate(struct ast_sdp_crypto *p, int suite_val, unsigned char *remote_key, int key_len, struct ast_rtp_instance *rtp)
+{
+	struct ast_srtp_policy *local_policy = NULL;
+	struct ast_srtp_policy *remote_policy = NULL;
+	struct ast_rtp_instance_stats stats = {0,};
+	int res = -1;
+
+	if (!p) {
+		return -1;
+	}
+
+	if (!(local_policy = policy_res.alloc())) {
+		return -1;
+	}
+
+	if (!(remote_policy = policy_res.alloc())) {
+		goto err;
+	}
+
+	if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_LOCAL_SSRC)) {
+		goto err;
+	}
+
+	if (set_crypto_policy(local_policy, suite_val, p->local_key, key_len, stats.local_ssrc, 0) < 0) {
+		goto err;
+	}
+
+	if (set_crypto_policy(remote_policy, suite_val, remote_key, key_len, 0, 1) < 0) {
+		goto err;
+	}
+
+	/* Add the SRTP policies */
+	if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy, 0)) {
+		ast_log(LOG_WARNING, "Could not set SRTP policies\n");
+		goto err;
+	}
+
+	ast_debug(1 , "SRTP policy activated\n");
+	res = 0;
+
+err:
+	if (local_policy) {
+		policy_res.destroy(local_policy);
+	}
+
+	if (remote_policy) {
+		policy_res.destroy(remote_policy);
+	}
+
+	return res;
+}
+
+static int res_sdp_crypto_parse_offer(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *srtp, const char *attr)
+{
+	char *str = NULL;
+	char *tag = NULL;
+	char *suite = NULL;
+	char *key_params = NULL;
+	char *key_param = NULL;
+	char *session_params = NULL;
+	char *key_salt = NULL;       /* The actual master key and key salt */
+	char *lifetime = NULL;       /* Key lifetime (# of RTP packets) */
+	char *mki = NULL;            /* Master Key Index */
+	int found = 0;
+	int key_len_from_sdp;
+	int key_len_expected;
+	int tag_from_sdp;
+	int suite_val = 0;
+	unsigned char remote_key[SRTP_MAX_KEY_LEN];
+	int taglen;
+	double sdes_lifetime;
+	struct ast_sdp_crypto *crypto;
+	struct ast_sdp_srtp *tmp;
+
+	str = ast_strdupa(attr);
+
+	tag = strsep(&str, " ");
+	suite = strsep(&str, " ");
+	key_params = strsep(&str, " ");
+	session_params = strsep(&str, " ");
+
+	if (!tag || !suite) {
+		ast_log(LOG_WARNING, "Unrecognized crypto attribute a=%s\n", attr);
+		return -1;
+	}
+
+	/* RFC4568 9.1 - tag is 1-9 digits, greater than zero */
+	if (sscanf(tag, "%30d", &tag_from_sdp) != 1 || tag_from_sdp <= 0 || tag_from_sdp > 999999999) {
+		ast_log(LOG_WARNING, "Unacceptable a=crypto tag: %s\n", tag);
+		return -1;
+	}
+
+	if (!ast_strlen_zero(session_params)) {
+		ast_log(LOG_WARNING, "Unsupported crypto parameters: %s\n", session_params);
+		return -1;
+	}
+
+	/* On egress, Asterisk sent several crypto lines in the SIP/SDP offer
+	   The remote party might have choosen another line than the first */
+	for (tmp = srtp; tmp && tmp->crypto && tmp->crypto->tag != tag_from_sdp;) {
+		tmp = AST_LIST_NEXT(tmp, sdp_srtp_list);
+	}
+	if (tmp) { /* tag matched an already created crypto line */
+		unsigned int flags = tmp->flags;
+
+		/* Make that crypto line the head of the list, not by changing the
+		   list structure but by exchanging the content of the list members */
+		crypto = tmp->crypto;
+		tmp->crypto = srtp->crypto;
+		tmp->flags = srtp->flags;
+		srtp->crypto = crypto;
+		srtp->flags = flags;
+	} else {
+		crypto = srtp->crypto;
+		crypto->tag = tag_from_sdp;
+	}
+
+	if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
+		suite_val = AST_AES_CM_128_HMAC_SHA1_80;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
+		key_len_expected = 30;
+	} else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
+		suite_val = AST_AES_CM_128_HMAC_SHA1_32;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
+		key_len_expected = 30;
+#ifdef HAVE_SRTP_192
+	} else if (!strcmp(suite, "AES_192_CM_HMAC_SHA1_80")) {
