Asterisk - The Open Source Telephony Project GIT-master-70eff7f
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res_pjsip_session.c
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1/*
2* Asterisk -- An open source telephony toolkit.
3*
4* Copyright (C) 2013, Digium, Inc.
5*
6* Mark Michelson <mmichelson@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/*** MODULEINFO
20 <depend>pjproject</depend>
21 <depend>res_pjsip</depend>
22 <support_level>core</support_level>
23 ***/
24
25#include "asterisk.h"
26
27#include <pjsip.h>
28#include <pjsip_ua.h>
29#include <pjlib.h>
30#include <pjmedia.h>
31
32#include "asterisk/res_pjsip.h"
35#include "asterisk/callerid.h"
36#include "asterisk/datastore.h"
37#include "asterisk/module.h"
38#include "asterisk/logger.h"
39#include "asterisk/res_pjsip.h"
40#include "asterisk/astobj2.h"
41#include "asterisk/lock.h"
42#include "asterisk/uuid.h"
43#include "asterisk/pbx.h"
45#include "asterisk/taskpool.h"
46#include "asterisk/causes.h"
47#include "asterisk/sdp_srtp.h"
48#include "asterisk/dsp.h"
49#include "asterisk/acl.h"
51#include "asterisk/pickup.h"
52#include "asterisk/test.h"
53#include "asterisk/stream.h"
54#include "asterisk/vector.h"
55
57
58#define SDP_HANDLER_BUCKETS 11
59
60#define MOD_DATA_ON_RESPONSE "on_response"
61
62/* Most common case is one audio and one video stream */
63#define DEFAULT_NUM_SESSION_MEDIA 2
64
65/* Some forward declarations */
67static void handle_session_end(struct ast_sip_session *session);
69static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata);
70static void handle_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata,
71 enum ast_sip_session_response_priority response_priority);
72static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata,
73 enum ast_sip_session_response_priority response_priority);
74static void handle_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata);
75static void handle_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata);
77 ast_sip_session_request_creation_cb on_request_creation,
78 ast_sip_session_sdp_creation_cb on_sdp_creation,
80 enum ast_sip_session_refresh_method method, int generate_new_sdp,
81 struct ast_sip_session_media_state *pending_media_state,
82 struct ast_sip_session_media_state *active_media_state,
83 int queued);
84
85/*! \brief NAT hook for modifying outgoing messages with SDP */
87
88/*!
89 * \brief Registered SDP stream handlers
90 *
91 * This container is keyed on stream types. Each
92 * object in the container is a linked list of
93 * handlers for the stream type.
94 */
96
97/*!
98 * These are the objects in the sdp_handlers container
99 */
101 /* The list of handlers to visit */
103 /* The handlers in this list handle streams of this type */
104 char stream_type[1];
105};
106
107static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer, const unsigned int ignore_active_stream_topology);
108
109static int sdp_handler_list_hash(const void *obj, int flags)
110{
111 const struct sdp_handler_list *handler_list = obj;
112 const char *stream_type = flags & OBJ_KEY ? obj : handler_list->stream_type;
113
115}
116
118{
119 if (!session) {
120 return "(null session)";
121 }
122 if (session->channel) {
123 return ast_channel_name(session->channel);
124 } else if (session->endpoint) {
125 return ast_sorcery_object_get_id(session->endpoint);
126 } else {
127 return "unknown";
128 }
129}
130
132{
133 return id->number.valid
134 && (session->endpoint->id.trust_outbound
136}
137
138static int sdp_handler_list_cmp(void *obj, void *arg, int flags)
139{
140 struct sdp_handler_list *handler_list1 = obj;
141 struct sdp_handler_list *handler_list2 = arg;
142 const char *stream_type2 = flags & OBJ_KEY ? arg : handler_list2->stream_type;
143
144 return strcmp(handler_list1->stream_type, stream_type2) ? 0 : CMP_MATCH | CMP_STOP;
145}
146
148{
149 RAII_VAR(struct sdp_handler_list *, handler_list,
152
153 if (handler_list) {
154 struct ast_sip_session_sdp_handler *iter;
155 /* Check if this handler is already registered for this stream type */
156 AST_LIST_TRAVERSE(&handler_list->list, iter, next) {
157 if (!strcmp(iter->id, handler->id)) {
158 ast_log(LOG_WARNING, "Handler '%s' already registered for stream type '%s'.\n", handler->id, stream_type);
159 return -1;
160 }
161 }
162 AST_LIST_INSERT_TAIL(&handler_list->list, handler, next);
163 ast_debug(1, "Registered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
164
165 return 0;
166 }
167
168 /* No stream of this type has been registered yet, so we need to create a new list */
169 handler_list = ao2_alloc(sizeof(*handler_list) + strlen(stream_type), NULL);
170 if (!handler_list) {
171 return -1;
172 }
173 /* Safe use of strcpy */
174 strcpy(handler_list->stream_type, stream_type);
175 AST_LIST_HEAD_INIT_NOLOCK(&handler_list->list);
176 AST_LIST_INSERT_TAIL(&handler_list->list, handler, next);
177 if (!ao2_link(sdp_handlers, handler_list)) {
178 return -1;
179 }
180 ast_debug(1, "Registered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
181
182 return 0;
183}
184
185static int remove_handler(void *obj, void *arg, void *data, int flags)
186{
187 struct sdp_handler_list *handler_list = obj;
189 struct ast_sip_session_sdp_handler *iter;
190 const char *stream_type = arg;
191
192 AST_LIST_TRAVERSE_SAFE_BEGIN(&handler_list->list, iter, next) {
193 if (!strcmp(iter->id, handler->id)) {
195 ast_debug(1, "Unregistered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
196 }
197 }
199
200 if (AST_LIST_EMPTY(&handler_list->list)) {
201 ast_debug(3, "No more handlers exist for stream type '%s'\n", stream_type);
202 return CMP_MATCH;
203 } else {
204 return CMP_STOP;
205 }
206}
207
212
214 const struct ast_rtp_instance_stats *vec_elem, const struct ast_rtp_instance_stats *srch)
215{
216 if (vec_elem->local_ssrc == srch->local_ssrc) {
217 return 1;
218 }
219
220 return 0;
221}
222
224 size_t sessions, size_t read_callbacks)
225{
226 struct ast_sip_session_media_state *media_state;
227
228 media_state = ast_calloc(1, sizeof(*media_state));
229 if (!media_state) {
230 return NULL;
231 }
232
233 if (AST_VECTOR_INIT(&media_state->sessions, sessions) < 0) {
234 ast_free(media_state);
235 return NULL;
236 }
237
238 if (AST_VECTOR_INIT(&media_state->read_callbacks, read_callbacks) < 0) {
239 AST_VECTOR_FREE(&media_state->sessions);
240 ast_free(media_state);
241 return NULL;
242 }
243
244 return media_state;
245}
246
252
254{
255 int i;
256 int ret;
257
258 if (!media_state || !sip_session) {
259 return;
260 }
261
262 for (i = 0; i < AST_VECTOR_SIZE(&media_state->sessions); i++) {
263 struct ast_rtp_instance_stats *stats_tmp = NULL;
264 struct ast_sip_session_media *media = AST_VECTOR_GET(&media_state->sessions, i);
265 if (!media || !media->rtp) {
266 continue;
267 }
268
269 stats_tmp = ast_calloc(1, sizeof(struct ast_rtp_instance_stats));
270 if (!stats_tmp) {
271 return;
272 }
273
275 if (ret) {
276 ast_free(stats_tmp);
277 continue;
278 }
279
280 /* remove all the duplicated stats if exist */
282
283 AST_VECTOR_APPEND(&sip_session->media_stats, stats_tmp);
284 }
285}
286
288{
289 int index;
290
291 if (!media_state) {
292 return;
293 }
294
295 AST_VECTOR_RESET(&media_state->sessions, ao2_cleanup);
297
298 for (index = 0; index < AST_MEDIA_TYPE_END; ++index) {
299 media_state->default_session[index] = NULL;
300 }
301
303 media_state->topology = NULL;
304}
305
307{
308 struct ast_sip_session_media_state *cloned;
309 int index;
310
311 if (!media_state) {
312 return NULL;
313 }
314
316 AST_VECTOR_SIZE(&media_state->sessions),
317 AST_VECTOR_SIZE(&media_state->read_callbacks));
318 if (!cloned) {
319 return NULL;
320 }
321
322 if (media_state->topology) {
323 cloned->topology = ast_stream_topology_clone(media_state->topology);
324 if (!cloned->topology) {
326 return NULL;
327 }
328 }
329
330 for (index = 0; index < AST_VECTOR_SIZE(&media_state->sessions); ++index) {
331 struct ast_sip_session_media *session_media = AST_VECTOR_GET(&media_state->sessions, index);
333
334 ao2_bump(session_media);
335 if (AST_VECTOR_REPLACE(&cloned->sessions, index, session_media)) {
336 ao2_cleanup(session_media);
337 }
339 !cloned->default_session[type]) {
340 cloned->default_session[type] = session_media;
341 }
342 }
343
344 for (index = 0; index < AST_VECTOR_SIZE(&media_state->read_callbacks); ++index) {
346
348 }
349
350 return cloned;
351}
352
354{
355 if (!media_state) {
356 return;
357 }
358
359 /* This will reset the internal state so we only have to free persistent things */
361
362 AST_VECTOR_FREE(&media_state->sessions);
363 AST_VECTOR_FREE(&media_state->read_callbacks);
364
365 ast_free(media_state);
366}
367
369{
370 int index;
371
372 if (!session->pending_media_state->topology) {
373 ast_log(LOG_WARNING, "Pending topology was NULL for channel '%s'\n",
374 session->channel ? ast_channel_name(session->channel) : "unknown");
375 return 0;
376 }
377
379 return 0;
380 }
381
382 for (index = 0; index < ast_stream_topology_get_count(session->pending_media_state->topology); ++index) {
383 if (ast_stream_get_type(ast_stream_topology_get_stream(session->pending_media_state->topology, index)) !=
384 ast_stream_get_type(stream)) {
385 continue;
386 }
387
388 return ast_stream_topology_get_stream(session->pending_media_state->topology, index) == stream ? 1 : 0;
389 }
390
391 return 0;
392}
393
396{
397 struct ast_sip_session_media_read_callback_state callback_state = {
398 .fd = fd,
399 .read_callback = callback,
400 .session = session_media,
401 };
402
403 /* The contents of the vector are whole structs and not pointers */
404 return AST_VECTOR_APPEND(&session->pending_media_state->read_callbacks, callback_state);
405}
406
409{
410 if (session_media->write_callback) {
411 if (session_media->write_callback == callback) {
412 return 0;
413 }
414
415 return -1;
416 }
417
418 session_media->write_callback = callback;
419
420 return 0;
421}
422
424{
425 int index;
426
427 if (!session->endpoint->media.bundle || ast_strlen_zero(session_media->mid)) {
428 return session_media;
429 }
430
431 for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
432 struct ast_sip_session_media *bundle_group_session_media;
433
434 bundle_group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
435
436 /* The first session which is in the bundle group is considered the authoritative session for transport */
437 if (bundle_group_session_media->bundle_group == session_media->bundle_group) {
438 return bundle_group_session_media;
439 }
440 }
441
442 return session_media;
443}
444
445/*!
446 * \brief Set an SDP stream handler for a corresponding session media.
447 *
448 * \note Always use this function to set the SDP handler for a session media.
449 *
450 * This function will properly free resources on the SDP handler currently being
451 * used by the session media, then set the session media to use the new SDP
452 * handler.
453 */
454static void session_media_set_handler(struct ast_sip_session_media *session_media,
456{
457 ast_assert(session_media->handler != handler);
458
459 if (session_media->handler) {
460 session_media->handler->stream_destroy(session_media);
461 }
462 session_media->handler = handler;
463}
464
465static int stream_destroy(void *obj, void *arg, int flags)
466{
467 struct sdp_handler_list *handler_list = obj;
468 struct ast_sip_session_media *session_media = arg;
470
471 AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
472 handler->stream_destroy(session_media);
473 }
474
475 return 0;
476}
477
478static void session_media_dtor(void *obj)
479{
480 struct ast_sip_session_media *session_media = obj;
481
482 /* It is possible for multiple handlers to have allocated memory on the
483 * session media (usually through a stream changing types). Therefore, we
484 * traverse all the SDP handlers and let them all call stream_destroy on
485 * the session_media
486 */
487 ao2_callback(sdp_handlers, 0, stream_destroy, session_media);
488
489 if (session_media->srtp) {
490 ast_sdp_srtp_destroy(session_media->srtp);
491 }
492
493 ast_free(session_media->mid);
494 ast_free(session_media->remote_mslabel);
495 ast_free(session_media->remote_label);
496 ast_free(session_media->stream_name);
497}
498
500 struct ast_sip_session_media_state *media_state, enum ast_media_type type, int position)
501{
502 struct ast_sip_session_media *session_media = NULL;
503 struct ast_sip_session_media *current_session_media = NULL;
504 SCOPE_ENTER(1, "%s Adding position %d\n", ast_sip_session_get_name(session), position);
505
506 /* It is possible for this media state to already contain a session for the stream. If this
507 * is the case we simply return it.
508 */
509 if (position < AST_VECTOR_SIZE(&media_state->sessions)) {
510 current_session_media = AST_VECTOR_GET(&media_state->sessions, position);
511 if (current_session_media && current_session_media->type == type) {
512 SCOPE_EXIT_RTN_VALUE(current_session_media, "Using existing media_session\n");
513 }
514 }
515
516 /* Determine if we can reuse the session media from the active media state if present */
517 if (position < AST_VECTOR_SIZE(&session->active_media_state->sessions)) {
518 session_media = AST_VECTOR_GET(&session->active_media_state->sessions, position);
519 /* A stream can never exist without an accompanying media session */
520 if (session_media->type == type) {
521 ao2_ref(session_media, +1);
522 ast_trace(1, "Reusing existing media session\n");
523 /*
524 * If this session_media was previously removed, its bundle group was probably reset
525 * to -1 so if bundling is enabled on the endpoint, we need to reset it to 0, set
526 * the bundled flag and reset its mid.
