Update for 13.16.0-rc2

Merge "chan_sip: Better ICE handling for RTCP-MUX" into 13.16
chan_sip: Better ICE handling for RTCP-MUX
2025-09-02 19:16:15 +00:00 · 2017-05-24 11:55:58 -05:00 · 2017-05-24 10:36:22 -05:00 · 2017-05-23 18:08:02 -05:00 · 2017-05-23 18:03:56 -05:00 · 2017-05-22 15:20:11 -05:00
19 changed files with 57154 additions and 42 deletions
--- a/.lastclean
+++ b/.lastclean
@@ -0,0 +1 @@
+40
--- a/.version
+++ b/.version
@@ -0,0 +1 @@
+13.16.0-rc2
--- a/51034
+++ b/51034
--- a/asterisk-13.16.0-rc2-summary.html
+++ b/asterisk-13.16.0-rc2-summary.html
@@ -0,0 +1,18 @@
+<!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-13.16.0-rc2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.16.0-rc2</h3><h3 align="center">Date: 2017-05-24</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="#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 release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-13.16.0-rc1.</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%">1 Sean Bright <sean.bright@gmail.com><br/>1 Kevin Harwell <kharwell@digium.com><br/></td><td width="33%"><td width="33%">1 Stefan Engström <stefanen@kth.se><br/>1 Javier Riveros  <goseeped@gmail.com><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: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0e5b5815c14d0e532ae00026ac5e440752d46f5">[f0e5b5815c]</a> Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX</li>
+</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0e5b5815c14d0e532ae00026ac5e440752d46f5">[f0e5b5815c]</a> Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a26431a198c58b8d7a7dcffa957705ee5f635848">[a26431a198]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</li>
+</ul><br><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a26431a198c58b8d7a7dcffa957705ee5f635848">[a26431a198]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</li>
+</ul><br><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>chan_sip.c |   36 +++++++++++++++++++++++++++++-------
+1 file changed, 29 insertions(+), 7 deletions(-)</pre><br></html>
--- a/asterisk-13.16.0-rc2-summary.txt
+++ b/asterisk-13.16.0-rc2-summary.txt
@@ -0,0 +1,103 @@
+                                Release Summary
+
+                              asterisk-13.16.0-rc2
+
+                                Date: 2017-05-24
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Closed Issues
+    4. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This release is a point release of an existing major version. The changes
+   included were made to address problems that have been identified in this
+   release series, or are minor, backwards compatible new features or
+   improvements. Users should be able to safely upgrade to this version if
+   this release series is already in use. Users considering upgrading from a
+   previous version are strongly encouraged to review the UPGRADE.txt
+   document as well as the CHANGES document for information about upgrading
+   to this release series.
+
+   The data in this summary reflects changes that have been made since the
+   previous release, asterisk-13.16.0-rc1.
+
+     ----------------------------------------------------------------------
+
+                                  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                
+   1 Sean Bright                                     1 Stefan EngstrAP:m      
+   1 Kevin Harwell                                   1 Javier Riveros         
+
+     ----------------------------------------------------------------------
+
+                                 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: Channels/chan_sip/General
+
+   ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion
+   failure/delay if client offers rtcp-mux as negotiable
+   Reported by: Stefan EngstrAP:m
+     * [f0e5b5815c] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
+
+    Category: Resources/res_rtp_asterisk
+
+   ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion
+   failure/delay if client offers rtcp-mux as negotiable
+   Reported by: Stefan EngstrAP:m
+     * [f0e5b5815c] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
+   ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
+   authentication failure 10 or 110
+   Reported by: Javier Riveros
+     * [a26431a198] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
+       wrong srtp unprotecting algorithm
+
+    Category: Resources/res_srtp
+
+   ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
+   authentication failure 10 or 110
+   Reported by: Javier Riveros
+     * [a26431a198] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
+       wrong srtp unprotecting algorithm
+
+     ----------------------------------------------------------------------
+
+                                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.
+
+ chan_sip.c |   36 +++++++++++++++++++++++++++++-------
+ 1 file changed, 29 insertions(+), 7 deletions(-)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1210,7 +1210,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 static int process_sdp_o(const char *o, struct sip_pvt *p);
 static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
 static int process_sdp_a_sendonly(const char *a, int *sendonly);
-static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
+static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance, int rtcp_mux);
 static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested);
 static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
 static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec);
@@ -10144,6 +10144,24 @@ static void set_ice_components(struct sip_pvt *p, struct ast_rtp_instance *insta
 	}
 }

