rtp_engine: Allow more than 32 dynamic payload types.

The dynamic range (96-127) allows 32 RTP Payload Types. RFC 3551 section 3
allows to reassign other ranges. Consequently, when the dynamic range is
exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This
enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload
types.

ASTERISK-26311 #close

Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964
(cherry picked from commit 9ac53877f688c06acaa7c377f15da8770e4ee88b)

2 files changed, 93 insertions, 16 deletions

diff --git a/main/asterisk.c b/main/asterisk.c
index de00f4faa..1c7a0e188 100644
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -249,6 +249,7 @@ int daemon(int, int);  /* defined in libresolv of all places */
 #include "asterisk/codec.h"
 #include "asterisk/format_cache.h"
 #include "asterisk/astdb.h"
+#include "asterisk/options.h"
 
 #include "../defaults.h"
 
@@ -331,6 +332,7 @@ unsigned int option_dtmfminduration;		/*!< Minimum duration of DTMF. */
 #if defined(HAVE_SYSINFO)
 long option_minmemfree;				/*!< Minimum amount of free system memory - stop accepting calls if free memory falls below this watermark */
 #endif
+unsigned int ast_option_rtpptdynamic;
 
 /*! @} */
 
@@ -670,6 +672,19 @@ static char *handle_show_settings(struct ast_cli_entry *e, int cmd, struct ast_c
 	ast_cli(a->fd, "  Generic PLC:                 %s\n", ast_test_flag(&ast_options, AST_OPT_FLAG_GENERIC_PLC) ? "Enabled" : "Disabled");
 	ast_cli(a->fd, "  Min DTMF duration::          %u\n", option_dtmfminduration);
 
+	if (ast_option_rtpptdynamic == AST_RTP_PT_LAST_REASSIGN) {
+		ast_cli(a->fd, "  RTP dynamic payload types:   %u,%u-%u\n",
+		        ast_option_rtpptdynamic,
+		        AST_RTP_PT_FIRST_DYNAMIC, AST_RTP_MAX_PT - 1);
+	} else if (ast_option_rtpptdynamic < AST_RTP_PT_LAST_REASSIGN) {
+		ast_cli(a->fd, "  RTP dynamic payload types:   %u-%u,%u-%u\n",
+		        ast_option_rtpptdynamic, AST_RTP_PT_LAST_REASSIGN,
+		        AST_RTP_PT_FIRST_DYNAMIC, AST_RTP_MAX_PT - 1);
+	} else {
+		ast_cli(a->fd, "  RTP dynamic payload types:   %u-%u\n",
+		        AST_RTP_PT_FIRST_DYNAMIC, AST_RTP_MAX_PT - 1);
+	}
+
 	ast_cli(a->fd, "\n* Subsystems\n");
 	ast_cli(a->fd, "  -------------\n");
 	ast_cli(a->fd, "  Manager (AMI):               %s\n", check_manager_enabled() ? "Enabled" : "Disabled");
@@ -3606,6 +3621,7 @@ static void ast_readconfig(void)
 
 	/* Set default value */
 	option_dtmfminduration = AST_MIN_DTMF_DURATION;
+	ast_option_rtpptdynamic = AST_RTP_PT_FIRST_DYNAMIC;
 