+		suite_val = AST_AES_CM_192_HMAC_SHA1_80;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
+		key_len_expected = 38;
+	} else if (!strcmp(suite, "AES_192_CM_HMAC_SHA1_32")) {
+		suite_val = AST_AES_CM_192_HMAC_SHA1_32;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
+		key_len_expected = 38;
+	/* RFC used a different name while in draft, some still use that */
+	} else if (!strcmp(suite, "AES_CM_192_HMAC_SHA1_80")) {
+		suite_val = AST_AES_CM_192_HMAC_SHA1_80;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
+		key_len_expected = 38;
+	} else if (!strcmp(suite, "AES_CM_192_HMAC_SHA1_32")) {
+		suite_val = AST_AES_CM_192_HMAC_SHA1_32;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
+		key_len_expected = 38;
+#endif
+#ifdef HAVE_SRTP_256
+	} else if (!strcmp(suite, "AES_256_CM_HMAC_SHA1_80")) {
+		suite_val = AST_AES_CM_256_HMAC_SHA1_80;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
+		key_len_expected = 46;
+	} else if (!strcmp(suite, "AES_256_CM_HMAC_SHA1_32")) {
+		suite_val = AST_AES_CM_256_HMAC_SHA1_32;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
+		key_len_expected = 46;
+	/* RFC used a different name while in draft, some still use that */
+	} else if (!strcmp(suite, "AES_CM_256_HMAC_SHA1_80")) {
+		suite_val = AST_AES_CM_256_HMAC_SHA1_80;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
+		key_len_expected = 46;
+	} else if (!strcmp(suite, "AES_CM_256_HMAC_SHA1_32")) {
+		suite_val = AST_AES_CM_256_HMAC_SHA1_32;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
+		key_len_expected = 46;
+#endif
+#ifdef HAVE_SRTP_GCM
+	} else if (!strcmp(suite, "AEAD_AES_128_GCM")) {
+		suite_val = AST_AES_GCM_128;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_16);
+		key_len_expected = AES_128_GCM_KEYSIZE_WSALT;
+	} else if (!strcmp(suite, "AEAD_AES_256_GCM")) {
+		suite_val = AST_AES_GCM_256;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_16);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
+		key_len_expected = AES_256_GCM_KEYSIZE_WSALT;
+	/* RFC contained a (too) short auth tag for RTP media, some still use that */
+	} else if (!strcmp(suite, "AEAD_AES_128_GCM_8")) {
+		suite_val = AST_AES_GCM_128_8;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_8);
+		key_len_expected = AES_128_GCM_KEYSIZE_WSALT;
+	} else if (!strcmp(suite, "AEAD_AES_256_GCM_8")) {
+		suite_val = AST_AES_GCM_256_8;
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_8);
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
+		key_len_expected = AES_256_GCM_KEYSIZE_WSALT;
+#endif
+	} else {
+		ast_verb(1, "Unsupported crypto suite: %s\n", suite);
+		return -1;
+	}
+
+	while ((key_param = strsep(&key_params, ";"))) {
+		unsigned int n_lifetime;
+		char *method = NULL;
+		char *info = NULL;
+
+		method = strsep(&key_param, ":");
+		info = strsep(&key_param, ";");
+		sdes_lifetime = 0;
+
+		if (strcmp(method, "inline")) {
+			continue;
+		}
+
+		key_salt = strsep(&info, "|");
+
+		/* The next parameter can be either lifetime or MKI */
+		lifetime = strsep(&info, "|");
+		if (!lifetime) {
+			found = 1;
+			break;
+		}
+
+		mki = strchr(lifetime, ':');
+		if (mki) {
+			mki = lifetime;
+			lifetime = NULL;
+		} else {
+			mki = strsep(&info, "|");
+		}
+
+		if (mki && *mki != '1') {
+			ast_log(LOG_NOTICE, "Crypto MKI handling is not supported: ignoring attribute %s\n", attr);
+			continue;
+		}
+
+		if (lifetime) {
+			if (!strncmp(lifetime, "2^", 2)) {
+				char *lifetime_val = lifetime + 2;
+
+				/* Exponential lifetime */
+				if (sscanf(lifetime_val, "%30u", &n_lifetime) != 1) {
+					ast_log(LOG_NOTICE, "Failed to parse lifetime value in crypto attribute: %s\n", attr);
+					continue;
+				}
+
+				if (n_lifetime > 48) {
