2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2008, Digium, Inc.
6 * Joshua Colp <jcolp@digium.com>
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.
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.
21 * \brief Pluggable RTP Architecture
23 * \author Joshua Colp <jcolp@digium.com>
27 <support_level>core</support_level>
32 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
36 #include "asterisk/channel.h"
37 #include "asterisk/frame.h"
38 #include "asterisk/module.h"
39 #include "asterisk/rtp_engine.h"
40 #include "asterisk/manager.h"
41 #include "asterisk/options.h"
42 #include "asterisk/astobj2.h"
43 #include "asterisk/pbx.h"
44 #include "asterisk/translate.h"
45 #include "asterisk/netsock2.h"
46 #include "asterisk/_private.h"
47 #include "asterisk/framehook.h"
49 struct ast_srtp_res *res_srtp = NULL;
50 struct ast_srtp_policy_res *res_srtp_policy = NULL;
52 /*! Structure that represents an RTP session (instance) */
53 struct ast_rtp_instance {
54 /*! Engine that is handling this RTP instance */
55 struct ast_rtp_engine *engine;
56 /*! Data unique to the RTP engine */
58 /*! RTP properties that have been set and their value */
59 int properties[AST_RTP_PROPERTY_MAX];
60 /*! Address that we are expecting RTP to come in to */
61 struct ast_sockaddr local_address;
62 /*! Address that we are sending RTP to */
63 struct ast_sockaddr remote_address;
64 /*! Alternate address that we are receiving RTP from */
65 struct ast_sockaddr alt_remote_address;
66 /*! Instance that we are bridged to if doing remote or local bridging */
67 struct ast_rtp_instance *bridged;
68 /*! Payload and packetization information */
69 struct ast_rtp_codecs codecs;
70 /*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
72 /*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
74 /*! RTP keepalive interval */
76 /*! Glue currently in use */
77 struct ast_rtp_glue *glue;
78 /*! Channel associated with the instance */
79 struct ast_channel *chan;
80 /*! SRTP info associated with the instance */
81 struct ast_srtp *srtp;
84 /*! List of RTP engines that are currently registered */
85 static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
87 /*! List of RTP glues */
88 static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
90 /*! The following array defines the MIME Media type (and subtype) for each
91 of our codecs, or RTP-specific data type. */
92 static struct ast_rtp_mime_type {
93 struct ast_rtp_payload_type payload_type;
96 unsigned int sample_rate;
97 } ast_rtp_mime_types[128]; /* This will Likely not need to grow any time soon. */
98 static ast_rwlock_t mime_types_lock;
99 static int mime_types_len = 0;
102 * \brief Mapping between Asterisk codecs and rtp payload types
104 * Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
105 * also, our own choices for dynamic payload types. This is our master
106 * table for transmission
108 * See http://www.iana.org/assignments/rtp-parameters for a list of
111 static struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT];
112 static ast_rwlock_t static_RTP_PT_lock;
114 int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
116 struct ast_rtp_engine *current_engine;
118 /* Perform a sanity check on the engine structure to make sure it has the basics */
119 if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
120 ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
124 /* Link owner module to the RTP engine for reference counting purposes */
125 engine->mod = module;
127 AST_RWLIST_WRLOCK(&engines);
129 /* Ensure that no two modules with the same name are registered at the same time */
130 AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
131 if (!strcmp(current_engine->name, engine->name)) {
132 ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
133 AST_RWLIST_UNLOCK(&engines);
138 /* The engine survived our critique. Off to the list it goes to be used */
139 AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
141 AST_RWLIST_UNLOCK(&engines);
143 ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
148 int ast_rtp_engine_unregister(struct ast_rtp_engine *engine)
150 struct ast_rtp_engine *current_engine = NULL;
152 AST_RWLIST_WRLOCK(&engines);
154 if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
155 ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
158 AST_RWLIST_UNLOCK(&engines);
160 return current_engine ? 0 : -1;
163 int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module)
165 struct ast_rtp_glue *current_glue = NULL;
167 if (ast_strlen_zero(glue->type)) {
173 AST_RWLIST_WRLOCK(&glues);
175 AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
176 if (!strcasecmp(current_glue->type, glue->type)) {
177 ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
178 AST_RWLIST_UNLOCK(&glues);
183 AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
185 AST_RWLIST_UNLOCK(&glues);
187 ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
192 int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
194 struct ast_rtp_glue *current_glue = NULL;
196 AST_RWLIST_WRLOCK(&glues);
198 if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
199 ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
202 AST_RWLIST_UNLOCK(&glues);
204 return current_glue ? 0 : -1;
207 static void instance_destructor(void *obj)
209 struct ast_rtp_instance *instance = obj;
211 /* Pass us off to the engine to destroy */
212 if (instance->data && instance->engine->destroy(instance)) {
213 ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
217 if (instance->srtp) {
218 res_srtp->destroy(instance->srtp);
221 /* Drop our engine reference */
222 ast_module_unref(instance->engine->mod);
224 ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
227 int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
229 ao2_ref(instance, -1);
234 struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name,
235 struct ast_sched_context *sched, const struct ast_sockaddr *sa,
238 struct ast_sockaddr address = {{0,}};
239 struct ast_rtp_instance *instance = NULL;
240 struct ast_rtp_engine *engine = NULL;
242 AST_RWLIST_RDLOCK(&engines);
244 /* If an engine name was specified try to use it or otherwise use the first one registered */
245 if (!ast_strlen_zero(engine_name)) {
246 AST_RWLIST_TRAVERSE(&engines, engine, entry) {
247 if (!strcmp(engine->name, engine_name)) {
252 engine = AST_RWLIST_FIRST(&engines);
255 /* If no engine was actually found bail out now */
257 ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
258 AST_RWLIST_UNLOCK(&engines);
262 /* Bump up the reference count before we return so the module can not be unloaded */
263 ast_module_ref(engine->mod);
265 AST_RWLIST_UNLOCK(&engines);
267 /* Allocate a new RTP instance */
268 if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
269 ast_module_unref(engine->mod);
272 instance->engine = engine;
273 ast_sockaddr_copy(&instance->local_address, sa);
274 ast_sockaddr_copy(&address, sa);
276 ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
278 /* And pass it off to the engine to setup */
279 if (instance->engine->new(instance, sched, &address, data)) {
280 ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
281 ao2_ref(instance, -1);
285 ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
290 void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
292 instance->data = data;
295 void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance)
297 return instance->data;
300 int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
302 return instance->engine->write(instance, frame);
305 struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp)
307 return instance->engine->read(instance, rtcp);
310 int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance,
311 const struct ast_sockaddr *address)
313 ast_sockaddr_copy(&instance->local_address, address);
317 int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance,
318 const struct ast_sockaddr *address)
