main/rtp_engine.c

   1 /*
   2  * Asterisk -- An open source telephony toolkit.
   3  *
   4  * Copyright (C) 1999 - 2008, Digium, Inc.
   5  *
   6  * Joshua Colp <jcolp@digium.com>
   7  *
   8  * See http://www.asterisk.org for more information about
   9  * the Asterisk project. Please do not directly contact
  10  * any of the maintainers of this project for assistance;
  11  * the project provides a web site, mailing lists and IRC
  12  * channels for your use.
  13  *
  14  * This program is free software, distributed under the terms of
  15  * the GNU General Public License Version 2. See the LICENSE file
  16  * at the top of the source tree.
  17  */
  18
  19 /*! \file
  20  *
  21  * \brief Pluggable RTP Architecture
  22  *
  23  * \author Joshua Colp <jcolp@digium.com>
  24  */
  25
  26 /*** MODULEINFO
  27         <support_level>core</support_level>
  28 ***/
  29
  30 /*** DOCUMENTATION
  31         <managerEvent language="en_US" name="RTCPSent">
  32                 <managerEventInstance class="EVENT_FLAG_REPORTING">
  33                         <synopsis>Raised when an RTCP packet is sent.</synopsis>
  34                         <syntax>
  35                                 <channel_snapshot/>
  36                                 <parameter name="SSRC">
  37                                         <para>The SSRC identifier for our stream</para>
  38                                 </parameter>
  39                                 <parameter name="PT">
  40                                         <para>The type of packet for this RTCP report.</para>
  41                                         <enumlist>
  42                                                 <enum name="200(SR)"/>
  43                                                 <enum name="201(RR)"/>
  44                                         </enumlist>
  45                                 </parameter>
  46                                 <parameter name="To">
  47                                         <para>The address the report is sent to.</para>
  48                                 </parameter>
  49                                 <parameter name="ReportCount">
  50                                         <para>The number of reports that were sent.</para>
  51                                         <para>The report count determines the number of ReportX headers in
  52                                         the message. The X for each set of report headers will range from 0 to
  53                                         <literal>ReportCount - 1</literal>.</para>
  54                                 </parameter>
  55                                 <parameter name="SentNTP" required="false">
  56                                         <para>The time the sender generated the report. Only valid when
  57                                         PT is <literal>200(SR)</literal>.</para>
  58                                 </parameter>
  59                                 <parameter name="SentRTP" required="false">
  60                                         <para>The sender's last RTP timestamp. Only valid when PT is
  61                                         <literal>200(SR)</literal>.</para>
  62                                 </parameter>
  63                                 <parameter name="SentPackets" required="false">
  64                                         <para>The number of packets the sender has sent. Only valid when PT
  65                                         is <literal>200(SR)</literal>.</para>
  66                                 </parameter>
  67                                 <parameter name="SentOctets" required="false">
  68                                         <para>The number of bytes the sender has sent. Only valid when PT is
  69                                         <literal>200(SR)</literal>.</para>
  70                                 </parameter>
  71                                 <parameter name="ReportXSourceSSRC">
  72                                         <para>The SSRC for the source of this report block.</para>
  73                                 </parameter>
  74                                 <parameter name="ReportXFractionLost">
  75                                         <para>The fraction of RTP data packets from <literal>ReportXSourceSSRC</literal>
  76                                         lost since the previous SR or RR report was sent.</para>
  77                                 </parameter>
  78                                 <parameter name="ReportXCumulativeLost">
  79                                         <para>The total number of RTP data packets from <literal>ReportXSourceSSRC</literal>
  80                                         lost since the beginning of reception.</para>
  81                                 </parameter>
  82                                 <parameter name="ReportXHighestSequence">
  83                                         <para>The highest sequence number received in an RTP data packet from
  84                                         <literal>ReportXSourceSSRC</literal>.</para>
  85                                 </parameter>
  86                                 <parameter name="ReportXSequenceNumberCycles">
  87                                         <para>The number of sequence number cycles seen for the RTP data
  88                                         received from <literal>ReportXSourceSSRC</literal>.</para>
  89                                 </parameter>
  90                                 <parameter name="ReportXIAJitter">
  91                                         <para>An estimate of the statistical variance of the RTP data packet
  92                                         interarrival time, measured in timestamp units.</para>
  93                                 </parameter>
  94                                 <parameter name="ReportXLSR">
  95                                         <para>The last SR timestamp received from <literal>ReportXSourceSSRC</literal>.
  96                                         If no SR has been received from <literal>ReportXSourceSSRC</literal>,
  97                                         then 0.</para>
  98                                 </parameter>
  99                                 <parameter name="ReportXDLSR">
 100                                         <para>The delay, expressed in units of 1/65536 seconds, between
 101                                         receiving the last SR packet from <literal>ReportXSourceSSRC</literal>
 102                                         and sending this report.</para>
 103                                 </parameter>
 104                         </syntax>
 105                 </managerEventInstance>
 106         </managerEvent>
 107         <managerEvent language="en_US" name="RTCPReceived">
 108                 <managerEventInstance class="EVENT_FLAG_REPORTING">
 109                         <synopsis>Raised when an RTCP packet is received.</synopsis>
 110                         <syntax>
 111                                 <channel_snapshot/>
 112                                 <parameter name="SSRC">
 113                                         <para>The SSRC identifier for the remote system</para>
 114                                 </parameter>
 115                                 <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='PT'])" />
 116                                 <parameter name="From">
 117                                         <para>The address the report was received from.</para>
 118                                 </parameter>
 119                                 <parameter name="RTT">
 120                                         <para>Calculated Round-Trip Time in seconds</para>
 121                                 </parameter>
 122                                 <parameter name="ReportCount">
 123                                         <para>The number of reports that were received.</para>
 124                                         <para>The report count determines the number of ReportX headers in
 125                                         the message. The X for each set of report headers will range from 0 to
 126                                         <literal>ReportCount - 1</literal>.</para>
 127                                 </parameter>
 128                                 <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentNTP'])" />
 129                                 <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentRTP'])" />
 130                                 <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentPackets'])" />
 131                                 <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentOctets'])" />
 132                                 <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[contains(@name, 'ReportX')])" />
 133                         </syntax>
 134                 </managerEventInstance>
 135         </managerEvent>
 136  ***/
 137
 138 #include "asterisk.h"
 139
 140 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 141
 142 #include <math.h>
 143
 144 #include "asterisk/channel.h"
 145 #include "asterisk/frame.h"
 146 #include "asterisk/module.h"
 147 #include "asterisk/rtp_engine.h"
 148 #include "asterisk/manager.h"
 149 #include "asterisk/options.h"
 150 #include "asterisk/astobj2.h"
 151 #include "asterisk/pbx.h"
 152 #include "asterisk/translate.h"
 153 #include "asterisk/netsock2.h"
 154 #include "asterisk/_private.h"
 155 #include "asterisk/framehook.h"
 156 #include "asterisk/stasis.h"
 157 #include "asterisk/json.h"
 158 #include "asterisk/stasis_channels.h"
 159
 160 struct ast_srtp_res *res_srtp = NULL;
 161 struct ast_srtp_policy_res *res_srtp_policy = NULL;
 162
 163 /*! Structure that represents an RTP session (instance) */
 164 struct ast_rtp_instance {
 165         /*! Engine that is handling this RTP instance */
 166         struct ast_rtp_engine *engine;
 167         /*! Data unique to the RTP engine */
 168         void *data;
 169         /*! RTP properties that have been set and their value */
 170         int properties[AST_RTP_PROPERTY_MAX];
 171         /*! Address that we are expecting RTP to come in to */
 172         struct ast_sockaddr local_address;
 173         /*! Address that we are sending RTP to */
 174         struct ast_sockaddr remote_address;
 175         /*! Instance that we are bridged to if doing remote or local bridging */
 176         struct ast_rtp_instance *bridged;
 177         /*! Payload and packetization information */
 178         struct ast_rtp_codecs codecs;
 179         /*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
 180         int timeout;
 181         /*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
 182         int holdtimeout;
 183         /*! RTP keepalive interval */
 184         int keepalive;
 185         /*! Glue currently in use */
 186         struct ast_rtp_glue *glue;
 187         /*! SRTP info associated with the instance */
 188         struct ast_srtp *srtp;
 189         /*! Channel unique ID */
 190         char channel_uniqueid[AST_MAX_UNIQUEID];
 191 };
 192
 193 /*! List of RTP engines that are currently registered */
 194 static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
 195
 196 /*! List of RTP glues */
 197 static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
 198
 199 /*! The following array defines the MIME Media type (and subtype) for each
 200    of our codecs, or RTP-specific data type. */
 201 static struct ast_rtp_mime_type {
 202         struct ast_rtp_payload_type payload_type;
 203         char *type;
 204         char *subtype;
 205         unsigned int sample_rate;
 206 } ast_rtp_mime_types[128]; /* This will Likely not need to grow any time soon. */
 207 static ast_rwlock_t mime_types_lock;
 208 static int mime_types_len = 0;
 209
 210 /*!
