rtp_engine.h: No sense allowing payload types larger than RFC allows.
[asterisk/asterisk.git] / 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_REGISTER_FILE()
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         /*! The original source address */
174         struct ast_sockaddr requested_target_address;
175         /*! Address that we are sending RTP to */
176         struct ast_sockaddr incoming_source_address;
177         /*! Instance that we are bridged to if doing remote or local bridging */
178         struct ast_rtp_instance *bridged;
179         /*! Payload and packetization information */
180         struct ast_rtp_codecs codecs;
181         /*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
182         int timeout;
183         /*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
184         int holdtimeout;
185         /*! RTP keepalive interval */
186         int keepalive;
187         /*! Glue currently in use */
188         struct ast_rtp_glue *glue;
189         /*! SRTP info associated with the instance */
190         struct ast_srtp *srtp;
191         /*! Channel unique ID */
192         char channel_uniqueid[AST_MAX_UNIQUEID];
193         /*! Time of last packet sent */
194         time_t last_tx;
195         /*! Time of last packet received */
196         time_t last_rx;
197 };
198
199 /*! List of RTP engines that are currently registered */
200 static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
201
202 /*! List of RTP glues */
203 static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
204
205 #define MAX_RTP_MIME_TYPES 128
206
207 /*! The following array defines the MIME Media type (and subtype) for each
208    of our codecs, or RTP-specific data type. */
209 static struct ast_rtp_mime_type {
210         /*! \brief A mapping object between the Asterisk codec and this RTP payload */
211         struct ast_rtp_payload_type payload_type;
212         /*! \brief The media type */
213         char type[16];
214         /*! \brief The format type */
215         char subtype[64];
216         /*! \brief Expected sample rate of the /c subtype */
217         unsigned int sample_rate;
218 } ast_rtp_mime_types[128]; /* This will Likely not need to grow any time soon. */
219 static ast_rwlock_t mime_types_lock;
220 static int mime_types_len = 0;
221
222 /*!
223  * \brief Mapping between Asterisk codecs and rtp payload types
224  *
225  * Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
226  * also, our own choices for dynamic payload types.  This is our master
227  * table for transmission
228  *
229  * See http://www.iana.org/assignments/rtp-parameters for a list of
230  * assigned values
231  */
232 static struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT];
233 static ast_rwlock_t static_RTP_PT_lock;
234
235 /*! \brief \ref stasis topic for RTP related messages */
236 static struct stasis_topic *rtp_topic;
237
238
239 /*!
240  * \internal
241  * \brief Destructor for \c ast_rtp_payload_type
242  */
243 static void rtp_payload_type_dtor(void *obj)
244 {
245         struct ast_rtp_payload_type *payload = obj;
246
247         ao2_cleanup(payload->format);
248 }
249
250 struct ast_rtp_payload_type *ast_rtp_engine_alloc_payload_type(void)
251 {
252         struct ast_rtp_payload_type *payload;
253
254         payload = ao2_alloc_options(sizeof(*payload), rtp_payload_type_dtor,
255                 AO2_ALLOC_OPT_LOCK_NOLOCK);
256
257         return payload;
258 }
259
260 int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
261 {
262         struct ast_rtp_engine *current_engine;
263
264         /* Perform a sanity check on the engine structure to make sure it has the basics */
265         if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
266                 ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
267                 return -1;
268         }
269
270         /* Link owner module to the RTP engine for reference counting purposes */
271         engine->mod = module;
272
273         AST_RWLIST_WRLOCK(&engines);
274
275         /* Ensure that no two modules with the same name are registered at the same time */
276         AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
277                 if (!strcmp(current_engine->name, engine->name)) {
278                         ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
279                         AST_RWLIST_UNLOCK(&engines);
280                         return -1;
281                 }
282         }
283
284         /* The engine survived our critique. Off to the list it goes to be used */
285         AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
286
287         AST_RWLIST_UNLOCK(&engines);
288
289         ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
290
291         return 0;
292 }
293
294 int ast_rtp_engine_unregister(struct ast_rtp_engine *engine)
295 {
296         struct ast_rtp_engine *current_engine = NULL;
297
298         AST_RWLIST_WRLOCK(&engines);
299
300         if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
301                 ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
302         }
303
304         AST_RWLIST_UNLOCK(&engines);
305
306         return current_engine ? 0 : -1;
307 }
308
309 int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module)
310 {
311         struct ast_rtp_glue *current_glue = NULL;
312
313         if (ast_strlen_zero(glue->type)) {
314                 return -1;
315         }
316
317         glue->mod = module;
318
319         AST_RWLIST_WRLOCK(&glues);
320
321         AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
322                 if (!strcasecmp(current_glue->type, glue->type)) {
323                         ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
324                         AST_RWLIST_UNLOCK(&glues);
325                         return -1;
326                 }
327         }
328
329         AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
330
331         AST_RWLIST_UNLOCK(&glues);
332
333         ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
334
335         return 0;
336 }
337
338 int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
339 {
340         struct ast_rtp_glue *current_glue = NULL;
341
342         AST_RWLIST_WRLOCK(&glues);
343
344         if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
345                 ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
346         }
347
348         AST_RWLIST_UNLOCK(&glues);
349
350         return current_glue ? 0 : -1;
351 }
352
353 static void instance_destructor(void *obj)
354 {
355         struct ast_rtp_instance *instance = obj;
356
357         /* Pass us off to the engine to destroy */
358         if (instance->data && instance->engine->destroy(instance)) {
359                 ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
360                 return;
361         }
362
363         if (instance->srtp) {
364                 res_srtp->destroy(instance->srtp);
365         }
366
367         ast_rtp_codecs_payloads_destroy(&instance->codecs);
368
369         /* Drop our engine reference */
370         ast_module_unref(instance->engine->mod);
371
372         ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
373 }
374
375 int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
376 {
377         ao2_ref(instance, -1);
378
379         return 0;
380 }
381
382 struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name,
383                 struct ast_sched_context *sched, const struct ast_sockaddr *sa,
384                 void *data)
385 {
386         struct ast_sockaddr address = {{0,}};
387         struct ast_rtp_instance *instance = NULL;
388         struct ast_rtp_engine *engine = NULL;
389
390         AST_RWLIST_RDLOCK(&engines);
391
392         /* If an engine name was specified try to use it or otherwise use the first one registered */
393         if (!ast_strlen_zero(engine_name)) {
394                 AST_RWLIST_TRAVERSE(&engines, engine, entry) {
395                         if (!strcmp(engine->name, engine_name)) {
396                                 break;
397                         }
398                 }
399         } else {
400                 engine = AST_RWLIST_FIRST(&engines);
401         }
402
403         /* If no engine was actually found bail out now */
404         if (!engine) {
405                 ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
406                 AST_RWLIST_UNLOCK(&engines);
407                 return NULL;
408         }
409
410         /* Bump up the reference count before we return so the module can not be unloaded */
411         ast_module_ref(engine->mod);
412
413         AST_RWLIST_UNLOCK(&engines);
414
415         /* Allocate a new RTP instance */
416         if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
417                 ast_module_unref(engine->mod);
418                 return NULL;
419         }
420         instance->engine = engine;
421         ast_sockaddr_copy(&instance->local_address, sa);
422         ast_sockaddr_copy(&address, sa);
423
424         if (ast_rtp_codecs_payloads_initialize(&instance->codecs)) {
425                 ao2_ref(instance, -1);
426                 return NULL;
427         }
428
429         ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
430
431         /* And pass it off to the engine to setup */
432         if (instance->engine->new(instance, sched, &address, data)) {
433                 ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
434                 ao2_ref(instance, -1);
435                 return NULL;
436         }
437
438         ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
439
440         return instance;
441 }
442
443 const char *ast_rtp_instance_get_channel_id(struct ast_rtp_instance *instance)
444 {
445         return instance->channel_uniqueid;
446 }
447
448 void ast_rtp_instance_set_channel_id(struct ast_rtp_instance *instance, const char *uniqueid)
449 {
450         ast_copy_string(instance->channel_uniqueid, uniqueid, sizeof(instance->channel_uniqueid));
451 }
452
453 void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
454 {
455         instance->data = data;
456 }
457
458 void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance)
459 {
460         return instance->data;
461 }
462
463 int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
464 {
465         return instance->engine->write(instance, frame);
466 }
467
468 struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp)
469 {
470         return instance->engine->read(instance, rtcp);
471 }
472
473 int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance,
474                 const struct ast_sockaddr *address)
475 {
476         ast_sockaddr_copy(&instance->local_address, address);
477         return 0;
478 }
479
480 int ast_rtp_instance_set_incoming_source_address(struct ast_rtp_instance *instance,
481                                                  const struct ast_sockaddr *address)
482 {
483         ast_sockaddr_copy(&instance->incoming_source_address, address);
484
485         /* moo */
486
487         if (instance->engine->remote_address_set) {
488                 instance->engine->remote_address_set(instance, &instance->incoming_source_address);
489         }
490
491         return 0;
492 }
493
494 int ast_rtp_instance_set_requested_target_address(struct ast_rtp_instance *instance,
495                                                   const struct ast_sockaddr *address)
496 {