320 ast_sockaddr_copy(&instance->remote_address, address);
324 if (instance->engine->remote_address_set) {
325 instance->engine->remote_address_set(instance, &instance->remote_address);
331 int ast_rtp_instance_set_alt_remote_address(struct ast_rtp_instance *instance,
332 const struct ast_sockaddr *address)
334 ast_sockaddr_copy(&instance->alt_remote_address, address);
338 if (instance->engine->alt_remote_address_set) {
339 instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address);
345 int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance,
346 struct ast_sockaddr *address)
348 if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
349 ast_sockaddr_copy(address, &instance->local_address);
356 void ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance,
357 struct ast_sockaddr *address)
359 ast_sockaddr_copy(address, &instance->local_address);
362 int ast_rtp_instance_get_and_cmp_remote_address(struct ast_rtp_instance *instance,
363 struct ast_sockaddr *address)
365 if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) {
366 ast_sockaddr_copy(address, &instance->remote_address);
373 void ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance,
374 struct ast_sockaddr *address)
376 ast_sockaddr_copy(address, &instance->remote_address);
379 void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
381 if (instance->engine->extended_prop_set) {
382 instance->engine->extended_prop_set(instance, property, value);
386 void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property)
388 if (instance->engine->extended_prop_get) {
389 return instance->engine->extended_prop_get(instance, property);
395 void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
397 instance->properties[property] = value;
399 if (instance->engine->prop_set) {
400 instance->engine->prop_set(instance, property, value);
404 int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property)
406 return instance->properties[property];
409 struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance)
411 return &instance->codecs;
414 void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
418 for (i = 0; i < AST_RTP_MAX_PT; i++) {
419 codecs->payloads[i].asterisk_format = 0;
420 codecs->payloads[i].rtp_code = 0;
421 ast_format_clear(&codecs->payloads[i].format);
422 if (instance && instance->engine && instance->engine->payload_set) {
423 instance->engine->payload_set(instance, i, 0, NULL, 0);
428 void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
432 ast_rwlock_rdlock(&static_RTP_PT_lock);
433 for (i = 0; i < AST_RTP_MAX_PT; i++) {
434 if (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
436 codecs->payloads[i].asterisk_format = static_RTP_PT[i].asterisk_format;
437 codecs->payloads[i].rtp_code = static_RTP_PT[i].rtp_code;
438 ast_format_copy(&codecs->payloads[i].format, &static_RTP_PT[i].format);
439 if (instance && instance->engine && instance->engine->payload_set) {
440 instance->engine->payload_set(instance, i, codecs->payloads[i].asterisk_format, &codecs->payloads[i].format, codecs->payloads[i].rtp_code);
444 ast_rwlock_unlock(&static_RTP_PT_lock);
447 void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
451 for (i = 0; i < AST_RTP_MAX_PT; i++) {
452 if (src->payloads[i].rtp_code || src->payloads[i].asterisk_format) {
453 ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
454 dest->payloads[i].asterisk_format = src->payloads[i].asterisk_format;
455 dest->payloads[i].rtp_code = src->payloads[i].rtp_code;
456 ast_format_copy(&dest->payloads[i].format, &src->payloads[i].format);
457 if (instance && instance->engine && instance->engine->payload_set) {
458 instance->engine->payload_set(instance, i, dest->payloads[i].asterisk_format, &dest->payloads[i].format, dest->payloads[i].rtp_code);
464 void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
467 ast_rwlock_rdlock(&static_RTP_PT_lock);
468 if (payload < 0 || payload >= AST_RTP_MAX_PT || (!static_RTP_PT[payload].rtp_code && !static_RTP_PT[payload].asterisk_format)) {
469 ast_rwlock_unlock(&static_RTP_PT_lock);
473 codecs->payloads[payload].asterisk_format = static_RTP_PT[payload].asterisk_format;
474 codecs->payloads[payload].rtp_code = static_RTP_PT[payload].rtp_code;
475 ast_format_copy(&codecs->payloads[payload].format, &static_RTP_PT[payload].format);
477 ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
479 if (instance && instance->engine && instance->engine->payload_set) {
480 instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, &codecs->payloads[payload].format, codecs->payloads[payload].rtp_code);
482 ast_rwlock_unlock(&static_RTP_PT_lock);
485 int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
486 char *mimetype, char *mimesubtype,
487 enum ast_rtp_options options,
488 unsigned int sample_rate)
493 if (pt < 0 || pt >= AST_RTP_MAX_PT)
494 return -1; /* bogus payload type */
496 ast_rwlock_rdlock(&mime_types_lock);
497 for (i = 0; i < mime_types_len; ++i) {
498 const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
500 if (strcasecmp(mimesubtype, t->subtype)) {
504 if (strcasecmp(mimetype, t->type)) {
508 /* if both sample rates have been supplied, and they don't match,
509 * then this not a match; if one has not been supplied, then the
510 * rates are not compared */
511 if (sample_rate && t->sample_rate &&
512 (sample_rate != t->sample_rate)) {
517 codecs->payloads[pt] = t->payload_type;
519 if ((t->payload_type.format.id == AST_FORMAT_G726) && t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
520 ast_format_set(&codecs->payloads[pt].format, AST_FORMAT_G726_AAL2, 0);
523 if (instance && instance->engine && instance->engine->payload_set) {
524 instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, &codecs->payloads[i].format, codecs->payloads[i].rtp_code);
529 ast_rwlock_unlock(&mime_types_lock);
531 return (found ? 0 : -2);
534 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)
536 return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
539 void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
541 if (payload < 0 || payload >= AST_RTP_MAX_PT) {
545 ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
547 codecs->payloads[payload].asterisk_format = 0;
548 codecs->payloads[payload].rtp_code = 0;
549 ast_format_clear(&codecs->payloads[payload].format);
551 if (instance && instance->engine && instance->engine->payload_set) {
552 instance->engine->payload_set(instance, payload, 0, NULL, 0);
556 struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
558 struct ast_rtp_payload_type result = { .asterisk_format = 0, };
560 if (payload < 0 || payload >= AST_RTP_MAX_PT) {
564 result.asterisk_format = codecs->payloads[payload].asterisk_format;
565 result.rtp_code = codecs->payloads[payload].rtp_code;
566 ast_format_copy(&result.format, &codecs->payloads[payload].format);
568 if (!result.rtp_code && !result.asterisk_format) {
569 ast_rwlock_rdlock(&static_RTP_PT_lock);
570 result = static_RTP_PT[payload];
571 ast_rwlock_unlock(&static_RTP_PT_lock);
578 struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload)
580 if (payload < 0 || payload >= AST_RTP_MAX_PT) {
583 if (!codecs->payloads[payload].asterisk_format) {
586 return &codecs->payloads[payload].format;
589 void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
593 ast_format_cap_remove_all(astformats);
596 for (i = 0; i < AST_RTP_MAX_PT; i++) {
597 if (codecs->payloads[i].rtp_code || codecs->payloads[i].asterisk_format) {
598 ast_debug(1, "Incorporating payload %d on %p\n", i, codecs);
600 if (codecs->payloads[i].asterisk_format) {
601 ast_format_cap_add(astformats, &codecs->payloads[i].format);
603 *nonastformats |= codecs->payloads[i].rtp_code;
608 int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
612 for (i = 0; i < AST_RTP_MAX_PT; i++) {
613 if (codecs->payloads[i].asterisk_format && asterisk_format && format &&
614 (ast_format_cmp(format, &codecs->payloads[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