527 */
528 if (session->endpoint->media.bundle && session_media->bundle_group == -1) {
529 session_media->bundled = session->endpoint->media.webrtc;
530 session_media->bundle_group = 0;
531 ast_free(session_media->mid);
532 if (ast_asprintf(&session_media->mid, "%s-%d", ast_codec_media_type2str(type), position) < 0) {
533 ao2_ref(session_media, -1);
534 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't alloc mid\n");
535 }
536 }
537 } else {
538 ast_trace(1, "Can't reuse existing media session because the types are different. %s <> %s\n",
540 session_media = NULL;
541 }
542 }
543
544 if (!session_media) {
545 /* No existing media session we can use so create a new one */
546 session_media = ao2_alloc_options(sizeof(*session_media), session_media_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
547 if (!session_media) {
548 return NULL;
549 }
550 ast_trace(1, "Creating new media session\n");
551
552 session_media->encryption = session->endpoint->media.rtp.encryption;
553 session_media->remote_ice = session->endpoint->media.rtp.ice_support;
554 session_media->remote_rtcp_mux = session->endpoint->media.rtcp_mux;
555 session_media->keepalive_sched_id = -1;
556 session_media->timeout_sched_id = -1;
557 session_media->type = type;
558 session_media->stream_num = position;
559
560 if (session->endpoint->media.bundle) {
561 /* This is a new stream so create a new mid based on media type and position, which makes it unique.
562 * If this is the result of an offer the mid will just end up getting replaced.
563 */
564 if (ast_asprintf(&session_media->mid, "%s-%d", ast_codec_media_type2str(type), position) < 0) {
565 ao2_ref(session_media, -1);
566 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't alloc mid\n");
567 }
568 session_media->bundle_group = 0;
569
570 /* Some WebRTC clients can't handle an offer to bundle media streams. Instead they expect them to
571 * already be bundled. Every client handles this scenario though so if WebRTC is enabled just go
572 * ahead and treat the streams as having already been bundled.
573 */
574 session_media->bundled = session->endpoint->media.webrtc;
575 } else {
576 session_media->bundle_group = -1;
577 }
578 }
579
580 ast_free(session_media->stream_name);
581 session_media->stream_name = ast_strdup(ast_stream_get_name(ast_stream_topology_get_stream(media_state->topology, position)));
582
583 if (AST_VECTOR_REPLACE(&media_state->sessions, position, session_media)) {
584 ao2_ref(session_media, -1);
585
586 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't replace media_session\n");
587 }
588
589 ao2_cleanup(current_session_media);
590
591 /* If this stream will be active in some way and it is the first of this type then consider this the default media session to match */
593 ast_trace(1, "Setting media session as default for %s\n", ast_codec_media_type2str(session_media->type));
594
595 media_state->default_session[type] = session_media;
596 }
597
598 SCOPE_EXIT_RTN_VALUE(session_media, "Done\n");
599}
600
601static int is_stream_limitation_reached(enum ast_media_type type, const struct ast_sip_endpoint *endpoint, int *type_streams)
602{
603 switch (type) {
605 return !(type_streams[type] < endpoint->media.max_audio_streams);
607 return !(type_streams[type] < endpoint->media.max_video_streams);
609 /* We don't have an option for image (T.38) streams so cap it to one. */
610 return (type_streams[type] > 0);
613 default:
614 /* We don't want any unknown or "other" streams on our endpoint,
615 * so always just say we've reached the limit
616 */
617 return 1;
618 }
619}
620
621static int get_mid_bundle_group(const pjmedia_sdp_session *sdp, const char *mid)
622{
623 int bundle_group = 0;
624 int index;
625
626 for (index = 0; index < sdp->attr_count; ++index) {
627 pjmedia_sdp_attr *attr = sdp->attr[index];
628 char value[pj_strlen(&attr->value) + 1], *mids = value, *attr_mid;
629
630 if (pj_strcmp2(&attr->name, "group") || pj_strncmp2(&attr->value, "BUNDLE", 6)) {
631 continue;
632 }
633
634 ast_copy_pj_str(value, &attr->value, sizeof(value));
635
636 /* Skip the BUNDLE at the front */
637 mids += 7;
638
639 while ((attr_mid = strsep(&mids, " "))) {
640 if (!strcmp(attr_mid, mid)) {
641 /* The ordering of attributes determines our internal identification of the bundle group based on number,
642 * with -1 being not in a bundle group. Since this is only exposed internally for response purposes it's
643 * actually even fine if things move around.
644 */
645 return bundle_group;
646 }
647 }
648
649 bundle_group++;
650 }
651
652 return -1;
653}
654
656 struct ast_sip_session_media *session_media,
657 const pjmedia_sdp_session *sdp,
658 const struct pjmedia_sdp_media *stream)
659{
660 pjmedia_sdp_attr *attr;
661
662 if (!session->endpoint->media.bundle) {
663 return 0;
664 }
665
666 /* By default on an incoming negotiation we assume no mid and bundle group is present */
667 ast_free(session_media->mid);
668 session_media->mid = NULL;
669 session_media->bundle_group = -1;
670 session_media->bundled = 0;
671
672 /* Grab the media identifier for the stream */
673 attr = pjmedia_sdp_media_find_attr2(stream, "mid", NULL);
674 if (!attr) {
675 return 0;
676 }
677
678 session_media->mid = ast_calloc(1, attr->value.slen + 1);
679 if (!session_media->mid) {
680 return 0;
681 }
682 ast_copy_pj_str(session_media->mid, &attr->value, attr->value.slen + 1);
683
684 /* Determine what bundle group this is part of */
685 session_media->bundle_group = get_mid_bundle_group(sdp, session_media->mid);
686
687 /* If this is actually part of a bundle group then the other side requested or accepted the bundle request */
688 session_media->bundled = session_media->bundle_group != -1;
689
690 return 0;
691}
692
694 struct ast_sip_session_media *session_media,
695 const pjmedia_sdp_session *sdp,
696 const struct pjmedia_sdp_media *stream,
697 struct ast_stream *asterisk_stream)
698{
699 int index;
700
701 ast_free(session_media->remote_mslabel);
702 session_media->remote_mslabel = NULL;
703 ast_free(session_media->remote_label);
704 session_media->remote_label = NULL;
705
706 for (index = 0; index < stream->attr_count; ++index) {
707 pjmedia_sdp_attr *attr = stream->attr[index];
708 char attr_value[pj_strlen(&attr->value) + 1];
709 char *ssrc_attribute_name, *ssrc_attribute_value = NULL;
710 char *msid, *tmp = attr_value;
711 static const pj_str_t STR_msid = { "msid", 4 };
712 static const pj_str_t STR_ssrc = { "ssrc", 4 };
713 static const pj_str_t STR_label = { "label", 5 };
714
715 if (!pj_strcmp(&attr->name, &STR_label)) {
716 ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
717 session_media->remote_label = ast_strdup(attr_value);
718 } else if (!pj_strcmp(&attr->name, &STR_msid)) {
719 ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
720 msid = strsep(&tmp, " ");
721 session_media->remote_mslabel = ast_strdup(msid);
722 break;
723 } else if (!pj_strcmp(&attr->name, &STR_ssrc)) {
724 ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
725
726 if ((ssrc_attribute_name = strchr(attr_value, ' '))) {
727 /* This has an actual attribute */
728 *ssrc_attribute_name++ = '\0';
729 ssrc_attribute_value = strchr(ssrc_attribute_name, ':');
730 if (ssrc_attribute_value) {
731 /* Values are actually optional according to the spec */
732 *ssrc_attribute_value++ = '\0';
733 }
734
735 if (!strcasecmp(ssrc_attribute_name, "mslabel") && !ast_strlen_zero(ssrc_attribute_value)) {
736 session_media->remote_mslabel = ast_strdup(ssrc_attribute_value);
737 break;
738 }
739 }
740 }
741 }
742
743 if (ast_strlen_zero(session_media->remote_mslabel)) {
744 return;
745 }
746
747 /* Iterate through the existing streams looking for a match and if so then group this with it */
748 for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
749 struct ast_sip_session_media *group_session_media;
750
751 group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
752
753 if (ast_strlen_zero(group_session_media->remote_mslabel) ||
754 strcmp(group_session_media->remote_mslabel, session_media->remote_mslabel)) {
755 continue;
756 }
757
758 ast_stream_set_group(asterisk_stream, index);
759 break;
760 }
761}
762
763static void remove_stream_from_bundle(struct ast_sip_session_media *session_media,
764 struct ast_stream *stream)
765{
767 ast_free(session_media->mid);
768 session_media->mid = NULL;
769 session_media->bundle_group = -1;
770 session_media->bundled = 0;
771}
772
773static int handle_incoming_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
774{
775 int i;
776 int handled = 0;
777 int type_streams[AST_MEDIA_TYPE_END] = {0};
778 SCOPE_ENTER(3, "%s: Media count: %d\n", ast_sip_session_get_name(session), sdp->media_count);
779
780 if (session->inv_session && session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
781 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to handle incoming SDP. Session has been already disconnected\n",
783 }
784
785 /* It is possible for SDP deferral to have already created a pending topology */
786 if (!session->pending_media_state->topology) {
787 session->pending_media_state->topology = ast_stream_topology_alloc();
788 if (!session->pending_media_state->topology) {
789 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc pending topology\n",
791 }
792 }
793
794 for (i = 0; i < sdp->media_count; ++i) {
795 /* See if there are registered handlers for this media stream type */
796 char media[20];
798 RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
799 struct ast_sip_session_media *session_media = NULL;
800 int res;
801 enum ast_media_type type;
802 struct ast_stream *stream = NULL;
803 pjmedia_sdp_media *remote_stream = sdp->media[i];
804 SCOPE_ENTER(4, "%s: Processing stream %d\n", ast_sip_session_get_name(session), i);
805
806 /* We need a null-terminated version of the media string */
807 ast_copy_pj_str(media, &sdp->media[i]->desc.media, sizeof(media));
809
810 /* See if we have an already existing stream, which can occur from SDP deferral checking */
811 if (i < ast_stream_topology_get_count(session->pending_media_state->topology)) {
812 stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
813 ast_trace(-1, "%s: Using existing pending stream %s\n", ast_sip_session_get_name(session),
814 ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
815 }
816 if (!stream) {
817 struct ast_stream *existing_stream = NULL;
818 char *stream_name = NULL, *stream_name_allocated = NULL;
819 const char *stream_label = NULL;
820
821 if (session->active_media_state->topology &&
822 (i < ast_stream_topology_get_count(session->active_media_state->topology))) {
823 existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, i);
824 ast_trace(-1, "%s: Found existing active stream %s\n", ast_sip_session_get_name(session),
825 ast_str_tmp(128, ast_stream_to_str(existing_stream, &STR_TMP)));
826
827 if (ast_stream_get_state(existing_stream) != AST_STREAM_STATE_REMOVED) {
828 stream_name = (char *)ast_stream_get_name(existing_stream);
829 stream_label = ast_stream_get_metadata(existing_stream, "SDP:LABEL");
830 }
831 }
832
833 if (ast_strlen_zero(stream_name)) {
834 if (ast_asprintf(&stream_name_allocated, "%s-%d", ast_codec_media_type2str(type), i) < 0) {
835 handled = 0;
836 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't alloc stream name\n",
838
839 }
840 stream_name = stream_name_allocated;
841 ast_trace(-1, "%s: Using %s for new stream name\n", ast_sip_session_get_name(session),
842 stream_name);
843 }
844
845 stream = ast_stream_alloc(stream_name, type);
846 ast_free(stream_name_allocated);
847 if (!stream) {
848 handled = 0;
849 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't alloc stream\n",
851 }
852
853 if (!ast_strlen_zero(stream_label)) {
854 ast_stream_set_metadata(stream, "SDP:LABEL", stream_label);
855 ast_trace(-1, "%s: Using %s for new stream label\n", ast_sip_session_get_name(session),
856 stream_label);
857
858 }
859
860 if (ast_stream_topology_set_stream(session->pending_media_state->topology, i, stream)) {
861 ast_stream_free(stream);
862 handled = 0;
863 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't set stream in topology\n",
865 }
866
867 /* For backwards compatibility with the core the default audio stream is always sendrecv */
868 if (!ast_sip_session_is_pending_stream_default(session, stream) || strcmp(media, "audio")) {
869 if (pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
870 /* Stream state reflects our state of a stream, so in the case of
871 * sendonly and recvonly we store the opposite since that is what ours
872 * is.