497         ast_sockaddr_copy(&instance->requested_target_address, address);
498
499         return ast_rtp_instance_set_incoming_source_address(instance, address);
500 }
501
502 int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance,
503                 struct ast_sockaddr *address)
504 {
505         if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
506                 ast_sockaddr_copy(address, &instance->local_address);
507                 return 1;
508         }
509
510         return 0;
511 }
512
513 void ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance,
514                 struct ast_sockaddr *address)
515 {
516         ast_sockaddr_copy(address, &instance->local_address);
517 }
518
519 int ast_rtp_instance_get_and_cmp_requested_target_address(struct ast_rtp_instance *instance,
520                 struct ast_sockaddr *address)
521 {
522         if (ast_sockaddr_cmp(address, &instance->requested_target_address) != 0) {
523                 ast_sockaddr_copy(address, &instance->requested_target_address);
524                 return 1;
525         }
526
527         return 0;
528 }
529
530 void ast_rtp_instance_get_incoming_source_address(struct ast_rtp_instance *instance,
531                                                   struct ast_sockaddr *address)
532 {
533         ast_sockaddr_copy(address, &instance->incoming_source_address);
534 }
535
536 void ast_rtp_instance_get_requested_target_address(struct ast_rtp_instance *instance,
537                                                    struct ast_sockaddr *address)
538 {
539         ast_sockaddr_copy(address, &instance->requested_target_address);
540 }
541
542 void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
543 {
544         if (instance->engine->extended_prop_set) {
545                 instance->engine->extended_prop_set(instance, property, value);
546         }
547 }
548
549 void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property)
550 {
551         if (instance->engine->extended_prop_get) {
552                 return instance->engine->extended_prop_get(instance, property);
553         }
554
555         return NULL;
556 }
557
558 void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
559 {
560         instance->properties[property] = value;
561
562         if (instance->engine->prop_set) {
563                 instance->engine->prop_set(instance, property, value);
564         }
565 }
566
567 int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property)
568 {
569         return instance->properties[property];
570 }
571
572 struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance)
573 {
574         return &instance->codecs;
575 }
576
577 int ast_rtp_codecs_payloads_initialize(struct ast_rtp_codecs *codecs)
578 {
579         int res;
580
581         codecs->framing = 0;
582         ast_rwlock_init(&codecs->codecs_lock);
583         res = AST_VECTOR_INIT(&codecs->payloads, AST_RTP_MAX_PT);
584
585         return res;
586 }
587
588 void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
589 {
590         int i;
591
592         for (i = 0; i < AST_VECTOR_SIZE(&codecs->payloads); i++) {
593                 struct ast_rtp_payload_type *type;
594
595                 type = AST_VECTOR_GET(&codecs->payloads, i);
596                 ao2_t_cleanup(type, "destroying ast_rtp_codec");
597         }
598         AST_VECTOR_FREE(&codecs->payloads);
599
600         ast_rwlock_destroy(&codecs->codecs_lock);
601 }
602
603 void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
604 {
605         ast_rtp_codecs_payloads_destroy(codecs);
606
607         if (instance && instance->engine && instance->engine->payload_set) {
608                 int i;
609                 for (i = 0; i < AST_RTP_MAX_PT; i++) {
610                         instance->engine->payload_set(instance, i, 0, NULL, 0);
611                 }
612         }
613
614         ast_rtp_codecs_payloads_initialize(codecs);
615 }
616
617 void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
618 {
619         int i;
620
621         ast_rwlock_rdlock(&src->codecs_lock);
622         ast_rwlock_wrlock(&dest->codecs_lock);
623
624         for (i = 0; i < AST_VECTOR_SIZE(&src->payloads); i++) {
625                 struct ast_rtp_payload_type *type;
626
627                 type = AST_VECTOR_GET(&src->payloads, i);
628                 if (!type) {
629                         continue;
630                 }
631                 if (i < AST_VECTOR_SIZE(&dest->payloads)) {
632                         ao2_t_cleanup(AST_VECTOR_GET(&dest->payloads, i), "cleaning up vector element about to be replaced");
633                 }
634                 ast_debug(2, "Copying payload %d (%p) from %p to %p\n", i, type, src, dest);
635                 ao2_bump(type);
636                 AST_VECTOR_REPLACE(&dest->payloads, i, type);
637
638                 if (instance && instance->engine && instance->engine->payload_set) {
639                         instance->engine->payload_set(instance, i, type->asterisk_format, type->format, type->rtp_code);
640                 }
641         }
642         dest->framing = src->framing;
643         ast_rwlock_unlock(&dest->codecs_lock);
644         ast_rwlock_unlock(&src->codecs_lock);
645 }
646
647 void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
648 {
649         struct ast_rtp_payload_type *new_type;
650
651         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
652                 return;
653         }
654
655         new_type = ast_rtp_engine_alloc_payload_type();
656         if (!new_type) {
657                 return;
658         }
659
660         ast_rwlock_rdlock(&static_RTP_PT_lock);
661         ast_rwlock_wrlock(&codecs->codecs_lock);
662         if (payload < AST_VECTOR_SIZE(&codecs->payloads)) {
663                 ao2_t_cleanup(AST_VECTOR_GET(&codecs->payloads, payload), "cleaning up replaced payload type");
664         }
665
666         new_type->asterisk_format = static_RTP_PT[payload].asterisk_format;
667         new_type->rtp_code = static_RTP_PT[payload].rtp_code;
668         new_type->payload = payload;
669         new_type->format = ao2_bump(static_RTP_PT[payload].format);
670
671         ast_debug(1, "Setting payload %d (%p) based on m type on %p\n", payload, new_type, codecs);
672         AST_VECTOR_REPLACE(&codecs->payloads, payload, new_type);
673
674         if (instance && instance->engine && instance->engine->payload_set) {
675                 instance->engine->payload_set(instance, payload, new_type->asterisk_format, new_type->format, new_type->rtp_code);
676         }
677
678         ast_rwlock_unlock(&codecs->codecs_lock);
679         ast_rwlock_unlock(&static_RTP_PT_lock);
680 }
681
682 int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
683                                  char *mimetype, char *mimesubtype,
684                                  enum ast_rtp_options options,
685                                  unsigned int sample_rate)
686 {
687         unsigned int i;
688         int found = 0;
689
690         if (pt < 0 || pt >= AST_RTP_MAX_PT) {
691                 return -1; /* bogus payload type */
692         }
693
694         ast_rwlock_rdlock(&mime_types_lock);
695         ast_rwlock_wrlock(&codecs->codecs_lock);
696         for (i = 0; i < mime_types_len; ++i) {
697                 const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
698                 struct ast_rtp_payload_type *new_type;
699
700                 if (strcasecmp(mimesubtype, t->subtype)) {
701                         continue;
702                 }
703
704                 if (strcasecmp(mimetype, t->type)) {
705                         continue;
706                 }
707
708                 /* if both sample rates have been supplied, and they don't match,
709                  * then this not a match; if one has not been supplied, then the
710                  * rates are not compared */
711                 if (sample_rate && t->sample_rate &&
712                     (sample_rate != t->sample_rate)) {
713                         continue;
714                 }
715
716                 found = 1;
717
718                 new_type = ast_rtp_engine_alloc_payload_type();
719                 if (!new_type) {
720                         continue;
721                 }
722
723                 if (pt < AST_VECTOR_SIZE(&codecs->payloads)) {
724                         ao2_t_cleanup(AST_VECTOR_GET(&codecs->payloads, pt), "cleaning up replaced payload type");
725                 }
726
727                 new_type->payload = pt;
728                 new_type->asterisk_format = t->payload_type.asterisk_format;
729                 new_type->rtp_code = t->payload_type.rtp_code;
730                 if ((ast_format_cmp(t->payload_type.format, ast_format_g726) == AST_FORMAT_CMP_EQUAL) &&
731                                 t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
732                         new_type->format = ao2_bump(ast_format_g726_aal2);
733                 } else {
734                         new_type->format = ao2_bump(t->payload_type.format);
735                 }
736                 AST_VECTOR_REPLACE(&codecs->payloads, pt, new_type);
737
738                 if (instance && instance->engine && instance->engine->payload_set) {
739                         instance->engine->payload_set(instance, pt, new_type->asterisk_format, new_type->format, new_type->rtp_code);
740                 }
741
742                 break;
743         }
744         ast_rwlock_unlock(&codecs->codecs_lock);
745         ast_rwlock_unlock(&mime_types_lock);
746
747         return (found ? 0 : -2);
748 }
749
750 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)
751 {
752         return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
753 }
754
755 void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
756 {
757         struct ast_rtp_payload_type *type;
758
759         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
760                 return;
761         }
762
763         ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
764
765         ast_rwlock_wrlock(&codecs->codecs_lock);
766         if (payload < AST_VECTOR_SIZE(&codecs->payloads)) {
767                 type = AST_VECTOR_GET(&codecs->payloads, payload);
768                 ao2_cleanup(type);
769                 AST_VECTOR_REPLACE(&codecs->payloads, payload, NULL);
770         }
771
772         if (instance && instance->engine && instance->engine->payload_set) {
773                 instance->engine->payload_set(instance, payload, 0, NULL, 0);
774         }
775
776         ast_rwlock_unlock(&codecs->codecs_lock);
777 }
778
779 struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *codecs, int payload)
780 {
781         struct ast_rtp_payload_type *type = NULL;
782
783         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
784                 return NULL;
785         }
786
787         ast_rwlock_rdlock(&codecs->codecs_lock);
788         if (payload < AST_VECTOR_SIZE(&codecs->payloads)) {
789                 type = AST_VECTOR_GET(&codecs->payloads, payload);
790                 ao2_bump(type);
791         }
792         ast_rwlock_unlock(&codecs->codecs_lock);
793
794         if (!type) {
795                 type = ast_rtp_engine_alloc_payload_type();
796                 if (!type) {
797                         return NULL;
798                 }
799                 ast_rwlock_rdlock(&static_RTP_PT_lock);
800                 type->asterisk_format = static_RTP_PT[payload].asterisk_format;