616 } else if (!codecs->payloads[i].asterisk_format && !asterisk_format &&
617 (codecs->payloads[i].rtp_code == code)) {
622 ast_rwlock_rdlock(&static_RTP_PT_lock);
623 for (i = 0; i < AST_RTP_MAX_PT; i++) {
624 if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
625 (ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
628 } else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
629 (static_RTP_PT[i].rtp_code == code)) {
634 ast_rwlock_unlock(&static_RTP_PT_lock);
639 const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, struct ast_format *format, int code, enum ast_rtp_options options)
642 const char *res = "";
644 ast_rwlock_rdlock(&mime_types_lock);
645 for (i = 0; i < mime_types_len; i++) {
646 if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
647 (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
648 if ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
652 res = ast_rtp_mime_types[i].subtype;
655 } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
656 ast_rtp_mime_types[i].payload_type.rtp_code == code) {
658 res = ast_rtp_mime_types[i].subtype;
662 ast_rwlock_unlock(&mime_types_lock);
667 unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format *format, int code)
670 unsigned int res = 0;
672 ast_rwlock_rdlock(&mime_types_lock);
673 for (i = 0; i < mime_types_len; ++i) {
674 if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
675 (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
676 res = ast_rtp_mime_types[i].sample_rate;
678 } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
679 ast_rtp_mime_types[i].payload_type.rtp_code == code) {
680 res = ast_rtp_mime_types[i].sample_rate;
684 ast_rwlock_unlock(&mime_types_lock);
689 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)
698 if (asterisk_format) {
699 struct ast_format tmp_fmt;
700 ast_format_cap_iter_start(ast_format_capability);
701 while (!ast_format_cap_iter_next(ast_format_capability, &tmp_fmt)) {
702 name = ast_rtp_lookup_mime_subtype2(asterisk_format, &tmp_fmt, 0, options);
703 ast_str_append(&buf, 0, "%s|", name);
706 ast_format_cap_iter_end(ast_format_capability);
710 ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
711 for (x = 1; x < AST_RTP_MAX; x <<= 1) {
712 if (rtp_capability & x) {
713 name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
714 ast_str_append(&buf, 0, "%s|", name);
720 ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
722 return ast_str_buffer(buf);
725 void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
727 codecs->pref = *prefs;
729 if (instance && instance->engine->packetization_set) {
730 instance->engine->packetization_set(instance, &instance->codecs.pref);
734 int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
736 return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
739 int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
741 return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
743 int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
745 return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
748 int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
750 return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
753 enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
755 return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
758 void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
760 if (instance->engine->update_source) {
761 instance->engine->update_source(instance);
765 void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
767 if (instance->engine->change_source) {
768 instance->engine->change_source(instance);
772 int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
774 return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
777 void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
779 if (instance->engine->stop) {
780 instance->engine->stop(instance);
784 int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
786 return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
789 struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
791 struct ast_rtp_glue *glue = NULL;
793 AST_RWLIST_RDLOCK(&glues);
795 AST_RWLIST_TRAVERSE(&glues, glue, entry) {
796 if (!strcasecmp(glue->type, type)) {
801 AST_RWLIST_UNLOCK(&glues);
806 static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
808 enum ast_bridge_result res = AST_BRIDGE_FAILED;
809 struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
810 struct ast_frame *fr = NULL;
812 /* Start locally bridging both instances */
813 if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
814 ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c0), ast_channel_name(c1));
815 ast_channel_unlock(c0);
816 ast_channel_unlock(c1);
817 return AST_BRIDGE_FAILED_NOWARN;
819 if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
820 ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c1), ast_channel_name(c0));
821 if (instance0->engine->local_bridge) {
822 instance0->engine->local_bridge(instance0, NULL);
824 ast_channel_unlock(c0);
825 ast_channel_unlock(c1);
826 return AST_BRIDGE_FAILED_NOWARN;
829 ast_channel_unlock(c0);
830 ast_channel_unlock(c1);
832 instance0->bridged = instance1;
833 instance1->bridged = instance0;
835 ast_poll_channel_add(c0, c1);
837 /* Hop into a loop waiting for a frame from either channel */
842 /* If the underlying formats have changed force this bridge to break */
843 if ((ast_format_cmp(ast_channel_rawreadformat(c0), ast_channel_rawwriteformat(c1)) == AST_FORMAT_CMP_NOT_EQUAL) ||
844 (ast_format_cmp(ast_channel_rawreadformat(c1), ast_channel_rawwriteformat(c0)) == AST_FORMAT_CMP_NOT_EQUAL)) {
845 ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
846 res = AST_BRIDGE_FAILED_NOWARN;
849 /* Check if anything changed */
850 if ((ast_channel_tech_pvt(c0) != pvt0) ||
851 (ast_channel_tech_pvt(c1) != pvt1) ||
852 (ast_channel_masq(c0) || ast_channel_masqr(c0) || ast_channel_masq(c1) || ast_channel_masqr(c1)) ||
853 (ast_channel_monitor(c0) || ast_channel_audiohooks(c0) || ast_channel_monitor(c1) || ast_channel_audiohooks(c1)) ||
854 (!ast_framehook_list_is_empty(ast_channel_framehooks(c0)) || !ast_framehook_list_is_empty(ast_channel_framehooks(c1)))) {
855 ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
856 /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
857 if ((ast_channel_masq(c0) || ast_channel_masqr(c0)) && (fr = ast_read(c0))) {
860 if ((ast_channel_masq(c1) || ast_channel_masqr(c1)) && (fr = ast_read(c1))) {
863 res = AST_BRIDGE_RETRY;
866 /* Wait on a channel to feed us a frame */
867 if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
869 res = AST_BRIDGE_RETRY;
872 ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
873 if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
878 /* Read in frame from channel */
880 other = (who == c0) ? c1 : c0;
881 /* Depending on the frame we may need to break out of our bridge */
882 if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
883 ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
884 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
885 /* Record received frame and who */
888 ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
889 res = AST_BRIDGE_COMPLETE;
891 } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
892 if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
893 (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
894 (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
895 (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
896 (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) ||
897 (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
898 /* If we are going on hold, then break callback mode and P2P bridging */
899 if (fr->subclass.integer == AST_CONTROL_HOLD) {
900 if (instance0->engine->local_bridge) {
901 instance0->engine->local_bridge(instance0, NULL);
903 if (instance1->engine->local_bridge) {
904 instance1->engine->local_bridge(instance1, NULL);