873 */
875 } else if (pjmedia_sdp_media_find_attr2(remote_stream, "recvonly", NULL)) {
877 } else if (pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
879 } else {
881 }
882 } else {
884 }
885 ast_trace(-1, "%s: Using new stream %s\n", ast_sip_session_get_name(session),
886 ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
887 }
888
889 session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_media_type_from_str(media), i);
890 if (!session_media) {
891 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't alloc session media\n",
893 }
894
895 /* If this stream is already declined mark it as such, or mark it as such if we've reached the limit */
896 if (!remote_stream->desc.port || is_stream_limitation_reached(type, session->endpoint, type_streams)) {
897 remove_stream_from_bundle(session_media, stream);
898 SCOPE_EXIT_EXPR(continue, "%s: Declining incoming SDP media stream %s'\n",
900 }
901
902 set_mid_and_bundle_group(session, session_media, sdp, remote_stream);
903 set_remote_mslabel_and_stream_group(session, session_media, sdp, remote_stream, stream);
904
905 if (session_media->handler) {
906 handler = session_media->handler;
907 ast_trace(-1, "%s: Negotiating incoming SDP media stream %s using %s SDP handler\n",
909 session_media->handler->id);
910 res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
911 if (res < 0) {
912 /* Catastrophic failure. Abort! */
913 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't negotiate stream %s\n",
915 } else if (res == 0) {
916 remove_stream_from_bundle(session_media, stream);
917 SCOPE_EXIT_EXPR(continue, "%s: Declining incoming SDP media stream %s\n",
919 } else if (res > 0) {
920 handled = 1;
921 ++type_streams[type];
922 /* Handled by this handler. Move to the next stream */
923 SCOPE_EXIT_EXPR(continue, "%s: Media stream %s handled by %s\n",
925 session_media->handler->id);
926 }
927 }
928
929 handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
930 if (!handler_list) {
931 SCOPE_EXIT_EXPR(continue, "%s: Media stream %s has no registered handlers\n",
933 }
934 AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
935 if (handler == session_media->handler) {
936 continue;
937 }
938 ast_trace(-1, "%s: Negotiating incoming SDP media stream %s using %s SDP handler\n",
940 handler->id);
941
942 res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
943 if (res < 0) {
944 /* Catastrophic failure. Abort! */
945 handled = 0;
946 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't negotiate stream %s\n",
948 } else if (res == 0) {
949 remove_stream_from_bundle(session_media, stream);
950 ast_trace(-1, "%s: Declining incoming SDP media stream %s\n",
952 continue;
953 } else if (res > 0) {
954 session_media_set_handler(session_media, handler);
955 handled = 1;
956 ++type_streams[type];
957 ast_trace(-1, "%s: Media stream %s handled by %s\n",
959 session_media->handler->id);
960 break;
961 }
962 }
963
964 SCOPE_EXIT("%s: Done with stream %s\n", ast_sip_session_get_name(session),
965 ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
966 }
967
968end:
969 SCOPE_EXIT_RTN_VALUE(handled ? 0 : -1, "%s: Handled? %s\n", ast_sip_session_get_name(session),
970 handled ? "yes" : "no");
971}
972
974 struct ast_sip_session *session, const pjmedia_sdp_session *local,
975 const pjmedia_sdp_session *remote, int index, struct ast_stream *asterisk_stream)
976{
977 /* See if there are registered handlers for this media stream type */
978 struct pjmedia_sdp_media *local_stream = local->media[index];
979 char media[20];
981 RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
982 int res;
983 SCOPE_ENTER(1, "%s\n", session ? ast_sip_session_get_name(session) : "unknown");
984
985 /* We need a null-terminated version of the media string */
986 ast_copy_pj_str(media, &local->media[index]->desc.media, sizeof(media));
987
988 /* For backwards compatibility we only reflect the stream state correctly on
989 * the non-default streams and any non-audio streams. This is because the stream
990 * state of the default audio stream is also used for signaling that someone has
991 * placed us on hold. This situation is not handled currently and can result in
992 * the remote side being sorted of placed on hold too.
993 */
994 if (!ast_sip_session_is_pending_stream_default(session, asterisk_stream) || strcmp(media, "audio")) {
995 /* Determine the state of the stream based on our local SDP */
996 if (pjmedia_sdp_media_find_attr2(local_stream, "sendonly", NULL)) {
998 } else if (pjmedia_sdp_media_find_attr2(local_stream, "recvonly", NULL)) {
1000 } else if (pjmedia_sdp_media_find_attr2(local_stream, "inactive", NULL)) {
1002 } else {
1004 }
1005 } else {
1007 }
1008
1009 set_mid_and_bundle_group(session, session_media, remote, remote->media[index]);
1010 set_remote_mslabel_and_stream_group(session, session_media, remote, remote->media[index], asterisk_stream);
1011
1012 handler = session_media->handler;
1013 if (handler) {
1014 ast_debug(4, "%s: Applying negotiated SDP media stream '%s' using %s SDP handler\n",
1016 handler->id);
1017 res = handler->apply_negotiated_sdp_stream(session, session_media, local, remote, index, asterisk_stream);
1018 if (res >= 0) {
1019 ast_debug(4, "%s: Applied negotiated SDP media stream '%s' using %s SDP handler\n",
1021 handler->id);
1022 SCOPE_EXIT_RTN_VALUE(0, "%s: Applied negotiated SDP media stream '%s' using %s SDP handler\n",
1024 handler->id);
1025 }
1026 SCOPE_EXIT_RTN_VALUE(-1, "%s: Failed to apply negotiated SDP media stream '%s' using %s SDP handler\n",
1028 handler->id);
1029 }
1030
1031 handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
1032 if (!handler_list) {
1033 ast_debug(4, "%s: No registered SDP handlers for media type '%s'\n", ast_sip_session_get_name(session), media);
1034 return -1;
1035 }
1036 AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
1037 if (handler == session_media->handler) {
1038 continue;
1039 }
1040 ast_debug(4, "%s: Applying negotiated SDP media stream '%s' using %s SDP handler\n",
1042 handler->id);
1043 res = handler->apply_negotiated_sdp_stream(session, session_media, local, remote, index, asterisk_stream);
1044 if (res < 0) {
1045 /* Catastrophic failure. Abort! */
1046 SCOPE_EXIT_RTN_VALUE(-1, "%s: Handler '%s' returned %d\n",
1048 }
1049 if (res > 0) {
1050 ast_debug(4, "%s: Applied negotiated SDP media stream '%s' using %s SDP handler\n",
1052 handler->id);
1053 /* Handled by this handler. Move to the next stream */
1054 session_media_set_handler(session_media, handler);
1055 SCOPE_EXIT_RTN_VALUE(0, "%s: Handler '%s' handled this sdp stream\n",
1057 }
1058 }
1059
1060 res = 0;
1061 if (session_media->handler && session_media->handler->stream_stop) {
1062 ast_debug(4, "%s: Stopping SDP media stream '%s' as it is not currently negotiated\n",
1064 session_media->handler->stream_stop(session_media);
1065 }
1066
1067 SCOPE_EXIT_RTN_VALUE(0, "%s: Media type '%s' %s\n",
1069 res ? "not negotiated. Stopped" : "handled");
1070}
1071
1072static int handle_negotiated_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *local, const pjmedia_sdp_session *remote)
1073{
1074 int i;
1075 struct ast_stream_topology *topology;
1076 unsigned int changed = 0; /* 0 = unchanged, 1 = new source, 2 = new topology */
1078
1079 if (!session->pending_media_state->topology) {
1080 if (session->active_media_state->topology) {
1081 /*
1082 * This happens when we have negotiated media after receiving a 183,
1083 * and we're now receiving a 200 with a new SDP. In this case, there
1084 * is active_media_state, but the pending_media_state has been reset.
1085 */
1086 struct ast_sip_session_media_state *active_media_state_clone;
1087
1088 active_media_state_clone =
1089 ast_sip_session_media_state_clone(session->active_media_state);
1090 if (!active_media_state_clone) {
1091 ast_log(LOG_WARNING, "%s: Unable to clone active media state\n",
1093 return -1;
1094 }
1095
1096 ast_sip_session_media_state_free(session->pending_media_state);
1097 session->pending_media_state = active_media_state_clone;
1098 } else {
1099 ast_log(LOG_WARNING, "%s: No pending or active media state\n",
1101 return -1;
1102 }
1103 }
1104
1105 /* If we're handling negotiated streams, then we should already have set
1106 * up session media instances (and Asterisk streams) that correspond to
1107 * the local SDP, and there should be the same number of session medias
1108 * and streams as there are local SDP streams
1109 */
1110 if (ast_stream_topology_get_count(session->pending_media_state->topology) != local->media_count
1111 || AST_VECTOR_SIZE(&session->pending_media_state->sessions) != local->media_count) {
1112 ast_log(LOG_WARNING, "%s: Local SDP contains %d media streams while we expected it to contain %u\n",
1114 ast_stream_topology_get_count(session->pending_media_state->topology), local->media_count);
1115 SCOPE_EXIT_RTN_VALUE(-1, "Media stream count mismatch\n");
1116 }
1117
1118 AST_VECTOR_RESET(&session->pending_media_state->read_callbacks, AST_VECTOR_ELEM_CLEANUP_NOOP);
1119
1120 for (i = 0; i < local->media_count; ++i) {
1121 struct ast_sip_session_media *session_media;
1122 struct ast_stream *stream;
1123
1124 if (!remote->media[i]) {
1125 continue;
1126 }
1127
1128 session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, i);
1129 stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
1130
1131 /* Make sure that this stream is in the correct state. If we need to change
1132 * the state to REMOVED, then our work here is done, so go ahead and move on
1133 * to the next stream.
1134 */
1135 if (!remote->media[i]->desc.port) {
1137 continue;
1138 }
1139
1140 /* If the stream state is REMOVED, nothing needs to be done, so move on to the
1141 * next stream. This can occur if an internal thing has requested it to be
1142 * removed, or if we remove it as a result of the stream limit being reached.
1143 */
1145 /*
1146 * Defer removing the handler until we are ready to activate
1147 * the new topology. The channel's thread may still be using
1148 * the stream and we could crash before we are ready.
1149 */
1150 continue;
1151 }
1152
1153 if (handle_negotiated_sdp_session_media(session_media, session, local, remote, i, stream)) {
1154 SCOPE_EXIT_RTN_VALUE(-1, "Unable to handle negotiated session media\n");
1155 }
1156
1157 changed |= session_media->changed;
1158 session_media->changed = 0;
1159 }
1160
1161 /* Apply the pending media state to the channel and make it active */
1162 ast_channel_lock(session->channel);
1163
1164 /* Now update the stream handler for any declined/removed streams */
1165 for (i = 0; i < local->media_count; ++i) {
1166 struct ast_sip_session_media *session_media;
1167 struct ast_stream *stream;
1168
1169 if (!remote->media[i]) {
1170 continue;
1171 }
1172
1173 session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, i);
1174 stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
1175
1177 && session_media->handler) {
1178 /*
1179 * This stream is no longer being used and the channel's thread
1180 * is held off because we have the channel lock so release any
1181 * resources the handler may have on it.
1182 */
1183 session_media_set_handler(session_media, NULL);
1184 }
1185 }
1186
1187 /* Update the topology on the channel to match the accepted one */
1188 topology = ast_stream_topology_clone(session->pending_media_state->topology);
1189 if (topology) {
1190 ast_channel_set_stream_topology(session->channel, topology);
1191 /* If this is a remotely done renegotiation that has changed the stream topology notify what is
1192 * currently handling this channel. Note that fax uses its own process, so if we are transitioning
1193 * between audio and fax or vice versa we don't notify.
1194 */
1195 if (pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE &&
1196 session->active_media_state && session->active_media_state->topology &&
1197 !ast_stream_topology_equal(session->active_media_state->topology, topology) &&
1198 !session->active_media_state->default_session[AST_MEDIA_TYPE_IMAGE] &&
1199 !session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE]) {
1200 changed = 2;
1201 }
1202 }
1203
1204 /* Remove all current file descriptors from the channel */
1205 for (i = 0; i < AST_VECTOR_SIZE(&session->active_media_state->read_callbacks); ++i) {
1207 }
1208
1209 /* Add all the file descriptors from the pending media state */
1210 for (i = 0; i < AST_VECTOR_SIZE(&session->pending_media_state->read_callbacks); ++i) {
1211 struct ast_sip_session_media_read_callback_state *callback_state;
1212
1213 callback_state = AST_VECTOR_GET_ADDR(&session->pending_media_state->read_callbacks, i);
1214 ast_channel_internal_fd_set(session->channel, i + AST_EXTENDED_FDS, callback_state->fd);
1215 }
1216
1217 /* Active and pending flip flop as needed */
1218 ast_sip_session_media_stats_save(session, session->active_media_state);
1219 SWAP(session->active_media_state, session->pending_media_state);
1220 ast_sip_session_media_state_reset(session->pending_media_state);
1221
1222 ast_channel_unlock(session->channel);
1223
1224 if (changed == 1) {
1226
1227 ast_queue_frame(session->channel, &f);
1228 } else if (changed == 2) {
1230 } else {
1232 }
1233
1235}
1236
1237#define DATASTORE_BUCKETS 53
1238#define MEDIA_BUCKETS 7
1239
1240static void session_datastore_destroy(void *obj)
1241{
1242 struct ast_datastore *datastore = obj;
1243
1244 /* Using the destroy function (if present) destroy the data */
1245 if (datastore->info->destroy != NULL && datastore->data != NULL) {
1246 datastore->info->destroy(datastore->data);
1247 datastore->data = NULL;
1248 }
1249
1250 ast_free((void *) datastore->uid);
1251 datastore->uid = NULL;
1252}
1253
1255{
1256 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
1257 char uuid_buf[AST_UUID_STR_LEN];
1258 const char *uid_ptr = uid;
1259
1260 if (!info) {
1261 return NULL;
1262 }
1263
1264 datastore = ao2_alloc(sizeof(*datastore), session_datastore_destroy);
1265 if (!datastore) {
1266 return NULL;
1267 }
1268
1269 datastore->info = info;
1270 if (ast_strlen_zero(uid)) {
1271 /* They didn't provide an ID so we'll provide one ourself */
1272 uid_ptr = ast_uuid_generate_str(uuid_buf, sizeof(uuid_buf));
1273 }
1274
1275 datastore->uid = ast_strdup(uid_ptr);
1276 if (!datastore->uid) {
1277 return NULL;
1278 }
1279
1280 ao2_ref(datastore, +1);
1281 return datastore;
1282}
1283
1285{
1286 ast_assert(datastore != NULL);
1287 ast_assert(datastore->info != NULL);
1288 ast_assert(ast_strlen_zero(datastore->uid) == 0);
1289
1290 if (!ao2_link(session->datastores, datastore)) {
1291 return -1;
1292 }
1293 return 0;
1294}
1295
1297{
1298 return ao2_find(session->datastores, name, OBJ_KEY);
1299}
1300
1302{
1303 ao2_callback(session->datastores, OBJ_KEY | OBJ_UNLINK | OBJ_NODATA, NULL, (void *) name);
1304}
1305
1311
1312/*!
1313 * \internal
1314 * \brief Convert delayed method enum value to a string.
1315 * \since 13.3.0
1316 *
1317 * \param method Delayed method enum value to convert to a string.
1318 *
1319 * \return String value of delayed method.
1320 */
1322{
1323 const char *str = "<unknown>";
1324
1325 switch (method) {
1327 str = "INVITE";
1328 break;
1330 str = "UPDATE";
1331 break;
1332 case DELAYED_METHOD_BYE:
1333 str = "BYE";
1334 break;
1335 }
1336
1337 return str;
1338}
1339
1340/*!
1341 * \brief Structure used for sending delayed requests
1342 *
1343 * Requests are typically delayed because the current transaction
1344 * state of an INVITE. Once the pending INVITE transaction terminates,
1345 * the delayed request will be sent
1346 */
1348 /*! Method of the request */
1350 /*! Callback to call when the delayed request is created. */
1352 /*! Callback to call when the delayed request SDP is created */
1354 /*! Callback to call when the delayed request receives a response */
1356 /*! Whether to generate new SDP */
1358 /*! Requested media state for the SDP */
1360 /*! Active media state at the time of the original request */
1362
1364};
1365
1389
1396
1397/*
1398 * Delayed requests own pending/active media states. Keep cleanup centralized so
1399 * timeout-driven teardown releases the same state as normal session teardown.