801                 type->rtp_code = static_RTP_PT[payload].rtp_code;
802                 type->payload = payload;
803                 type->format = ao2_bump(static_RTP_PT[payload].format);
804                 ast_rwlock_unlock(&static_RTP_PT_lock);
805         }
806
807         return type;
808 }
809
810 int ast_rtp_codecs_payload_replace_format(struct ast_rtp_codecs *codecs, int payload, struct ast_format *format)
811 {
812         struct ast_rtp_payload_type *type;
813
814         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
815                 return -1;
816         }
817
818         ast_rwlock_wrlock(&codecs->codecs_lock);
819         if (payload < AST_VECTOR_SIZE(&codecs->payloads)) {
820                 type = AST_VECTOR_GET(&codecs->payloads, payload);
821                 if (type && type->asterisk_format) {
822                         ao2_replace(type->format, format);
823                 }
824         }
825         ast_rwlock_unlock(&codecs->codecs_lock);
826
827         return 0;
828 }
829
830 struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload)
831 {
832         struct ast_rtp_payload_type *type;
833         struct ast_format *format = NULL;
834
835         if (payload < 0 || payload >= AST_RTP_MAX_PT) {
836                 return NULL;
837         }
838
839         ast_rwlock_rdlock(&codecs->codecs_lock);
840         if (payload < AST_VECTOR_SIZE(&codecs->payloads)) {
841                 type = AST_VECTOR_GET(&codecs->payloads, payload);
842                 if (type && type->asterisk_format) {
843                         format = ao2_bump(type->format);
844                 }
845         }
846         ast_rwlock_unlock(&codecs->codecs_lock);
847
848         return format;
849 }
850
851 void ast_rtp_codecs_set_framing(struct ast_rtp_codecs *codecs, unsigned int framing)
852 {
853         if (!framing) {
854                 return;
855         }
856
857         ast_rwlock_wrlock(&codecs->codecs_lock);
858         codecs->framing = framing;
859         ast_rwlock_unlock(&codecs->codecs_lock);
860 }
861
862 unsigned int ast_rtp_codecs_get_framing(struct ast_rtp_codecs *codecs)
863 {
864         unsigned int framing;
865
866         ast_rwlock_rdlock(&codecs->codecs_lock);
867         framing = codecs->framing;
868         ast_rwlock_unlock(&codecs->codecs_lock);
869
870         return framing;
871 }
872
873 void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
874 {
875         int i;
876
877         ast_format_cap_remove_by_type(astformats, AST_MEDIA_TYPE_UNKNOWN);
878         *nonastformats = 0;
879
880         ast_rwlock_rdlock(&codecs->codecs_lock);
881         for (i = 0; i < AST_VECTOR_SIZE(&codecs->payloads); i++) {
882                 struct ast_rtp_payload_type *type;
883
884                 type = AST_VECTOR_GET(&codecs->payloads, i);
885                 if (!type) {
886                         continue;
887                 }
888
889                 if (type->asterisk_format) {
890                         ast_format_cap_append(astformats, type->format, 0);
891                 } else {
892                         *nonastformats |= type->rtp_code;
893                 }
894         }
895
896         if (codecs->framing) {
897                 ast_format_cap_set_framing(astformats, codecs->framing);
898         }
899
900         ast_rwlock_unlock(&codecs->codecs_lock);
901 }
902
903 int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
904 {
905         struct ast_rtp_payload_type *type;
906         int i;
907         int payload = -1;
908
909         ast_rwlock_rdlock(&codecs->codecs_lock);
910         for (i = 0; i < AST_VECTOR_SIZE(&codecs->payloads); i++) {
911                 type = AST_VECTOR_GET(&codecs->payloads, i);
912                 if (!type) {
913                         continue;
914                 }
915
916                 if ((asterisk_format && format && ast_format_cmp(format, type->format) == AST_FORMAT_CMP_EQUAL)
917                         || (!asterisk_format && type->rtp_code == code)) {
918                         payload = i;
919                         break;
920                 }
921         }
922         ast_rwlock_unlock(&codecs->codecs_lock);
923
924         if (payload < 0) {
925                 ast_rwlock_rdlock(&static_RTP_PT_lock);
926                 for (i = 0; i < AST_RTP_MAX_PT; i++) {
927                         if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
928                                 (ast_format_cmp(format, static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
929                                 payload = i;
930                                 break;
931                         } else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
932                                 (static_RTP_PT[i].rtp_code == code)) {
933                                 payload = i;
934                                 break;
935                         }
936                 }
937                 ast_rwlock_unlock(&static_RTP_PT_lock);
938         }
939
940         return payload;
941 }
942
943 int ast_rtp_codecs_find_payload_code(struct ast_rtp_codecs *codecs, int code)
944 {
945         struct ast_rtp_payload_type *type;
946         int res = -1;
947
948         ast_rwlock_rdlock(&codecs->codecs_lock);
949         if (code < AST_VECTOR_SIZE(&codecs->payloads)) {
950                 type = AST_VECTOR_GET(&codecs->payloads, code);
951                 if (type) {
952                         res = type->payload;
953                 }
954         }
955         ast_rwlock_unlock(&codecs->codecs_lock);
956
957         return res;
958 }
959
960 const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, struct ast_format *format, int code, enum ast_rtp_options options)
961 {
962         int i;
963         const char *res = "";
964
965         ast_rwlock_rdlock(&mime_types_lock);
966         for (i = 0; i < mime_types_len; i++) {
967                 if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
968                         (ast_format_cmp(format, ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
969                         if ((ast_format_cmp(format, ast_format_g726_aal2) == AST_FORMAT_CMP_EQUAL) &&
970                                         (options & AST_RTP_OPT_G726_NONSTANDARD)) {
971                                 res = "G726-32";
972                                 break;
973                         } else {
974                                 res = ast_rtp_mime_types[i].subtype;
975                                 break;
976                         }
977                 } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
978                         ast_rtp_mime_types[i].payload_type.rtp_code == code) {
979
980                         res = ast_rtp_mime_types[i].subtype;
981                         break;
982                 }
983         }
984         ast_rwlock_unlock(&mime_types_lock);
985
986         return res;
987 }
988
989 unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format *format, int code)
990 {
991         unsigned int i;
992         unsigned int res = 0;
993
994         ast_rwlock_rdlock(&mime_types_lock);
995         for (i = 0; i < mime_types_len; ++i) {
996                 if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
997                         (ast_format_cmp(format, ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
998                         res = ast_rtp_mime_types[i].sample_rate;
999                         break;
1000                 } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
1001                         ast_rtp_mime_types[i].payload_type.rtp_code == code) {
1002                         res = ast_rtp_mime_types[i].sample_rate;
1003                         break;
1004                 }
1005         }
1006         ast_rwlock_unlock(&mime_types_lock);
1007
1008         return res;
1009 }
1010
1011 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)
1012 {
1013         int found = 0;
1014         const char *name;
1015         if (!buf) {
1016                 return NULL;
1017         }
1018
1019
1020         if (asterisk_format) {
1021                 int x;
1022                 struct ast_format *tmp_fmt;
1023                 for (x = 0; x < ast_format_cap_count(ast_format_capability); x++) {
1024                         tmp_fmt = ast_format_cap_get_format(ast_format_capability, x);
1025                         name = ast_rtp_lookup_mime_subtype2(asterisk_format, tmp_fmt, 0, options);
1026                         ao2_ref(tmp_fmt, -1);
1027                         ast_str_append(&buf, 0, "%s|", name);
1028                         found = 1;
1029                 }
1030         } else {
1031                 int x;
1032                 ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
1033                 for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
1034                         if (rtp_capability & x) {
1035                                 name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
1036                                 ast_str_append(&buf, 0, "%s|", name);
1037                                 found = 1;
1038                         }
1039                 }
1040         }
1041
1042         ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
1043
1044         return ast_str_buffer(buf);
1045 }
1046
1047 int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
1048 {
1049         return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
1050 }
1051
1052 int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
1053 {
1054         return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
1055 }
1056 int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
1057 {
1058         return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
1059 }
1060
1061 int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
1062 {
1063         return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
1064 }
1065
1066 enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
1067 {
1068         return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
1069 }
1070
1071 void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
1072 {
1073         if (instance->engine->update_source) {
1074                 instance->engine->update_source(instance);
1075         }
1076 }
1077
1078 void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
1079 {
1080         if (instance->engine->change_source) {
1081                 instance->engine->change_source(instance);
1082         }
1083 }
1084
1085 int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
1086 {
1087         return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
1088 }
1089
1090 void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
1091 {
1092         if (instance->engine->stop) {
1093                 instance->engine->stop(instance);
1094         }
1095 }
1096
1097 int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
1098 {
1099         return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
1100 }
1101
1102 struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
1103 {
1104         struct ast_rtp_glue *glue = NULL;
1105
1106         AST_RWLIST_RDLOCK(&glues);
1107
1108         AST_RWLIST_TRAVERSE(&glues, glue, entry) {
1109                 if (!strcasecmp(glue->type, type)) {
1110                         break;
1111                 }
1112         }
1113
1114         AST_RWLIST_UNLOCK(&glues);
1115
1116         return glue;
1117 }
1118
1119 /*!