906 instance0->bridged = NULL;
907 instance1->bridged = NULL;
908 } else if (fr->subclass.integer == AST_CONTROL_UNHOLD) {
909 if (instance0->engine->local_bridge) {
910 instance0->engine->local_bridge(instance0, instance1);
912 if (instance1->engine->local_bridge) {
913 instance1->engine->local_bridge(instance1, instance0);
915 instance0->bridged = instance1;
916 instance1->bridged = instance0;
918 /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
919 if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
920 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
923 } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
924 if (ast_channel_connected_line_sub(who, other, fr, 1) &&
925 ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
926 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
929 } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
930 if (ast_channel_redirecting_sub(who, other, fr, 1) &&
931 ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
932 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
935 } else if (fr->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
936 ast_channel_hangupcause_hash_set(other, fr->data.ptr);
941 ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, ast_channel_name(who));
942 res = AST_BRIDGE_COMPLETE;
946 if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
947 (fr->frametype == AST_FRAME_DTMF_END) ||
948 (fr->frametype == AST_FRAME_VOICE) ||
949 (fr->frametype == AST_FRAME_VIDEO) ||
950 (fr->frametype == AST_FRAME_IMAGE) ||
951 (fr->frametype == AST_FRAME_HTML) ||
952 (fr->frametype == AST_FRAME_MODEM) ||
953 (fr->frametype == AST_FRAME_TEXT)) {
954 ast_write(other, fr);
965 /* Stop locally bridging both instances */
966 if (instance0->engine->local_bridge) {
967 instance0->engine->local_bridge(instance0, NULL);
969 if (instance1->engine->local_bridge) {
970 instance1->engine->local_bridge(instance1, NULL);
973 instance0->bridged = NULL;
974 instance1->bridged = NULL;
976 ast_poll_channel_del(c0, c1);
981 static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0,
982 struct ast_channel *c1,
983 struct ast_rtp_instance *instance0,
984 struct ast_rtp_instance *instance1,
985 struct ast_rtp_instance *vinstance0,
986 struct ast_rtp_instance *vinstance1,
987 struct ast_rtp_instance *tinstance0,
988 struct ast_rtp_instance *tinstance1,
989 struct ast_rtp_glue *glue0,
990 struct ast_rtp_glue *glue1,
991 struct ast_format_cap *cap0,
992 struct ast_format_cap *cap1,
995 struct ast_frame **fo,
996 struct ast_channel **rc,
1000 enum ast_bridge_result res = AST_BRIDGE_FAILED;
1001 struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
1002 struct ast_format_cap *oldcap0 = ast_format_cap_dup(cap0);
1003 struct ast_format_cap *oldcap1 = ast_format_cap_dup(cap1);
1004 struct ast_sockaddr ac1 = {{0,}}, vac1 = {{0,}}, tac1 = {{0,}}, ac0 = {{0,}}, vac0 = {{0,}}, tac0 = {{0,}};
1005 struct ast_sockaddr t1 = {{0,}}, vt1 = {{0,}}, tt1 = {{0,}}, t0 = {{0,}}, vt0 = {{0,}}, tt0 = {{0,}};
1006 struct ast_frame *fr = NULL;
1008 if (!oldcap0 || !oldcap1) {
1009 ast_channel_unlock(c0);
1010 ast_channel_unlock(c1);
1011 goto remote_bridge_cleanup;
1013 /* Test the first channel */
1014 if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0))) {
1015 ast_rtp_instance_get_remote_address(instance1, &ac1);
1017 ast_rtp_instance_get_remote_address(vinstance1, &vac1);
1020 ast_rtp_instance_get_remote_address(tinstance1, &tac1);
1023 ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
1026 /* Test the second channel */
1027 if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0))) {
1028 ast_rtp_instance_get_remote_address(instance0, &ac0);
1030 ast_rtp_instance_get_remote_address(instance0, &vac0);
1033 ast_rtp_instance_get_remote_address(instance0, &tac0);
1036 ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c1), ast_channel_name(c0));
1039 ast_channel_unlock(c0);
1040 ast_channel_unlock(c1);
1042 instance0->bridged = instance1;
1043 instance1->bridged = instance0;
1045 ast_poll_channel_add(c0, c1);
1047 /* Go into a loop handling any stray frames that may come in */
1052 /* Check if anything changed */
1053 if ((ast_channel_tech_pvt(c0) != pvt0) ||
1054 (ast_channel_tech_pvt(c1) != pvt1) ||
1055 (ast_channel_masq(c0) || ast_channel_masqr(c0) || ast_channel_masq(c1) || ast_channel_masqr(c1)) ||
1056 (ast_channel_monitor(c0) || ast_channel_audiohooks(c0) || ast_channel_monitor(c1) || ast_channel_audiohooks(c1)) ||
1057 (!ast_framehook_list_is_empty(ast_channel_framehooks(c0)) || !ast_framehook_list_is_empty(ast_channel_framehooks(c1)))) {
1058 ast_debug(1, "Oooh, something is weird, backing out\n");
1059 res = AST_BRIDGE_RETRY;
1063 /* Check if they have changed their address */
1064 ast_rtp_instance_get_remote_address(instance1, &t1);
1066 ast_rtp_instance_get_remote_address(vinstance1, &vt1);
1069 ast_rtp_instance_get_remote_address(tinstance1, &tt1);
1071 if (glue1->get_codec) {
1072 ast_format_cap_remove_all(cap1);
1073 glue1->get_codec(c1, cap1);
1076 ast_rtp_instance_get_remote_address(instance0, &t0);
1078 ast_rtp_instance_get_remote_address(vinstance0, &vt0);
1081 ast_rtp_instance_get_remote_address(tinstance0, &tt0);
1083 if (glue0->get_codec) {
1084 ast_format_cap_remove_all(cap0);
1085 glue0->get_codec(c0, cap0);
1088 if ((ast_sockaddr_cmp(&t1, &ac1)) ||
1089 (vinstance1 && ast_sockaddr_cmp(&vt1, &vac1)) ||
1090 (tinstance1 && ast_sockaddr_cmp(&tt1, &tac1)) ||
1091 (!ast_format_cap_identical(cap1, oldcap1))) {
1092 char tmp_buf[512] = { 0, };
1093 ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
1094 ast_channel_name(c1), ast_sockaddr_stringify(&t1),
1095 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
1096 ast_debug(1, "Oooh, '%s' changed end vaddress to %s (format %s)\n",
1097 ast_channel_name(c1), ast_sockaddr_stringify(&vt1),
1098 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
1099 ast_debug(1, "Oooh, '%s' changed end taddress to %s (format %s)\n",
1100 ast_channel_name(c1), ast_sockaddr_stringify(&tt1),
1101 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
1102 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
1103 ast_channel_name(c1), ast_sockaddr_stringify(&ac1),
1104 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
1105 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
1106 ast_channel_name(c1), ast_sockaddr_stringify(&vac1),
1107 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
1108 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
1109 ast_channel_name(c1), ast_sockaddr_stringify(&tac1),
1110 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
1111 if (glue0->update_peer(c0,
1112 ast_sockaddr_isnull(&t1) ? NULL : instance1,
1113 ast_sockaddr_isnull(&vt1) ? NULL : vinstance1,
1114 ast_sockaddr_isnull(&tt1) ? NULL : tinstance1,
1116 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
1118 ast_sockaddr_copy(&ac1, &t1);
1119 ast_sockaddr_copy(&vac1, &vt1);
1120 ast_sockaddr_copy(&tac1, &tt1);
1121 ast_format_cap_copy(oldcap1, cap1);
1123 if ((ast_sockaddr_cmp(&t0, &ac0)) ||
1124 (vinstance0 && ast_sockaddr_cmp(&vt0, &vac0)) ||
1125 (tinstance0 && ast_sockaddr_cmp(&tt0, &tac0)) ||
1126 (!ast_format_cap_identical(cap0, oldcap0))) {
1127 char tmp_buf[512] = { 0, };
1128 ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
1129 ast_channel_name(c0), ast_sockaddr_stringify(&t0),
1130 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap0));
1131 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
1132 ast_channel_name(c0), ast_sockaddr_stringify(&ac0),
1133 ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap0));
1134 if (glue1->update_peer(c1, t0.len ? instance0 : NULL,
1135 vt0.len ? vinstance0 : NULL,
1136 tt0.len ? tinstance0 : NULL,
1138 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", ast_channel_name(c1), ast_channel_name(c0));
1140 ast_sockaddr_copy(&ac0, &t0);
1141 ast_sockaddr_copy(&vac0, &vt0);