1400 */
1402{
1403 struct ast_sip_session_delayed_request *delay;
1404
1405 while ((delay = AST_LIST_REMOVE_HEAD(&session->delayed_requests, next))) {
1406 delayed_request_free(delay);
1407 }
1408}
1409
1410/*!
1411 * \internal
1412 * \brief Send a delayed request
1413 *
1414 * \retval -1 failure
1415 * \retval 0 success
1416 * \retval 1 refresh request not sent as no change would occur
1417 */
1419{
1420 int res;
1421 SCOPE_ENTER(3, "%s: sending delayed %s request\n",
1423 delayed_method2str(delay->method));
1424
1425 switch (delay->method) {
1428 delay->on_sdp_creation, delay->on_response,
1430 delay->active_media_state, 1);
1431 /* Ownership of media state transitions to ast_sip_session_refresh */
1432 delay->pending_media_state = NULL;
1433 delay->active_media_state = NULL;
1437 delay->on_sdp_creation, delay->on_response,
1439 delay->active_media_state, 1);
1440 /* Ownership of media state transitions to ast_sip_session_refresh */
1441 delay->pending_media_state = NULL;
1442 delay->active_media_state = NULL;
1444 case DELAYED_METHOD_BYE:
1445 /* The delayed BYE is being sent now; timeout cleanup is no longer needed. */
1446 session->terminate_on_invite_timeout = 0;
1448 SCOPE_EXIT_RTN_VALUE(0, "%s: Terminating session on delayed BYE\n", ast_sip_session_get_name(session));
1449 }
1450
1451 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Don't know how to send delayed %s(%d) request.\n",
1453 delayed_method2str(delay->method), delay->method);
1454}
1455
1456/*!
1457 * \internal
1458 * \brief The current INVITE transaction is in the PROCEEDING state.
1459 * \since 13.3.0
1460 *
1461 * \param vsession Session object.
1462 *
1463 * \retval 0 on success.
1464 * \retval -1 on error.
1465 */
1466static int invite_proceeding(void *vsession)
1467{
1468 struct ast_sip_session *session = vsession;
1469 struct ast_sip_session_delayed_request *delay;
1470 int found = 0;
1471 int res = 0;
1473
1474 AST_LIST_TRAVERSE_SAFE_BEGIN(&session->delayed_requests, delay, next) {
1475 switch (delay->method) {
1477 break;
1480 ast_trace(-1, "%s: Sending delayed %s request\n", ast_sip_session_get_name(session),
1481 delayed_method2str(delay->method));
1482 res = send_delayed_request(session, delay);
1483 delayed_request_free(delay);
1484 if (!res) {
1485 found = 1;
1486 }
1487 break;
1488 case DELAYED_METHOD_BYE:
1489 /* A BYE is pending so don't bother anymore. */
1490 found = 1;
1491 break;
1492 }
1493 if (found) {
1494 break;
1495 }
1496 }
1498
1499 ao2_ref(session, -1);
1501}
1502
1503/*!
1504 * \internal
1505 * \brief The current INVITE transaction is in the TERMINATED state.
1506 * \since 13.3.0
1507 *
1508 * \param vsession Session object.
1509 *
1510 * \retval 0 on success.
1511 * \retval -1 on error.
1512 */
1513static int invite_terminated(void *vsession)
1514{
1515 struct ast_sip_session *session = vsession;
1516 struct ast_sip_session_delayed_request *delay;
1517 int found = 0;
1518 int res = 0;
1519 int timer_running;
1521
1522 /* re-INVITE collision timer running? */
1523 timer_running = pj_timer_entry_running(&session->rescheduled_reinvite);
1524
1525 AST_LIST_TRAVERSE_SAFE_BEGIN(&session->delayed_requests, delay, next) {
1526 switch (delay->method) {
1528 if (!timer_running) {
1529 found = 1;
1530 }
1531 break;
1533 case DELAYED_METHOD_BYE:
1534 found = 1;
1535 break;
1536 }
1537 if (found) {
1539 ast_trace(-1, "%s: Sending delayed %s request\n", ast_sip_session_get_name(session),
1540 delayed_method2str(delay->method));
1541 res = send_delayed_request(session, delay);
1542 delayed_request_free(delay);
1543 if (!res) {
1544 break;
1545 }
1546 }
1547 }
1549
1550 ao2_ref(session, -1);
1552}
1553
1554/*!
1555 * \internal
1556 * \brief INVITE collision timeout.
1557 * \since 13.3.0
1558 *
1559 * \param vsession Session object.
1560 *
1561 * \retval 0 on success.
1562 * \retval -1 on error.
1563 */
1564static int invite_collision_timeout(void *vsession)
1565{
1566 struct ast_sip_session *session = vsession;
1567 int res;
1569
1570 if (session->inv_session->invite_tsx) {
1571 /*
1572 * INVITE transaction still active. Let it send
1573 * the collision re-INVITE when it terminates.
1574 */
1575 ao2_ref(session, -1);
1576 res = 0;
1577 } else {
1579 }
1580
1582}
1583
1584/*!
1585 * \internal
1586 * \brief The current UPDATE transaction is in the COMPLETED state.
1587 * \since 13.3.0
1588 *
1589 * \param vsession Session object.
1590 *
1591 * \retval 0 on success.
1592 * \retval -1 on error.
1593 */
1594static int update_completed(void *vsession)
1595{
1596 struct ast_sip_session *session = vsession;
1597 int res;
1598
1599 if (session->inv_session->invite_tsx) {
1601 } else {
1603 }
1604
1605 return res;
1606}
1607
1609 int (*cb)(void *vsession))
1610{
1611 ao2_ref(session, +1);
1612 if (ast_sip_push_task(session->serializer, cb, session)) {
1613 ao2_ref(session, -1);
1614 }
1615}
1616
1619 ast_sip_session_sdp_creation_cb on_sdp_creation,
1620 ast_sip_session_response_cb on_response,
1621 int generate_new_sdp,
1625 int queue_head)
1626{
1631
1632 if (!delay) {
1635 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "Unable to allocate delay request\n");
1636 }
1637
1638 if (method == DELAYED_METHOD_BYE || queue_head) {
1639 /* Send BYE as early as possible */
1640 AST_LIST_INSERT_HEAD(&session->delayed_requests, delay, next);
1641 } else {
1642 AST_LIST_INSERT_TAIL(&session->delayed_requests, delay, next);
1643 }
1645}
1646
1647/*
1648 * A UAC re-INVITE that has received a provisional response may no longer have
1649 * Timer B running. If its final response is malformed and rejected before
1650 * transaction processing, invite_tsx can keep the session alive indefinitely.
1651 * Arm PJPROJECT's INVITE timeout so delayed BYE cleanup has a bounded wait.
1652 */
1654{
1655 pjsip_transaction *tsx;
1656 pj_status_t status;
1657
1658 if (!session->inv_session || !session->inv_session->invite_tsx) {
1659 return 0;
1660 }
1661
1662 tsx = session->inv_session->invite_tsx;
1663 if (tsx->role != PJSIP_ROLE_UAC || tsx->method.id != PJSIP_INVITE_METHOD
1664 || tsx->state >= PJSIP_TSX_STATE_COMPLETED) {
1665 return 0;
1666 }
1667
1668 status = pjsip_tsx_set_timeout(tsx, pjsip_cfg()->tsx.td);
1669 if (status != PJ_SUCCESS && status != PJ_EEXISTS) {
1670 char errmsg[PJ_ERR_MSG_SIZE];
1671
1672 pj_strerror(status, errmsg, sizeof(errmsg));
1673 ast_log(LOG_WARNING, "%s: Failed to set timeout on outstanding INVITE transaction: %s\n",
1675 return 0;
1676 }
1677
1678 return 1;
1679}
1680
1681/*
1682 * PJPROJECT treats 408/481 on in-dialog UAC requests as dialog terminating.
1683 * When that happens after our timeout, drop Asterisk's queued BYE instead of
1684 * sending a duplicate BYE.
1685 */
1686static int uac_invite_tsx_terminates_dialog(pjsip_transaction *tsx)
1687{
1688 if (tsx->role != PJSIP_ROLE_UAC || tsx->method.id != PJSIP_INVITE_METHOD
1689 || tsx->state < PJSIP_TSX_STATE_COMPLETED) {
1690 return 0;
1691 }
1692
1693 return tsx->status_code == PJSIP_SC_CALL_TSX_DOES_NOT_EXIST
1694 || (tsx->status_code == PJSIP_SC_REQUEST_TIMEOUT
1695 && !pjsip_cfg()->endpt.keep_inv_after_tsx_timeout);
1696}
1697
1698static pjmedia_sdp_session *generate_session_refresh_sdp(struct ast_sip_session *session)
1699{
1700 pjsip_inv_session *inv_session = session->inv_session;
1701 const pjmedia_sdp_session *previous_sdp = NULL;
1703
1704 if (inv_session->neg) {
1705 if (pjmedia_sdp_neg_was_answer_remote(inv_session->neg)) {
1706 pjmedia_sdp_neg_get_active_remote(inv_session->neg, &previous_sdp);
1707 } else {
1708 pjmedia_sdp_neg_get_active_local(inv_session->neg, &previous_sdp);
1709 }
1710 }
1711 SCOPE_EXIT_RTN_VALUE(create_local_sdp(inv_session, session, previous_sdp, 0));
1712}
1713
1715{
1716 struct ast_party_id effective_id;
1717 struct ast_party_id connected_id;
1718 pj_pool_t *dlg_pool;
1719 pjsip_fromto_hdr *dlg_info;
1720 pjsip_contact_hdr *dlg_contact;
1721 pjsip_name_addr *dlg_info_name_addr;
1722 pjsip_sip_uri *dlg_info_uri;
1723 pjsip_sip_uri *dlg_contact_uri;
1724 int restricted;
1725 const char *pjsip_from_domain;
1726
1727 if (!session->channel || session->saved_from_hdr) {
1728 return;
1729 }
1730
1731 /* We need to save off connected_id for RPID/PAI generation */
1732 ast_party_id_init(&connected_id);
1733 ast_channel_lock(session->channel);
1734 effective_id = ast_channel_connected_effective_id(session->channel);
1735 ast_party_id_copy(&connected_id, &effective_id);
1736 ast_channel_unlock(session->channel);
1737
1738 restricted =
1740
1741 /* Now set up dlg->local.info so pjsip can correctly generate From */
1742
1743 dlg_pool = session->inv_session->dlg->pool;
1744 dlg_info = session->inv_session->dlg->local.info;
1745 dlg_contact = session->inv_session->dlg->local.contact;
1746 dlg_info_name_addr = (pjsip_name_addr *) dlg_info->uri;
1747 dlg_info_uri = pjsip_uri_get_uri(dlg_info_name_addr);
1748 dlg_contact_uri = (pjsip_sip_uri*)pjsip_uri_get_uri(dlg_contact->uri);
1749
1750 if (session->endpoint->id.trust_outbound || !restricted) {
1751 ast_sip_modify_id_header(dlg_pool, dlg_info, &connected_id);
1752 if (ast_sip_get_use_callerid_contact() && ast_strlen_zero(session->endpoint->contact_user)) {
1753 pj_strdup2(dlg_pool, &dlg_contact_uri->user, S_COR(connected_id.number.valid, connected_id.number.str, ""));
1754 }
1755 }
1756
1757 ast_party_id_free(&connected_id);
1758
1759 if (!ast_strlen_zero(session->endpoint->fromuser)) {
1760 dlg_info_name_addr->display.ptr = NULL;
1761 dlg_info_name_addr->display.slen = 0;
1762 pj_strdup2(dlg_pool, &dlg_info_uri->user, session->endpoint->fromuser);
1763 }
1764
1765 if (!ast_strlen_zero(session->endpoint->fromdomain)) {
1766 pj_strdup2(dlg_pool, &dlg_info_uri->host, session->endpoint->fromdomain);
1767 }
1768
1769 /*
1770 * Channel variable for compatibility with chan_sip SIPFROMDOMAIN
1771 */
1772 ast_channel_lock(session->channel);
1773 pjsip_from_domain = pbx_builtin_getvar_helper(session->channel, "SIPFROMDOMAIN");
1774 if (!ast_strlen_zero(pjsip_from_domain)) {
1775 ast_debug(3, "%s: From header domain reset by channel variable SIPFROMDOMAIN (%s)\n",
1776 ast_sip_session_get_name(session), pjsip_from_domain);
1777 pj_strdup2(dlg_pool, &dlg_info_uri->host, pjsip_from_domain);
1778 }
1779 ast_channel_unlock(session->channel);
1780
1781 /* We need to save off the non-anonymized From for RPID/PAI generation (for domain) */
1782 session->saved_from_hdr = pjsip_hdr_clone(dlg_pool, dlg_info);
1783 ast_sip_add_usereqphone(session->endpoint, dlg_pool, session->saved_from_hdr->uri);
1784
1785 /* In chan_sip, fromuser and fromdomain trump restricted so we only
1786 * anonymize if they're not set.