1120  * \brief Conditionally unref an rtp instance
1121  */
1122 static void unref_instance_cond(struct ast_rtp_instance **instance)
1123 {
1124         if (*instance) {
1125                 ao2_ref(*instance, -1);
1126                 *instance = NULL;
1127         }
1128 }
1129
1130 struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
1131 {
1132         return instance->bridged;
1133 }
1134
1135 void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged)
1136 {
1137         instance->bridged = bridged;
1138 }
1139
1140 void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c_dst, struct ast_channel *c_src)
1141 {
1142         struct ast_rtp_instance *instance_dst = NULL, *instance_src = NULL,
1143                 *vinstance_dst = NULL, *vinstance_src = NULL,
1144                 *tinstance_dst = NULL, *tinstance_src = NULL;
1145         struct ast_rtp_glue *glue_dst, *glue_src;
1146         enum ast_rtp_glue_result audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
1147         enum ast_rtp_glue_result audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
1148         struct ast_format_cap *cap_dst = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1149         struct ast_format_cap *cap_src = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1150
1151         /* Lock both channels so we can look for the glue that binds them together */
1152         ast_channel_lock_both(c_dst, c_src);
1153
1154         if (!cap_src || !cap_dst) {
1155                 goto done;
1156         }
1157
1158         /* Grab glue that binds each channel to something using the RTP engine */
1159         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))) {
1160                 ast_debug(1, "Can't find native functions for channel '%s'\n", glue_dst ? ast_channel_name(c_src) : ast_channel_name(c_dst));
1161                 goto done;
1162         }
1163
1164         audio_glue_dst_res = glue_dst->get_rtp_info(c_dst, &instance_dst);
1165         video_glue_dst_res = glue_dst->get_vrtp_info ? glue_dst->get_vrtp_info(c_dst, &vinstance_dst) : AST_RTP_GLUE_RESULT_FORBID;
1166
1167         audio_glue_src_res = glue_src->get_rtp_info(c_src, &instance_src);
1168         video_glue_src_res = glue_src->get_vrtp_info ? glue_src->get_vrtp_info(c_src, &vinstance_src) : AST_RTP_GLUE_RESULT_FORBID;
1169
1170         /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
1171         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)) {
1172                 audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
1173         }
1174         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)) {
1175                 audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
1176         }
1177         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) {
1178                 glue_dst->get_codec(c_dst, cap_dst);
1179         }
1180         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) {
1181                 glue_src->get_codec(c_src, cap_src);
1182         }
1183
1184         /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
1185         if (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE) {
1186                 goto done;
1187         }
1188
1189         /* Make sure we have matching codecs */
1190         if (!ast_format_cap_iscompatible(cap_dst, cap_src)) {
1191                 goto done;
1192         }
1193
1194         ast_rtp_codecs_payloads_copy(&instance_src->codecs, &instance_dst->codecs, instance_dst);
1195
1196         if (vinstance_dst && vinstance_src) {
1197                 ast_rtp_codecs_payloads_copy(&vinstance_src->codecs, &vinstance_dst->codecs, vinstance_dst);
1198         }
1199         if (tinstance_dst && tinstance_src) {
1200                 ast_rtp_codecs_payloads_copy(&tinstance_src->codecs, &tinstance_dst->codecs, tinstance_dst);
1201         }
1202
1203         if (glue_dst->update_peer(c_dst, instance_src, vinstance_src, tinstance_src, cap_src, 0)) {
1204                 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
1205                         ast_channel_name(c_dst), ast_channel_name(c_src));
1206         } else {
1207                 ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
1208                         ast_channel_name(c_dst), ast_channel_name(c_src));
1209         }
1210
1211 done:
1212         ast_channel_unlock(c_dst);
1213         ast_channel_unlock(c_src);
1214
1215         ao2_cleanup(cap_dst);
1216         ao2_cleanup(cap_src);
1217
1218         unref_instance_cond(&instance_dst);
1219         unref_instance_cond(&instance_src);
1220         unref_instance_cond(&vinstance_dst);
1221         unref_instance_cond(&vinstance_src);
1222         unref_instance_cond(&tinstance_dst);
1223         unref_instance_cond(&tinstance_src);
1224 }
1225
1226 int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
1227 {
1228         struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
1229                         *vinstance0 = NULL, *vinstance1 = NULL,
1230                         *tinstance0 = NULL, *tinstance1 = NULL;
1231         struct ast_rtp_glue *glue0, *glue1;
1232         enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1233         enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1234         struct ast_format_cap *cap0 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1235         struct ast_format_cap *cap1 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1236
1237         /* If there is no second channel just immediately bail out, we are of no use in that scenario */
1238         if (!c1 || !cap1 || !cap0) {
1239                 ao2_cleanup(cap0);
1240                 ao2_cleanup(cap1);
1241                 return -1;
1242         }
1243
1244         /* Lock both channels so we can look for the glue that binds them together */
1245         ast_channel_lock_both(c0, c1);
1246
1247         /* Grab glue that binds each channel to something using the RTP engine */
1248         if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
1249                 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
1250                 goto done;
1251         }
1252
1253         audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
1254         video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
1255
1256         audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
1257         video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
1258
1259         /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
1260         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)) {
1261                 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1262         }
1263         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)) {
1264                 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1265         }
1266         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) {
1267                 glue0->get_codec(c0, cap0);
1268         }
1269         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) {
1270                 glue1->get_codec(c1, cap1);
1271         }
1272
1273         /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
1274         if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
1275                 goto done;
1276         }
1277
1278         /* Make sure we have matching codecs */
1279         if (!ast_format_cap_iscompatible(cap0, cap1)) {
1280                 goto done;
1281         }
1282
1283         /* Bridge media early */
1284         if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
1285                 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
1286         }
1287
1288 done:
1289         ast_channel_unlock(c0);
1290         ast_channel_unlock(c1);
1291
1292         ao2_cleanup(cap0);
1293         ao2_cleanup(cap1);
1294
1295         unref_instance_cond(&instance0);
1296         unref_instance_cond(&instance1);
1297         unref_instance_cond(&vinstance0);
1298         unref_instance_cond(&vinstance1);
1299         unref_instance_cond(&tinstance0);
1300         unref_instance_cond(&tinstance1);
1301
1302         ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
1303
1304         return 0;
1305 }
1306
1307 int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
1308 {
1309         return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
1310 }
1311
1312 int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
1313 {
1314         return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
1315 }
1316
1317 int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
1318 {
1319         return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
1320 }
1321
1322 char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size)
1323 {
1324         struct ast_rtp_instance_stats stats = { 0, };
1325         enum ast_rtp_instance_stat stat;
1326
1327         /* Determine what statistics we will need to retrieve based on field passed in */
1328         if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
1329                 stat = AST_RTP_INSTANCE_STAT_ALL;
1330         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
1331                 stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
1332         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
1333                 stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
1334         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
1335                 stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
1336         } else {
1337                 return NULL;
1338         }
1339
1340         /* Attempt to actually retrieve the statistics we need to generate the quality string */
1341         if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
1342                 return NULL;
1343         }
1344
1345         /* Now actually fill the buffer with the good information */
1346         if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
1347                 snprintf(buf, size, "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
1348                          stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.rxjitter, stats.rxcount, stats.txjitter, stats.txcount, stats.txploss, stats.rtt);
1349         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
1350                 snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
1351                          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));
1352         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
1353                 snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
1354                          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));
1355         } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
1356                 snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
1357         }
1358
1359         return buf;
1360 }
1361
1362 void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
1363 {
1364         char quality_buf[AST_MAX_USER_FIELD];
1365         char *quality;
1366         struct ast_channel *bridge = ast_channel_bridge_peer(chan);
1367
1368         ast_channel_lock(chan);
1369         ast_channel_stage_snapshot(chan);
1370         ast_channel_unlock(chan);
1371         if (bridge) {
1372                 ast_channel_lock(bridge);
1373                 ast_channel_stage_snapshot(bridge);
1374                 ast_channel_unlock(bridge);
1375         }
1376
1377         quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY,
1378                 quality_buf, sizeof(quality_buf));
1379         if (quality) {
1380                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
1381                 if (bridge) {