+					/* Yeah... that's a bit big. */
+					ast_log(LOG_NOTICE, "Crypto lifetime exponent of '%u' is a bit large; using 48\n", n_lifetime);
+					n_lifetime = 48;
+				}
+				sdes_lifetime = pow(2, n_lifetime);
+			} else {
+				/* Decimal lifetime */
+				if (sscanf(lifetime, "%30u", &n_lifetime) != 1) {
+					ast_log(LOG_NOTICE, "Failed to parse lifetime value in crypto attribute: %s\n", attr);
+					continue;
+				}
+				sdes_lifetime = n_lifetime;
+			}
+
+			/* Accept anything above 10 hours. Less than 10; reject. */
+			if (sdes_lifetime < 1800000) {
+				ast_log(LOG_NOTICE, "Rejecting crypto attribute '%s': lifetime '%f' too short\n", attr, sdes_lifetime);
+				continue;
+			}
+		}
+
+		ast_debug(2, "Crypto attribute '%s' accepted with lifetime '%f', MKI '%s'\n",
+			attr, sdes_lifetime, mki ? mki : "-");
+
+		found = 1;
+		break;
+	}
+
+	if (!found) {
+		ast_log(LOG_NOTICE, "SRTP crypto offer not acceptable: '%s'\n", attr);
+		return -1;
+	}
+
+	key_len_from_sdp = ast_base64decode(remote_key, key_salt, sizeof(remote_key));
+	if (key_len_from_sdp != key_len_expected) {
+		ast_log(LOG_WARNING, "SRTP descriptions key length is '%d', not '%d'\n",
+			key_len_from_sdp, key_len_expected);
+		return -1;
+	}
+
+	/* on default, the key is 30 (AES-128); throw that away (only) when the suite changed actually */
+	/* ingress: optional, but saves one expensive call to get_random(.) */
+	/*  egress: required, because the local key was communicated before the remote key is processed */
+	if (crypto->key_len != key_len_from_sdp) {
+		if (!crypto_init_keys(crypto, key_len_from_sdp)) {
+			return -1;
+		}
+	} else if (!memcmp(crypto->remote_key, remote_key, key_len_from_sdp)) {
+		ast_debug(1, "SRTP remote key unchanged; maintaining current policy\n");
+		ast_set_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
+		return 0;
+	}
+
+	if (key_len_from_sdp > sizeof(crypto->remote_key)) {
+		ast_log(LOG_ERROR,
+			"SRTP key buffer is %zu although it must be at least %d bytes\n",
+			sizeof(crypto->remote_key), key_len_from_sdp);
+		return -1;
+	}
+	memcpy(crypto->remote_key, remote_key, key_len_from_sdp);
+
+	if (crypto_activate(crypto, suite_val, remote_key, key_len_from_sdp, rtp) < 0) {
+		return -1;
+	}
+
+	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_32)) {
+		taglen = 32;
+	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_16)) {
+		taglen = 16;
+	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_8)) {
+		taglen = 8;
+	} else {
+		taglen = 80;
+	}
+	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_256)) {
+		taglen |= 0x0200;
+	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_192)) {
+		taglen |= 0x0100;
+	}
+	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME)) {
+		taglen |= 0x0080;
+	}
+
+	/* Finally, rebuild the crypto line */
+	if (res_sdp_crypto_build_offer(crypto, taglen)) {
+		return -1;
+	}
+
+	ast_set_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
+	return 0;
+}
+
+static const char *res_sdp_srtp_get_attr(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
+{
+	int taglen;
+
+	if (!srtp) {
+		return NULL;
+	}
+
+	/* Set encryption properties */
+	if (!srtp->crypto) {
+		if (AST_LIST_NEXT(srtp, sdp_srtp_list)) {
+			srtp->crypto = res_sdp_crypto_alloc();
+			ast_log(LOG_ERROR, "SRTP SDP list was not empty\n");
+		} else {
+			const int len = default_taglen_32 ? AST_SRTP_CRYPTO_TAG_32 : AST_SRTP_CRYPTO_TAG_80;
+			const int attr[][3] = {
+			/* This array creates the following list:
+			 * a=crypto:1 AES_CM_128_HMAC_SHA1_ ...
+			 * a=crypto:2 AEAD_AES_128_GCM ...
+			 * a=crypto:3 AES_256_CM_HMAC_SHA1_ ...
+			 * a=crypto:4 AEAD_AES_256_GCM ...
+			 * a=crypto:5 AES_192_CM_HMAC_SHA1_ ...