1142 ast_sockaddr_copy(&tac0, &tt0);
1143 ast_format_cap_copy(oldcap0, cap0);
1146 /* Wait for frame to come in on the channels */
1147 if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
1149 res = AST_BRIDGE_RETRY;
1152 ast_debug(1, "Ooh, empty read...\n");
1153 if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
1159 other = (who == c0) ? c1 : c0;
1160 if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
1161 (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
1162 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
1163 /* Break out of bridge */
1166 ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
1167 res = AST_BRIDGE_COMPLETE;
1169 } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
1170 if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
1171 (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
1172 (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
1173 (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
1174 (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) ||
1175 (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
1176 if (fr->subclass.integer == AST_CONTROL_HOLD) {
1177 /* If we someone went on hold we want the other side to reinvite back to us */
1179 glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
1181 glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
1183 } else if (fr->subclass.integer == AST_CONTROL_UNHOLD ||
1184 fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER) {
1185 /* If they went off hold they should go back to being direct, or if we have
1186 * been told to force a peer update, go ahead and do it. */
1188 glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0);
1190 glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0);
1193 /* Update local address information */
1194 ast_rtp_instance_get_remote_address(instance0, &t0);
1195 ast_sockaddr_copy(&ac0, &t0);
1196 ast_rtp_instance_get_remote_address(instance1, &t1);
1197 ast_sockaddr_copy(&ac1, &t1);
1198 /* Update codec information */
1199 if (glue0->get_codec && ast_channel_tech_pvt(c0)) {
1200 ast_format_cap_remove_all(cap0);
1201 ast_format_cap_remove_all(oldcap0);
1202 glue0->get_codec(c0, cap0);
1203 ast_format_cap_append(oldcap0, cap0);
1206 if (glue1->get_codec && ast_channel_tech_pvt(c1)) {
1207 ast_format_cap_remove_all(cap1);
1208 ast_format_cap_remove_all(oldcap1);
1209 glue0->get_codec(c1, cap1);
1210 ast_format_cap_append(oldcap1, cap1);
1212 /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
1213 if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
1214 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
1217 } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
1218 if (ast_channel_connected_line_sub(who, other, fr, 1) &&
1219 ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
1220 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
1223 } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
1224 if (ast_channel_redirecting_sub(who, other, fr, 1) &&
1225 ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
1226 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
1229 } else if (fr->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
1230 ast_channel_hangupcause_hash_set(other, fr->data.ptr);
1235 ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, ast_channel_name(who));
1236 res = AST_BRIDGE_COMPLETE;
1237 goto remote_bridge_cleanup;
1240 if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
1241 (fr->frametype == AST_FRAME_DTMF_END) ||
1242 (fr->frametype == AST_FRAME_VOICE) ||
1243 (fr->frametype == AST_FRAME_VIDEO) ||
1244 (fr->frametype == AST_FRAME_IMAGE) ||
1245 (fr->frametype == AST_FRAME_HTML) ||
1246 (fr->frametype == AST_FRAME_MODEM) ||
1247 (fr->frametype == AST_FRAME_TEXT)) {
1248 ast_write(other, fr);
1258 if (ast_test_flag(ast_channel_flags(c0), AST_FLAG_ZOMBIE)) {
1259 ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", ast_channel_name(c0));
1260 } else if (ast_channel_tech_pvt(c0) != pvt0) {
1261 ast_debug(1, "Channel c0->'%s' pvt changed, in bridge with c1->'%s'\n", ast_channel_name(c0), ast_channel_name(c1));
1262 } else if (glue0 != ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) {
1263 ast_debug(1, "Channel c0->'%s' technology changed, in bridge with c1->'%s'\n", ast_channel_name(c0), ast_channel_name(c1));
1264 } else if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
1265 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", ast_channel_name(c0));
1267 if (ast_test_flag(ast_channel_flags(c1), AST_FLAG_ZOMBIE)) {
1268 ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", ast_channel_name(c1));
1269 } else if (ast_channel_tech_pvt(c1) != pvt1) {
1270 ast_debug(1, "Channel c1->'%s' pvt changed, in bridge with c0->'%s'\n", ast_channel_name(c1), ast_channel_name(c0));
1271 } else if (glue1 != ast_rtp_instance_get_glue(ast_channel_tech(c1)->type)) {
1272 ast_debug(1, "Channel c1->'%s' technology changed, in bridge with c0->'%s'\n", ast_channel_name(c1), ast_channel_name(c0));
1273 } else if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
1274 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", ast_channel_name(c1));
1277 instance0->bridged = NULL;
1278 instance1->bridged = NULL;
1280 ast_poll_channel_del(c0, c1);
1282 remote_bridge_cleanup:
1283 ast_format_cap_destroy(oldcap0);
1284 ast_format_cap_destroy(oldcap1);
1290 * \brief Conditionally unref an rtp instance
1292 static void unref_instance_cond(struct ast_rtp_instance **instance)
1295 ao2_ref(*instance, -1);
1300 enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
1302 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
1303 *vinstance0 = NULL, *vinstance1 = NULL,
1304 *tinstance0 = NULL, *tinstance1 = NULL;
1305 struct ast_rtp_glue *glue0, *glue1;
1306 struct ast_sockaddr addr1 = { {0, }, }, addr2 = { {0, }, };
1307 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1308 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1309 enum ast_bridge_result res = AST_BRIDGE_FAILED;
1310 enum ast_rtp_dtmf_mode dmode;
1311 struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
1312 struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
1313 int unlock_chans = 1;
1315 if (!cap0 || !cap1) {
1320 /* Lock both channels so we can look for the glue that binds them together */
1321 ast_channel_lock(c0);
1322 while (ast_channel_trylock(c1)) {
1323 ast_channel_unlock(c0);
1325 ast_channel_lock(c0);
1328 /* Ensure neither channel got hungup during lock avoidance */
1329 if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
1330 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
1334 /* Grab glue that binds each channel to something using the RTP engine */
1335 if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
1336 ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
1340 audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
1341 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
1343 audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
1344 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
1346 /* If the channels are of the same technology, they might have limitations on remote bridging */
1347 if (ast_channel_tech(c0) == ast_channel_tech(c1)) {
1348 if (audio_glue0_res == audio_glue1_res && audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
1349 if (glue0->allow_rtp_remote && !(glue0->allow_rtp_remote(c0, c1))) {
1350 /* If the allow_rtp_remote indicates that remote isn't allowed, revert to local bridge */
1351 audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
1354 if (video_glue0_res == video_glue1_res && video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
1355 if (glue0->allow_vrtp_remote && !(glue0->allow_vrtp_remote(c0, c1))) {
1356 /* if the allow_vrtp_remote indicates that remote isn't allowed, revert to local bridge */
1357 video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
1362 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
