[asterisk-commits] rtp engine: Allow more than 32 dynamic payload types. (asterisk[14])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 7 07:07:06 CST 2016
Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/3682 )
Change subject: rtp_engine: Allow more than 32 dynamic payload types.
......................................................................
rtp_engine: Allow more than 32 dynamic payload types.
Since adding all remaining rates of Signed Linear (ASTERISK-24274) and SILK
(Gerrit 3136), only one 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, 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
---
M CHANGES
M configs/samples/asterisk.conf.sample
M include/asterisk/options.h
M include/asterisk/rtp_engine.h
M main/asterisk.c
M main/rtp_engine.c
6 files changed, 115 insertions(+), 15 deletions(-)
Approvals:
Mark Michelson: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
diff --git a/CHANGES b/CHANGES
index d20984a..302170d 100644
--- a/CHANGES
+++ b/CHANGES
@@ -39,6 +39,15 @@
Enable/disable debugging of an ARI application. When debugged, verbose
information will be sent to the Asterisk CLI.
+RTP
+------------------
+ * New setting "rtp_pt_dynamic = 96" in asterisk.conf:
+ Normally the Dynamic RTP Payload Type numbers are 96-127, which allow 32
+ formats. When you use more and receive the message "No Dynamic RTP mapping
+ available", extend the dynamic range by going for rtp_pt_dynamic = 35 (or 0)
+ instead of 96. This allows 29 (or 64) additional formats. On default this is
+ disabled and the range is 96-127 because any number below might be rejected
+ by a remote implementation; although no such broken implementation is known.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ----------
diff --git a/configs/samples/asterisk.conf.sample b/configs/samples/asterisk.conf.sample
index 6d6d2f0..54670b4 100644
--- a/configs/samples/asterisk.conf.sample
+++ b/configs/samples/asterisk.conf.sample
@@ -97,6 +97,15 @@
; This is currently is used by DUNDi and
; Exchanging Device and Mailbox State
; using protocols: XMPP, Corosync and PJSIP.
+;rtp_pt_dynamic = 96 ; Normally the Dynamic RTP Payload Type numbers
+ ; are 96-127, which allow 32 formats. When you
+ ; use more and receive the message "No Dynamic
+ ; RTP mapping available", extend the dynamic
+ ; range by going for 35 (or 0) instead of 96.
+ ; This allows 29 (or 64) more formats. 96 is the
+ ; default because any number below might be
+ ; rejected by a remote implementation; although
+ ; no such broken implementation is known, yet.
; Changing the following lines may compromise your security.
;[files]
diff --git a/include/asterisk/options.h b/include/asterisk/options.h
index e2709f9..345bacf 100644
--- a/include/asterisk/options.h
+++ b/include/asterisk/options.h
@@ -155,6 +155,8 @@
extern int ast_language_is_prefix;
+extern unsigned int ast_option_rtpptdynamic;
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index a40472e..017bb7b 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -84,6 +84,9 @@
/*! First dynamic RTP payload type */
#define AST_RTP_PT_FIRST_DYNAMIC 96
+/*! Last reassignable RTP payload type */
+#define AST_RTP_PT_LAST_REASSIGN 63
+
/*! Maximum number of generations */
#define AST_RED_MAX_GENERATION 5
diff --git a/main/asterisk.c b/main/asterisk.c
index 377f421..4341b74 100644
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -250,6 +250,7 @@
#include "asterisk/format_cache.h"
#include "asterisk/media_cache.h"
#include "asterisk/astdb.h"
+#include "asterisk/options.h"
#include "../defaults.h"
@@ -338,6 +339,7 @@
#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;
/*! @} */
@@ -659,6 +661,19 @@
ast_cli(a->fd, " Transmit silence during rec: %s\n", ast_test_flag(&ast_options, AST_OPT_FLAG_TRANSMIT_SILENCE) ? "Enabled" : "Disabled");
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");
@@ -3509,6 +3524,7 @@
/* Set default value */
option_dtmfminduration = AST_MIN_DTMF_DURATION;
+ ast_option_rtpptdynamic = AST_RTP_PT_FIRST_DYNAMIC;
/* init with buildtime config */
ast_copy_string(cfg_paths.config_dir, DEFAULT_CONFIG_DIR, sizeof(cfg_paths.config_dir));
@@ -3664,6 +3680,12 @@
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 b91bc41..1b72af1 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -145,23 +145,36 @@
ASTERISK_REGISTER_FILE()
-#include <math.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/channel.h"
-#include "asterisk/frame.h"
-#include "asterisk/module.h"
-#include "asterisk/rtp_engine.h"
+#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;
@@ -2303,6 +2316,48 @@
}
}
+ /* 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",
--
To view, visit https://gerrit.asterisk.org/3682
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964
Gerrit-PatchSet: 5
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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