1382                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
1383                 }
1384         }
1385
1386         quality = ast_rtp_instance_get_quality(instance,
1387                 AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf));
1388         if (quality) {
1389                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
1390                 if (bridge) {
1391                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
1392                 }
1393         }
1394
1395         quality = ast_rtp_instance_get_quality(instance,
1396                 AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf));
1397         if (quality) {
1398                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
1399                 if (bridge) {
1400                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
1401                 }
1402         }
1403
1404         quality = ast_rtp_instance_get_quality(instance,
1405                 AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf));
1406         if (quality) {
1407                 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
1408                 if (bridge) {
1409                         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
1410                 }
1411         }
1412
1413         ast_channel_lock(chan);
1414         ast_channel_stage_snapshot_done(chan);
1415         ast_channel_unlock(chan);
1416         if (bridge) {
1417                 ast_channel_lock(bridge);
1418                 ast_channel_stage_snapshot_done(bridge);
1419                 ast_channel_unlock(bridge);
1420                 ast_channel_unref(bridge);
1421         }
1422 }
1423
1424 int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)
1425 {
1426         return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
1427 }
1428
1429 int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, struct ast_format *format)
1430 {
1431         return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
1432 }
1433
1434 int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer)
1435 {
1436         struct ast_rtp_glue *glue;
1437         struct ast_rtp_instance *peer_instance = NULL;
1438         int res = -1;
1439
1440         if (!instance->engine->make_compatible) {
1441                 return -1;
1442         }
1443
1444         ast_channel_lock(peer);
1445
1446         if (!(glue = ast_rtp_instance_get_glue(ast_channel_tech(peer)->type))) {
1447                 ast_channel_unlock(peer);
1448                 return -1;
1449         }
1450
1451         glue->get_rtp_info(peer, &peer_instance);
1452         if (!peer_instance) {
1453                 ast_log(LOG_ERROR, "Unable to get_rtp_info for peer type %s\n", glue->type);
1454                 ast_channel_unlock(peer);
1455                 return -1;
1456         }
1457         if (peer_instance->engine != instance->engine) {
1458                 ast_log(LOG_ERROR, "Peer engine mismatch for type %s\n", glue->type);
1459                 ast_channel_unlock(peer);
1460                 ao2_ref(peer_instance, -1);
1461                 return -1;
1462         }
1463
1464         res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
1465
1466         ast_channel_unlock(peer);
1467
1468         ao2_ref(peer_instance, -1);
1469         peer_instance = NULL;
1470
1471         return res;
1472 }
1473
1474 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)
1475 {
1476         if (instance->engine->available_formats) {
1477                 instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
1478                 if (ast_format_cap_count(result)) {
1479                         return;
1480                 }
1481         }
1482
1483         ast_translate_available_formats(to_endpoint, to_asterisk, result);
1484 }
1485
1486 int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
1487 {
1488         return instance->engine->activate ? instance->engine->activate(instance) : 0;
1489 }
1490
1491 void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance,
1492                                    struct ast_sockaddr *suggestion,
1493                                    const char *username)
1494 {
1495         if (instance->engine->stun_request) {
1496                 instance->engine->stun_request(instance, suggestion, username);
1497         }
1498 }
1499
1500 void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout)
1501 {
1502         instance->timeout = timeout;
1503 }
1504
1505 void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout)
1506 {
1507         instance->holdtimeout = timeout;
1508 }
1509
1510 void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int interval)
1511 {
1512         instance->keepalive = interval;
1513 }
1514
1515 int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
1516 {
1517         return instance->timeout;
1518 }
1519
1520 int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
1521 {
1522         return instance->holdtimeout;
1523 }
1524
1525 int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance)
1526 {
1527         return instance->keepalive;
1528 }
1529
1530 struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
1531 {
1532         return instance->engine;
1533 }
1534
1535 struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *instance)
1536 {
1537         return instance->glue;
1538 }
1539
1540 int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res)
1541 {
1542         if (res_srtp || res_srtp_policy) {
1543                 return -1;
1544         }
1545         if (!srtp_res || !policy_res) {
1546                 return -1;
1547         }
1548
1549         res_srtp = srtp_res;
1550         res_srtp_policy = policy_res;
1551
1552         return 0;
1553 }
1554
1555 void ast_rtp_engine_unregister_srtp(void)
1556 {
1557         res_srtp = NULL;
1558         res_srtp_policy = NULL;
1559 }
1560
1561 int ast_rtp_engine_srtp_is_registered(void)
1562 {
1563         return res_srtp && res_srtp_policy;
1564 }
1565
1566 int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy)
1567 {
1568         int res = 0;
1569
1570         if (!res_srtp) {
1571                 return -1;
1572         }
1573
1574         if (!instance->srtp) {
1575                 res = res_srtp->create(&instance->srtp, instance, remote_policy);
1576         } else {
1577                 res = res_srtp->replace(&instance->srtp, instance, remote_policy);
1578         }
1579         if (!res) {
1580                 res = res_srtp->add_stream(instance->srtp, local_policy);
1581         }
1582
1583         return res;
1584 }
1585
1586 struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
1587 {
1588         return instance->srtp;
1589 }
1590
1591 int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
1592 {
1593         if (instance->engine->sendcng) {
1594                 return instance->engine->sendcng(instance, level);
1595         }
1596
1597         return -1;
1598 }
1599
1600 struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *instance)
1601 {
1602         return instance->engine->ice;
1603 }
1604
1605 struct ast_rtp_engine_dtls *ast_rtp_instance_get_dtls(struct ast_rtp_instance *instance)
1606 {
1607         return instance->engine->dtls;
1608 }
1609
1610 int ast_rtp_dtls_cfg_parse(struct ast_rtp_dtls_cfg *dtls_cfg, const char *name, const char *value)
1611 {
1612         if (!strcasecmp(name, "dtlsenable")) {
1613                 dtls_cfg->enabled = ast_true(value) ? 1 : 0;
1614         } else if (!strcasecmp(name, "dtlsverify")) {
1615                 if (!strcasecmp(value, "yes")) {
1616                         dtls_cfg->verify = AST_RTP_DTLS_VERIFY_FINGERPRINT | AST_RTP_DTLS_VERIFY_CERTIFICATE;
1617                 } else if (!strcasecmp(value, "fingerprint")) {
1618                         dtls_cfg->verify = AST_RTP_DTLS_VERIFY_FINGERPRINT;
1619                 } else if (!strcasecmp(value, "certificate")) {
1620                         dtls_cfg->verify = AST_RTP_DTLS_VERIFY_CERTIFICATE;
1621                 } else if (!strcasecmp(value, "no")) {
1622                         dtls_cfg->verify = AST_RTP_DTLS_VERIFY_NONE;
1623                 } else {
1624                         return -1;
1625                 }
1626         } else if (!strcasecmp(name, "dtlsrekey")) {
1627                 if (sscanf(value, "%30u", &dtls_cfg->rekey) != 1) {
1628                         return -1;
1629                 }
1630         } else if (!strcasecmp(name, "dtlscertfile")) {
1631                 ast_free(dtls_cfg->certfile);
1632                 dtls_cfg->certfile = ast_strdup(value);
1633         } else if (!strcasecmp(name, "dtlsprivatekey")) {
1634                 ast_free(dtls_cfg->pvtfile);
1635                 dtls_cfg->pvtfile = ast_strdup(value);
1636         } else if (!strcasecmp(name, "dtlscipher")) {
1637                 ast_free(dtls_cfg->cipher);
1638                 dtls_cfg->cipher = ast_strdup(value);
1639         } else if (!strcasecmp(name, "dtlscafile")) {
1640                 ast_free(dtls_cfg->cafile);
1641                 dtls_cfg->cafile = ast_strdup(value);
1642         } else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
1643                 ast_free(dtls_cfg->capath);
1644                 dtls_cfg->capath = ast_strdup(value);
1645         } else if (!strcasecmp(name, "dtlssetup")) {
1646                 if (!strcasecmp(value, "active")) {
1647                         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTIVE;
1648                 } else if (!strcasecmp(value, "passive")) {
1649                         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_PASSIVE;
1650                 } else if (!strcasecmp(value, "actpass")) {
1651                         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTPASS;
1652                 }
1653         } else if (!strcasecmp(name, "dtlsfingerprint")) {
1654                 if (!strcasecmp(value, "sha-256")) {
1655                         dtls_cfg->hash = AST_RTP_DTLS_HASH_SHA256;
1656                 } else if (!strcasecmp(value, "sha-1")) {
1657                         dtls_cfg->hash = AST_RTP_DTLS_HASH_SHA1;
1658                 }
1659         } else {
1660                 return -1;
1661         }
1662
1663         return 0;
1664 }
1665
1666 void ast_rtp_dtls_cfg_copy(const struct ast_rtp_dtls_cfg *src_cfg, struct ast_rtp_dtls_cfg *dst_cfg)
1667 {
1668         ast_rtp_dtls_cfg_free(dst_cfg);         /* Prevent a double-call leaking memory via ast_strdup */
1669
1670         dst_cfg->enabled = src_cfg->enabled;
1671         dst_cfg->verify = src_cfg->verify;
1672         dst_cfg->rekey = src_cfg->rekey;
1673         dst_cfg->suite = src_cfg->suite;
1674         dst_cfg->hash = src_cfg->hash;
1675         dst_cfg->certfile = ast_strdup(src_cfg->certfile);
1676         dst_cfg->pvtfile = ast_strdup(src_cfg->pvtfile);
1677         dst_cfg->cipher = ast_strdup(src_cfg->cipher);
1678         dst_cfg->cafile = ast_strdup(src_cfg->cafile);
1679         dst_cfg->capath = ast_strdup(src_cfg->capath);
1680         dst_cfg->default_setup = src_cfg->default_setup;
1681 }
1682
1683 void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg)
1684 {
1685         ast_free(dtls_cfg->certfile);
1686         dtls_cfg->certfile = NULL;
1687         ast_free(dtls_cfg->pvtfile);
1688         dtls_cfg->pvtfile = NULL;
1689         ast_free(dtls_cfg->cipher);
1690         dtls_cfg->cipher = NULL;
1691         ast_free(dtls_cfg->cafile);
1692         dtls_cfg->cafile = NULL;
1693         ast_free(dtls_cfg->capath);
1694         dtls_cfg->capath = NULL;
1695 }
1696
1697 /*! \internal
1698  * \brief Small helper routine that cleans up entry i in
1699  * \c static_RTP_PT.