 211  * \brief Mapping between Asterisk codecs and rtp payload types
 212  *
 213  * Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
 214  * also, our own choices for dynamic payload types.  This is our master
 215  * table for transmission
 216  *
 217  * See http://www.iana.org/assignments/rtp-parameters for a list of
 218  * assigned values
 219  */
 220 static struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT];
 221 static ast_rwlock_t static_RTP_PT_lock;
 222
 223 /*! \brief \ref stasis topic for RTP related messages */
 224 static struct stasis_topic *rtp_topic;
 225
 226 int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
 227 {
 228         struct ast_rtp_engine *current_engine;
 229
 230         /* Perform a sanity check on the engine structure to make sure it has the basics */
 231         if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
 232                 ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
 233                 return -1;
 234         }
 235
 236         /* Link owner module to the RTP engine for reference counting purposes */
 237         engine->mod = module;
 238
 239         AST_RWLIST_WRLOCK(&engines);
 240
 241         /* Ensure that no two modules with the same name are registered at the same time */
 242         AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
 243                 if (!strcmp(current_engine->name, engine->name)) {
 244                         ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
 245                         AST_RWLIST_UNLOCK(&engines);
 246                         return -1;
 247                 }
 248         }
 249
 250         /* The engine survived our critique. Off to the list it goes to be used */
 251         AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
 252
 253         AST_RWLIST_UNLOCK(&engines);
 254
 255         ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
 256
 257         return 0;
 258 }
 259
 260 int ast_rtp_engine_unregister(struct ast_rtp_engine *engine)
 261 {
 262         struct ast_rtp_engine *current_engine = NULL;
 263
 264         AST_RWLIST_WRLOCK(&engines);
 265
 266         if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
 267                 ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
 268         }
 269
 270         AST_RWLIST_UNLOCK(&engines);
 271
 272         return current_engine ? 0 : -1;
 273 }
 274
 275 int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module)
 276 {
 277         struct ast_rtp_glue *current_glue = NULL;
 278
 279         if (ast_strlen_zero(glue->type)) {
 280                 return -1;
 281         }
 282
 283         glue->mod = module;
 284
 285         AST_RWLIST_WRLOCK(&glues);
 286
 287         AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
 288                 if (!strcasecmp(current_glue->type, glue->type)) {
 289                         ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
 290                         AST_RWLIST_UNLOCK(&glues);
 291                         return -1;
 292                 }
 293         }
 294
 295         AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
 296
 297         AST_RWLIST_UNLOCK(&glues);
 298
 299         ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
 300
 301         return 0;
 302 }
 303
 304 int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
 305 {
 306         struct ast_rtp_glue *current_glue = NULL;
 307
 308         AST_RWLIST_WRLOCK(&glues);
 309
 310         if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
 311                 ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
 312         }
 313
 314         AST_RWLIST_UNLOCK(&glues);
 315
 316         return current_glue ? 0 : -1;
 317 }
 318
 319 static void instance_destructor(void *obj)
 320 {
 321         struct ast_rtp_instance *instance = obj;
 322
 323         /* Pass us off to the engine to destroy */
 324         if (instance->data && instance->engine->destroy(instance)) {
 325                 ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
 326                 return;
 327         }
 328
 329         if (instance->srtp) {
 330                 res_srtp->destroy(instance->srtp);
 331         }
 332
 333         ast_rtp_codecs_payloads_destroy(&instance->codecs);
 334
 335         /* Drop our engine reference */
 336         ast_module_unref(instance->engine->mod);
 337
 338         ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
 339 }
 340
 341 int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
 342 {
 343         ao2_ref(instance, -1);
 344
 345         return 0;
 346 }
 347
 348 struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name,
 349                 struct ast_sched_context *sched, const struct ast_sockaddr *sa,
 350                 void *data)
 351 {
 352         struct ast_sockaddr address = {{0,}};
 353         struct ast_rtp_instance *instance = NULL;
 354         struct ast_rtp_engine *engine = NULL;
 355
 356         AST_RWLIST_RDLOCK(&engines);
 357
 358         /* If an engine name was specified try to use it or otherwise use the first one registered */
 359         if (!ast_strlen_zero(engine_name)) {
 360                 AST_RWLIST_TRAVERSE(&engines, engine, entry) {
 361                         if (!strcmp(engine->name, engine_name)) {
 362                                 break;
 363                         }
 364                 }
 365         } else {
 366                 engine = AST_RWLIST_FIRST(&engines);
 367         }
 368
 369         /* If no engine was actually found bail out now */
 370         if (!engine) {
 371                 ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
 372                 AST_RWLIST_UNLOCK(&engines);
 373                 return NULL;
 374         }
 375
 376         /* Bump up the reference count before we return so the module can not be unloaded */
 377         ast_module_ref(engine->mod);
 378
 379         AST_RWLIST_UNLOCK(&engines);
 380
 381         /* Allocate a new RTP instance */
 382         if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
 383                 ast_module_unref(engine->mod);
 384                 return NULL;
 385         }
 386         instance->engine = engine;
 387         ast_sockaddr_copy(&instance->local_address, sa);
 388         ast_sockaddr_copy(&address, sa);
 389
 390         if (ast_rtp_codecs_payloads_initialize(&instance->codecs)) {
 391                 ao2_ref(instance, -1);
 392                 return NULL;
 393         }
 394
 395         ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
 396
 397         /* And pass it off to the engine to setup */
 398         if (instance->engine->new(instance, sched, &address, data)) {
 399                 ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
 400                 ao2_ref(instance, -1);
 401                 return NULL;
 402         }
 403
 404         ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
 405
 406         return instance;
 407 }
 408
 409 const char *ast_rtp_instance_get_channel_id(struct ast_rtp_instance *instance)
 410 {
 411         return instance->channel_uniqueid;
 412 }
 413
 414 void ast_rtp_instance_set_channel_id(struct ast_rtp_instance *instance, const char *uniqueid)
 415 {
 416         ast_copy_string(instance->channel_uniqueid, uniqueid, sizeof(instance->channel_uniqueid));
 417 }
 418
 419 void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
 420 {
 421         instance->data = data;
 422 }
 423
 424 void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance)
 425 {
 426         return instance->data;
 427 }
 428
 429 int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
 430 {
 431         return instance->engine->write(instance, frame);
 432 }
 433
 434 struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp)
 435 {
 436         return instance->engine->read(instance, rtcp);
 437 }
 438
 439 int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance,
 440                 const struct ast_sockaddr *address)
 441 {
 442         ast_sockaddr_copy(&instance->local_address, address);
 443         return 0;
 444 }
 445
 446 int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance,
 447                 const struct ast_sockaddr *address)
 448 {
 449         ast_sockaddr_copy(&instance->remote_address, address);
 450
 451         /* moo */
 452
 453         if (instance->engine->remote_address_set) {
 454                 instance->engine->remote_address_set(instance, &instance->remote_address);
 455         }
 456
 457         return 0;
 458 }
 459
 460 int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance,
 461                 struct ast_sockaddr *address)
 462 {
 463         if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
 464                 ast_sockaddr_copy(address, &instance->local_address);
 465                 return 1;
 466         }
 467
 468         return 0;
 469 }
 470
 471 void ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance,
 472                 struct ast_sockaddr *address)
 473 {
 474         ast_sockaddr_copy(address, &instance->local_address);
 475 }
 476
 477 int ast_rtp_instance_get_and_cmp_remote_address(struct ast_rtp_instance *instance,
 478                 struct ast_sockaddr *address)
 479 {
 480         if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) {
 481                 ast_sockaddr_copy(address, &instance->remote_address);
 482                 return 1;
 483         }
 484
 485         return 0;
 486 }
 487
 488 void ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance,
 489                 struct ast_sockaddr *address)
 490 {
 491         ast_sockaddr_copy(address, &instance->remote_address);
 492 }
 493
 494 void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
 495 {
 496         if (instance->engine->extended_prop_set) {
 497                 instance->engine->extended_prop_set(instance, property, value);
 498         }
 499 }
 500
 501 void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property)
 502 {
 503         if (instance->engine->extended_prop_get) {
 504                 return instance->engine->extended_prop_get(instance, property);
 505         }
 506
 507         return NULL;
 508 }
 509
 510 void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
 511 {
 512         instance->properties[property] = value;
 513
 514         if (instance->engine->prop_set) {
 515                 instance->engine->prop_set(instance, property, value);
 516         }
 517 }
 518
 519 int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property)
 520 {
 521         return instance->properties[property];
 522 }
 523
 524 struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance)
 525 {
 526         return &instance->codecs;
 527 }
 528
 529 static int rtp_payload_type_hash(const void *obj, const int flags)
 530 {
 531         const struct ast_rtp_payload_type *type = obj;
 532         const int *payload = obj;
 533
 534         return (flags & OBJ_KEY) ? *payload : type->payload;
 535 }
 536
 537 static int rtp_payload_type_cmp(void *obj, void *arg, int flags)
 538 {
 539         struct ast_rtp_payload_type *type1 = obj, *type2 = arg;
 540         const int *payload = arg;
 541
 542         return (type1->payload == (OBJ_KEY ? *payload : type2->payload)) ? CMP_MATCH | CMP_STOP : 0;
 543 }
 544
 545 int ast_rtp_codecs_payloads_initialize(struct ast_rtp_codecs *codecs)
 546 {
 547         if (!(codecs->payloads = ao2_container_alloc(AST_RTP_MAX_PT, rtp_payload_type_hash, rtp_payload_type_cmp))) {
 548                 return -1;
 549         }
 550
 551         return 0;
 552 }
 553
 554 void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
 555 {
 556         ao2_cleanup(codecs->payloads);
 557 }
 558
 559 void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
 560 {
 561         ast_rtp_codecs_payloads_destroy(codecs);
 562
 563         if (instance && instance->engine && instance->engine->payload_set) {
 564                 int i;
 565                 for (i = 0; i < AST_RTP_MAX_PT; i++) {
 566                         instance->engine->payload_set(instance, i, 0, NULL, 0);
 567                 }
 568         }
 569
 570         ast_rtp_codecs_payloads_initialize(codecs);
 571 }
 572
 573 void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
 574 {
 575         int i;
 576
 577         ast_rwlock_rdlock(&static_RTP_PT_lock);
 578         for (i = 0; i < AST_RTP_MAX_PT; i++) {
 579                 if (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
 580                         struct ast_rtp_payload_type *type;
 581
 582                         if (!(type = ao2_alloc(sizeof(*type), NULL))) {
 583                                 /* Unfortunately if this occurs the payloads container will not contain all possible default payloads
 584                                  * but we err on the side of doing what we can in the hopes that the extreme memory conditions which
 585                                  * caused this to occur will go away.