1363 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)) {
1364 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1366 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)) {
1367 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1370 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
1371 if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
1372 res = AST_BRIDGE_FAILED_NOWARN;
1377 /* If address families differ, force a local bridge */
1378 ast_rtp_instance_get_remote_address(instance0, &addr1);
1379 ast_rtp_instance_get_remote_address(instance1, &addr2);
1381 if (addr1.ss.ss_family != addr2.ss.ss_family ||
1382 (ast_sockaddr_is_ipv4_mapped(&addr1) != ast_sockaddr_is_ipv4_mapped(&addr2))) {
1383 audio_glue0_res = AST_RTP_GLUE_RESULT_LOCAL;
1384 audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
1387 /* If we need to get DTMF see if we can do it outside of the RTP stream itself */
1388 dmode = ast_rtp_instance_dtmf_mode_get(instance0);
1389 if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && dmode) {
1390 res = AST_BRIDGE_FAILED_NOWARN;
1393 dmode = ast_rtp_instance_dtmf_mode_get(instance1);
1394 if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && dmode) {
1395 res = AST_BRIDGE_FAILED_NOWARN;
1399 /* If we have gotten to a local bridge make sure that both sides have the same local bridge callback and that they are DTMF compatible */
1400 if ((audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) && ((instance0->engine->local_bridge != instance1->engine->local_bridge) || (instance0->engine->dtmf_compatible && !instance0->engine->dtmf_compatible(c0, instance0, c1, instance1)))) {
1401 res = AST_BRIDGE_FAILED_NOWARN;
1405 /* Make sure that codecs match */
1406 if (glue0->get_codec){
1407 glue0->get_codec(c0, cap0);
1409 if (glue1->get_codec) {
1410 glue1->get_codec(c1, cap1);
1412 if (!ast_format_cap_is_empty(cap0) && !ast_format_cap_is_empty(cap1) && !ast_format_cap_has_joint(cap0, cap1)) {
1413 char tmp0[256] = { 0, };
1414 char tmp1[256] = { 0, };
1415 ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n",
1416 ast_getformatname_multiple(tmp0, sizeof(tmp0), cap0),
1417 ast_getformatname_multiple(tmp1, sizeof(tmp1), cap1));
1418 res = AST_BRIDGE_FAILED_NOWARN;
1422 instance0->glue = glue0;
1423 instance1->glue = glue1;
1424 instance0->chan = c0;
1425 instance1->chan = c1;
1427 /* Depending on the end result for bridging either do a local bridge or remote bridge */
1428 if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
1429 ast_verb(3, "Locally bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
1430 res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
1432 ast_verb(3, "Remotely bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
1433 res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
1434 tinstance0, tinstance1, glue0, glue1, cap0, cap1, timeoutms, flags,
1435 fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
1438 instance0->glue = NULL;
1439 instance1->glue = NULL;
1440 instance0->chan = NULL;
1441 instance1->chan = NULL;
1447 ast_channel_unlock(c0);
1448 ast_channel_unlock(c1);
1450 ast_format_cap_destroy(cap1);
1451 ast_format_cap_destroy(cap0);
1453 unref_instance_cond(&instance0);
1454 unref_instance_cond(&instance1);
1455 unref_instance_cond(&vinstance0);
1456 unref_instance_cond(&vinstance1);
1457 unref_instance_cond(&tinstance0);
1458 unref_instance_cond(&tinstance1);
1463 struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
1465 return instance->bridged;
1468 void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
1470 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
1471 *vinstance0 = NULL, *vinstance1 = NULL,
1472 *tinstance0 = NULL, *tinstance1 = NULL;
1473 struct ast_rtp_glue *glue0, *glue1;
1474 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1475 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1476 struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
1477 struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
1479 /* Lock both channels so we can look for the glue that binds them together */
1480 ast_channel_lock_both(c0, c1);
1482 if (!cap1 || !cap0) {
1486 /* Grab glue that binds each channel to something using the RTP engine */
1487 if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
1488 ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
1492 audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
1493 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
1495 audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
1496 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
1498 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
1499 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)) {
1500 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1502 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)) {
1503 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1505 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) {
1506 glue0->get_codec(c0, cap0);
1508 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) {
1509 glue1->get_codec(c1, cap1);
1512 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
1513 if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
1517 /* Make sure we have matching codecs */
1518 if (!ast_format_cap_has_joint(cap0, cap1)) {
1522 ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
1524 if (vinstance0 && vinstance1) {
1525 ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
1527 if (tinstance0 && tinstance1) {
1528 ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
1531 if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
1532 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
1533 ast_channel_name(c0), ast_channel_name(c1));
1535 ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
1536 ast_channel_name(c0), ast_channel_name(c1));
1540 ast_channel_unlock(c0);
1541 ast_channel_unlock(c1);
1543 ast_format_cap_destroy(cap0);
1544 ast_format_cap_destroy(cap1);
1546 unref_instance_cond(&instance0);
1547 unref_instance_cond(&instance1);
1548 unref_instance_cond(&vinstance0);
1549 unref_instance_cond(&vinstance1);
1550 unref_instance_cond(&tinstance0);
1551 unref_instance_cond(&tinstance1);
1554 int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
1556 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
1557 *vinstance0 = NULL, *vinstance1 = NULL,
1558 *tinstance0 = NULL, *tinstance1 = NULL;
1559 struct ast_rtp_glue *glue0, *glue1;
1560 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1561 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1562 struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
1563 struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
1566 /* If there is no second channel just immediately bail out, we are of no use in that scenario */
1568 ast_format_cap_destroy(cap0);
1569 ast_format_cap_destroy(cap1);
1573 /* Lock both channels so we can look for the glue that binds them together */
1574 ast_channel_lock(c0);
1575 while (ast_channel_trylock(c1)) {
1576 ast_channel_unlock(c0);
1578 ast_channel_lock(c0);
1581 if (!cap1 || !cap0) {
1585 /* Grab glue that binds each channel to something using the RTP engine */
1586 if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
1587 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
1591 audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
1592 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
1594 audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
1595 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
1597 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
1598 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)) {
1599 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
1601 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)) {
1602 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
1604 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) {
1605 glue0->get_codec(c0, cap0);