1700  */
1701 static void rtp_engine_static_RTP_PT_cleanup(int i)
1702 {
1703         ao2_cleanup(static_RTP_PT[i].format);
1704         memset(&static_RTP_PT[i], 0, sizeof(struct ast_rtp_payload_type));
1705 }
1706
1707 /*! \internal
1708  * \brief Small helper routine that cleans up entry i in
1709  * \c ast_rtp_mime_types.
1710  */
1711 static void rtp_engine_mime_type_cleanup(int i)
1712 {
1713         ao2_cleanup(ast_rtp_mime_types[i].payload_type.format);
1714         memset(&ast_rtp_mime_types[i], 0, sizeof(struct ast_rtp_mime_type));
1715 }
1716
1717 static void set_next_mime_type(struct ast_format *format, int rtp_code, const char *type, const char *subtype, unsigned int sample_rate)
1718 {
1719         int x = mime_types_len;
1720         if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
1721                 return;
1722         }
1723
1724         ast_rwlock_wrlock(&mime_types_lock);
1725         /* Make sure any previous value in ast_rtp_mime_types is cleaned up */
1726         memset(&ast_rtp_mime_types[x], 0, sizeof(struct ast_rtp_mime_type));    
1727         if (format) {
1728                 ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
1729                 ast_rtp_mime_types[x].payload_type.format = ao2_bump(format);
1730         } else {
1731                 ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
1732         }
1733         ast_copy_string(ast_rtp_mime_types[x].type, type, sizeof(ast_rtp_mime_types[x].type));
1734         ast_copy_string(ast_rtp_mime_types[x].subtype, subtype, sizeof(ast_rtp_mime_types[x].subtype));
1735         ast_rtp_mime_types[x].sample_rate = sample_rate;
1736         mime_types_len++;
1737         ast_rwlock_unlock(&mime_types_lock);
1738 }
1739
1740 static void add_static_payload(int map, struct ast_format *format, int rtp_code)
1741 {
1742         int x;
1743
1744         ast_assert(map < ARRAY_LEN(static_RTP_PT));
1745
1746         ast_rwlock_wrlock(&static_RTP_PT_lock);
1747         if (map < 0) {
1748                 /* find next available dynamic payload slot */
1749                 for (x = AST_RTP_PT_FIRST_DYNAMIC; x < AST_RTP_MAX_PT; ++x) {
1750                         if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
1751                                 map = x;
1752                                 break;
1753                         }
1754                 }
1755                 if (map < 0) {
1756                         ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n",
1757                                 ast_format_get_name(format));
1758                         ast_rwlock_unlock(&static_RTP_PT_lock);
1759                         return;
1760                 }
1761         }
1762
1763         if (format) {
1764                 static_RTP_PT[map].asterisk_format = 1;
1765                 static_RTP_PT[map].format = ao2_bump(format);
1766         } else {
1767                 static_RTP_PT[map].rtp_code = rtp_code;
1768         }
1769         ast_rwlock_unlock(&static_RTP_PT_lock);
1770 }
1771
1772 int ast_rtp_engine_load_format(struct ast_format *format)
1773 {
1774         char *codec_name = ast_strdupa(ast_format_get_name(format));
1775
1776         codec_name = ast_str_to_upper(codec_name);
1777
1778         set_next_mime_type(format,
1779                 0,
1780                 ast_codec_media_type2str(ast_format_get_type(format)),
1781                 codec_name,
1782                 ast_format_get_sample_rate(format));
1783         add_static_payload(-1, format, 0);
1784
1785         return 0;
1786 }
1787
1788 int ast_rtp_engine_unload_format(struct ast_format *format)
1789 {
1790         int x;
1791         int y = 0;
1792
1793         ast_rwlock_wrlock(&static_RTP_PT_lock);
1794         /* remove everything pertaining to this format id from the lists */
1795         for (x = 0; x < AST_RTP_MAX_PT; x++) {
1796                 if (ast_format_cmp(static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
1797                         rtp_engine_static_RTP_PT_cleanup(x);
1798                 }
1799         }
1800         ast_rwlock_unlock(&static_RTP_PT_lock);
1801
1802         ast_rwlock_wrlock(&mime_types_lock);
1803         /* rebuild the list skipping the items matching this id */
1804         for (x = 0; x < mime_types_len; x++) {
1805                 if (ast_format_cmp(ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
1806                         rtp_engine_mime_type_cleanup(x);
1807                         continue;
1808                 }
1809                 if (x != y) {
1810                         ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
1811                 }
1812                 y++;
1813         }
1814         mime_types_len = y;
1815         ast_rwlock_unlock(&mime_types_lock);
1816         return 0;
1817 }
1818
1819 /*!
1820  * \internal
1821  * \brief \ref stasis message payload for RTCP messages
1822  */
1823 struct rtcp_message_payload {
1824         struct ast_channel_snapshot *snapshot;  /*< The channel snapshot, if available */
1825         struct ast_rtp_rtcp_report *report;     /*< The RTCP report */
1826         struct ast_json *blob;                  /*< Extra JSON data to publish */
1827 };
1828
1829 static void rtcp_message_payload_dtor(void *obj)
1830 {
1831         struct rtcp_message_payload *payload = obj;
1832
1833         ao2_cleanup(payload->report);
1834         ao2_cleanup(payload->snapshot);
1835         ast_json_unref(payload->blob);
1836 }
1837
1838 static struct ast_manager_event_blob *rtcp_report_to_ami(struct stasis_message *msg)
1839 {
1840         struct rtcp_message_payload *payload = stasis_message_data(msg);
1841         RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
1842         RAII_VAR(struct ast_str *, packet_string, ast_str_create(512), ast_free);
1843         unsigned int ssrc = payload->report->ssrc;
1844         unsigned int type = payload->report->type;
1845         unsigned int report_count = payload->report->reception_report_count;
1846         int i;
1847
1848         if (!packet_string) {
1849                 return NULL;
1850         }
1851
1852         if (payload->snapshot) {
1853                 channel_string = ast_manager_build_channel_state_string(payload->snapshot);
1854                 if (!channel_string) {
1855                         return NULL;
1856                 }
1857         }
1858
1859         if (payload->blob) {
1860                 /* Optional data */
1861                 struct ast_json *to = ast_json_object_get(payload->blob, "to");
1862                 struct ast_json *from = ast_json_object_get(payload->blob, "from");
1863                 struct ast_json *rtt = ast_json_object_get(payload->blob, "rtt");
1864                 if (to) {
1865                         ast_str_append(&packet_string, 0, "To: %s\r\n", ast_json_string_get(to));
1866                 }
1867                 if (from) {
1868                         ast_str_append(&packet_string, 0, "From: %s\r\n", ast_json_string_get(from));
1869                 }
1870                 if (rtt) {
1871                         ast_str_append(&packet_string, 0, "RTT: %4.4f\r\n", ast_json_real_get(rtt));
1872                 }
1873         }
1874
1875         ast_str_append(&packet_string, 0, "SSRC: 0x%.8x\r\n", ssrc);
1876         ast_str_append(&packet_string, 0, "PT: %u(%s)\r\n", type, type== AST_RTP_RTCP_SR ? "SR" : "RR");
1877         ast_str_append(&packet_string, 0, "ReportCount: %u\r\n", report_count);
1878         if (type == AST_RTP_RTCP_SR) {
1879                 ast_str_append(&packet_string, 0, "SentNTP: %lu.%06lu\r\n",
1880                         (unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec,
1881                         (unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec * 4096);
1882                 ast_str_append(&packet_string, 0, "SentRTP: %u\r\n",
1883                                 payload->report->sender_information.rtp_timestamp);
1884                 ast_str_append(&packet_string, 0, "SentPackets: %u\r\n",
1885                                 payload->report->sender_information.packet_count);
1886                 ast_str_append(&packet_string, 0, "SentOctets: %u\r\n",
1887                                 payload->report->sender_information.octet_count);
1888         }
1889
1890         for (i = 0; i < report_count; i++) {
1891                 RAII_VAR(struct ast_str *, report_string, NULL, ast_free);