1787 */
1788 if (restricted) {
1789 /* fromuser doesn't provide a display name so we always set it */
1790 pj_strdup2(dlg_pool, &dlg_info_name_addr->display, "Anonymous");
1791
1792 if (ast_strlen_zero(session->endpoint->fromuser)) {
1793 pj_strdup2(dlg_pool, &dlg_info_uri->user, "anonymous");
1794 }
1795
1796 if (ast_sip_get_use_callerid_contact() && ast_strlen_zero(session->endpoint->contact_user)) {
1797 pj_strdup2(dlg_pool, &dlg_contact_uri->user, "anonymous");
1798 }
1799
1800 if (ast_strlen_zero(session->endpoint->fromdomain)) {
1801 pj_strdup2(dlg_pool, &dlg_info_uri->host, "anonymous.invalid");
1802 }
1803 } else {
1804 ast_sip_add_usereqphone(session->endpoint, dlg_pool, dlg_info->uri);
1805 }
1806}
1807
1808/*
1809 * Helper macros for merging and validating media states
1810 */
1811#define STREAM_REMOVED(_stream) (ast_stream_get_state(_stream) == AST_STREAM_STATE_REMOVED)
1812#define STATE_REMOVED(_stream_state) (_stream_state == AST_STREAM_STATE_REMOVED)
1813#define STATE_NONE(_stream_state) (_stream_state == AST_STREAM_STATE_END)
1814#define GET_STREAM_SAFE(_topology, _i) (_i < ast_stream_topology_get_count(_topology) ? ast_stream_topology_get_stream(_topology, _i) : NULL)
1815#define GET_STREAM_STATE_SAFE(_stream) (_stream ? ast_stream_get_state(_stream) : AST_STREAM_STATE_END)
1816#define GET_STREAM_NAME_SAFE(_stream) (_stream ? ast_stream_get_name(_stream) : "")
1817
1818/*!
1819 * \internal
1820 * \brief Validate a media state
1821 *
1822 * \param session_name For log messages
1823 * \param state Media state
1824 *
1825 * \retval 1 The media state is valid
1826 * \retval 0 The media state is NOT valid
1827 *
1828 */
1829static int is_media_state_valid(const char *session_name, struct ast_sip_session_media_state *state)
1830{
1831 int stream_count = ast_stream_topology_get_count(state->topology);
1832 int session_count = AST_VECTOR_SIZE(&state->sessions);
1833 int i;
1834 int res = 0;
1835 SCOPE_ENTER(3, "%s: Topology: %s\n", session_name,
1836 ast_str_tmp(256, ast_stream_topology_to_str(state->topology, &STR_TMP)));
1837
1838 if (session_count != stream_count) {
1839 SCOPE_EXIT_RTN_VALUE(0, "%s: %d media sessions but %d streams\n", session_name,
1840 session_count, stream_count);
1841 }
1842
1843 for (i = 0; i < stream_count; i++) {
1844 struct ast_sip_session_media *media = NULL;
1845 struct ast_stream *stream = ast_stream_topology_get_stream(state->topology, i);
1846 const char *stream_name = NULL;
1847 int j;
1848 SCOPE_ENTER(4, "%s: Checking stream %s\n", session_name, ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
1849
1850 if (!stream) {
1851 SCOPE_EXIT_EXPR(goto end, "%s: stream %d is null\n", session_name, i);
1852 }
1853 stream_name = ast_stream_get_name(stream);
1854
1855 for (j = 0; j < stream_count; j++) {
1856 struct ast_stream *possible_dup = ast_stream_topology_get_stream(state->topology, j);
1857 if (j == i || !possible_dup) {
1858 continue;
1859 }
1860 if (!STREAM_REMOVED(stream) && ast_strings_equal(stream_name, GET_STREAM_NAME_SAFE(possible_dup))) {
1861 SCOPE_EXIT_EXPR(goto end, "%s: stream %i %s is duplicated to %d\n", session_name,
1862 i, stream_name, j);
1863 }
1864 }
1865
1866 media = AST_VECTOR_GET(&state->sessions, i);
1867 if (!media) {
1868 SCOPE_EXIT_EXPR(continue, "%s: media %d is null\n", session_name, i);
1869 }
1870
1871 for (j = 0; j < session_count; j++) {
1872 struct ast_sip_session_media *possible_dup = AST_VECTOR_GET(&state->sessions, j);
1873 if (j == i || !possible_dup) {
1874 continue;
1875 }
1876 if (!ast_strlen_zero(media->label) && !ast_strlen_zero(possible_dup->label)
1877 && ast_strings_equal(media->label, possible_dup->label)) {
1878 SCOPE_EXIT_EXPR(goto end, "%s: media %d %s is duplicated to %d\n", session_name,
1879 i, media->label, j);
1880 }
1881 }
1882
1883 if (media->stream_num != i) {
1884 SCOPE_EXIT_EXPR(goto end, "%s: media %d has stream_num %d\n", session_name,
1885 i, media->stream_num);
1886 }
1887
1888 if (media->type != ast_stream_get_type(stream)) {
1889 SCOPE_EXIT_EXPR(goto end, "%s: media %d has type %s but stream has type %s\n", stream_name,
1891 }
1892 SCOPE_EXIT("%s: Done with stream %s\n", session_name, ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
1893 }
1894
1895 res = 1;
1896end:
1897 SCOPE_EXIT_RTN_VALUE(res, "%s: %s\n", session_name, res ? "Valid" : "NOT Valid");
1898}
1899
1900/*!
1901 * \internal
1902 * \brief Merge media states for a delayed session refresh
1903 *
1904 * \param session_name For log messages
1905 * \param delayed_pending_state The pending media state at the time the request was queued
1906 * \param delayed_active_state The active media state at the time the request was queued
1907 * \param current_active_state The current active media state
1908 * \param run_post_validation Whether to run validation on the resulting media state or not
1909 *
1910 * \returns New merged topology or NULL if there's an error
1911 *
1912 */
1914 const char *session_name,
1915 struct ast_sip_session_media_state *delayed_pending_state,
1916 struct ast_sip_session_media_state *delayed_active_state,
1917 struct ast_sip_session_media_state *current_active_state,
1918 int run_post_validation)
1919{
1921 struct ast_sip_session_media_state *returned_media_state = NULL;
1922 struct ast_stream_topology *delayed_pending = delayed_pending_state->topology;
1923 struct ast_stream_topology *delayed_active = delayed_active_state->topology;
1924 struct ast_stream_topology *current_active = current_active_state->topology;
1925 struct ast_stream_topology *new_pending = NULL;
1926 int i;
1927 int max_stream_count;
1928 int res;
1929 SCOPE_ENTER(2, "%s: DP: %s DA: %s CA: %s\n", session_name,
1930 ast_str_tmp(256, ast_stream_topology_to_str(delayed_pending, &STR_TMP)),
1931 ast_str_tmp(256, ast_stream_topology_to_str(delayed_active, &STR_TMP)),
1932 ast_str_tmp(256, ast_stream_topology_to_str(current_active, &STR_TMP))
1933 );
1934
1935 max_stream_count = MAX(ast_stream_topology_get_count(delayed_pending),
1936 ast_stream_topology_get_count(delayed_active));
1937 max_stream_count = MAX(max_stream_count, ast_stream_topology_get_count(current_active));
1938
1939 /*
1940 * The new_pending_state is always based on the currently negotiated state because
1941 * the stream ordering in its topology must be preserved.
1942 */
1943 new_pending_state = ast_sip_session_media_state_clone(current_active_state);
1944 if (!new_pending_state) {
1945 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "%s: Couldn't clone current_active_state to new_pending_state\n", session_name);
1946 }
1947 new_pending = new_pending_state->topology;
1948
1949 for (i = 0; i < max_stream_count; i++) {
1950 struct ast_stream *dp_stream = GET_STREAM_SAFE(delayed_pending, i);
1951 struct ast_stream *da_stream = GET_STREAM_SAFE(delayed_active, i);
1952 struct ast_stream *ca_stream = GET_STREAM_SAFE(current_active, i);
1953 struct ast_stream *np_stream = GET_STREAM_SAFE(new_pending, i);
1954 struct ast_stream *found_da_stream = NULL;
1955 struct ast_stream *found_np_stream = NULL;
1956 enum ast_stream_state dp_state = GET_STREAM_STATE_SAFE(dp_stream);
1957 enum ast_stream_state da_state = GET_STREAM_STATE_SAFE(da_stream);
1958 enum ast_stream_state ca_state = GET_STREAM_STATE_SAFE(ca_stream);
1959 enum ast_stream_state np_state = GET_STREAM_STATE_SAFE(np_stream);
1960 enum ast_stream_state found_da_state = AST_STREAM_STATE_END;
1961 enum ast_stream_state found_np_state = AST_STREAM_STATE_END;
1962 const char *da_name = GET_STREAM_NAME_SAFE(da_stream);
1963 const char *dp_name = GET_STREAM_NAME_SAFE(dp_stream);
1964 const char *ca_name = GET_STREAM_NAME_SAFE(ca_stream);
1965 const char *np_name = GET_STREAM_NAME_SAFE(np_stream);
1966 const char *found_da_name __attribute__((unused)) = "";
1967 const char *found_np_name __attribute__((unused)) = "";
1968 int found_da_slot __attribute__((unused)) = -1;
1969 int found_np_slot = -1;
1970 int removed_np_slot = -1;
1971 int j;
1972 SCOPE_ENTER(3, "%s: slot: %d DP: %s DA: %s CA: %s\n", session_name, i,
1973 ast_str_tmp(128, ast_stream_to_str(dp_stream, &STR_TMP)),
1974 ast_str_tmp(128, ast_stream_to_str(da_stream, &STR_TMP)),
1975 ast_str_tmp(128, ast_stream_to_str(ca_stream, &STR_TMP)));
1976
1977 if (STATE_NONE(da_state) && STATE_NONE(dp_state) && STATE_NONE(ca_state)) {
1978 SCOPE_EXIT_EXPR(break, "%s: All gone\n", session_name);
1979 }
1980
1981 /*
1982 * Simple cases are handled first to avoid having to search the NP and DA
1983 * topologies for streams with the same name but not in the same position.
1984 */
1985
1986 if (STATE_NONE(dp_state) && !STATE_NONE(da_state)) {
1987 /*
1988 * The slot in the delayed pending topology can't be empty if the delayed
1989 * active topology has a stream there. Streams can't just go away. They
1990 * can be reused or marked "removed" but they can't go away.
1991 */
1992 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: DP slot is empty but DA is not\n", session_name);
1993 }
1994
1995 if (STATE_NONE(dp_state)) {
1996 /*
1997 * The current active topology can certainly have streams that weren't
1998 * in existence when the delayed request was queued. In this case,
1999 * no action is needed since we already copied the current active topology
2000 * to the new pending one.
2001 */
2002 SCOPE_EXIT_EXPR(continue, "%s: No DP stream so use CA stream as is\n", session_name);
2003 }
2004
2005 if (ast_strings_equal(dp_name, da_name) && ast_strings_equal(da_name, ca_name)) {
2006 /*
2007 * The delayed pending stream in this slot matches by name, the streams
2008 * in the same slot in the other two topologies. Easy case.
2009 */
2010 ast_trace(-1, "%s: Same stream in all 3 states\n", session_name);
2011 if (dp_state == da_state && da_state == ca_state) {
2012 /* All the same state, no need to update. */
2013 SCOPE_EXIT_EXPR(continue, "%s: All in the same state so nothing to do\n", session_name);
2014 }
2015 if (da_state != ca_state) {
2016 /*
2017 * Something set the CA state between the time this request was queued
2018 * and now. The CA state wins so we don't do anything.
2019 */
2020 SCOPE_EXIT_EXPR(continue, "%s: Ignoring request to change state from %s to %s\n",
2021 session_name, ast_stream_state2str(ca_state), ast_stream_state2str(dp_state));
2022 }
2023 if (dp_state != da_state) {
2024 /* DP needs to update the state */
2025 ast_stream_set_state(np_stream, dp_state);
2026 SCOPE_EXIT_EXPR(continue, "%s: Changed NP stream state from %s to %s\n",
2027 session_name, ast_stream_state2str(ca_state), ast_stream_state2str(dp_state));
2028 }
2029 }
2030
2031 /*
2032 * We're done with the simple cases. For the rest, we need to identify if the
2033 * DP stream we're trying to take action on is already in the other topologies
2034 * possibly in a different slot. To do that, if the stream in the DA or CA slots
2035 * doesn't match the current DP stream, we need to iterate over the topology
2036 * looking for a stream with the same name.
2037 */
2038
2039 /*
2040 * Since we already copied all of the CA streams to the NP topology, we'll use it
2041 * instead of CA because we'll be updating the NP as we go.
2042 */
2043 if (!ast_strings_equal(dp_name, np_name)) {
2044 /*
2045 * The NP stream in this slot doesn't have the same name as the DP stream
2046 * so we need to see if it's in another NP slot. We're not going to stop
2047 * when we find a matching stream because we also want to find the first
2048 * removed removed slot, if any, so we can re-use this slot. We'll break
2049 * early if we find both before we reach the end.
2050 */
2051 ast_trace(-1, "%s: Checking if DP is already in NP somewhere\n", session_name);
2052 for (j = 0; j < ast_stream_topology_get_count(new_pending); j++) {
2053 struct ast_stream *possible_existing = ast_stream_topology_get_stream(new_pending, j);
2054 const char *possible_existing_name = GET_STREAM_NAME_SAFE(possible_existing);
2055
2056 ast_trace(-1, "%s: Checking %s against %s\n", session_name, dp_name, possible_existing_name);
2057 if (found_np_slot == -1 && ast_strings_equal(dp_name, possible_existing_name)) {
2058 ast_trace(-1, "%s: Pending stream %s slot %d is in NP slot %d\n", session_name,
2059 dp_name, i, j);
2060 found_np_slot = j;
2061 found_np_stream = possible_existing;
2062 found_np_state = ast_stream_get_state(possible_existing);
2063 found_np_name = ast_stream_get_name(possible_existing);
2064 }
2065 if (STREAM_REMOVED(possible_existing) && removed_np_slot == -1) {
2066 removed_np_slot = j;
2067 }
2068 if (removed_np_slot >= 0 && found_np_slot >= 0) {
2069 break;
2070 }
2071 }
2072 } else {
2073 /* Makes the subsequent code easier */
2074 found_np_slot = i;
2075 found_np_stream = np_stream;
2076 found_np_state = np_state;
2077 found_np_name = np_name;
2078 }
2079
2080 if (!ast_strings_equal(dp_name, da_name)) {
2081 /*
2082 * The DA stream in this slot doesn't have the same name as the DP stream
2083 * so we need to see if it's in another DA slot. In real life, the DA stream
2084 * in this slot could have a different name but there shouldn't be a case
2085 * where the DP stream is another slot in the DA topology. Just in case though.
2086 * We don't care about removed slots in the DA topology.