1607 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) {
1608 glue1->get_codec(c1, cap1);
1611 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
1612 if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
1616 /* Make sure we have matching codecs */
1617 if (!ast_format_cap_has_joint(cap0, cap1)) {
1621 /* Bridge media early */
1622 if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
1623 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
1629 ast_channel_unlock(c0);
1630 ast_channel_unlock(c1);
1632 ast_format_cap_destroy(cap0);
1633 ast_format_cap_destroy(cap1);
1635 unref_instance_cond(&instance0);
1636 unref_instance_cond(&instance1);
1637 unref_instance_cond(&vinstance0);
1638 unref_instance_cond(&vinstance1);
1639 unref_instance_cond(&tinstance0);
1640 unref_instance_cond(&tinstance1);
1643 ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
1649 int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
1651 return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
1654 int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
1656 return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
1659 int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
1661 return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
1664 char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size)
1666 struct ast_rtp_instance_stats stats = { 0, };
1667 enum ast_rtp_instance_stat stat;
1669 /* Determine what statistics we will need to retrieve based on field passed in */
1670 if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
1671 stat = AST_RTP_INSTANCE_STAT_ALL;
1672 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
1673 stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
1674 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
1675 stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
1676 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
1677 stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
1682 /* Attempt to actually retrieve the statistics we need to generate the quality string */
1683 if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
1687 /* Now actually fill the buffer with the good information */
1688 if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
1689 snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
1690 stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
1691 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
1692 snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
1693 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));
1694 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
1695 snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
1696 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));
1697 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
1698 snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
1704 void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
1706 char quality_buf[AST_MAX_USER_FIELD], *quality;
1707 struct ast_channel *bridge = ast_bridged_channel(chan);
1709 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
1710 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
1712 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
1716 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
1717 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
1719 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
1723 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
1724 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
1726 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
1730 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
1731 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
1733 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
1738 int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)
1740 return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
1743 int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, struct ast_format *format)
1745 return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
1748 int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer)
1750 struct ast_rtp_glue *glue;
1751 struct ast_rtp_instance *peer_instance = NULL;
1754 if (!instance->engine->make_compatible) {
1758 ast_channel_lock(peer);
1760 if (!(glue = ast_rtp_instance_get_glue(ast_channel_tech(peer)->type))) {
1761 ast_channel_unlock(peer);
1765 glue->get_rtp_info(peer, &peer_instance);
1767 if (!peer_instance || peer_instance->engine != instance->engine) {
1768 ast_channel_unlock(peer);
1769 ao2_ref(peer_instance, -1);
1770 peer_instance = NULL;
1774 res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
1776 ast_channel_unlock(peer);
1778 ao2_ref(peer_instance, -1);
1779 peer_instance = NULL;
1784 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)
1786 if (instance->engine->available_formats) {
1787 instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
1788 if (!ast_format_cap_is_empty(result)) {
1793 ast_translate_available_formats(to_endpoint, to_asterisk, result);
1796 int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
1798 return instance->engine->activate ? instance->engine->activate(instance) : 0;
1801 void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance,
1802 struct ast_sockaddr *suggestion,
1803 const char *username)
1805 if (instance->engine->stun_request) {
1806 instance->engine->stun_request(instance, suggestion, username);
1810 void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout)
1812 instance->timeout = timeout;
1815 void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout)
1817 instance->holdtimeout = timeout;
1820 void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int interval)
1822 instance->keepalive = interval;
1825 int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
1827 return instance->timeout;
1830 int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
1832 return instance->holdtimeout;
1835 int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance)
1837 return instance->keepalive;
1840 struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
1842 return instance->engine;
1845 struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *instance)
1847 return instance->glue;
1850 struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance)
1852 return instance->chan;
1855 int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res)
1857 if (res_srtp || res_srtp_policy) {
1860 if (!srtp_res || !policy_res) {
1864 res_srtp = srtp_res;
1865 res_srtp_policy = policy_res;
1870 void ast_rtp_engine_unregister_srtp(void)
1873 res_srtp_policy = NULL;
1876 int ast_rtp_engine_srtp_is_registered(void)
1878 return res_srtp && res_srtp_policy;
1881 int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy)
1889 if (!instance->srtp) {
1890 res = res_srtp->create(&instance->srtp, instance, remote_policy);
1892 res = res_srtp->replace(&instance->srtp, instance, remote_policy);
1895 res = res_srtp->add_stream(instance->srtp, local_policy);
1901 struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
1903 return instance->srtp;
1906 int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
1908 if (instance->engine->sendcng) {
1909 return instance->engine->sendcng(instance, level);
1915 static void set_next_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate)
1917 int x = mime_types_len;
1918 if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
1922 ast_rwlock_wrlock(&mime_types_lock);
1924 ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
1925 ast_format_copy(&ast_rtp_mime_types[x].payload_type.format, format);
1927 ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
1929 ast_rtp_mime_types[x].type = type;
1930 ast_rtp_mime_types[x].subtype = subtype;
1931 ast_rtp_mime_types[x].sample_rate = sample_rate;
1933 ast_rwlock_unlock(&mime_types_lock);
1936 static void add_static_payload(int map, const struct ast_format *format, int rtp_code)
1939 ast_rwlock_wrlock(&static_RTP_PT_lock);