+			 * something like 'AEAD_AES_192_GCM' is not specified by the RFCs
+			 *
+			 * If you want to prefer another crypto suite or you want to
+			 * exclude a suite, change this array and recompile Asterisk.
+			 * This list cannot be changed from rtp.conf because you should
+			 * know what you are doing. Especially AES-192 and AES-GCM are
+			 * broken in many VoIP clients, see
+			 * https://github.com/cisco/libsrtp/pull/170
+			 * https://github.com/cisco/libsrtp/pull/184
+			 * Furthermore, AES-GCM uses a shorter crypto-suite string which
+			 * causes Nokia phones based on Symbian/S60 to reject the whole
+			 * INVITE with status 500, even if a matching suite was offered.
+			 * AES-256 might just waste your processor cycles, especially if
+			 * your TLS transport is not secured with equivalent grade, see
+			 * https://security.stackexchange.com/q/61361
+			 * Therefore, AES-128 was preferred here.
+			 *
+			 * If you want to enable one of those defines, please, go for
+			 * CFLAGS='-DENABLE_SRTP_AES_GCM' ./configure && sudo make install
+			 */
+				{ len, 0, 30 },
+#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM)
+				{ AST_SRTP_CRYPTO_TAG_16, 0, AES_128_GCM_KEYSIZE_WSALT },
+#endif
+#if defined(HAVE_SRTP_256) && defined(ENABLE_SRTP_AES_256)
+				{ len, AST_SRTP_CRYPTO_AES_256, 46 },
+#endif
+#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM) && defined(ENABLE_SRTP_AES_256)
+				{ AST_SRTP_CRYPTO_TAG_16, AST_SRTP_CRYPTO_AES_256, AES_256_GCM_KEYSIZE_WSALT },
+#endif
+#if defined(HAVE_SRTP_192) && defined(ENABLE_SRTP_AES_192)
+				{ len, AST_SRTP_CRYPTO_AES_192, 38 },
+#endif
+			};
+			struct ast_sdp_srtp *tmp = srtp;
+			int i;
+
+			for (i = 0; i < ARRAY_LEN(attr); i++) {
+				if (attr[i][0]) {
+					ast_set_flag(tmp, attr[i][0]);
+				}
+				if (attr[i][1]) {
+					ast_set_flag(tmp, attr[i][1]);
+				}
+				tmp->crypto = sdp_crypto_alloc(attr[i][2]); /* key_len */
+				tmp->crypto->tag = (i + 1); /* tag starts at 1 */
+
+				if (i < ARRAY_LEN(attr) - 1) {
+					AST_LIST_NEXT(tmp, sdp_srtp_list) = ast_sdp_srtp_alloc();
+					tmp = AST_LIST_NEXT(tmp, sdp_srtp_list);
+				}
+			}
+		}
+	}
+
+	if (dtls_enabled) {
+		/* If DTLS-SRTP is enabled the key details will be pulled from TLS */
+		return NULL;
+	}
+
+	/* set the key length based on INVITE or settings */
+	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_80)) {
+		taglen = 80;
+	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_32)) {
+		taglen = 32;
+	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_16)) {
+		taglen = 16;
+	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_8)) {
+		taglen = 8;
+	} else {
+		taglen = default_taglen_32 ? 32 : 80;
+	}
+	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_256)) {
+		taglen |= 0x0200;
+	} else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_192)) {
+		taglen |= 0x0100;
+	}
+	if (ast_test_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME)) {
+		taglen |= 0x0080;
+	}
+
+	if (srtp->crypto && (res_sdp_crypto_build_offer(srtp->crypto, taglen) >= 0)) {
+		return srtp->crypto->a_crypto;
+	}
+
+	ast_log(LOG_WARNING, "No SRTP key management enabled\n");
+	return NULL;
+}
+
+static struct ast_sdp_crypto_api res_sdp_crypto_api = {
+	.dtor = res_sdp_crypto_dtor,
+	.alloc = res_sdp_crypto_alloc,
+	.build_offer = res_sdp_crypto_build_offer,
+	.parse_offer = res_sdp_crypto_parse_offer,
+	.get_attr = res_sdp_srtp_get_attr,
+};
+
 static void res_srtp_shutdown(void)
 {
-	srtp_install_event_handler(NULL);
+	ast_sdp_crypto_unregister(&res_sdp_crypto_api);
 	ast_rtp_engine_unregister_srtp();
+	srtp_install_event_handler(NULL);
 #ifdef HAVE_SRTP_SHUTDOWN
 	srtp_shutdown();
 #endif
@@ -607,6 +1186,12 @@ static int res_srtp_init(void)
 		return -1;
 	}