 586                                  */
 587                                 continue;
 588                         }
 589
 590                         type->payload = i;
 591                         type->asterisk_format = static_RTP_PT[i].asterisk_format;
 592                         type->rtp_code = static_RTP_PT[i].rtp_code;
 593                         ast_format_copy(&type->format, &static_RTP_PT[i].format);
 594
 595                         ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
 596
 597                         if (instance && instance->engine && instance->engine->payload_set) {
 598                                 instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
 599                         }
 600
 601                         ao2_ref(type, -1);
 602                 }
 603         }
 604         ast_rwlock_unlock(&static_RTP_PT_lock);
 605 }
 606
 607 void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
 608 {
 609         int i;
 610         struct ast_rtp_payload_type *type;
 611
 612         for (i = 0; i < AST_RTP_MAX_PT; i++) {
 613                 struct ast_rtp_payload_type *new_type;
 614
 615                 if (!(type = ao2_find(src->payloads, &i, OBJ_KEY | OBJ_NOLOCK))) {
 616                         continue;
 617                 }
 618
 619                 if (!(new_type = ao2_alloc(sizeof(*new_type), NULL))) {
 620                         continue;
 621                 }
 622
 623                 ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
 624
 625                 new_type->payload = i;
 626                 *new_type = *type;
 627
 628                 ao2_link_flags(dest->payloads, new_type, OBJ_NOLOCK);
 629
 630                 ao2_ref(new_type, -1);
 631
 632                 if (instance && instance->engine && instance->engine->payload_set) {
 633                         instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
 634                 }
 635
 636                 ao2_ref(type, -1);
 637         }
 638 }
 639
 640 void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
 641 {
 642         struct ast_rtp_payload_type *type;
 643
 644         ast_rwlock_rdlock(&static_RTP_PT_lock);
 645
 646         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
 647                 ast_rwlock_unlock(&static_RTP_PT_lock);
 648                 return;
 649         }
 650
 651         if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
 652                 if (!(type = ao2_alloc(sizeof(*type), NULL))) {
 653                         ast_rwlock_unlock(&static_RTP_PT_lock);
 654                         return;
 655                 }
 656                 type->payload = payload;
 657                 ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
 658         }
 659
 660         type->asterisk_format = static_RTP_PT[payload].asterisk_format;
 661         type->rtp_code = static_RTP_PT[payload].rtp_code;
 662         type->payload = payload;
 663         ast_format_copy(&type->format, &static_RTP_PT[payload].format);
 664
 665         ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
 666
 667         if (instance && instance->engine && instance->engine->payload_set) {
 668                 instance->engine->payload_set(instance, payload, type->asterisk_format, &type->format, type->rtp_code);
 669         }
 670
 671         ao2_ref(type, -1);
 672
 673         ast_rwlock_unlock(&static_RTP_PT_lock);
 674 }
 675
 676 int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
 677                                  char *mimetype, char *mimesubtype,
 678                                  enum ast_rtp_options options,
 679                                  unsigned int sample_rate)
 680 {
 681         unsigned int i;
 682         int found = 0;
 683
 684         if (pt < 0 || pt >= AST_RTP_MAX_PT)
 685                 return -1; /* bogus payload type */
 686
 687         ast_rwlock_rdlock(&mime_types_lock);
 688         for (i = 0; i < mime_types_len; ++i) {
 689                 const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
 690                 struct ast_rtp_payload_type *type;
 691
 692                 if (strcasecmp(mimesubtype, t->subtype)) {
 693                         continue;
 694                 }
 695
 696                 if (strcasecmp(mimetype, t->type)) {
 697                         continue;
 698                 }
 699
 700                 /* if both sample rates have been supplied, and they don't match,
 701                  * then this not a match; if one has not been supplied, then the
 702                  * rates are not compared */
 703                 if (sample_rate && t->sample_rate &&
 704                     (sample_rate != t->sample_rate)) {
 705                         continue;
 706                 }
 707
 708                 found = 1;
 709
 710                 if (!(type = ao2_find(codecs->payloads, &pt, OBJ_KEY | OBJ_NOLOCK))) {
 711                         if (!(type = ao2_alloc(sizeof(*type), NULL))) {
 712                                 continue;
 713                         }
 714                         type->payload = pt;
 715                         ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
 716                 }
 717
 718                 *type = t->payload_type;
 719                 type->payload = pt;
 720
 721                 if ((t->payload_type.format.id == AST_FORMAT_G726) && t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
 722                         ast_format_set(&type->format, AST_FORMAT_G726_AAL2, 0);
 723                 }
 724
 725                 if (instance && instance->engine && instance->engine->payload_set) {
 726                         instance->engine->payload_set(instance, pt, type->asterisk_format, &type->format, type->rtp_code);
 727                 }
 728
 729                 ao2_ref(type, -1);
 730
 731                 break;
 732         }
 733         ast_rwlock_unlock(&mime_types_lock);
 734
 735         return (found ? 0 : -2);
 736 }
 737
 738 int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options)
 739 {
 740         return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
 741 }
 742
 743 void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
 744 {
 745         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
 746                 return;
 747         }
 748
 749         ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
 750
 751         ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK | OBJ_NODATA | OBJ_UNLINK);
 752
 753         if (instance && instance->engine && instance->engine->payload_set) {
 754                 instance->engine->payload_set(instance, payload, 0, NULL, 0);
 755         }
 756 }
 757
 758 struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
 759 {
 760         struct ast_rtp_payload_type result = { .asterisk_format = 0, }, *type;
 761
 762         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
 763                 return result;
 764         }
 765
 766         if ((type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
 767                 result = *type;
 768                 ao2_ref(type, -1);
 769         }
 770
 771         if (!result.rtp_code && !result.asterisk_format) {
 772                 ast_rwlock_rdlock(&static_RTP_PT_lock);
 773                 result = static_RTP_PT[payload];
 774                 ast_rwlock_unlock(&static_RTP_PT_lock);
 775         }
 776
 777         return result;
 778 }
 779
 780
 781 struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload)
 782 {
 783         struct ast_rtp_payload_type *type;
 784         struct ast_format *format;
 785
 786         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
 787                 return NULL;
 788         }
 789
 790         if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
 791                 return NULL;
 792         }
 793
 794         format = type->asterisk_format ? &type->format : NULL;
 795
 796         ao2_ref(type, -1);
 797
 798         return format;
 799 }
 800
 801 static int rtp_payload_type_add_ast(void *obj, void *arg, int flags)
 802 {
 803         struct ast_rtp_payload_type *type = obj;
 804         struct ast_format_cap *astformats = arg;
 805
 806         if (type->asterisk_format) {
 807                 ast_format_cap_add(astformats, &type->format);
 808         }
 809
 810         return 0;
 811 }
 812
 813 static int rtp_payload_type_add_nonast(void *obj, void *arg, int flags)
 814 {
 815         struct ast_rtp_payload_type *type = obj;
 816         int *nonastformats = arg;
 817
 818         if (!type->asterisk_format) {
 819                 *nonastformats |= type->rtp_code;
 820         }
 821
 822         return 0;
 823 }
 824
 825 void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
 826 {
 827         ast_format_cap_remove_all(astformats);