+static int has_media_level_attribute(int start, struct sip_request *req, const char *attr)
+{
+	int next = start;
+	char type;
+	const char *value;
+
+	/* We don't care about the return result here */
+	get_sdp_iterate(&next, req, "m");
+
+	while ((type = get_sdp_line(&start, next, req, &value)) != '\0') {
+		if (type == 'a' && !strcasecmp(value, attr)) {
+			return 1;
+		}
+	}
+
+	return 0;
+}
+
 /*! \brief Process SIP SDP offer, select formats and activate media channels
 	If offer is rejected, we will not change any properties of the call
 	Return 0 on success, a negative value on errors.
@@ -10286,13 +10304,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			else if (process_sdp_a_image(value, p))
 				processed = TRUE;

-			if (process_sdp_a_ice(value, p, p->rtp)) {
+			if (process_sdp_a_ice(value, p, p->rtp, 0)) {
 				processed = TRUE;
 			}
-			if (process_sdp_a_ice(value, p, p->vrtp)) {
+			if (process_sdp_a_ice(value, p, p->vrtp, 0)) {
 				processed = TRUE;
 			}
-			if (process_sdp_a_ice(value, p, p->trtp)) {
+			if (process_sdp_a_ice(value, p, p->trtp, 0)) {
 				processed = TRUE;
 			}

@@ -10332,6 +10350,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		int image = FALSE;
 		int text = FALSE;
 		int processed_crypto = FALSE;
+		int rtcp_mux_offered = 0;
 		char protocol[18] = {0,};
 		unsigned int x;
 		struct ast_rtp_engine_dtls *dtls;
@@ -10351,6 +10370,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		AST_LIST_INSERT_TAIL(&p->offered_media, offer, next);
 		offer->type = SDP_UNKNOWN;

+		/* We need to check for this ahead of time */
+		rtcp_mux_offered = has_media_level_attribute(iterator, req, "rtcp-mux");
+
 		/* Check for 'audio' media offer */
 		if (strncmp(m, "audio ", 6) == 0) {
 			if ((sscanf(m, "audio %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
@@ -10717,7 +10739,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			case 'a':
 				/* Audio specific scanning */
 				if (audio) {
-					if (process_sdp_a_ice(value, p, p->rtp)) {
+					if (process_sdp_a_ice(value, p, p->rtp, rtcp_mux_offered)) {
 						processed = TRUE;
 					} else if (process_sdp_a_dtls(value, p, p->rtp)) {
 						processed_crypto = TRUE;
@@ -10738,7 +10760,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				}
 				/* Video specific scanning */
 				else if (video) {
-					if (process_sdp_a_ice(value, p, p->vrtp)) {
+					if (process_sdp_a_ice(value, p, p->vrtp, rtcp_mux_offered)) {
 						processed = TRUE;
 					} else if (process_sdp_a_dtls(value, p, p->vrtp)) {
 						processed_crypto = TRUE;
@@ -10757,7 +10779,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				}
 				/* Text (T.140) specific scanning */
 				else if (text) {
-					if (process_sdp_a_ice(value, p, p->trtp)) {
+					if (process_sdp_a_ice(value, p, p->trtp, rtcp_mux_offered)) {
 						processed = TRUE;
 					} else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
 						processed = TRUE;
@@ -11269,7 +11291,7 @@ static int process_sdp_a_sendonly(const char *a, int *sendonly)
 	return found;
 }

-static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
+static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance, int rtcp_mux_offered)
 {
 	struct ast_rtp_engine_ice *ice;
 	int found = FALSE;
@@ -11289,6 +11311,12 @@ static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_in
 		found = TRUE;
 	} else if (sscanf(a, "candidate: %31s %30u %3s %30u %23s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport, (unsigned *)&candidate.priority,
 			  address, &port, cand_type, relay_address, &relay_port) >= 7) {
+
+		if (rtcp_mux_offered && ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX) && candidate.id > 1) {
+			/* If we support RTCP-MUX and they offered it, don't consider RTCP candidates */
+			return TRUE;
+		}
+
 		candidate.foundation = foundation;
 		candidate.transport = transport;