+	if (ast_sdp_crypto_register(&res_sdp_crypto_api)) {
+		ast_log(AST_LOG_WARNING, "Failed to register SDP SRTP crypto API\n");
+		res_srtp_shutdown();
+		return -1;
+	}
+
 	g_initialized = 1;
 	return 0;
 }
--- a/third-party/pjproject/Makefile.rules
+++ b/third-party/pjproject/Makefile.rules
@@ -1,9 +1,34 @@
 PJPROJECT_URL = http://www.pjsip.org/release/$(PJPROJECT_VERSION)

 # Even though we're not installing pjproject, we're setting prefix to /opt/pjproject to be safe
-PJPROJECT_CONFIG_OPTS = --prefix=/opt/pjproject --disable-speex-codec --disable-speex-aec \
-	--disable-gsm-codec --disable-video --disable-v4l2 --disable-sound --disable-opencore-amr \
-	--disable-ilbc-codec --without-libyuv --disable-g7221-codec --disable-resample
+
+PJPROJECT_CONFIG_OPTS = --prefix=/opt/pjproject \
+	--disable-speex-codec \
+	--disable-speex-aec \
+	--disable-speex-aec \
+	--disable-gsm-codec \
+	--disable-ilbc-codec \
+	--disable-l16-codec \
+	--disable-g711-codec \
+	--disable-g722-codec \
+	--disable-g7221-codec \
+	--disable-opencore-amr \
+	--disable-webrtc \
+	--disable-silk \
+	--disable-opus \
+	--disable-video \
+	--disable-v4l2 \
+	--disable-sound \
+	--disable-ext-sound \
+	--disable-oss \
+	--disable-sdl \
+	--disable-libyuv \
+	--disable-resample \
+	--disable-ffmpeg \
+	--disable-openh264 \
+	--disable-ipp \
+	--without-external-pa \
+	--with-external-srtp

 ifeq ($(shell uname -s),Linux)
   PJPROJECT_CONFIG_OPTS +=  --enable-epoll
--- a/third-party/pjproject/patches/0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch
+++ b/third-party/pjproject/patches/0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch
@@ -1,73 +0,0 @@
-From a5030c9b33b2c936879fbacb1d2ea5edc2979181 Mon Sep 17 00:00:00 2001
-From: George Joseph <gjoseph@digium.com>
-Date: Sat, 18 Jun 2016 10:14:34 -0600
-Subject: [PATCH] evsub:  Add APIs to add/decrement an event  subscription's
- group lock
-
-These APIs can be used to ensure that the evsub isn't destroyed before
-an application is finished using it.
---
- pjsip/include/pjsip-simple/evsub.h | 20 ++++++++++++++++++++
- pjsip/src/pjsip-simple/evsub.c     | 14 ++++++++++++++
- 2 files changed, 34 insertions(+)
-
-diff --git a/pjsip/include/pjsip-simple/evsub.h b/pjsip/include/pjsip-simple/evsub.h
-index 2dc4d69..31f85f8 100644
--- a/pjsip/include/pjsip-simple/evsub.h
-+++ b/pjsip/include/pjsip-simple/evsub.h
-@@ -490,6 +490,26 @@ PJ_DECL(void) pjsip_evsub_set_mod_data( pjsip_evsub *sub, unsigned mod_id,
- PJ_DECL(void*) pjsip_evsub_get_mod_data( pjsip_evsub *sub, unsigned mod_id );
- 
- 
-+/**
-+ * Increment the event subscription's group lock.
-+ *
-+ * @param sub		The server subscription instance.
-+ *
-+ * @return		PJ_SUCCESS on success.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_add_ref(pjsip_evsub *sub);
-+
-+
-+/**
-+ * Decrement the event subscription's group lock.
-+ *
-+ * @param sub		The server subscription instance.
-+ *
-+ * @return		PJ_SUCCESS on success.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_dec_ref(pjsip_evsub *sub);
-+
-+
- 
- PJ_END_DECL
- 
-diff --git a/pjsip/src/pjsip-simple/evsub.c b/pjsip/src/pjsip-simple/evsub.c
-index 7cd8859..68a9564 100644
--- a/pjsip/src/pjsip-simple/evsub.c
-+++ b/pjsip/src/pjsip-simple/evsub.c
-@@ -831,7 +831,21 @@ static pj_status_t evsub_create( pjsip_dialog *dlg,
-     return PJ_SUCCESS;
- }
- 
-+/*
-+ * Increment the event subscription's group lock.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_add_ref(pjsip_evsub *sub)
-+{
-+    return pj_grp_lock_add_ref(sub->grp_lock);
-+}
- 
-+/*
-+ * Decrement the event subscription's group lock.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_dec_ref(pjsip_evsub *sub)
-+{
-+    return pj_grp_lock_dec_ref(sub->grp_lock);
-+}
- 
- /*
-  * Create client subscription session.
