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 9ac53877f6)
changes/81/3681/10
Alexander Traud 9 years ago committed by George Joseph
parent 18974927e5
commit 0cf1778eed

@ -39,6 +39,15 @@ CLI Commands
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 13.11.0 to Asterisk 13.12.0 ----------

@ -96,6 +96,15 @@ documentation_language = en_US ; Set the language you want documentation
; 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]

@ -155,6 +155,8 @@ extern int dahdi_chan_name_len;
extern int ast_language_is_prefix;
extern unsigned int ast_option_rtpptdynamic;
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif

@ -84,6 +84,9 @@ extern "C" {
/*! 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

@ -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;

@ -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",

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