 828         *nonastformats = 0;
 829
 830         ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_ast, astformats);
 831         ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_nonast, nonastformats);
 832 }
 833
 834 static int rtp_payload_type_find_format(void *obj, void *arg, int flags)
 835 {
 836         struct ast_rtp_payload_type *type = obj;
 837         struct ast_format *format = arg;
 838
 839         return (type->asterisk_format && (ast_format_cmp(&type->format, format) != AST_FORMAT_CMP_NOT_EQUAL)) ? CMP_MATCH | CMP_STOP : 0;
 840 }
 841
 842 static int rtp_payload_type_find_nonast_format(void *obj, void *arg, int flags)
 843 {
 844         struct ast_rtp_payload_type *type = obj;
 845         int *rtp_code = arg;
 846
 847         return ((!type->asterisk_format && (type->rtp_code == *rtp_code)) ? CMP_MATCH | CMP_STOP : 0);
 848 }
 849
 850 int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
 851 {
 852         struct ast_rtp_payload_type *type;
 853         int i, res = -1;
 854
 855         if (asterisk_format && format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_format, (void*)format))) {
 856                 res = type->payload;
 857                 ao2_ref(type, -1);
 858                 return res;
 859         } else if (!asterisk_format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_nonast_format, (void*)&code))) {
 860                 res = type->payload;
 861                 ao2_ref(type, -1);
 862                 return res;
 863         }
 864
 865         ast_rwlock_rdlock(&static_RTP_PT_lock);
 866         for (i = 0; i < AST_RTP_MAX_PT; i++) {
 867                 if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
 868                         (ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
 869                         res = i;
 870                         break;
 871                 } else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
 872                         (static_RTP_PT[i].rtp_code == code)) {
 873                         res = i;
 874                         break;
 875                 }
 876         }
 877         ast_rwlock_unlock(&static_RTP_PT_lock);
 878
 879         return res;
 880 }
 881 int ast_rtp_codecs_find_payload_code(struct ast_rtp_codecs *codecs, int code)
 882 {
 883         struct ast_rtp_payload_type *type;
 884         int res = -1;
 885
 886         /* Search the payload type in the codecs passed */
 887         if ((type = ao2_find(codecs->payloads, &code, OBJ_NOLOCK | OBJ_KEY)))
 888         {
 889                 res = type->payload;
 890                 ao2_ref(type, -1);
 891                 return res;
 892         }
 893
 894         return res;
 895 }
 896 const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, struct ast_format *format, int code, enum ast_rtp_options options)
 897 {
 898         int i;
 899         const char *res = "";
 900
 901         ast_rwlock_rdlock(&mime_types_lock);
 902         for (i = 0; i < mime_types_len; i++) {
 903                 if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
 904                         (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
 905                         if ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
 906                                 res = "G726-32";
 907                                 break;
 908                         } else {
 909                                 res = ast_rtp_mime_types[i].subtype;
 910                                 break;
 911                         }
 912                 } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
 913                         ast_rtp_mime_types[i].payload_type.rtp_code == code) {
 914
 915                         res = ast_rtp_mime_types[i].subtype;
 916                         break;
 917                 }
 918         }
 919         ast_rwlock_unlock(&mime_types_lock);
 920
 921         return res;
 922 }
 923
 924 unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format *format, int code)
 925 {
 926         unsigned int i;
 927         unsigned int res = 0;
 928
 929         ast_rwlock_rdlock(&mime_types_lock);
 930         for (i = 0; i < mime_types_len; ++i) {
 931                 if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
 932                         (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
 933                         res = ast_rtp_mime_types[i].sample_rate;
 934                         break;
 935                 } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
 936                         ast_rtp_mime_types[i].payload_type.rtp_code == code) {
 937                         res = ast_rtp_mime_types[i].sample_rate;
 938                         break;
 939                 }
 940         }
 941         ast_rwlock_unlock(&mime_types_lock);
 942
 943         return res;
 944 }
 945
 946 char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, struct ast_format_cap *ast_format_capability, int rtp_capability, const int asterisk_format, enum ast_rtp_options options)
 947 {
 948         int found = 0;
 949         const char *name;
 950         if (!buf) {
 951                 return NULL;
 952         }
 953
 954
 955         if (asterisk_format) {
 956                 struct ast_format tmp_fmt;
 957                 ast_format_cap_iter_start(ast_format_capability);
 958                 while (!ast_format_cap_iter_next(ast_format_capability, &tmp_fmt)) {
 959                         name = ast_rtp_lookup_mime_subtype2(asterisk_format, &tmp_fmt, 0, options);
 960                         ast_str_append(&buf, 0, "%s|", name);
 961                         found = 1;
 962                 }
 963                 ast_format_cap_iter_end(ast_format_capability);
 964
 965         } else {
 966                 int x;
 967                 ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
 968                 for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
 969                         if (rtp_capability & x) {
 970                                 name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
 971                                 ast_str_append(&buf, 0, "%s|", name);
 972                                 found = 1;
 973                         }
 974                 }
 975         }
 976
 977         ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
 978
 979         return ast_str_buffer(buf);
 980 }
 981
 982 void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
 983 {
 984         codecs->pref = *prefs;
 985
 986         if (instance && instance->engine->packetization_set) {
 987                 instance->engine->packetization_set(instance, &instance->codecs.pref);
 988         }
 989 }
 990
 991 int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
 992 {
 993         return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
 994 }
 995
 996 int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
 997 {
 998         return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
 999 }
1000 int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
1001 {
1002         return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
1003 }
1004
1005 int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
1006 {
1007         return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
1008 }
1009
1010 enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
1011 {
1012         return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
1013 }
1014
1015 void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
1016 {
1017         if (instance->engine->update_source) {
1018                 instance->engine->update_source(instance);
1019         }
1020 }
1021
1022 void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
1023 {
1024         if (instance->engine->change_source) {
1025                 instance->engine->change_source(instance);
1026         }
1027 }
1028
1029 int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
1030 {
1031         return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
1032 }
1033
1034 void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
1035 {
1036         if (instance->engine->stop) {
1037                 instance->engine->stop(instance);
1038         }
1039 }
1040
1041 int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
1042 {
1043         return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
1044 }
1045
1046 struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
1047 {
1048         struct ast_rtp_glue *glue = NULL;
1049
1050         AST_RWLIST_RDLOCK(&glues);
1051
1052         AST_RWLIST_TRAVERSE(&glues, glue, entry) {
1053                 if (!strcasecmp(glue->type, type)) {
1054                         break;
1055                 }
1056         }
1057
1058         AST_RWLIST_UNLOCK(&glues);
1059
1060         return glue;
1061 }
1062
1063 /*!