 	if (ast_opt_override_config) {
 		cfg = ast_config_load2(ast_config_AST_CONFIG_FILE, "" /* core, can't reload */, config_flags);
@@ -3755,6 +3771,12 @@ static void ast_readconfig(void)
 			if (sscanf(v->value, "%30u", &option_dtmfminduration) != 1) {
 				option_dtmfminduration = AST_MIN_DTMF_DURATION;
 			}
+		/* http://www.iana.org/assignments/rtp-parameters
+		 * RTP dynamic payload types start at 96 normally; extend down to 0 */
+		} else if (!strcasecmp(v->name, "rtp_pt_dynamic")) {
+			ast_parse_arg(v->value, PARSE_UINT32|PARSE_IN_RANGE|PARSE_DEFAULT,
+			              &ast_option_rtpptdynamic, AST_RTP_PT_FIRST_DYNAMIC,
+			              0, AST_RTP_PT_LAST_REASSIGN);
 		} else if (!strcasecmp(v->name, "maxcalls")) {
 			if ((sscanf(v->value, "%30d", &ast_option_maxcalls) != 1) || (ast_option_maxcalls < 0)) {
 				ast_option_maxcalls = 0;
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 7a8378346..051253103 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -145,23 +145,36 @@
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <math.h>
-
-#include "asterisk/channel.h"
-#include "asterisk/frame.h"
-#include "asterisk/module.h"
-#include "asterisk/rtp_engine.h"
+#include <math.h>                       /* for sqrt, MAX */
+#include <sched.h>                      /* for sched_yield */
+#include <sys/time.h>                   /* for timeval */
+#include <time.h>                       /* for time_t */
+
+#include "asterisk/_private.h"          /* for ast_rtp_engine_init prototype */
+#include "asterisk/astobj2.h"           /* for ao2_cleanup, ao2_ref, etc */
+#include "asterisk/channel.h"           /* for ast_channel_name, etc */
+#include "asterisk/codec.h"             /* for ast_codec_media_type2str, etc */
+#include "asterisk/format.h"            /* for ast_format_cmp, etc */
+#include "asterisk/format_cache.h"      /* for ast_format_adpcm, etc */
+#include "asterisk/format_cap.h"        /* for ast_format_cap_alloc, etc */
+#include "asterisk/json.h"              /* for ast_json_ref, etc */
+#include "asterisk/linkedlists.h"       /* for ast_rtp_engine::<anonymous>, etc */
+#include "asterisk/lock.h"              /* for ast_rwlock_unlock, etc */
+#include "asterisk/logger.h"            /* for ast_log, ast_debug, etc */
 #include "asterisk/manager.h"
-#include "asterisk/options.h"
-#include "asterisk/astobj2.h"
-#include "asterisk/pbx.h"
-#include "asterisk/translate.h"
-#include "asterisk/netsock2.h"
-#include "asterisk/_private.h"
-#include "asterisk/framehook.h"
-#include "asterisk/stasis.h"
-#include "asterisk/json.h"
-#include "asterisk/stasis_channels.h"
+#include "asterisk/module.h"            /* for ast_module_unref, etc */
+#include "asterisk/netsock2.h"          /* for ast_sockaddr_copy, etc */
+#include "asterisk/options.h"           /* for ast_option_rtpptdynamic */
+#include "asterisk/pbx.h"               /* for pbx_builtin_setvar_helper */
+#include "asterisk/res_srtp.h"          /* for ast_srtp_res */
+#include "asterisk/rtp_engine.h"        /* for ast_rtp_codecs, etc */
+#include "asterisk/stasis.h"            /* for stasis_message_data, etc */
+#include "asterisk/stasis_channels.h"   /* for ast_channel_stage_snapshot, etc */
+#include "asterisk/strings.h"           /* for ast_str_append, etc */
+#include "asterisk/time.h"              /* for ast_tvdiff_ms, ast_tvnow */
+#include "asterisk/translate.h"         /* for ast_translate_available_formats */
+#include "asterisk/utils.h"             /* for ast_free, ast_strdup, etc */
+#include "asterisk/vector.h"            /* for AST_VECTOR_GET, etc */
 
 struct ast_srtp_res *res_srtp = NULL;
 struct ast_srtp_policy_res *res_srtp_policy = NULL;
@@ -1796,6 +1809,48 @@ static void add_static_payload(int map, struct ast_format *format, int rtp_code)
 			}
 		}
 
+		/* http://www.iana.org/assignments/rtp-parameters
+		 * RFC 3551, Section 3: "[...] applications which need to define more
+		 * than 32 dynamic payload types MAY bind codes below 96, in which case
+		 * it is RECOMMENDED that unassigned payload type numbers be used
+		 * first". Updated by RFC 5761, Section 4: "[...] values in the range
+		 * 64-95 MUST NOT be used [to avoid conflicts with RTCP]". Summaries:
+		 * https://tools.ietf.org/html/draft-roach-mmusic-unified-plan#section-3.2.1.2
+		 * https://tools.ietf.org/html/draft-wu-avtcore-dynamic-pt-usage#section-3
+		 */
+		if (map < 0) {
+			for (x = MAX(ast_option_rtpptdynamic, 35); x <= AST_RTP_PT_LAST_REASSIGN; ++x) {
+				if (!static_RTP_PT[x]) {
+					map = x;
+					break;
+				}
+			}
+		}
+		/* Yet, reusing mappings below 35 is not supported in Asterisk because
+		 * when Compact Headers are activated, no rtpmap is send for those below
+		 * 35. If you want to use 35 and below
+		 * A) do not use Compact Headers,
+		 * B) remove that code in chan_sip/res_pjsip, or
+		 * C) add a flag that this RTP Payload Type got reassigned dynamically
+		 *    and requires a rtpmap even with Compact Headers enabled.
+		 */
+		if (map < 0) {
+			for (x = MAX(ast_option_rtpptdynamic, 20); x < 35; ++x) {
+				if (!static_RTP_PT[x]) {
+					map = x;
+					break;
+				}
+			}
+		}
+		if (map < 0) {
+			for (x = MAX(ast_option_rtpptdynamic, 0); x < 20; ++x) {
+				if (!static_RTP_PT[x]) {
+					map = x;
+					break;
+				}
+			}
+		}
+
 		if (map < 0) {
 			if (format) {
 				ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n",