2087 */
2088 ast_trace(-1, "%s: Checking if DP is already in DA somewhere\n", session_name);
2089 for (j = 0; j < ast_stream_topology_get_count(delayed_active); j++) {
2090 struct ast_stream *possible_existing = ast_stream_topology_get_stream(delayed_active, j);
2091 const char *possible_existing_name = GET_STREAM_NAME_SAFE(possible_existing);
2092
2093 ast_trace(-1, "%s: Checking %s against %s\n", session_name, dp_name, possible_existing_name);
2094 if (ast_strings_equal(dp_name, possible_existing_name)) {
2095 ast_trace(-1, "%s: Pending stream %s slot %d is already in delayed active slot %d\n",
2096 session_name, dp_name, i, j);
2097 found_da_slot = j;
2098 found_da_stream = possible_existing;
2099 found_da_state = ast_stream_get_state(possible_existing);
2100 found_da_name = ast_stream_get_name(possible_existing);
2101 break;
2102 }
2103 }
2104 } else {
2105 /* Makes the subsequent code easier */
2106 found_da_slot = i;
2107 found_da_stream = da_stream;
2108 found_da_state = da_state;
2109 found_da_name = da_name;
2110 }
2111
2112 ast_trace(-1, "%s: Found NP slot: %d Found removed NP slot: %d Found DA slot: %d\n",
2113 session_name, found_np_slot, removed_np_slot, found_da_slot);
2114
2115 /*
2116 * Now we know whether the DP stream is new or changing state and we know if the DP
2117 * stream exists in the other topologies and if so, where in those topologies it exists.
2118 */
2119
2120 if (!found_da_stream) {
2121 /*
2122 * The DP stream isn't in the DA topology which would imply that the intention of the
2123 * request was to add the stream, not change its state. It's possible though that
2124 * the stream was added by another request between the time this request was queued
2125 * and now so we need to check the CA topology as well.
2126 */
2127 ast_trace(-1, "%s: There was no corresponding DA stream so the request was to add a stream\n", session_name);
2128
2129 if (found_np_stream) {
2130 /*
2131 * We found it in the CA topology. Since the intention was to add it
2132 * and it's already there, there's nothing to do.
2133 */
2134 SCOPE_EXIT_EXPR(continue, "%s: New stream requested but it's already in CA\n", session_name);
2135 } else {
2136 /* OK, it's not in either which would again imply that the intention of the
2137 * request was to add the stream.
2138 */
2139 ast_trace(-1, "%s: There was no corresponding NP stream\n", session_name);
2140 if (STATE_REMOVED(dp_state)) {
2141 /*
2142 * How can DP request to remove a stream that doesn't seem to exist anythere?
2143 * It's not. It's possible that the stream was already removed and the slot
2144 * reused in the CA topology, but it would still have to exist in the DA
2145 * topology. Bail.
2146 */
2148 "%s: Attempting to remove stream %d:%s but it doesn't exist anywhere.\n", session_name, i, dp_name);
2149 } else {
2150 /*
2151 * We're now sure we want to add the the stream. Since we can re-use
2152 * slots in the CA topology that have streams marked as "removed", we
2153 * use the slot we saved in removed_np_slot if it exists.
2154 */
2155 ast_trace(-1, "%s: Checking for open slot\n", session_name);
2156 if (removed_np_slot >= 0) {
2157 struct ast_sip_session_media *old_media = AST_VECTOR_GET(&new_pending_state->sessions, removed_np_slot);
2158 res = ast_stream_topology_set_stream(new_pending, removed_np_slot, ast_stream_clone(dp_stream, NULL));
2159 if (res != 0) {
2160 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't set stream in new topology\n", session_name);
2161 }
2162 /*
2163 * Since we're reusing the removed_np_slot slot for something else, we need
2164 * to free and remove any session media already in it.
2165 * ast_stream_topology_set_stream() took care of freeing the old stream.
2166 */
2167 res = AST_VECTOR_REPLACE(&new_pending_state->sessions, removed_np_slot, NULL);
2168 if (res != 0) {
2169 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't replace media session\n", session_name);
2170 }
2171
2172 ao2_cleanup(old_media);
2173 SCOPE_EXIT_EXPR(continue, "%s: Replaced removed stream in slot %d\n",
2174 session_name, removed_np_slot);
2175 } else {
2176 int new_slot = ast_stream_topology_append_stream(new_pending, ast_stream_clone(dp_stream, NULL));
2177 if (new_slot < 0) {
2178 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't append stream in new topology\n", session_name);
2179 }
2180
2181 res = AST_VECTOR_REPLACE(&new_pending_state->sessions, new_slot, NULL);
2182 if (res != 0) {
2183 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't replace media session\n", session_name);
2184 }
2185 SCOPE_EXIT_EXPR(continue, "%s: Appended new stream to slot %d\n",
2186 session_name, new_slot);
2187 }
2188 }
2189 }
2190 } else {
2191 /*
2192 * The DP stream exists in the DA topology so it's a change of some sort.
2193 */
2194 ast_trace(-1, "%s: There was a corresponding DA stream so the request was to change/remove a stream\n", session_name);
2195 if (dp_state == found_da_state) {
2196 /* No change? Let's see if it's in CA */
2197 if (!found_np_stream) {
2198 /*
2199 * The DP and DA state are the same which would imply that the stream
2200 * already exists but it's not in the CA topology. It's possible that
2201 * between the time this request was queued and now the stream was removed
2202 * from the CA topology and the slot used for something else. Nothing
2203 * we can do here.
2204 */
2205 SCOPE_EXIT_EXPR(continue, "%s: Stream doesn't exist in CA so nothing to do\n", session_name);
2206 } else if (dp_state == found_np_state) {
2207 SCOPE_EXIT_EXPR(continue, "%s: States are the same all around so nothing to do\n", session_name);
2208 } else {
2209 SCOPE_EXIT_EXPR(continue, "%s: Something changed the CA state so we're going to leave it as is\n", session_name);
2210 }
2211 } else {
2212 /* We have a state change. */
2213 ast_trace(-1, "%s: Requesting state change to %s\n", session_name, ast_stream_state2str(dp_state));
2214 if (!found_np_stream) {
2215 SCOPE_EXIT_EXPR(continue, "%s: Stream doesn't exist in CA so nothing to do\n", session_name);
2216 } else if (da_state == found_np_state) {
2217 ast_stream_set_state(found_np_stream, dp_state);
2218 SCOPE_EXIT_EXPR(continue, "%s: Changed NP stream state from %s to %s\n",
2219 session_name, ast_stream_state2str(found_np_state), ast_stream_state2str(dp_state));
2220 } else {
2221 SCOPE_EXIT_EXPR(continue, "%s: Something changed the CA state so we're going to leave it as is\n",
2222 session_name);
2223 }
2224 }
2225 }
2226
2227 SCOPE_EXIT("%s: Done with slot %d\n", session_name, i);
2228 }
2229
2230 ast_trace(-1, "%s: Resetting default media states\n", session_name);
2231 for (i = 0; i < AST_MEDIA_TYPE_END; i++) {
2232 int j;
2233 new_pending_state->default_session[i] = NULL;
2234 for (j = 0; j < AST_VECTOR_SIZE(&new_pending_state->sessions); j++) {
2235 struct ast_sip_session_media *media = AST_VECTOR_GET(&new_pending_state->sessions, j);
2236 struct ast_stream *stream = ast_stream_topology_get_stream(new_pending_state->topology, j);
2237
2238 if (media && media->type == i && !STREAM_REMOVED(stream)) {
2239 new_pending_state->default_session[i] = media;
2240 break;
2241 }
2242 }
2243 }
2244
2245 if (run_post_validation) {
2246 ast_trace(-1, "%s: Running post-validation\n", session_name);
2247 if (!is_media_state_valid(session_name, new_pending_state)) {
2248 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "State not consistent\n");
2249 }
2250 }
2251
2252 /*
2253 * We need to move the new pending state to another variable and set new_pending_state to NULL
2254 * so RAII_VAR doesn't free it.
2255 */
2256 returned_media_state = new_pending_state;
2257 new_pending_state = NULL;
2258 SCOPE_EXIT_RTN_VALUE(returned_media_state, "%s: NP: %s\n", session_name,
2259 ast_str_tmp(256, ast_stream_topology_to_str(new_pending, &STR_TMP)));
2260}
2261
2263 ast_sip_session_request_creation_cb on_request_creation,
2264 ast_sip_session_sdp_creation_cb on_sdp_creation,
2265 ast_sip_session_response_cb on_response,
2266 enum ast_sip_session_refresh_method method, int generate_new_sdp,
2267 struct ast_sip_session_media_state *pending_media_state,
2268 struct ast_sip_session_media_state *active_media_state,
2269 int queued)
2270{
2271 pjsip_inv_session *inv_session = session->inv_session;
2272 pjmedia_sdp_session *new_sdp = NULL;
2273 pjsip_tx_data *tdata;
2274 int res = -1;
2275 SCOPE_ENTER(3, "%s: New SDP? %s Queued? %s DP: %s DA: %s\n", ast_sip_session_get_name(session),
2276 generate_new_sdp ? "yes" : "no", queued ? "yes" : "no",
2277 pending_media_state ? ast_str_tmp(256, ast_stream_topology_to_str(pending_media_state->topology, &STR_TMP)) : "none",
2278 active_media_state ? ast_str_tmp(256, ast_stream_topology_to_str(active_media_state->topology, &STR_TMP)) : "none");
2279
2280 if (pending_media_state && (!pending_media_state->topology || !generate_new_sdp)) {
2281
2282 ast_sip_session_media_state_free(pending_media_state);
2283 ast_sip_session_media_state_free(active_media_state);
2284 SCOPE_EXIT_RTN_VALUE(-1, "%s: Not sending reinvite because %s%s\n", ast_sip_session_get_name(session),
2285 pending_media_state->topology == NULL ? "pending topology is null " : "",
2286 !generate_new_sdp ? "generate_new_sdp is false" : "");
2287 }
2288
2289 if (inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
2290 /* Don't try to do anything with a hung-up call */
2291 ast_sip_session_media_state_free(pending_media_state);
2292 ast_sip_session_media_state_free(active_media_state);
2293 SCOPE_EXIT_RTN_VALUE(0, "%s: Not sending reinvite because of disconnected state\n",
2295 }
2296
2297 /* If the dialog has not yet been established we have to defer until it has */
2298 if (inv_session->dlg->state != PJSIP_DIALOG_STATE_ESTABLISHED) {
2299 res = delay_request(session, on_request_creation, on_sdp_creation, on_response,
2300 generate_new_sdp,
2303 pending_media_state, active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
2304 SCOPE_EXIT_RTN_VALUE(res, "%s: Delay sending reinvite because dialog has not been established\n",
2306 }
2307
2309 if (inv_session->invite_tsx) {
2310 /* We can't send a reinvite yet, so delay it */
2311 res = delay_request(session, on_request_creation, on_sdp_creation,
2312 on_response, generate_new_sdp, DELAYED_METHOD_INVITE, pending_media_state,
2313 active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
2314 SCOPE_EXIT_RTN_VALUE(res, "%s: Delay sending reinvite because of outstanding transaction\n",
2316 } else if (inv_session->state != PJSIP_INV_STATE_CONFIRMED) {
2317 /* Initial INVITE transaction failed to progress us to a confirmed state
2318 * which means re-invites are not possible
2319 */
2320 ast_sip_session_media_state_free(pending_media_state);
2321 ast_sip_session_media_state_free(active_media_state);
2322 SCOPE_EXIT_RTN_VALUE(0, "%s: Not sending reinvite because not in confirmed state\n",
2324 }
2325 }
2326
2327 if (generate_new_sdp) {
2328 /* SDP can only be generated if current negotiation has already completed */
2329 if (inv_session->neg
2330 && pjmedia_sdp_neg_get_state(inv_session->neg)
2331 != PJMEDIA_SDP_NEG_STATE_DONE) {
2332 res = delay_request(session, on_request_creation, on_sdp_creation,
2333 on_response, generate_new_sdp,
2335 ? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE, pending_media_state,
2336 active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
2337 SCOPE_EXIT_RTN_VALUE(res, "%s: Delay session refresh with new SDP because SDP negotiation is not yet done\n",
2339 }
2340
2341 /* If an explicitly requested media state has been provided use it instead of any pending one */
2342 if (pending_media_state) {
2343 int index;
2344 int type_streams[AST_MEDIA_TYPE_END] = {0};
2345
2346 ast_trace(-1, "%s: Pending media state exists\n", ast_sip_session_get_name(session));
2347
2348 /* Media state conveys a desired media state, so if there are outstanding
2349 * delayed requests we need to ensure we go into the queue and not jump
2350 * ahead. If we sent this media state now then updates could go out of
2351 * order.
2352 */
2353 if (!queued && !AST_LIST_EMPTY(&session->delayed_requests)) {
2354 res = delay_request(session, on_request_creation, on_sdp_creation,
2355 on_response, generate_new_sdp,
2357 ? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE, pending_media_state,
2358 active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
2359 SCOPE_EXIT_RTN_VALUE(res, "%s: Delay sending reinvite because of outstanding requests\n",
2361 }
2362
2363 /*
2364 * Attempt to resolve only if objects are available, and it's not
2365 * switching to or from an image type.
2366 */
2367 if (active_media_state && active_media_state->topology &&
2368 (!active_media_state->default_session[AST_MEDIA_TYPE_IMAGE] ==
2369 !pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE])) {
2370
2371 struct ast_sip_session_media_state *new_pending_state;
2372
2373 ast_trace(-1, "%s: Active media state exists and is%s equal to pending\n", ast_sip_session_get_name(session),
2374 !ast_stream_topology_equal(active_media_state->topology,pending_media_state->topology) ? " not" : "");
2375 ast_trace(-1, "%s: DP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(pending_media_state->topology, &STR_TMP)));
2376 ast_trace(-1, "%s: DA: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(active_media_state->topology, &STR_TMP)));
2377 ast_trace(-1, "%s: CP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)));
2378 ast_trace(-1, "%s: CA: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
2379
2381 pending_media_state, active_media_state, session->active_media_state, 1);
2382 if (new_pending_state) {
2383 ast_trace(-1, "%s: NP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(new_pending_state->topology, &STR_TMP)));
2384 ast_sip_session_media_state_free(pending_media_state);
2385 pending_media_state = new_pending_state;
2386 } else {
2387 ast_sip_session_media_state_reset(pending_media_state);
2388 ast_sip_session_media_state_free(active_media_state);
2389 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Unable to merge media states\n", ast_sip_session_get_name(session));
2390 }
2391 }
2392
2393 /* Prune the media state so the number of streams fit within the configured limits - we do it here
2394 * so that the index of the resulting streams in the SDP match. If we simply left the streams out
2395 * of the SDP when producing it we'd be in trouble. We also enforce formats here for media types that
2396 * are configurable on the endpoint.