1064  * \brief Conditionally unref an rtp instance
1065  */
1066 static void unref_instance_cond(struct ast_rtp_instance **instance)
1067 {
1068         if (*instance) {
1069                 ao2_ref(*instance, -1);
1070                 *instance = NULL;
1071         }
1072 }
1073
1074 struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
1075 {
1076         return instance->bridged;
1077 }
1078
1079 void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged)
1080 {
1081         instance->bridged = bridged;
1082 }
1083
1084 void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c_dst, struct ast_channel *c_src)
1085 {
1086         struct ast_rtp_instance *instance_dst = NULL, *instance_src = NULL,
1087                 *vinstance_dst = NULL, *vinstance_src = NULL,
1088                 *tinstance_dst = NULL, *tinstance_src = NULL;
1089         struct ast_rtp_glue *glue_dst, *glue_src;
1090         enum ast_rtp_glue_result audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
1091         enum ast_rtp_glue_result audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
1092         struct ast_format_cap *cap_dst = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
1093         struct ast_format_cap *cap_src = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
1094
1095         /* Lock both channels so we can look for the glue that binds them together */
1096         ast_channel_lock_both(c_dst, c_src);
1097
1098         if (!cap_src || !cap_dst) {
1099                 goto done;
1100         }
1101
1102         /* Grab glue that binds each channel to something using the RTP engine */
1103         if (!(glue_dst = ast_rtp_instance_get_glue(ast_channel_tech(c_dst)->type)) || !(glue_src = ast_rtp_instance_get_glue(ast_channel_tech(c_src)->type))) {
1104                 ast_debug(1, "Can't find native functions for channel '%s'\n", glue_dst ? ast_channel_name(c_src) : ast_channel_name(c_dst));
1105                 goto done;
1106         }
1107
1108         audio_glue_dst_res = glue_dst->get_rtp_info(c_dst, &instance_dst);
1109         video_glue_dst_res = glue_dst->get_vrtp_info ? glue_dst->get_vrtp_info(c_dst, &vinstance_dst) : AST_RTP_GLUE_RESULT_FORBID;
1110
1111         audio_glue_src_res = glue_src->get_rtp_info(c_src, &instance_src);
1112         video_glue_src_res = glue_src->get_vrtp_info ? glue_src->get_vrtp_info(c_src, &vinstance_src) : AST_RTP_GLUE_RESULT_FORBID;
1113
1114         /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
1115         if (video_glue_dst_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE)) {
1116                 audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
1117         }
1118         if (video_glue_src_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE)) {
1119                 audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
1120         }
1121         if (audio_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_dst_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_dst->get_codec) {
1122                 glue_dst->get_codec(c_dst, cap_dst);
1123         }
1124         if (audio_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_src_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_src->get_codec) {
1125                 glue_src->get_codec(c_src, cap_src);
1126         }
1127
1128         /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
1129         if (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE) {
1130                 goto done;
1131         }
1132
1133         /* Make sure we have matching codecs */
1134         if (!ast_format_cap_has_joint(cap_dst, cap_src)) {
1135                 goto done;
1136         }
1137
1138         ast_rtp_codecs_payloads_copy(&instance_src->codecs, &instance_dst->codecs, instance_dst);
1139
1140         if (vinstance_dst && vinstance_src) {
1141                 ast_rtp_codecs_payloads_copy(&vinstance_src->codecs, &vinstance_dst->codecs, vinstance_dst);
1142         }
1143         if (tinstance_dst && tinstance_src) {
1144                 ast_rtp_codecs_payloads_copy(&tinstance_src->codecs, &tinstance_dst->codecs, tinstance_dst);
1145         }
1146
1147         if (glue_dst->update_peer(c_dst, instance_src, vinstance_src, tinstance_src, cap_src, 0)) {
1148                 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
1149                         ast_channel_name(c_dst), ast_channel_name(c_src));
1150         } else {
1151                 ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
1152                         ast_channel_name(c_dst), ast_channel_name(c_src));
1153         }
1154
1155 done:
1156         ast_channel_unlock(c_dst);
1157         ast_channel_unlock(c_src);
1158
1159         ast_format_cap_destroy(cap_dst);
1160         ast_format_cap_destroy(cap_src);
1161
1162         unref_instance_cond(&instance_dst);
1163         unref_instance_cond(&instance_src);
1164         unref_instance_cond(&vinstance_dst);
1165         unref_instance_cond(&vinstance_src);
1166         unref_instance_cond(&tinstance_dst);
1167         unref_instance_cond(&tinstance_src);
1168 }
1169
1170 int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
1171 {
1172         struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
1173                         *vinstance0 = NULL, *vinstance1 = NULL,
1174                         *tinstance0 = NULL, *tinstance1 = NULL;
1175         struct ast_rtp_glue *glue0, *glue1;
1176         enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1177         enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1178         struct ast_format_cap *cap0 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
1179         struct ast_format_cap *cap1 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
1180
1181         /* If there is no second channel just immediately bail out, we are of no use in that scenario */
1182         if (!c1 || !cap1 || !cap0) {
1183                 ast_format_cap_destroy(cap0);
1184                 ast_format_cap_destroy(cap1);
1185                 return -1;
1186         }
1187
1188         /* Lock both channels so we can look for the glue that binds them together */
1189         ast_channel_lock_both(c0, c1);
1190
1191         /* Grab glue that binds each channel to something using the RTP engine */
1192         if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
1193                 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
1194                 goto done;
1195         }
1196
1197         audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
1198         video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
1199
1200         audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
1201         video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
1202
1203         /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
1204         if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
1205                 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1206         }
1207         if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
1208                 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1209         }
1210         if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec) {
1211                 glue0->get_codec(c0, cap0);
1212         }
1213         if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec) {
1214                 glue1->get_codec(c1, cap1);
1215         }
1216
1217         /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
1218         if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
1219                 goto done;
1220         }
1221
1222         /* Make sure we have matching codecs */
1223         if (!ast_format_cap_has_joint(cap0, cap1)) {
1224                 goto done;
1225         }
1226
1227         /* Bridge media early */
1228         if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
1229                 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
1230         }
1231
1232 done:
1233         ast_channel_unlock(c0);
1234         ast_channel_unlock(c1);
1235
1236         ast_format_cap_destroy(cap0);
1237         ast_format_cap_destroy(cap1);
1238
1239         unref_instance_cond(&instance0);
1240         unref_instance_cond(&instance1);
1241         unref_instance_cond(&vinstance0);
1242         unref_instance_cond(&vinstance1);
1243         unref_instance_cond(&tinstance0);
1244         unref_instance_cond(&tinstance1);
1245
1246         ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
1247
1248         return 0;
1249 }
1250
1251 int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
1252 {
1253         return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
1254 }
1255
1256 int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
1257 {
1258         return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
1259 }
1260
1261 int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
1262 {
1263         return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
1264 }
1265
1266 char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size)
1267 {
1268         struct ast_rtp_instance_stats stats = { 0, };
1269         enum ast_rtp_instance_stat stat;
1270
1271         /* Determine what statistics we will need to retrieve based on field passed in */
1272         if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
1273                 stat = AST_RTP_INSTANCE_STAT_ALL;
1274         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
1275                 stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
1276         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
1277                 stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
1278         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
1279                 stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
1280         } else {
1281                 return NULL;
1282         }
1283
1284         /* Attempt to actually retrieve the statistics we need to generate the quality string */
1285         if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
1286                 return NULL;
1287         }
1288
1289         /* Now actually fill the buffer with the good information */
1290         if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
1291                 snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
1292                          stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.rxjitter, stats.rxcount, stats.txjitter, stats.txcount, stats.txploss, stats.rtt);
1293         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
1294                 snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
1295                          stats.local_minjitter, stats.local_maxjitter, stats.local_normdevjitter, sqrt(stats.local_stdevjitter), stats.remote_minjitter, stats.remote_maxjitter, stats.remote_normdevjitter, sqrt(stats.remote_stdevjitter));
1296         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
1297                 snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
1298                          stats.local_minrxploss, stats.local_maxrxploss, stats.local_normdevrxploss, sqrt(stats.local_stdevrxploss), stats.remote_minrxploss, stats.remote_maxrxploss, stats.remote_normdevrxploss, sqrt(stats.remote_stdevrxploss));
1299         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
1300                 snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
1301         }
1302
1303         return buf;
1304 }
1305
1306 void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
1307 {
1308         char quality_buf[AST_MAX_USER_FIELD], *quality;
1309         RAII_VAR(struct ast_channel *, bridge, ast_channel_bridge_peer(chan), ast_channel_cleanup);
1310
1311         if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
1312                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
1313                 if (bridge) {
1314                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
1315                 }
1316         }
1317
1318         if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
1319                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
1320                 if (bridge) {
1321                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
1322                 }
1323         }
1324
1325         if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
1326                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
1327                 if (bridge) {
1328                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
1329                 }
1330         }
1331
1332         if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