--- a/contrib/realtime/mssql/mssql_cdr.sql
+++ b/contrib/realtime/mssql/mssql_cdr.sql
@@ -0,0 +1,44 @@
+BEGIN TRANSACTION;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL
+);
+
+GO
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20) NULL, 
+    src VARCHAR(80) NULL, 
+    dst VARCHAR(80) NULL, 
+    dcontext VARCHAR(80) NULL, 
+    clid VARCHAR(80) NULL, 
+    channel VARCHAR(80) NULL, 
+    dstchannel VARCHAR(80) NULL, 
+    lastapp VARCHAR(80) NULL, 
+    lastdata VARCHAR(80) NULL, 
+    start DATETIME NULL, 
+    answer DATETIME NULL, 
+    [end] DATETIME NULL, 
+    duration INTEGER NULL, 
+    billsec INTEGER NULL, 
+    disposition VARCHAR(45) NULL, 
+    amaflags VARCHAR(45) NULL, 
+    userfield VARCHAR(256) NULL, 
+    uniqueid VARCHAR(150) NULL, 
+    linkedid VARCHAR(150) NULL, 
+    peeraccount VARCHAR(20) NULL, 
+    sequence INTEGER NULL
+);
+
+GO
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+GO
+
+COMMIT;
+
+GO
+
--- a/contrib/realtime/mssql/mssql_config.sql
+++ b/contrib/realtime/mssql/mssql_config.sql
--- a/contrib/realtime/mssql/mssql_voicemail.sql
+++ b/contrib/realtime/mssql/mssql_voicemail.sql
@@ -0,0 +1,54 @@
+BEGIN TRANSACTION;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL
+);
+
+GO
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80) NULL, 
+    macrocontext VARCHAR(80) NULL, 
+    callerid VARCHAR(80) NULL, 
+    origtime INTEGER NULL, 
+    duration INTEGER NULL, 
+    recording IMAGE NULL, 
+    flag VARCHAR(30) NULL, 
+    category VARCHAR(30) NULL, 
+    mailboxuser VARCHAR(30) NULL, 
+    mailboxcontext VARCHAR(30) NULL, 
+    msg_id VARCHAR(40) NULL
+);
+
+GO
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+GO
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+GO
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+GO
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages ALTER COLUMN recording IMAGE;
+
+GO
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+GO
+
+COMMIT;
+
+GO
+
--- a/contrib/realtime/mysql/mysql_cdr.sql
+++ b/contrib/realtime/mysql/mysql_cdr.sql
@@ -0,0 +1,32 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start DATETIME, 
+    answer DATETIME, 
+    end DATETIME, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
--- a/contrib/realtime/mysql/mysql_config.sql
+++ b/contrib/realtime/mysql/mysql_config.sql
--- a/contrib/realtime/mysql/mysql_voicemail.sql
+++ b/contrib/realtime/mysql/mysql_voicemail.sql
@@ -0,0 +1,34 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BLOB, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
--- a/contrib/realtime/oracle/oracle_cdr.sql
+++ b/contrib/realtime/oracle/oracle_cdr.sql
@@ -0,0 +1,38 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR2(32 CHAR) NOT NULL
+)
+
+/
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR2(20 CHAR), 
+    src VARCHAR2(80 CHAR), 
+    dst VARCHAR2(80 CHAR), 
+    dcontext VARCHAR2(80 CHAR), 
+    clid VARCHAR2(80 CHAR), 
+    channel VARCHAR2(80 CHAR), 
+    dstchannel VARCHAR2(80 CHAR), 
+    lastapp VARCHAR2(80 CHAR), 
+    lastdata VARCHAR2(80 CHAR), 
+    "start" DATE, 
+    answer DATE, 
+    end DATE, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR2(45 CHAR), 
+    amaflags VARCHAR2(45 CHAR), 
+    userfield VARCHAR2(256 CHAR), 
+    uniqueid VARCHAR2(150 CHAR), 
+    linkedid VARCHAR2(150 CHAR), 
+    peeraccount VARCHAR2(20 CHAR), 
+    sequence INTEGER
+)
+
+/
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d')
+
+/
+
--- a/contrib/realtime/oracle/oracle_config.sql
+++ b/contrib/realtime/oracle/oracle_config.sql
--- a/contrib/realtime/oracle/oracle_voicemail.sql
+++ b/contrib/realtime/oracle/oracle_voicemail.sql
@@ -0,0 +1,48 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR2(32 CHAR) NOT NULL
+)
+
+/
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR2(255 CHAR) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR2(80 CHAR), 
+    macrocontext VARCHAR2(80 CHAR), 
+    callerid VARCHAR2(80 CHAR), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BLOB, 
+    flag VARCHAR2(30 CHAR), 
+    category VARCHAR2(30 CHAR), 
+    mailboxuser VARCHAR2(30 CHAR), 
+    mailboxcontext VARCHAR2(30 CHAR), 
+    msg_id VARCHAR2(40 CHAR)
+)
+
+/
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum)
+
+/
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir)
+
+/
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e')
+
+/
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages MODIFY recording BLOB
+
+/
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e'
+
+/
+
--- a/contrib/realtime/postgresql/postgresql_cdr.sql
+++ b/contrib/realtime/postgresql/postgresql_cdr.sql
@@ -0,0 +1,36 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start TIMESTAMP WITHOUT TIME ZONE, 
+    answer TIMESTAMP WITHOUT TIME ZONE, 
+    "end" TIMESTAMP WITHOUT TIME ZONE, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+COMMIT;
+
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
--- a/contrib/realtime/postgresql/postgresql_voicemail.sql
+++ b/contrib/realtime/postgresql/postgresql_voicemail.sql
@@ -0,0 +1,38 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BYTEA, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+COMMIT;
+
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -2373,6 +2373,39 @@ error:
 }
 #endif