2397 */
2398 ast_trace(-1, "%s: Pruning and checking formats of streams\n", ast_sip_session_get_name(session));
2399
2400 for (index = 0; index < ast_stream_topology_get_count(pending_media_state->topology); ++index) {
2401 struct ast_stream *existing_stream = NULL;
2402 struct ast_stream *stream = ast_stream_topology_get_stream(pending_media_state->topology, index);
2403 SCOPE_ENTER(4, "%s: Checking stream %s\n", ast_sip_session_get_name(session),
2404 ast_stream_get_name(stream));
2405
2406 if (session->active_media_state->topology &&
2407 index < ast_stream_topology_get_count(session->active_media_state->topology)) {
2408 existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, index);
2409 ast_trace(-1, "%s: Found existing stream %s\n", ast_sip_session_get_name(session),
2410 ast_stream_get_name(existing_stream));
2411 }
2412
2413 if (is_stream_limitation_reached(ast_stream_get_type(stream), session->endpoint, type_streams)) {
2414 if (index < AST_VECTOR_SIZE(&pending_media_state->sessions)) {
2415 struct ast_sip_session_media *session_media = AST_VECTOR_GET(&pending_media_state->sessions, index);
2416
2417 ao2_cleanup(session_media);
2418 AST_VECTOR_REMOVE(&pending_media_state->sessions, index, 1);
2419 }
2420
2421 ast_stream_topology_del_stream(pending_media_state->topology, index);
2422 ast_trace(-1, "%s: Dropped overlimit stream %s\n", ast_sip_session_get_name(session),
2423 ast_stream_get_name(stream));
2424
2425 /* A stream has potentially moved into our spot so we need to jump back so we process it */
2426 index -= 1;
2427 SCOPE_EXIT_EXPR(continue);
2428 }
2429
2430 /* No need to do anything with stream if it's media state is removed */
2432 /* If there is no existing stream we can just not have this stream in the topology at all. */
2433 if (!existing_stream) {
2434 ast_trace(-1, "%s: Dropped removed stream %s\n", ast_sip_session_get_name(session),
2435 ast_stream_get_name(stream));
2436 ast_stream_topology_del_stream(pending_media_state->topology, index);
2437 /* TODO: Do we need to remove the corresponding media state? */
2438 index -= 1;
2439 }
2440 SCOPE_EXIT_EXPR(continue);
2441 }
2442
2443 /* Enforce the configured allowed codecs on audio and video streams */
2445 !ast_stream_get_metadata(stream, "pjsip_session_refresh")) {
2446 struct ast_format_cap *joint_cap;
2447
2449 if (!joint_cap) {
2450 ast_sip_session_media_state_free(pending_media_state);
2451 ast_sip_session_media_state_free(active_media_state);
2452 res = -1;
2453 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Unable to alloc format caps\n", ast_sip_session_get_name(session));
2454 }
2455 ast_format_cap_get_compatible(ast_stream_get_formats(stream), session->endpoint->media.codecs, joint_cap);
2456 if (!ast_format_cap_count(joint_cap)) {
2457 ao2_ref(joint_cap, -1);
2458
2459 if (!existing_stream) {
2460 /* If there is no existing stream we can just not have this stream in the topology
2461 * at all.
2462 */
2463 ast_stream_topology_del_stream(pending_media_state->topology, index);
2464 index -= 1;
2465 SCOPE_EXIT_EXPR(continue, "%s: Dropped incompatible stream %s\n",
2467 } else if (ast_stream_get_state(stream) != ast_stream_get_state(existing_stream) ||
2468 strcmp(ast_stream_get_name(stream), ast_stream_get_name(existing_stream))) {
2469 /* If the underlying stream is a different type or different name then we have to
2470 * mark it as removed, as it is replacing an existing stream. We do this so order
2471 * is preserved.
2472 */
2474 SCOPE_EXIT_EXPR(continue, "%s: Dropped incompatible stream %s\n",
2476 } else {
2477 /* However if the stream is otherwise remaining the same we can keep the formats
2478 * that exist on it already which allows media to continue to flow. We don't modify
2479 * the format capabilities but do need to cast it so that ao2_bump can raise the
2480 * reference count.
2481 */
2482 joint_cap = ao2_bump((struct ast_format_cap *)ast_stream_get_formats(existing_stream));
2483 }
2484 }
2485 ast_stream_set_formats(stream, joint_cap);
2486 ao2_cleanup(joint_cap);
2487 }
2488
2489 ++type_streams[ast_stream_get_type(stream)];
2490
2491 SCOPE_EXIT();
2492 }
2493
2494 if (session->active_media_state->topology) {
2495 /* SDP is a fun thing. Take for example the fact that streams are never removed. They just become
2496 * declined. To better handle this in the case where something requests a topology change for fewer
2497 * streams than are currently present we fill in the topology to match the current number of streams
2498 * that are active.
2499 */
2500
2501 for (index = ast_stream_topology_get_count(pending_media_state->topology);
2502 index < ast_stream_topology_get_count(session->active_media_state->topology); ++index) {
2503 struct ast_stream *stream = ast_stream_topology_get_stream(session->active_media_state->topology, index);
2504 struct ast_stream *cloned;
2505 int position;
2506 SCOPE_ENTER(4, "%s: Stream %s not in pending\n", ast_sip_session_get_name(session),
2507 ast_stream_get_name(stream));
2508
2509 cloned = ast_stream_clone(stream, NULL);
2510 if (!cloned) {
2511 ast_sip_session_media_state_free(pending_media_state);
2512 ast_sip_session_media_state_free(active_media_state);
2513 res = -1;
2514 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Unable to clone stream %s\n",
2516 }
2517
2519 position = ast_stream_topology_append_stream(pending_media_state->topology, cloned);
2520 if (position < 0) {
2521 ast_stream_free(cloned);
2522 ast_sip_session_media_state_free(pending_media_state);
2523 ast_sip_session_media_state_free(active_media_state);
2524 res = -1;
2525 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Unable to append cloned stream\n",
2527 }
2528 SCOPE_EXIT("%s: Appended empty stream in position %d to make counts match\n",
2530 }
2531
2532 /*
2533 * We can suppress this re-invite if the pending topology is equal to the currently
2534 * active topology.
2535 */
2536 if (ast_stream_topology_equal(session->active_media_state->topology, pending_media_state->topology)) {
2537 ast_trace(-1, "%s: CA: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
2538 ast_trace(-1, "%s: NP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(pending_media_state->topology, &STR_TMP)));
2539 ast_sip_session_media_state_free(pending_media_state);
2540 ast_sip_session_media_state_free(active_media_state);
2541 /* For external consumers we return 0 to say success, but internally for
2542 * send_delayed_request we return a separate value to indicate that this
2543 * session refresh would be redundant so we didn't send it
2544 */
2545 SCOPE_EXIT_RTN_VALUE(queued ? 1 : 0, "%s: Topologies are equal. Not sending re-invite\n",
2547 }
2548 }
2549
2550 ast_sip_session_media_state_free(session->pending_media_state);
2551 session->pending_media_state = pending_media_state;
2552 }
2553
2555 if (!new_sdp) {
2556 ast_sip_session_media_state_reset(session->pending_media_state);
2557 ast_sip_session_media_state_free(active_media_state);
2558 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Failed to generate session refresh SDP. Not sending session refresh\n",
2560 }
2561 if (on_sdp_creation) {
2562 if (on_sdp_creation(session, new_sdp)) {
2563 ast_sip_session_media_state_reset(session->pending_media_state);
2564 ast_sip_session_media_state_free(active_media_state);
2565 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: on_sdp_creation failed\n", ast_sip_session_get_name(session));
2566 }
2567 }
2568 }
2569
2571 if (pjsip_inv_reinvite(inv_session, NULL, new_sdp, &tdata)) {
2572 if (generate_new_sdp) {
2573 ast_sip_session_media_state_reset(session->pending_media_state);
2574 }
2575 ast_sip_session_media_state_free(active_media_state);
2576 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Failed to create reinvite properly\n", ast_sip_session_get_name(session));
2577 }
2578 } else if (pjsip_inv_update(inv_session, NULL, new_sdp, &tdata)) {
2579 if (generate_new_sdp) {
2580 ast_sip_session_media_state_reset(session->pending_media_state);
2581 }
2582 ast_sip_session_media_state_free(active_media_state);
2583 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Failed to create UPDATE properly\n", ast_sip_session_get_name(session));
2584 }
2585 if (on_request_creation) {
2586 if (on_request_creation(session, tdata)) {
2587 if (generate_new_sdp) {
2588 ast_sip_session_media_state_reset(session->pending_media_state);
2589 }
2590 ast_sip_session_media_state_free(active_media_state);
2591 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: on_request_creation failed.\n", ast_sip_session_get_name(session));
2592 }
2593 }
2594 ast_sip_session_send_request_with_cb(session, tdata, on_response);
2595 ast_sip_session_media_state_free(active_media_state);
2596
2597end:
2598 SCOPE_EXIT_RTN_VALUE(res, "%s: Sending session refresh SDP via %s\n", ast_sip_session_get_name(session),
2599 method == AST_SIP_SESSION_REFRESH_METHOD_INVITE ? "re-INVITE" : "UPDATE");
2600}
2601
2603 ast_sip_session_request_creation_cb on_request_creation,
2604 ast_sip_session_sdp_creation_cb on_sdp_creation,
2605 ast_sip_session_response_cb on_response,
2606 enum ast_sip_session_refresh_method method, int generate_new_sdp,
2607 struct ast_sip_session_media_state *media_state)
2608{
2609 return sip_session_refresh(session, on_request_creation, on_sdp_creation,
2610 on_response, method, generate_new_sdp, media_state, NULL, 0);
2611}
2612
2614 ast_sip_session_sdp_creation_cb on_sdp_creation)
2615{
2616 pjsip_inv_session *inv_session = session->inv_session;
2617 pjmedia_sdp_session *new_answer = NULL;
2618 const pjmedia_sdp_session *previous_offer = NULL;
2620
2621 /* The SDP answer can only be regenerated if it is still pending to be sent */
2622 if (!inv_session->neg || (pjmedia_sdp_neg_get_state(inv_session->neg) != PJMEDIA_SDP_NEG_STATE_REMOTE_OFFER &&
2623 pjmedia_sdp_neg_get_state(inv_session->neg) != PJMEDIA_SDP_NEG_STATE_WAIT_NEGO)) {
2624 ast_log(LOG_WARNING, "Requested to regenerate local SDP answer for channel '%s' but negotiation in state '%s'\n",
2625 ast_channel_name(session->channel), pjmedia_sdp_neg_state_str(pjmedia_sdp_neg_get_state(inv_session->neg)));
2626 SCOPE_EXIT_RTN_VALUE(-1, "Bad negotiation state\n");
2627 }
2628
2629 pjmedia_sdp_neg_get_neg_remote(inv_session->neg, &previous_offer);
2630 if (pjmedia_sdp_neg_get_state(inv_session->neg) == PJMEDIA_SDP_NEG_STATE_WAIT_NEGO) {
2631 /* Transition the SDP negotiator back to when it received the remote offer */
2632 pjmedia_sdp_neg_negotiate(inv_session->pool, inv_session->neg, 0);
2633 pjmedia_sdp_neg_set_remote_offer(inv_session->pool, inv_session->neg, previous_offer);
2634 }
2635
2636 new_answer = create_local_sdp(inv_session, session, previous_offer, 0);
2637 if (!new_answer) {
2638 ast_log(LOG_WARNING, "Could not create a new local SDP answer for channel '%s'\n",
2639 ast_channel_name(session->channel));
2640 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create new SDP\n");
2641 }
2642
2643 if (on_sdp_creation) {
2644 if (on_sdp_creation(session, new_answer)) {
2645 SCOPE_EXIT_RTN_VALUE(-1, "Callback failed\n");
2646 }
2647 }
2648
2649 pjsip_inv_set_sdp_answer(inv_session, new_answer);
2650
2652}
2653
2654void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
2655{
2656 pjsip_dialog *dlg = pjsip_tdata_get_dlg(tdata);
2657 RAII_VAR(struct ast_sip_session *, dlg_session, dlg ? ast_sip_dialog_get_session(dlg) : NULL, ao2_cleanup);
2658 if (!dlg_session) {
2659 /* If the dialog has a session, handle_outgoing_response will be called
2660 from session_on_tx_response. If it does not, call it from here. */
2662 }
2663 pjsip_inv_send_msg(session->inv_session, tdata);
2664 return;
2665}
2666
2667static pj_bool_t session_on_rx_request(pjsip_rx_data *rdata);
2668static pj_bool_t session_on_rx_response(pjsip_rx_data *rdata);
2669static pj_status_t session_on_tx_response(pjsip_tx_data *tdata);
2670static void session_on_tsx_state(pjsip_transaction *tsx, pjsip_event *e);
2671
2672static pjsip_module session_module = {
2673 .name = {"Session Module", 14},
2674 .priority = PJSIP_MOD_PRIORITY_APPLICATION,
2675 .on_rx_request = session_on_rx_request,
2676 .on_rx_response = session_on_rx_response,
2677 .on_tsx_state = session_on_tsx_state,
2678 .on_tx_response = session_on_tx_response,
2679};
2680
2681/*! \brief Determine whether the SDP provided requires deferral of negotiating or not
2682 *
2683 * \retval 1 re-invite should be deferred and resumed later
2684 * \retval 0 re-invite should not be deferred
2685 */
2686static int sdp_requires_deferral(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
2687{
2688 int i;
2689
2690 if (!session->pending_media_state->topology) {
2691 session->pending_media_state->topology = ast_stream_topology_alloc();
2692 if (!session->pending_media_state->topology) {
2693 return -1;
2694 }
2695 }
2696
2697 for (i = 0; i < sdp->media_count; ++i) {
2698 /* See if there are registered handlers for this media stream type */
2699 char media[20];
2701 RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
2702 struct ast_stream *existing_stream = NULL;
2703 struct ast_stream *stream;
2704 enum ast_media_type type;
2705 struct ast_sip_session_media *session_media = NULL;
2707 pjmedia_sdp_media *remote_stream = sdp->media[i];
2708
2709 /* We need a null-terminated version of the media string */
2710 ast_copy_pj_str(media, &sdp->media[i]->desc.media, sizeof(media));
2711
2712 if (session->active_media_state->topology &&
2713 (i < ast_stream_topology_get_count(session->active_media_state->topology))) {
2714 existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, i);
2715 }
2716
2718 stream = ast_stream_alloc(existing_stream ? ast_stream_get_name(existing_stream) : ast_codec_media_type2str(type), type);
2719 if (!stream) {
2720 return -1;
2721 }
2722
2723 /* As this is only called on an incoming SDP offer before processing it is not possible
2724 * for streams and their media sessions to exist.