1333                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
1334                 if (bridge) {
1335                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
1336                 }
1337         }
1338 }
1339
1340 int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)
1341 {
1342         return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
1343 }
1344
1345 int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, struct ast_format *format)
1346 {
1347         return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
1348 }
1349
1350 int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer)
1351 {
1352         struct ast_rtp_glue *glue;
1353         struct ast_rtp_instance *peer_instance = NULL;
1354         int res = -1;
1355
1356         if (!instance->engine->make_compatible) {
1357                 return -1;
1358         }
1359
1360         ast_channel_lock(peer);
1361
1362         if (!(glue = ast_rtp_instance_get_glue(ast_channel_tech(peer)->type))) {
1363                 ast_channel_unlock(peer);
1364                 return -1;
1365         }
1366
1367         glue->get_rtp_info(peer, &peer_instance);
1368         if (!peer_instance) {
1369                 ast_log(LOG_ERROR, "Unable to get_rtp_info for peer type %s\n", glue->type);
1370                 ast_channel_unlock(peer);
1371                 return -1;
1372         }
1373         if (peer_instance->engine != instance->engine) {
1374                 ast_log(LOG_ERROR, "Peer engine mismatch for type %s\n", glue->type);
1375                 ast_channel_unlock(peer);
1376                 ao2_ref(peer_instance, -1);
1377                 return -1;
1378         }
1379
1380         res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
1381
1382         ast_channel_unlock(peer);
1383
1384         ao2_ref(peer_instance, -1);
1385         peer_instance = NULL;
1386
1387         return res;
1388 }
1389
1390 void ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, struct ast_format_cap *to_endpoint, struct ast_format_cap *to_asterisk, struct ast_format_cap *result)
1391 {
1392         if (instance->engine->available_formats) {
1393                 instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
1394                 if (!ast_format_cap_is_empty(result)) {
1395                         return;
1396                 }
1397         }
1398
1399         ast_translate_available_formats(to_endpoint, to_asterisk, result);
1400 }
1401
1402 int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
1403 {
1404         return instance->engine->activate ? instance->engine->activate(instance) : 0;
1405 }
1406
1407 void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance,
1408                                    struct ast_sockaddr *suggestion,
1409                                    const char *username)
1410 {
1411         if (instance->engine->stun_request) {
1412                 instance->engine->stun_request(instance, suggestion, username);
1413         }
1414 }
1415
1416 void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout)
1417 {
1418         instance->timeout = timeout;
1419 }
1420
1421 void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout)
1422 {
1423         instance->holdtimeout = timeout;
1424 }
1425
1426 void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int interval)
1427 {
1428         instance->keepalive = interval;
1429 }
1430
1431 int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
1432 {
1433         return instance->timeout;
1434 }
1435
1436 int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
1437 {
1438         return instance->holdtimeout;
1439 }
1440
1441 int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance)
1442 {
1443         return instance->keepalive;
1444 }
1445
1446 struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
1447 {
1448         return instance->engine;
1449 }
1450
1451 struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *instance)
1452 {
1453         return instance->glue;
1454 }
1455
1456 int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res)
1457 {
1458         if (res_srtp || res_srtp_policy) {
1459                 return -1;
1460         }
1461         if (!srtp_res || !policy_res) {
1462                 return -1;
1463         }
1464
1465         res_srtp = srtp_res;
1466         res_srtp_policy = policy_res;
1467
1468         return 0;
1469 }
1470
1471 void ast_rtp_engine_unregister_srtp(void)
1472 {
1473         res_srtp = NULL;
1474         res_srtp_policy = NULL;
1475 }
1476
1477 int ast_rtp_engine_srtp_is_registered(void)
1478 {
1479         return res_srtp && res_srtp_policy;
1480 }
1481
1482 int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy)
1483 {
1484         int res = 0;
1485
1486         if (!res_srtp) {
1487                 return -1;
1488         }
1489
1490         if (!instance->srtp) {
1491                 res = res_srtp->create(&instance->srtp, instance, remote_policy);
1492         } else {
1493                 res = res_srtp->replace(&instance->srtp, instance, remote_policy);
1494         }
1495         if (!res) {
1496                 res = res_srtp->add_stream(instance->srtp, local_policy);
1497         }
1498
1499         return res;
1500 }
1501
1502 struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
1503 {
1504         return instance->srtp;
1505 }
1506
1507 int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
1508 {
1509         if (instance->engine->sendcng) {
1510                 return instance->engine->sendcng(instance, level);
1511         }
1512
1513         return -1;
1514 }
1515
1516 struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *instance)
1517 {
1518         return instance->engine->ice;
1519 }
1520
1521 struct ast_rtp_engine_dtls *ast_rtp_instance_get_dtls(struct ast_rtp_instance *instance)
1522 {
1523         return instance->engine->dtls;
1524 }
1525
1526 int ast_rtp_dtls_cfg_parse(struct ast_rtp_dtls_cfg *dtls_cfg, const char *name, const char *value)
1527 {
1528         if (!strcasecmp(name, "dtlsenable")) {
1529                 dtls_cfg->enabled = ast_true(value) ? 1 : 0;
1530         } else if (!strcasecmp(name, "dtlsverify")) {
1531                 dtls_cfg->verify = ast_true(value) ? 1 : 0;
1532         } else if (!strcasecmp(name, "dtlsrekey")) {
1533                 if (sscanf(value, "%30u", &dtls_cfg->rekey) != 1) {
1534                         return -1;
1535                 }
1536         } else if (!strcasecmp(name, "dtlscertfile")) {
1537                 ast_free(dtls_cfg->certfile);
1538                 dtls_cfg->certfile = ast_strdup(value);
1539         } else if (!strcasecmp(name, "dtlsprivatekey")) {
1540                 ast_free(dtls_cfg->pvtfile);
1541                 dtls_cfg->pvtfile = ast_strdup(value);
1542         } else if (!strcasecmp(name, "dtlscipher")) {
1543                 ast_free(dtls_cfg->cipher);
1544                 dtls_cfg->cipher = ast_strdup(value);
1545         } else if (!strcasecmp(name, "dtlscafile")) {
1546                 ast_free(dtls_cfg->cafile);
1547                 dtls_cfg->cafile = ast_strdup(value);
1548         } else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
1549                 ast_free(dtls_cfg->capath);
1550                 dtls_cfg->capath = ast_strdup(value);
1551         } else if (!strcasecmp(name, "dtlssetup")) {
1552                 if (!strcasecmp(value, "active")) {
1553                         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTIVE;
1554                 } else if (!strcasecmp(value, "passive")) {
1555                         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_PASSIVE;
1556                 } else if (!strcasecmp(value, "actpass")) {
1557                         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTPASS;
1558                 }
1559         } else {
1560                 return -1;
1561         }
1562
1563         return 0;
1564 }
1565
1566 void ast_rtp_dtls_cfg_copy(const struct ast_rtp_dtls_cfg *src_cfg, struct ast_rtp_dtls_cfg *dst_cfg)
1567 {
1568         dst_cfg->enabled = src_cfg->enabled;
1569         dst_cfg->verify = src_cfg->verify;
1570         dst_cfg->rekey = src_cfg->rekey;
1571         dst_cfg->suite = src_cfg->suite;
1572         dst_cfg->certfile = ast_strdup(src_cfg->certfile);
1573         dst_cfg->pvtfile = ast_strdup(src_cfg->pvtfile);
1574         dst_cfg->cipher = ast_strdup(src_cfg->cipher);
1575         dst_cfg->cafile = ast_strdup(src_cfg->cafile);
1576         dst_cfg->capath = ast_strdup(src_cfg->capath);
1577         dst_cfg->default_setup = src_cfg->default_setup;
1578 }
1579
1580 void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg)
1581 {
1582         ast_free(dtls_cfg->certfile);
1583         ast_free(dtls_cfg->pvtfile);
1584         ast_free(dtls_cfg->cipher);
1585         ast_free(dtls_cfg->cafile);
1586         ast_free(dtls_cfg->capath);
1587 }
1588
1589 static void set_next_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate)
1590 {
1591         int x = mime_types_len;
1592         if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
1593                 return;
1594         }
1595
1596         ast_rwlock_wrlock(&mime_types_lock);
1597         if (format) {
1598                 ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
1599                 ast_format_copy(&ast_rtp_mime_types[x].payload_type.format, format);
1600         } else {
1601                 ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
1602         }
1603         ast_rtp_mime_types[x].type = type;
1604         ast_rtp_mime_types[x].subtype = subtype;
1605         ast_rtp_mime_types[x].sample_rate = sample_rate;
1606         mime_types_len++;
1607         ast_rwlock_unlock(&mime_types_lock);
1608 }
1609
1610 static void add_static_payload(int map, const struct ast_format *format, int rtp_code)
1611 {
1612         int x;
1613         ast_rwlock_wrlock(&static_RTP_PT_lock);
1614         if (map < 0) {
1615                 /* find next available dynamic payload slot */
1616                 for (x = 96; x < 127; x++) {
1617                         if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
1618                                 map = x;
1619                                 break;
1620                         }
1621                 }
1622         }
1623
1624         if (map < 0) {
1625                 ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n" ,ast_getformatname(format));
1626                 ast_rwlock_unlock(&static_RTP_PT_lock);
1627                 return;
1628         }
1629
1630         if (format) {
1631                 static_RTP_PT[map].asterisk_format = 1;
1632                 ast_format_copy(&static_RTP_PT[map].format, format);
1633         } else {
1634                 static_RTP_PT[map].rtp_code = rtp_code;
1635         }
1636         ast_rwlock_unlock(&static_RTP_PT_lock);
1637 }
1638
1639 int ast_rtp_engine_load_format(const struct ast_format *format)
1640 {
1641         switch (format->id) {
1642         case AST_FORMAT_SILK:
1643                 set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
1644                 add_static_payload(-1, format, 0);
1645                 break;
1646         case AST_FORMAT_CELT:
1647                 set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
1648                 add_static_payload(-1, format, 0);
1649                 break;
1650         default:
1651                 break;
1652         }
1653
1654         return 0;
1655 }
1656
1657 int ast_rtp_engine_unload_format(const struct ast_format *format)
1658 {
1659         int x;
1660         int y = 0;
1661
1662         ast_rwlock_wrlock(&static_RTP_PT_lock);
1663         /* remove everything pertaining to this format id from the lists */
1664         for (x = 0; x < AST_RTP_MAX_PT; x++) {
1665                 if (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
1666                         memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
1667                 }
1668         }
1669         ast_rwlock_unlock(&static_RTP_PT_lock);
1670
1671
1672         ast_rwlock_wrlock(&mime_types_lock);
1673         /* rebuild the list skipping the items matching this id */
1674         for (x = 0; x < mime_types_len; x++) {
1675                 if (ast_format_cmp(&ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
1676                         continue;
1677                 }
1678                 ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
1679                 y++;
1680         }
1681         mime_types_len = y;
1682         ast_rwlock_unlock(&mime_types_lock);
1683         return 0;
1684 }
1685
1686 /*!