1941 /* find next available dynamic payload slot */
1942 for (x = 96; x < 127; x++) {
1943 if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
1951 ast_log(LOG_WARNING, "No Dynamic RTP mapping avaliable for format %s\n" ,ast_getformatname(format));
1952 ast_rwlock_unlock(&static_RTP_PT_lock);
1957 static_RTP_PT[map].asterisk_format = 1;
1958 ast_format_copy(&static_RTP_PT[map].format, format);
1960 static_RTP_PT[map].rtp_code = rtp_code;
1962 ast_rwlock_unlock(&static_RTP_PT_lock);
1965 int ast_rtp_engine_load_format(const struct ast_format *format)
1967 switch (format->id) {
1968 case AST_FORMAT_SILK:
1969 set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
1970 add_static_payload(-1, format, 0);
1972 case AST_FORMAT_CELT:
1973 set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
1974 add_static_payload(-1, format, 0);
1983 int ast_rtp_engine_unload_format(const struct ast_format *format)
1988 ast_rwlock_wrlock(&static_RTP_PT_lock);
1989 /* remove everything pertaining to this format id from the lists */
1990 for (x = 0; x < AST_RTP_MAX_PT; x++) {
1991 if (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
1992 memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
1995 ast_rwlock_unlock(&static_RTP_PT_lock);
1998 ast_rwlock_wrlock(&mime_types_lock);
1999 /* rebuild the list skipping the items matching this id */
2000 for (x = 0; x < mime_types_len; x++) {
2001 if (ast_format_cmp(&ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
2004 ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
2008 ast_rwlock_unlock(&mime_types_lock);
2012 int ast_rtp_engine_init()
2014 struct ast_format tmpfmt;
2016 ast_rwlock_init(&mime_types_lock);
2017 ast_rwlock_init(&static_RTP_PT_lock);
2019 /* Define all the RTP mime types available */
2020 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0, "audio", "G723", 8000);
2021 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0, "audio", "GSM", 8000);
2022 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "PCMU", 8000);
2023 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "G711U", 8000);
2024 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "PCMA", 8000);
2025 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "G711A", 8000);
2026 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0, "audio", "G726-32", 8000);
2027 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0, "audio", "DVI4", 8000);
2028 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0, "audio", "L16", 8000);
2029 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16", 16000);
2030 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16-256", 16000);
2031 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0, "audio", "LPC", 8000);
2032 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729", 8000);
2033 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729A", 8000);
2034 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G.729", 8000);
2035 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0, "audio", "speex", 8000);
2036 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0, "audio", "speex", 16000);
2037 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0, "audio", "speex", 32000);
2038 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0, "audio", "iLBC", 8000);
2039 /* this is the sample rate listed in the RTP profile for the G.722 codec, *NOT* the actual sample rate of the media stream */
2040 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0, "audio", "G722", 8000);
2041 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0, "audio", "AAL2-G726-32", 8000);
2042 set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
2043 set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
2044 set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
2045 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0, "video", "JPEG", 90000);
2046 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_PNG, 0), 0, "video", "PNG", 90000);
2047 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0, "video", "H261", 90000);
2048 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0, "video", "H263", 90000);
2049 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0, "video", "h263-1998", 90000);
2050 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0, "video", "H264", 90000);
2051 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0, "video", "MP4V-ES", 90000);
2052 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0, "text", "RED", 1000);
2053 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0, "text", "T140", 1000);
2054 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
2055 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
2056 set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
2058 /* Define the static rtp payload mappings */
2059 add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
2060 #ifdef USE_DEPRECATED_G726
2061 add_static_payload(2, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
2063 add_static_payload(3, ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0);
2064 add_static_payload(4, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0);
2065 add_static_payload(5, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0);/* 8 kHz */
2066 add_static_payload(6, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 16 kHz */
2067 add_static_payload(7, ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0);
2068 add_static_payload(8, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0);
2069 add_static_payload(9, ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0);
2070 add_static_payload(10, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 2 channels */
2071 add_static_payload(11, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 1 channel */
2072 add_static_payload(13, NULL, AST_RTP_CN);
2073 add_static_payload(16, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 11.025 kHz */
2074 add_static_payload(17, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 22.050 kHz */
2075 add_static_payload(18, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0);
2076 add_static_payload(19, NULL, AST_RTP_CN); /* Also used for CN */
2077 add_static_payload(26, ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0);
2078 add_static_payload(31, ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0);
2079 add_static_payload(34, ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0);
2080 add_static_payload(97, ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0);
2081 add_static_payload(98, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
2082 add_static_payload(99, ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0);
2083 add_static_payload(101, NULL, AST_RTP_DTMF);
2084 add_static_payload(102, ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0);
2085 add_static_payload(103, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
2086 add_static_payload(104, ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0);
2087 add_static_payload(105, ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0); /* Real time text chat (with redundancy encoding) */
2088 add_static_payload(106, ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0); /* Real time text chat */
2089 add_static_payload(110, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0);
2090 add_static_payload(111, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);
2091 add_static_payload(112, ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0);
2092 add_static_payload(115, ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0);
2093 add_static_payload(116, ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0);
2094 add_static_payload(117, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0);
2095 add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
2096 add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
2097 add_static_payload(121, NULL, AST_RTP_CISCO_DTMF); /* Must be type 121 */