rtp_engine: Allow more than 32 dynamic payload types.

Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK
(Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the
dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges.
Consequently, when the dynamic range is exhausted, this change utilizes payload
types in the range between 35 and 63 giving room for another 29 payload types.

ASTERISK-26311 #close

Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964

1 files changed, 71 insertions, 16 deletions

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index c9d228c56..4fc1414f0 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -143,23 +143,36 @@
 
 #include "asterisk.h"
 
-#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;
@@ -2301,6 +2314,48 @@ 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",