1892
1893                 if (!payload->report->report_block[i]) {
1894                         break;
1895                 }
1896
1897                 report_string = ast_str_create(256);
1898                 if (!report_string) {
1899                         return NULL;
1900                 }
1901
1902                 ast_str_append(&report_string, 0, "Report%dSourceSSRC: 0x%.8x\r\n",
1903                                 i, payload->report->report_block[i]->source_ssrc);
1904                 ast_str_append(&report_string, 0, "Report%dFractionLost: %d\r\n",
1905                                 i, payload->report->report_block[i]->lost_count.fraction);
1906                 ast_str_append(&report_string, 0, "Report%dCumulativeLost: %u\r\n",
1907                                 i, payload->report->report_block[i]->lost_count.packets);
1908                 ast_str_append(&report_string, 0, "Report%dHighestSequence: %u\r\n",
1909                                 i, payload->report->report_block[i]->highest_seq_no & 0xffff);
1910                 ast_str_append(&report_string, 0, "Report%dSequenceNumberCycles: %u\r\n",
1911                                 i, payload->report->report_block[i]->highest_seq_no >> 16);
1912                 ast_str_append(&report_string, 0, "Report%dIAJitter: %u\r\n",
1913                                 i, payload->report->report_block[i]->ia_jitter);
1914                 ast_str_append(&report_string, 0, "Report%dLSR: %u\r\n",
1915                                 i, payload->report->report_block[i]->lsr);
1916                 ast_str_append(&report_string, 0, "Report%dDLSR: %4.4f\r\n",
1917                                 i, ((double)payload->report->report_block[i]->dlsr) / 65536);
1918                 ast_str_append(&packet_string, 0, "%s", ast_str_buffer(report_string));
1919         }
1920
1921         return ast_manager_event_blob_create(EVENT_FLAG_REPORTING,
1922                 stasis_message_type(msg) == ast_rtp_rtcp_received_type() ? "RTCPReceived" : "RTCPSent",
1923                 "%s%s",
1924                 AS_OR(channel_string, ""),
1925                 ast_str_buffer(packet_string));
1926 }
1927
1928 static struct ast_json *rtcp_report_to_json(struct stasis_message *msg,
1929         const struct stasis_message_sanitizer *sanitize)
1930 {
1931         struct rtcp_message_payload *payload = stasis_message_data(msg);
1932         RAII_VAR(struct ast_json *, json_rtcp_report, NULL, ast_json_unref);
1933         RAII_VAR(struct ast_json *, json_rtcp_report_blocks, NULL, ast_json_unref);
1934         RAII_VAR(struct ast_json *, json_rtcp_sender_info, NULL, ast_json_unref);
1935         RAII_VAR(struct ast_json *, json_channel, NULL, ast_json_unref);
1936         int i;
1937
1938         json_rtcp_report_blocks = ast_json_array_create();
1939         if (!json_rtcp_report_blocks) {
1940                 return NULL;
1941         }
1942
1943         for (i = 0; i < payload->report->reception_report_count && payload->report->report_block[i]; i++) {
1944                 struct ast_json *json_report_block;
1945                 char str_lsr[32];
1946                 snprintf(str_lsr, sizeof(str_lsr), "%u", payload->report->report_block[i]->lsr);
1947                 json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: s, s: i}",
1948                                 "source_ssrc", payload->report->report_block[i]->source_ssrc,
1949                                 "fraction_lost", payload->report->report_block[i]->lost_count.fraction,
1950                                 "packets_lost", payload->report->report_block[i]->lost_count.packets,
1951                                 "highest_seq_no", payload->report->report_block[i]->highest_seq_no,
1952                                 "ia_jitter", payload->report->report_block[i]->ia_jitter,
1953                                 "lsr", str_lsr,
1954                                 "dlsr", payload->report->report_block[i]->dlsr);
1955                 if (!json_report_block) {
1956                         return NULL;
1957                 }
1958
1959                 if (ast_json_array_append(json_rtcp_report_blocks, json_report_block)) {
1960                         return NULL;
1961                 }
1962         }
1963
1964         if (payload->report->type == AST_RTP_RTCP_SR) {
1965                 char sec[32];
1966                 char usec[32];
1967                 snprintf(sec, sizeof(sec), "%lu", (unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec);
1968                 snprintf(usec, sizeof(usec), "%lu", (unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec);
1969                 json_rtcp_sender_info = ast_json_pack("{s: s, s: s, s: i, s: i, s: i}",
1970                                 "ntp_timestamp_sec", sec,
1971                                 "ntp_timestamp_usec", usec,
1972                                 "rtp_timestamp", payload->report->sender_information.rtp_timestamp,
1973                                 "packets", payload->report->sender_information.packet_count,
1974                                 "octets", payload->report->sender_information.octet_count);
1975                 if (!json_rtcp_sender_info) {
1976                         return NULL;
1977                 }
1978         }
1979
1980         json_rtcp_report = ast_json_pack("{s: i, s: i, s: i, s: O, s: O}",
1981                         "ssrc", payload->report->ssrc,
1982                         "type", payload->report->type,
1983                         "report_count", payload->report->reception_report_count,
1984                         "sender_information", json_rtcp_sender_info ? json_rtcp_sender_info : ast_json_null(),
1985                         "report_blocks", json_rtcp_report_blocks);
1986         if (!json_rtcp_report) {
1987                 return NULL;
1988         }
1989
1990         if (payload->snapshot) {
1991                 json_channel = ast_channel_snapshot_to_json(payload->snapshot, sanitize);
1992                 if (!json_channel) {
1993                         return NULL;
1994                 }
1995         }
1996
1997         return ast_json_pack("{s: O, s: O, s: O}",
1998                 "channel", payload->snapshot ? json_channel : ast_json_null(),
1999                 "rtcp_report", json_rtcp_report,
2000                 "blob", payload->blob);
2001 }
2002
2003 static void rtp_rtcp_report_dtor(void *obj)
2004 {
2005         int i;
2006         struct ast_rtp_rtcp_report *rtcp_report = obj;
2007
2008         for (i = 0; i < rtcp_report->reception_report_count; i++) {
2009                 ast_free(rtcp_report->report_block[i]);
2010         }
2011 }
2012
2013 struct ast_rtp_rtcp_report *ast_rtp_rtcp_report_alloc(unsigned int report_blocks)
2014 {
2015         struct ast_rtp_rtcp_report *rtcp_report;
2016
2017         /* Size of object is sizeof the report + the number of report_blocks * sizeof pointer */
2018         rtcp_report = ao2_alloc((sizeof(*rtcp_report) + report_blocks * sizeof(struct ast_rtp_rtcp_report_block *)),
2019                 rtp_rtcp_report_dtor);
2020
2021         return rtcp_report;
2022 }
2023
2024 void ast_rtp_publish_rtcp_message(struct ast_rtp_instance *rtp,
2025                 struct stasis_message_type *message_type,
2026                 struct ast_rtp_rtcp_report *report,
2027                 struct ast_json *blob)
2028 {
2029         RAII_VAR(struct rtcp_message_payload *, payload, NULL, ao2_cleanup);
2030         RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
2031
2032         if (!message_type) {
2033                 return;
2034         }
2035
2036         payload = ao2_alloc(sizeof(*payload), rtcp_message_payload_dtor);
2037         if (!payload || !report) {
2038                 return;
2039         }
2040
2041         if (!ast_strlen_zero(rtp->channel_uniqueid)) {
2042                 payload->snapshot = ast_channel_snapshot_get_latest(rtp->channel_uniqueid);
2043         }
2044         if (blob) {
2045                 payload->blob = blob;
2046                 ast_json_ref(blob);
2047         }
2048         ao2_ref(report, +1);
2049         payload->report = report;
2050
2051         message = stasis_message_create(message_type, payload);
2052         if (!message) {
2053                 return;
2054         }
2055
2056         stasis_publish(ast_rtp_topic(), message);
2057 }
2058
2059 /*!
2060  * @{ \brief Define RTCP/RTP message types.