-- 
-2.5.5
-
--- a/third-party/pjproject/patches/0001-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch
+++ b/third-party/pjproject/patches/0001-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch
@@ -1,48 +0,0 @@
-From b7cb93b0e1729589a71e8b30d9a9893f0918e2a2 Mon Sep 17 00:00:00 2001
-From: George Joseph <george.joseph@fairview5.com>
-Date: Mon, 30 May 2016 11:58:22 -0600
-Subject: [PATCH] sip_transport_tcp/tls:  Set factory on transports created
- from accept
-
-The ability to re-use tcp and tls transports when a factory is
-specified now depends on transport->factory being set which is a new field
-in 2.5.  This was being set only on new outgoing sockets not on
-incoming sockets.  The result was that a client REGISTER created a new
-socket but without the factory set, the next outgoing request to the
-client, OPTIONS, INVITE, etc, would attempt to create another socket
-which the client would refuse.
-
-This patch sets the factory on transports created as a result of an
-accept.
---
- pjsip/src/pjsip/sip_transport_tcp.c | 1 +
- pjsip/src/pjsip/sip_transport_tls.c | 1 +
- 2 files changed, 2 insertions(+)
-
-diff --git a/pjsip/src/pjsip/sip_transport_tcp.c b/pjsip/src/pjsip/sip_transport_tcp.c
-index 1bbb324..00eb8fc 100644
--- a/pjsip/src/pjsip/sip_transport_tcp.c
-+++ b/pjsip/src/pjsip/sip_transport_tcp.c
-@@ -713,6 +713,7 @@ static pj_status_t tcp_create( struct tcp_listener *listener,
-     tcp->base.send_msg = &tcp_send_msg;
-     tcp->base.do_shutdown = &tcp_shutdown;
-     tcp->base.destroy = &tcp_destroy_transport;
-+    tcp->base.factory = &listener->factory;
- 
-     /* Create group lock */
-     status = pj_grp_lock_create(pool, NULL, &tcp->grp_lock);
-diff --git a/pjsip/src/pjsip/sip_transport_tls.c b/pjsip/src/pjsip/sip_transport_tls.c
-index a83ac32..36ee70d 100644
--- a/pjsip/src/pjsip/sip_transport_tls.c
-+++ b/pjsip/src/pjsip/sip_transport_tls.c
-@@ -742,6 +742,7 @@ static pj_status_t tls_create( struct tls_listener *listener,
-     tls->base.send_msg = &tls_send_msg;
-     tls->base.do_shutdown = &tls_shutdown;
-     tls->base.destroy = &tls_destroy_transport;
-+    tls->base.factory = &listener->factory;
- 
-     tls->ssock = ssock;
- 
-- 
-2.5.5
-
--- a/third-party/pjproject/patches/0002-Fix-1946-Avoid-deinitialization-of-uninitialized-cli.patch
+++ b/third-party/pjproject/patches/0002-Fix-1946-Avoid-deinitialization-of-uninitialized-cli.patch
@@ -0,0 +1,56 @@
+From 33fd755e819dc85a96718abc0ae26a9b46f14800 Mon Sep 17 00:00:00 2001
+From: nanang <nanang@localhost>
+Date: Thu, 28 Jul 2016 08:21:45 +0000
+Subject: [PATCH 2/3] Fix #1946: Avoid deinitialization of uninitialized client
+ auth session.