1687  * \internal
1688  * \brief \ref stasis message payload for RTCP messages
1689  */
1690 struct rtcp_message_payload {
1691         struct ast_channel_snapshot *snapshot;  /*< The channel snapshot, if available */
1692         struct ast_rtp_rtcp_report *report;     /*< The RTCP report */
1693         struct ast_json *blob;                  /*< Extra JSON data to publish */
1694 };
1695
1696 static void rtcp_message_payload_dtor(void *obj)
1697 {
1698         struct rtcp_message_payload *payload = obj;
1699
1700         ao2_cleanup(payload->report);
1701         ao2_cleanup(payload->snapshot);
1702         ast_json_unref(payload->blob);
1703 }
1704
1705 static struct ast_manager_event_blob *rtcp_report_to_ami(struct stasis_message *msg)
1706 {
1707         struct rtcp_message_payload *payload = stasis_message_data(msg);
1708         RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
1709         RAII_VAR(struct ast_str *, packet_string, ast_str_create(512), ast_free);
1710         unsigned int ssrc = payload->report->ssrc;
1711         unsigned int type = payload->report->type;
1712         unsigned int report_count = payload->report->reception_report_count;
1713         int i;
1714
1715         if (!packet_string) {
1716                 return NULL;
1717         }
1718
1719         if (payload->snapshot) {
1720                 channel_string = ast_manager_build_channel_state_string(payload->snapshot);
1721                 if (!channel_string) {
1722                         return NULL;
1723                 }
1724         }
1725
1726         if (payload->blob) {
1727                 /* Optional data */
1728                 struct ast_json *to = ast_json_object_get(payload->blob, "to");
1729                 struct ast_json *from = ast_json_object_get(payload->blob, "from");
1730                 struct ast_json *rtt = ast_json_object_get(payload->blob, "rtt");
1731                 if (to) {
1732                         ast_str_append(&packet_string, 0, "To: %s\r\n", ast_json_string_get(to));
1733                 }
1734                 if (from) {
1735                         ast_str_append(&packet_string, 0, "From: %s\r\n", ast_json_string_get(from));
1736                 }
1737                 if (rtt) {
1738                         ast_str_append(&packet_string, 0, "RTT: %4.4f\r\n", ast_json_real_get(rtt));
1739                 }
1740         }
1741
1742         ast_str_append(&packet_string, 0, "SSRC: 0x%.8x\r\n", ssrc);
1743         ast_str_append(&packet_string, 0, "PT: %u(%s)\r\n", type, type== AST_RTP_RTCP_SR ? "SR" : "RR");
1744         ast_str_append(&packet_string, 0, "ReportCount: %u\r\n", report_count);
1745         if (type == AST_RTP_RTCP_SR) {
1746                 ast_str_append(&packet_string, 0, "SentNTP: %lu.%06lu\r\n",
1747                         (unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec,
1748                         (unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec * 4096);
1749                 ast_str_append(&packet_string, 0, "SentRTP: %u\r\n",
1750                                 payload->report->sender_information.rtp_timestamp);
1751                 ast_str_append(&packet_string, 0, "SentPackets: %u\r\n",
1752                                 payload->report->sender_information.packet_count);
1753                 ast_str_append(&packet_string, 0, "SentOctets: %u\r\n",
1754                                 payload->report->sender_information.octet_count);
1755         }
1756
1757         for (i = 0; i < report_count; i++) {
1758                 RAII_VAR(struct ast_str *, report_string, NULL, ast_free);
1759
1760                 if (!payload->report->report_block[i]) {
1761                         break;
1762                 }
1763
1764                 report_string = ast_str_create(256);
1765                 if (!report_string) {
1766                         return NULL;
1767                 }
1768
1769                 ast_str_append(&report_string, 0, "Report%dSourceSSRC: 0x%.8x\r\n",
1770                                 i, payload->report->report_block[i]->source_ssrc);
1771                 ast_str_append(&report_string, 0, "Report%dFractionLost: %u\r\n",
1772                                 i, payload->report->report_block[i]->lost_count.fraction);
1773                 ast_str_append(&report_string, 0, "Report%dCumulativeLost: %u\r\n",
1774                                 i, payload->report->report_block[i]->lost_count.packets);
1775                 ast_str_append(&report_string, 0, "Report%dHighestSequence: %u\r\n",
1776                                 i, payload->report->report_block[i]->highest_seq_no & 0xffff);
1777                 ast_str_append(&report_string, 0, "Report%dSequenceNumberCycles: %u\r\n",
1778                                 i, payload->report->report_block[i]->highest_seq_no >> 16);
1779                 ast_str_append(&report_string, 0, "Report%dIAJitter: %u\r\n",
1780                                 i, payload->report->report_block[i]->ia_jitter);
1781                 ast_str_append(&report_string, 0, "Report%dLSR: %u\r\n",
1782                                 i, payload->report->report_block[i]->lsr);
1783                 ast_str_append(&report_string, 0, "Report%dDLSR: %4.4f\r\n",
1784                                 i, ((double)payload->report->report_block[i]->dlsr) / 65536);
1785                 ast_str_append(&packet_string, 0, "%s", ast_str_buffer(report_string));
1786         }
1787
1788         return ast_manager_event_blob_create(EVENT_FLAG_REPORTING,
1789                 stasis_message_type(msg) == ast_rtp_rtcp_received_type() ? "RTCPReceived" : "RTCPSent",
1790                 "%s%s",
1791                 AS_OR(channel_string, ""),
1792                 ast_str_buffer(packet_string));
1793 }
1794
1795 static struct ast_json *rtcp_report_to_json(struct stasis_message *msg,
1796         const struct stasis_message_sanitizer *sanitize)
1797 {
1798         struct rtcp_message_payload *payload = stasis_message_data(msg);
1799         RAII_VAR(struct ast_json *, json_rtcp_report, NULL, ast_json_unref);
1800         RAII_VAR(struct ast_json *, json_rtcp_report_blocks, NULL, ast_json_unref);
1801         RAII_VAR(struct ast_json *, json_rtcp_sender_info, NULL, ast_json_unref);
1802         RAII_VAR(struct ast_json *, json_channel, NULL, ast_json_unref);
1803         int i;
1804
1805         json_rtcp_report_blocks = ast_json_array_create();
1806         if (!json_rtcp_report_blocks) {
1807                 return NULL;
1808         }
1809
1810         for (i = 0; i < payload->report->reception_report_count; i++) {
1811                 struct ast_json *json_report_block;
1812                 json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: i, s: i}",
1813                                 "source_ssrc", payload->report->report_block[i]->source_ssrc,
1814                                 "fraction_lost", payload->report->report_block[i]->lost_count.fraction,
1815                                 "packets_lost", payload->report->report_block[i]->lost_count.packets,
1816                                 "highest_seq_no", payload->report->report_block[i]->highest_seq_no,
1817                                 "ia_jitter", payload->report->report_block[i]->ia_jitter,
1818                                 "lsr", payload->report->report_block[i]->lsr,
1819                                 "dlsr", payload->report->report_block[i]->dlsr);
1820                 if (!json_report_block) {
1821                         return NULL;
1822                 }
1823
1824                 if (ast_json_array_append(json_rtcp_report_blocks, json_report_block)) {
1825                         return NULL;
1826                 }
1827         }
1828
1829         if (payload->report->type == AST_RTP_RTCP_SR) {
1830                 json_rtcp_sender_info = ast_json_pack("{s: i, s: i, s: i, s: i, s: i}",
1831                                 "ntp_timestamp_sec", payload->report->sender_information.ntp_timestamp.tv_sec,
1832                                 "ntp_timestamp_usec", payload->report->sender_information.ntp_timestamp.tv_usec,
1833                                 "rtp_timestamp", payload->report->sender_information.rtp_timestamp,
1834                                 "packets", payload->report->sender_information.packet_count,
1835                                 "octets", payload->report->sender_information.octet_count);
1836                 if (!json_rtcp_sender_info) {
1837                         return NULL;
1838                 }
1839         }
1840
1841         json_rtcp_report = ast_json_pack("{s: i, s: i, s: i, s: O, s: O}",
1842                         "ssrc", payload->report->ssrc,
1843                         "type", payload->report->type,
1844                         "report_count", payload->report->reception_report_count,
1845                         "sender_information", json_rtcp_sender_info ? json_rtcp_sender_info : ast_json_null(),
1846                         "report_blocks", json_rtcp_report_blocks);
1847         if (!json_rtcp_report) {
1848                 return NULL;
1849         }
1850
1851         if (payload->snapshot) {
1852                 json_channel = ast_channel_snapshot_to_json(payload->snapshot, sanitize);
1853                 if (!json_channel) {
1854                         return NULL;
1855                 }
1856         }
1857
1858         return ast_json_pack("{s: O, s: O, s: O}",
1859                 "channel", payload->snapshot ? json_channel : ast_json_null(),
1860                 "rtcp_report", json_rtcp_report,
1861                 "blob", payload->blob);
1862 }
1863
1864 static void rtp_rtcp_report_dtor(void *obj)
1865 {
1866         int i;
1867         struct ast_rtp_rtcp_report *rtcp_report = obj;
1868
1869         for (i = 0; i < rtcp_report->reception_report_count; i++) {
1870                 ast_free(rtcp_report->report_block[i]);
1871         }
1872 }
1873
1874 struct ast_rtp_rtcp_report *ast_rtp_rtcp_report_alloc(unsigned int report_blocks)
1875 {
1876         struct ast_rtp_rtcp_report *rtcp_report;
1877
1878         /* Size of object is sizeof the report + the number of report_blocks * sizeof pointer */