2061  */
2062 STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_sent_type,
2063                 .to_ami = rtcp_report_to_ami,
2064                 .to_json = rtcp_report_to_json,);
2065 STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_received_type,
2066                 .to_ami = rtcp_report_to_ami,
2067                 .to_json = rtcp_report_to_json,);
2068 /*! @} */
2069
2070 struct stasis_topic *ast_rtp_topic(void)
2071 {
2072         return rtp_topic;
2073 }
2074
2075 static void rtp_engine_shutdown(void)
2076 {
2077         int x;
2078
2079         ao2_cleanup(rtp_topic);
2080         rtp_topic = NULL;
2081         STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_received_type);
2082         STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_sent_type);
2083
2084         ast_rwlock_wrlock(&static_RTP_PT_lock);
2085         for (x = 0; x < AST_RTP_MAX_PT; x++) {
2086                 if (static_RTP_PT[x].format) {
2087                         rtp_engine_static_RTP_PT_cleanup(x);
2088                 }
2089         }
2090         ast_rwlock_unlock(&static_RTP_PT_lock);
2091
2092         ast_rwlock_wrlock(&mime_types_lock);
2093         for (x = 0; x < mime_types_len; x++) {
2094                 if (ast_rtp_mime_types[x].payload_type.format) {
2095                         rtp_engine_mime_type_cleanup(x);
2096                 }
2097         }
2098         ast_rwlock_unlock(&mime_types_lock);
2099 }
2100
2101 int ast_rtp_engine_init(void)
2102 {
2103         ast_rwlock_init(&mime_types_lock);
2104         ast_rwlock_init(&static_RTP_PT_lock);
2105
2106         rtp_topic = stasis_topic_create("rtp_topic");
2107         if (!rtp_topic) {
2108                 return -1;
2109         }
2110         STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_sent_type);
2111         STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_received_type);
2112         ast_register_cleanup(rtp_engine_shutdown);
2113
2114         /* Define all the RTP mime types available */
2115         set_next_mime_type(ast_format_g723, 0, "audio", "G723", 8000);
2116         set_next_mime_type(ast_format_gsm, 0, "audio", "GSM", 8000);
2117         set_next_mime_type(ast_format_ulaw, 0, "audio", "PCMU", 8000);
2118         set_next_mime_type(ast_format_ulaw, 0, "audio", "G711U", 8000);
2119         set_next_mime_type(ast_format_alaw, 0, "audio", "PCMA", 8000);
2120         set_next_mime_type(ast_format_alaw, 0, "audio", "G711A", 8000);
2121         set_next_mime_type(ast_format_g726, 0, "audio", "G726-32", 8000);
2122         set_next_mime_type(ast_format_adpcm, 0, "audio", "DVI4", 8000);
2123         set_next_mime_type(ast_format_slin, 0, "audio", "L16", 8000);
2124         set_next_mime_type(ast_format_slin16, 0, "audio", "L16", 16000);
2125         set_next_mime_type(ast_format_slin16, 0, "audio", "L16-256", 16000);
2126         set_next_mime_type(ast_format_slin12, 0, "audio", "L16", 12000);
2127         set_next_mime_type(ast_format_slin24, 0, "audio", "L16", 24000);
2128         set_next_mime_type(ast_format_slin32, 0, "audio", "L16", 32000);
2129         set_next_mime_type(ast_format_slin44, 0, "audio", "L16", 44000);
2130         set_next_mime_type(ast_format_slin48, 0, "audio", "L16", 48000);
2131         set_next_mime_type(ast_format_slin96, 0, "audio", "L16", 96000);
2132         set_next_mime_type(ast_format_slin192, 0, "audio", "L16", 192000);
2133         set_next_mime_type(ast_format_lpc10, 0, "audio", "LPC", 8000);
2134         set_next_mime_type(ast_format_g729, 0, "audio", "G729", 8000);
2135         set_next_mime_type(ast_format_g729, 0, "audio", "G729A", 8000);
2136         set_next_mime_type(ast_format_g729, 0, "audio", "G.729", 8000);
2137         set_next_mime_type(ast_format_speex, 0, "audio", "speex", 8000);
2138         set_next_mime_type(ast_format_speex16, 0,  "audio", "speex", 16000);
2139         set_next_mime_type(ast_format_speex32, 0,  "audio", "speex", 32000);
2140         set_next_mime_type(ast_format_ilbc, 0, "audio", "iLBC", 8000);
2141         /* this is the sample rate listed in the RTP profile for the G.722 codec, *NOT* the actual sample rate of the media stream */
2142         set_next_mime_type(ast_format_g722, 0, "audio", "G722", 8000);
2143         set_next_mime_type(ast_format_g726_aal2, 0, "audio", "AAL2-G726-32", 8000);
2144         set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
2145         set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
2146         set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
2147         set_next_mime_type(ast_format_jpeg, 0, "video", "JPEG", 90000);
2148         set_next_mime_type(ast_format_png, 0, "video", "PNG", 90000);
2149         set_next_mime_type(ast_format_h261, 0, "video", "H261", 90000);
2150         set_next_mime_type(ast_format_h263, 0, "video", "H263", 90000);
2151         set_next_mime_type(ast_format_h263p, 0, "video", "h263-1998", 90000);
2152         set_next_mime_type(ast_format_h264, 0, "video", "H264", 90000);
2153         set_next_mime_type(ast_format_mp4, 0, "video", "MP4V-ES", 90000);
2154         set_next_mime_type(ast_format_t140_red, 0, "text", "RED", 1000);
2155         set_next_mime_type(ast_format_t140, 0, "text", "T140", 1000);
2156         set_next_mime_type(ast_format_siren7, 0, "audio", "G7221", 16000);
2157         set_next_mime_type(ast_format_siren14, 0, "audio", "G7221", 32000);
2158         set_next_mime_type(ast_format_g719, 0, "audio", "G719", 48000);
2159         /* Opus and VP8 */
2160         set_next_mime_type(ast_format_opus, 0,  "audio", "opus", 48000);
2161         set_next_mime_type(ast_format_vp8, 0,  "video", "VP8", 90000);
2162
2163         /* Define the static rtp payload mappings */
2164         add_static_payload(0, ast_format_ulaw, 0);
2165         #ifdef USE_DEPRECATED_G726
2166         add_static_payload(2, ast_format_g726, 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
2167         #endif
2168         add_static_payload(3, ast_format_gsm, 0);
2169         add_static_payload(4, ast_format_g723, 0);
2170         add_static_payload(5, ast_format_adpcm, 0);/* 8 kHz */
2171         add_static_payload(6, ast_format_adpcm, 0); /* 16 kHz */
2172         add_static_payload(7, ast_format_lpc10, 0);
2173         add_static_payload(8, ast_format_alaw, 0);
2174         add_static_payload(9, ast_format_g722, 0);
2175         add_static_payload(10, ast_format_slin, 0); /* 2 channels */
2176         add_static_payload(11, ast_format_slin, 0); /* 1 channel */
2177         add_static_payload(13, NULL, AST_RTP_CN);
2178         add_static_payload(16, ast_format_adpcm, 0); /* 11.025 kHz */
2179         add_static_payload(17, ast_format_adpcm, 0); /* 22.050 kHz */
2180         add_static_payload(18, ast_format_g729, 0);
2181         add_static_payload(19, NULL, AST_RTP_CN);         /* Also used for CN */
2182         add_static_payload(26, ast_format_jpeg, 0);
2183         add_static_payload(31, ast_format_h261, 0);
2184         add_static_payload(34, ast_format_h263, 0);
2185         add_static_payload(97, ast_format_ilbc, 0);
2186         add_static_payload(98, ast_format_h263p, 0);
2187         add_static_payload(99, ast_format_h264, 0);
2188         add_static_payload(101, NULL, AST_RTP_DTMF);
2189         add_static_payload(102, ast_format_siren7, 0);
2190         add_static_payload(103, ast_format_h263p, 0);
2191         add_static_payload(104, ast_format_mp4, 0);
2192         add_static_payload(105, ast_format_t140_red, 0);   /* Real time text chat (with redundancy encoding) */
2193         add_static_payload(106, ast_format_t140, 0);     /* Real time text chat */
2194         add_static_payload(110, ast_format_speex, 0);
2195         add_static_payload(111, ast_format_g726, 0);
2196         add_static_payload(112, ast_format_g726_aal2, 0);
2197         add_static_payload(115, ast_format_siren14, 0);
2198         add_static_payload(116, ast_format_g719, 0);
2199         add_static_payload(117, ast_format_speex16, 0);
2200         add_static_payload(118, ast_format_slin16, 0); /* 16 Khz signed linear */
2201         add_static_payload(119, ast_format_speex32, 0);
2202         add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */
2203         add_static_payload(122, ast_format_slin12, 0);
2204         add_static_payload(123, ast_format_slin24, 0);
2205         add_static_payload(124, ast_format_slin32, 0);
2206         add_static_payload(125, ast_format_slin44, 0);
2207         add_static_payload(126, ast_format_slin48, 0);
2208         add_static_payload(127, ast_format_slin96, 0);
2209         /* payload types above 127 are not valid */
2210         add_static_payload(96, ast_format_slin192, 0);
2211         /* Opus and VP8 */
2212         add_static_payload(100, ast_format_vp8, 0);
2213         add_static_payload(107, ast_format_opus, 0);
2214
2215         return 0;
2216 }
2217
2218 time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp)
2219 {
2220         return rtp->last_tx;
2221 }
2222
2223 void ast_rtp_instance_set_last_tx(struct ast_rtp_instance *rtp, time_t time)
2224 {
2225         rtp->last_tx = time;
2226 }
2227
2228 time_t ast_rtp_instance_get_last_rx(const struct ast_rtp_instance *rtp)
2229 {
2230         return rtp->last_rx;
2231 }
2232
2233 void ast_rtp_instance_set_last_rx(struct ast_rtp_instance *rtp, time_t time)
2234 {
2235         rtp->last_rx = time;
2236 }