+
+---
+ pjsip/src/pjsip/sip_dialog.c | 18 ++++++------------
+ 1 file changed, 6 insertions(+), 12 deletions(-)
+
+diff --git a/pjsip/src/pjsip/sip_dialog.c b/pjsip/src/pjsip/sip_dialog.c
+index f03885d..421ddc4 100644
+--- a/pjsip/src/pjsip/sip_dialog.c
+++ b/pjsip/src/pjsip/sip_dialog.c
+@@ -92,6 +92,12 @@ static pj_status_t create_dialog( pjsip_user_agent *ua,
+     pj_list_init(&dlg->inv_hdr);
+     pj_list_init(&dlg->rem_cap_hdr);
+ 
+    /* Init client authentication session. */
+    status = pjsip_auth_clt_init(&dlg->auth_sess, dlg->endpt,
+				 dlg->pool, 0);
+    if (status != PJ_SUCCESS)
+	goto on_error;
+
+     status = pj_mutex_create_recursive(pool, dlg->obj_name, &dlg->mutex_);
+     if (status != PJ_SUCCESS)
+ 	goto on_error;
+@@ -283,12 +289,6 @@ PJ_DEF(pj_status_t) pjsip_dlg_create_uac( pjsip_user_agent *ua,
+     /* Initial route set is empty. */
+     pj_list_init(&dlg->route_set);
+ 
+-    /* Init client authentication session. */
+-    status = pjsip_auth_clt_init(&dlg->auth_sess, dlg->endpt,
+-				 dlg->pool, 0);
+-    if (status != PJ_SUCCESS)
+-	goto on_error;
+-
+     /* Register this dialog to user agent. */
+     status = pjsip_ua_register_dlg( ua, dlg );
+     if (status != PJ_SUCCESS)
+@@ -506,12 +506,6 @@ pj_status_t create_uas_dialog( pjsip_user_agent *ua,
+     }
+     dlg->route_set_frozen = PJ_TRUE;
+ 
+-    /* Init client authentication session. */
+-    status = pjsip_auth_clt_init(&dlg->auth_sess, dlg->endpt,
+-				 dlg->pool, 0);
+-    if (status != PJ_SUCCESS)
+-	goto on_error;
+-
+     /* Increment the dialog's lock since tsx may cause the dialog to be
+      * destroyed prematurely (such as in case of transport error).
+      */
+-- 
+2.7.4
+
--- a/third-party/pjproject/patches/config_site.h
+++ b/third-party/pjproject/patches/config_site.h
@@ -19,7 +19,7 @@
 #define PJ_SCANNER_USE_BITWISE	0
 #define PJ_OS_HAS_CHECK_STACK	0
 #define PJ_LOG_MAX_LEVEL		3
-#define PJ_ENABLE_EXTRA_CHECK	0
+#define PJ_ENABLE_EXTRA_CHECK	1
 #define PJSIP_MAX_TSX_COUNT		((64*1024)-1)
 #define PJSIP_MAX_DIALOG_COUNT	((64*1024)-1)
 #define PJSIP_UDP_SO_SNDBUF_SIZE	(512*1024)
--- a/third-party/versions.mak
+++ b/third-party/versions.mak
@@ -1,2 +1,2 @@

-PJPROJECT_VERSION = 2.5
+PJPROJECT_VERSION = 2.5.5
Author	SHA1	Message	Date
Joshua Colp	9528429f4c	ChangeLog: Updated for 14.0.0-beta2	2016-08-29 07:30:23 -05:00
Joshua Colp	9cdf44668d	Release summaries: Add summaries for 14.0.0-beta2	2016-08-29 07:29:55 -05:00
Joshua Colp	73d39f2029	Release summaries: Remove previous versions	2016-08-29 07:29:29 -05:00
Joshua Colp	e8a97775ee	.version: Update for 14.0.0-beta2	2016-08-29 07:29:29 -05:00
Joshua Colp	345409825a	.lastclean: Update for 14.0.0-beta2	2016-08-29 07:29:29 -05:00
Joshua Colp	105c1168f7	realtime: Add database scripts for 14.0.0-beta2	2016-08-29 07:29:28 -05:00
Joshua Colp	8927b52634	alembic: Fix downgrade path. The 3772f8f828da version was referencing a previous version that did not exist in the 14.0 branch. It has been fixed to reference the correct previous version. Change-Id: I004d0fcfdfe1d1bb6f01c6dac2b69f6b1f40ae51	2016-08-29 11:31:05 +00:00
Joshua Colp	fc68258037	Merge "res_pjsip: Fail global load if debug or default_from_user are empty" into 14.0	2016-08-12 16:35:21 -05:00
George Joseph	9a95c6dea3	res_pjsip: Fail global load if debug or default_from_user are empty If debug was specified in the global configuration but left blank, the logger would treat it as a wildcard and log all hosts. If default_from_user was empty, a crash would result. The global apply handler now checks for empty strings. ASTERISK-26239 #close ASTERISK-26238 #close Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336	2016-08-12 07:41:51 -05:00
George Joseph	aaee8160bc	res_pjsip_caller_id: Copy header name to short header name When compact_headers was set, we were sending a zero-length header name for PAI and RPID because we always forced the short header name length to 0. We did this because we cloned the header from "From" and wanted to clear "f" from the sname. By cloning however, we bypass pjproject's automatic logic that sets sname to name if there's no compact form of the header, which there isn't for PAI and RPID. So now we force sname to be the same as name right after we set name. res_pjsip_diversion needed the same treatment for the Diversion header. ASTERISK-26241 #close Change-Id: I633ec139630cd83809aae00336cee4a10077e467	2016-08-12 06:08:19 -05:00