+static int rtcp_mux(struct ast_rtp *rtp, const unsigned char *packet)
+{
+	uint8_t version;
+	uint8_t pt;
+	uint8_t m;
+
+	if (!rtp->rtcp || rtp->rtcp->type != AST_RTP_INSTANCE_RTCP_MUX) {
+		return 0;
+	}
+
+	version = (packet[0] & 0XC0) >> 6;
+	if (version == 0) {
+		/* version 0 indicates this is a STUN packet and shouldn't
+		 * be interpreted as a possible RTCP packet
+		 */
+		return 0;
+	}
+
+	/* The second octet of a packet will be one of the following:
+	 * For RTP: The marker bit (1 bit) and the RTP payload type (7 bits)
+	 * For RTCP: The payload type (8)
+	 *
+	 * RTP has a forbidden range of payload types (64-95) since these
+	 * will conflict with RTCP payload numbers if the marker bit is set.
+	 */
+	m = packet[1] & 0x80;
+	pt = packet[1] & 0x7F;
+	if (m && pt >= 64 && pt <= 95) {
+		return 1;
+	}
+	return 0;
+}
+
 /*! \pre instance is locked */
 static int __rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp)
 {
@@ -2495,7 +2528,8 @@ static int __rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t s
 	}
 #endif

-	if ((*in & 0xC0) && res_srtp && srtp && res_srtp->unprotect(srtp, buf, &len, rtcp) < 0) {
+	if ((*in & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
+		    srtp, buf, &len, rtcp || rtcp_mux(rtp, buf)) < 0) {
 	   return -1;
 	}

@@ -4859,39 +4893,6 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance,
 	return 0;
 }

-static int rtcp_mux(struct ast_rtp *rtp, const unsigned char *packet)
-{
-	uint8_t version;
-	uint8_t pt;
-	uint8_t m;
-
-	if (!rtp->rtcp || rtp->rtcp->type != AST_RTP_INSTANCE_RTCP_MUX) {
-		return 0;
-	}
-
-	version = (packet[0] & 0XC0) >> 6;
-	if (version == 0) {
-		/* version 0 indicates this is a STUN packet and shouldn't
-		 * be interpreted as a possible RTCP packet
-		 */
-		return 0;
-	}
-
-	/* The second octet of a packet will be one of the following:
-	 * For RTP: The marker bit (1 bit) and the RTP payload type (7 bits)
-	 * For RTCP: The payload type (8)
-	 *
-	 * RTP has a forbidden range of payload types (64-95) since these
-	 * will conflict with RTCP payload numbers if the marker bit is set.
-	 */
-	m = packet[1] & 0x80;
-	pt = packet[1] & 0x7F;
-	if (m && pt >= 64 && pt <= 95) {
-		return 1;
-	}
-	return 0;
-}
-
 /*! \pre instance is locked */
 static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
 {
Author	SHA1	Message	Date
Kevin Harwell	79ceeea8f8	Update for 13.16.0-rc2	2017-05-24 11:55:58 -05:00
Jenkins2	f291130cb6	Merge "chan_sip: Better ICE handling for RTCP-MUX" into 13.16	2017-05-24 10:36:22 -05:00
Sean Bright	f0e5b5815c	chan_sip: Better ICE handling for RTCP-MUX If we are offered or are offering RTCP-MUX, don't consider RTCP ICE candidates. This confuses certain browsers (current Firefox for example) and causes intial audio setup delays. ASTERISK-26982 #close Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91	2017-05-23 18:08:02 -05:00
Kevin Harwell	a26431a198	res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm When using rtcp mux if an rtcp payload came in it would still use the srtp unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp data was being passed to the rtp unprotect method this would result in an error. This patch ensures that the correct unprotect method is chosen by making sure the passed in rtcp flag is appropriately set when rtcp mux is enabled and an rtcp payload is received. ASTERISK-26979 #close Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241	2017-05-23 18:03:56 -05:00
Kevin Harwell	b634297c8c	Update for 13.16.0-rc1	2017-05-22 15:20:11 -05:00