1879         rtcp_report = ao2_alloc((sizeof(*rtcp_report) + report_blocks * sizeof(struct ast_rtp_rtcp_report_block *)),
1880                 rtp_rtcp_report_dtor);
1881
1882         return rtcp_report;
1883 }
1884
1885 void ast_rtp_publish_rtcp_message(struct ast_rtp_instance *rtp,
1886                 struct stasis_message_type *message_type,
1887                 struct ast_rtp_rtcp_report *report,
1888                 struct ast_json *blob)
1889 {
1890         RAII_VAR(struct rtcp_message_payload *, payload,
1891                         ao2_alloc(sizeof(*payload), rtcp_message_payload_dtor), ao2_cleanup);
1892         RAII_VAR(struct ast_channel_snapshot *, snapshot, NULL, ao2_cleanup);
1893         RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
1894
1895         if (!payload || !report) {
1896                 return;
1897         }
1898
1899         if (!ast_strlen_zero(rtp->channel_uniqueid)) {
1900                 snapshot = ast_channel_snapshot_get_latest(rtp->channel_uniqueid);
1901                 if (snapshot) {
1902                         ao2_ref(snapshot, +1);
1903                 }
1904         }
1905
1906         if (blob) {
1907                 ast_json_ref(blob);
1908         }
1909         ao2_ref(report, 1);
1910         payload->snapshot = snapshot;
1911         payload->blob = blob;
1912         payload->report = report;
1913
1914         message = stasis_message_create(message_type, payload);
1915         if (!message) {
1916                 return;
1917         }
1918
1919         stasis_publish(ast_rtp_topic(), message);
1920 }
1921
1922 /*!
1923  * @{ \brief Define RTCP/RTP message types.
1924  */
1925 STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_sent_type,
1926                 .to_ami = rtcp_report_to_ami,
1927                 .to_json = rtcp_report_to_json,);
1928 STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_received_type,
1929                 .to_ami = rtcp_report_to_ami,
1930                 .to_json = rtcp_report_to_json,);
1931 /*! @} */
1932
1933 struct stasis_topic *ast_rtp_topic(void)
1934 {
1935         return rtp_topic;
1936 }
1937
1938 static void rtp_engine_shutdown(void)
1939 {
1940         ao2_cleanup(rtp_topic);
1941         rtp_topic = NULL;
1942         STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_received_type);
1943         STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_sent_type);
1944 }
1945
1946 int ast_rtp_engine_init()
1947 {
1948         struct ast_format tmpfmt;
1949
1950         ast_rwlock_init(&mime_types_lock);
1951         ast_rwlock_init(&static_RTP_PT_lock);
1952
1953         rtp_topic = stasis_topic_create("rtp_topic");
1954         if (!rtp_topic) {
1955                 return -1;
1956         }
1957         STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_sent_type);
1958         STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_received_type);
1959         ast_register_atexit(rtp_engine_shutdown);
1960
1961         /* Define all the RTP mime types available */
1962         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0, "audio", "G723", 8000);
1963         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0, "audio", "GSM", 8000);
1964         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "PCMU", 8000);
1965         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "G711U", 8000);
1966         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "PCMA", 8000);
1967         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "G711A", 8000);
1968         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0, "audio", "G726-32", 8000);
1969         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0, "audio", "DVI4", 8000);
1970         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0, "audio", "L16", 8000);
1971         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16", 16000);
1972         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16-256", 16000);
1973         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0, "audio", "LPC", 8000);
1974         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729", 8000);
1975         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729A", 8000);
1976         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G.729", 8000);
1977         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0, "audio", "speex", 8000);
1978         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0,  "audio", "speex", 16000);
1979         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0,  "audio", "speex", 32000);
1980         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0, "audio", "iLBC", 8000);
1981         /* this is the sample rate listed in the RTP profile for the G.722 codec, *NOT* the actual sample rate of the media stream */
1982         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0, "audio", "G722", 8000);
1983         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0, "audio", "AAL2-G726-32", 8000);
1984         set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
1985         set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
1986         set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
1987         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0, "video", "JPEG", 90000);
1988         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_PNG, 0), 0, "video", "PNG", 90000);
1989         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0, "video", "H261", 90000);
1990         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0, "video", "H263", 90000);
1991         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0, "video", "h263-1998", 90000);
1992         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0, "video", "H264", 90000);
1993         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0, "video", "MP4V-ES", 90000);
1994         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0, "text", "RED", 1000);
1995         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0, "text", "T140", 1000);
1996         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
1997         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
1998         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
1999         /* Opus and VP8 */
2000         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0,  "audio", "opus", 48000);
2001         set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0,  "video", "VP8", 90000);
2002
2003         /* Define the static rtp payload mappings */
2004         add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
2005         #ifdef USE_DEPRECATED_G726
2006         add_static_payload(2, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
2007         #endif
2008         add_static_payload(3, ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0);
2009         add_static_payload(4, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0);
2010         add_static_payload(5, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0);/* 8 kHz */
2011         add_static_payload(6, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 16 kHz */
2012         add_static_payload(7, ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0);
2013         add_static_payload(8, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0);
2014         add_static_payload(9, ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0);
2015         add_static_payload(10, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 2 channels */
2016         add_static_payload(11, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 1 channel */
2017         add_static_payload(13, NULL, AST_RTP_CN);
2018         add_static_payload(16, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 11.025 kHz */
2019         add_static_payload(17, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 22.050 kHz */
2020         add_static_payload(18, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0);
2021         add_static_payload(19, NULL, AST_RTP_CN);         /* Also used for CN */
2022         add_static_payload(26, ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0);
2023         add_static_payload(31, ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0);
2024         add_static_payload(34, ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0);
2025         add_static_payload(97, ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0);
2026         add_static_payload(98, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
2027         add_static_payload(99, ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0);
2028         add_static_payload(101, NULL, AST_RTP_DTMF);
2029         add_static_payload(102, ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0);
2030         add_static_payload(103, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
2031         add_static_payload(104, ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0);
2032         add_static_payload(105, ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0);   /* Real time text chat (with redundancy encoding) */
2033         add_static_payload(106, ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0);     /* Real time text chat */
2034         add_static_payload(110, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0);
2035         add_static_payload(111, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);
2036         add_static_payload(112, ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0);
2037         add_static_payload(115, ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0);
2038         add_static_payload(116, ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0);
2039         add_static_payload(117, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0);
2040         add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
2041         add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
2042         add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */
2043         /* Opus and VP8 */
2044         add_static_payload(100, ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0);
2045         add_static_payload(107, ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0);
2046
2047         return 0;
2048 }