Joshua Colp	72e2d978ac	Merge "alembic: add auth_username to endpoint's identify_by enum" into 14.0	2016-08-12 04:47:14 -05:00
Joshua Colp	1877d36c95	Merge "res_resolver_unbound: Allow compilation with libunbound version < 1.5" into 14.0	2016-08-11 16:11:31 -05:00
George Joseph	7af0eac02a	autohints: Update CHANGES and extensions.conf.sample Make it clear that we're talking about device state hints and add an entry to the sample config. Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433	2016-08-11 15:00:44 -05:00
Kevin Harwell	ef0bf47bb3	alembic: add auth_username to endpoint's identify_by enum A new identify_by option was added recently, auth_username. However, this setting was not added as an allowable choice in the database enumeration value. This patch updates the current enumeration, adding in the new setting. ASTERISK-26268 #close Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8	2016-08-11 11:00:50 -05:00
zuul	be8fa4a81d	Merge "res_srtp: Move SDP SRTP code from the core to res_srtp." into 14.0	2016-08-11 06:57:06 -05:00
Richard Mudgett	a1d6b14c40	res_srtp: Move SDP SRTP code from the core to res_srtp. A patch made to the master branch (Now the 14 branch) inadvertently made libsrtp a required dependency in order to compile Asterisk. Rather than create dummy defines to substitute for the defines supplied by libsrtp when libsrtp is not available, most of the code in sdp_srtp.c is moved into res_srtp.c. This gets more code out of Asterisk's core that isn't used when SRTP is not available. This also makes another inadvertent required dependency on libsrtp by Asterisk's core unlikely. ASTERISK-26253 #close Reported by: Ben Merrills Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7	2016-08-10 17:43:45 -05:00
Kevin Harwell	a783e1e60d	alembic/sqlalchemy: auto increment only allowed on a single column The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b	2016-08-10 13:50:26 -05:00
George Joseph	9c56f798f6	res_resolver_unbound: Allow compilation with libunbound version < 1.5 libunbound at version 1.4.20 (which CentOS still uses) declared all of their string function parameters as as 'char '. 1.4.21 changed them all to 'const char '. Thankfully 1.4.21 also introduced the UNBOUND_VERSION_MAJOR define so configure now checks for that and sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks that and casts away the 'const' if it's not set. Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and Fedora24 (1.5.4). There are a few failing tests to be addressed though. ASTERISK-26283 #close Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148	2016-08-10 12:09:37 -05:00
zuul	01ee54ea1c	Merge "menuselect: Add an opaque "member_data" string to the acceptable xml" into 14.0	2016-08-02 18:20:07 -05:00
George Joseph	1ad00c1c30	menuselect: Add an opaque "member_data" string to the acceptable xml Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe	2016-08-02 09:56:56 -05:00
George Joseph	815b6f72f8	pjproject_bundled: Update for pjproject 2.5.5 Add more --disable-* switches to Makefile.rules including --disable-opus which was causing bundled pjproject to fail with "undefined reference" errors in libasteriskpj. Changed PJ_ENABLE_EXTRA_CHECK to 1. Removed 2 obsolete patches and added a new one. The new one was merged by Teluu on 6/27/2016. ASTERISK-26148 #close Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063 (cherry picked from commit `4cf02b5584`)	2016-08-02 09:56:20 -05:00
Mark Michelson	c95b611a73	Remove SILK payload mappings from Asterisk core. SILK is a bit of a hog when it comes to using up our limited number of dynamic payload types in the RTP engine. By freeing up four slots, it allows for other codecs to potentially take the place. Now, codec_silk.so will dynamically use the payload slots in the RTP engine when it loads. A better fix would be make RTP dynamic payload types actually dynamic. However, at this stage of Asterisk 14 development, this is a risky move that would be imprudent. Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612	2016-08-01 11:40:15 -05:00
Kevin Harwell	bc94ccbcdd	rtp_engine: Failed assertion and wrong name given for codec Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c	2016-07-27 12:49:28 -05:00