2725 */
2726 if (ast_stream_topology_set_stream(session->pending_media_state->topology, i, stream)) {
2727 ast_stream_free(stream);
2728 return -1;
2729 }
2730
2731 if (existing_stream) {
2732 const char *stream_label = ast_stream_get_metadata(existing_stream, "SDP:LABEL");
2733
2734 if (!ast_strlen_zero(stream_label)) {
2735 ast_stream_set_metadata(stream, "SDP:LABEL", stream_label);
2736 }
2737 }
2738
2739 session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_media_type_from_str(media), i);
2740 if (!session_media) {
2741 return -1;
2742 }
2743
2744 /* For backwards compatibility with the core the default audio stream is always sendrecv */
2745 if (!ast_sip_session_is_pending_stream_default(session, stream) || strcmp(media, "audio")) {
2746 if (pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
2747 /* Stream state reflects our state of a stream, so in the case of
2748 * sendonly and recvonly we store the opposite since that is what ours
2749 * is.
2750 */
2752 } else if (pjmedia_sdp_media_find_attr2(remote_stream, "recvonly", NULL)) {
2754 } else if (pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
2756 } else {
2758 }
2759 } else {
2761 }
2762
2763 if (session_media->handler) {
2764 handler = session_media->handler;
2765 if (handler->defer_incoming_sdp_stream) {
2766 res = handler->defer_incoming_sdp_stream(session, session_media, sdp,
2767 sdp->media[i]);
2768 switch (res) {
2770 break;
2772 return 0;
2774 break;
2776 return 1;
2777 }
2778 }
2779 /* Handled by this handler. Move to the next stream */
2780 continue;
2781 }
2782
2783 handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
2784 if (!handler_list) {
2785 ast_debug(3, "%s: No registered SDP handlers for media type '%s'\n", ast_sip_session_get_name(session), media);
2786 continue;
2787 }
2788 AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
2789 if (handler == session_media->handler) {
2790 continue;
2791 }
2792 if (!handler->defer_incoming_sdp_stream) {
2793 continue;
2794 }
2795 res = handler->defer_incoming_sdp_stream(session, session_media, sdp,
2796 sdp->media[i]);
2797 switch (res) {
2799 continue;
2801 session_media_set_handler(session_media, handler);
2802 return 0;
2804 /* Handled by this handler. */
2805 session_media_set_handler(session_media, handler);
2806 break;
2808 /* Handled by this handler. */
2809 session_media_set_handler(session_media, handler);
2810 return 1;
2811 }
2812 /* Move to the next stream */
2813 break;
2814 }
2815 }
2816 return 0;
2817}
2818
2819static pj_bool_t session_reinvite_on_rx_request(pjsip_rx_data *rdata)
2820{
2821 pjsip_dialog *dlg;
2823 pjsip_rdata_sdp_info *sdp_info;
2824 int deferred;
2825
2826 if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD ||
2827 !(dlg = pjsip_ua_find_dialog(&rdata->msg_info.cid->id, &rdata->msg_info.to->tag, &rdata->msg_info.from->tag, PJ_FALSE)) ||
2829 !session->channel) {
2830 return PJ_FALSE;
2831 }
2832
2833 if (session->inv_session->invite_tsx) {
2834 /* There's a transaction in progress so bail now and let pjproject send 491 */
2835 return PJ_FALSE;
2836 }
2837
2838 if (session->deferred_reinvite) {
2839 pj_str_t key, deferred_key;
2840 pjsip_tx_data *tdata;
2841
2842 /* We use memory from the new request on purpose so the deferred reinvite pool does not grow uncontrollably */
2843 pjsip_tsx_create_key(rdata->tp_info.pool, &key, PJSIP_ROLE_UAS, &rdata->msg_info.cseq->method, rdata);
2844 pjsip_tsx_create_key(rdata->tp_info.pool, &deferred_key, PJSIP_ROLE_UAS, &session->deferred_reinvite->msg_info.cseq->method,
2845 session->deferred_reinvite);
2846
2847 /* If this is a retransmission ignore it */
2848 if (!pj_strcmp(&key, &deferred_key)) {
2849 return PJ_TRUE;
2850 }
2851
2852 /* Otherwise this is a new re-invite, so reject it */
2853 if (pjsip_dlg_create_response(dlg, rdata, 491, NULL, &tdata) == PJ_SUCCESS) {
2854 if (pjsip_endpt_send_response2(ast_sip_get_pjsip_endpoint(), rdata, tdata, NULL, NULL) != PJ_SUCCESS) {
2855 pjsip_tx_data_dec_ref(tdata);
2856 }
2857 }
2858
2859 return PJ_TRUE;
2860 }
2861
2862 if (!(sdp_info = pjsip_rdata_get_sdp_info(rdata)) ||
2863 (sdp_info->sdp_err != PJ_SUCCESS)) {
2864 return PJ_FALSE;
2865 }
2866
2867 if (!sdp_info->sdp) {
2868 return PJ_FALSE;
2869 }
2870
2871 deferred = sdp_requires_deferral(session, sdp_info->sdp);
2872 if (deferred == -1) {
2873 ast_sip_session_media_state_reset(session->pending_media_state);
2874 return PJ_FALSE;
2875 } else if (!deferred) {
2876 return PJ_FALSE;
2877 }
2878
2879 pjsip_rx_data_clone(rdata, 0, &session->deferred_reinvite);
2880
2881 return PJ_TRUE;
2882}
2883
2885{
2886 if (!session->deferred_reinvite) {
2887 return;
2888 }
2889
2890 if (session->channel) {
2891 pjsip_endpt_process_rx_data(ast_sip_get_pjsip_endpoint(),
2892 session->deferred_reinvite, NULL, NULL);
2893 }
2894 pjsip_rx_data_free_cloned(session->deferred_reinvite);
2895 session->deferred_reinvite = NULL;
2896}
2897
2898static pjsip_module session_reinvite_module = {
2899 .name = { "Session Re-Invite Module", 24 },
2900 .priority = PJSIP_MOD_PRIORITY_UA_PROXY_LAYER - 1,
2901 .on_rx_request = session_reinvite_on_rx_request,
2902};
2903
2905 ast_sip_session_response_cb on_response)
2906{
2907 pjsip_inv_session *inv_session = session->inv_session;
2908
2909 /* For every request except BYE we disallow sending of the message when
2910 * the session has been disconnected. A BYE request is special though
2911 * because it can be sent again after the session is disconnected except
2912 * with credentials.
2913 */
2914 if (inv_session->state == PJSIP_INV_STATE_DISCONNECTED &&
2915 tdata->msg->line.req.method.id != PJSIP_BYE_METHOD) {
2916 return;
2917 }
2918
2919 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, session_module.id,
2920 MOD_DATA_ON_RESPONSE, on_response);
2921
2923 pjsip_inv_send_msg(session->inv_session, tdata);
2924
2925 return;
2926}
2927
2928void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
2929{
2931}
2932
2933int ast_sip_session_create_invite(struct ast_sip_session *session, pjsip_tx_data **tdata)
2934{
2935 pjmedia_sdp_session *offer;
2937
2938 if (!(offer = create_local_sdp(session->inv_session, session, NULL, 0))) {
2939 pjsip_inv_terminate(session->inv_session, 500, PJ_FALSE);
2940 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create offer\n");
2941 }
2942
2943 pjsip_inv_set_local_sdp(session->inv_session, offer);
2944 pjmedia_sdp_neg_set_prefer_remote_codec_order(session->inv_session->neg, PJ_FALSE);
2945#ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
2946 if (!session->endpoint->preferred_codec_only) {
2947 pjmedia_sdp_neg_set_answer_multiple_codecs(session->inv_session->neg, PJ_TRUE);
2948 }
2949#endif
2950
2951 /*
2952 * We MUST call set_from_header() before pjsip_inv_invite. If we don't, the
2953 * From in the initial INVITE will be wrong but the rest of the messages will be OK.
2954 */
2956
2957 if (pjsip_inv_invite(session->inv_session, tdata) != PJ_SUCCESS) {
2958 SCOPE_EXIT_RTN_VALUE(-1, "pjsip_inv_invite failed\n");
2959 }
2960
2962}
2963
2964static int datastore_hash(const void *obj, int flags)
2965{
2966 const struct ast_datastore *datastore = obj;
2967 const char *uid = flags & OBJ_KEY ? obj : datastore->uid;
2968
2969 ast_assert(uid != NULL);
2970
2971 return ast_str_hash(uid);
2972}
2973
2974static int datastore_cmp(void *obj, void *arg, int flags)
2975{
2976 const struct ast_datastore *datastore1 = obj;
2977 const struct ast_datastore *datastore2 = arg;
2978 const char *uid2 = flags & OBJ_KEY ? arg : datastore2->uid;
2979
2980 ast_assert(datastore1->uid != NULL);
2981 ast_assert(uid2 != NULL);
2982
2983 return strcmp(datastore1->uid, uid2) ? 0 : CMP_MATCH | CMP_STOP;
2984}
2985
2986static void session_destructor(void *obj)
2987{
2988 struct ast_sip_session *session = obj;
2989
2990#ifdef TEST_FRAMEWORK
2991 /* We dup the endpoint ID in case the endpoint gets freed out from under us */
2992 const char *endpoint_name = session->endpoint ?
2993 ast_strdupa(ast_sorcery_object_get_id(session->endpoint)) : "<none>";
2994#endif
2995
2996 ast_debug(3, "%s: Destroying SIP session\n", ast_sip_session_get_name(session));
2997
2998 ast_test_suite_event_notify("SESSION_DESTROYING",
2999 "Endpoint: %s\r\n"
3000 "AOR: %s\r\n"
3001 "Contact: %s"
3002 , endpoint_name
3003 , session->aor ? ast_sorcery_object_get_id(session->aor) : "<none>"
3004 , session->contact ? ast_sorcery_object_get_id(session->contact) : "<none>"
3005 );
3006
3007 /* fire session destroy handler */
3009
3010 /* remove all registered supplements */
3012 AST_LIST_HEAD_DESTROY(&session->supplements);
3013
3014 /* remove all saved media stats */
3015 AST_VECTOR_RESET(&session->media_stats, ast_free);
3016 AST_VECTOR_FREE(&session->media_stats);
3017
3019 ao2_cleanup(session->datastores);
3020 ast_sip_session_media_state_free(session->active_media_state);
3021 ast_sip_session_media_state_free(session->pending_media_state);
3022
3025 ao2_cleanup(session->endpoint);
3026 ao2_cleanup(session->aor);
3027 ao2_cleanup(session->contact);
3028 ao2_cleanup(session->direct_media_cap);
3029
3030 ast_dsp_free(session->dsp);
3031
3032 if (session->inv_session) {
3033 struct pjsip_dialog *dlg = session->inv_session->dlg;
3034
3035 /* The INVITE session uses the dialog pool for memory, so we need to
3036 * decrement its reference first before that of the dialog.
3037 */
3038
3039#ifdef HAVE_PJSIP_INV_SESSION_REF
3040 pjsip_inv_dec_ref(session->inv_session);
3041#endif
3042 pjsip_dlg_dec_session(dlg, &session_module);
3043 }
3044
3045 ast_test_suite_event_notify("SESSION_DESTROYED", "Endpoint: %s", endpoint_name);
3046}
3047
3048/*! \brief Destructor for SIP channel */
3049static void sip_channel_destroy(void *obj)
3050{
3051 struct ast_sip_channel_pvt *channel = obj;
3052
3053 ao2_cleanup(channel->pvt);
3054 ao2_cleanup(channel->session);
3055}
3056
3058{
3059 struct ast_sip_channel_pvt *channel = ao2_alloc(sizeof(*channel), sip_channel_destroy);
3060
3061 if (!channel) {
3062 return NULL;
3063 }
3064
3065 ao2_ref(pvt, +1);
3066 channel->pvt = pvt;
3067 ao2_ref(session, +1);
3068 channel->session = session;
3069
3070 return channel;
3071}
3072
3074 struct ast_sip_contact *contact, pjsip_inv_session *inv_session, pjsip_rx_data *rdata)
3075{
3077 struct ast_sip_session *ret_session;
3078 int dsp_features = 0;
3079
3081 if (!session) {
3082 return NULL;
3083 }
3084
3085 AST_LIST_HEAD_INIT(&session->supplements);
3086 AST_LIST_HEAD_INIT_NOLOCK(&session->delayed_requests);
3088
3090 if (!session->direct_media_cap) {
3091 return NULL;
3092 }
3095 if (!session->datastores) {
3096 return NULL;
3097 }
3098 session->active_media_state = ast_sip_session_media_state_alloc();
3099 if (!session->active_media_state) {
3100 return NULL;
3101 }
3102 session->pending_media_state = ast_sip_session_media_state_alloc();
3103 if (!session->pending_media_state) {
3104 return NULL;
3105 }
3106 if (AST_VECTOR_INIT(&session->media_stats, 1) < 0) {
3107 return NULL;
3108 }
3109
3111 dsp_features |= DSP_FEATURE_DIGIT_DETECT;
3112 }
3113 if (endpoint->faxdetect) {
3114 dsp_features |= DSP_FEATURE_FAX_DETECT;
3115 }
3116 if (dsp_features) {
3117 session->dsp = ast_dsp_new();
3118 if (!session->dsp) {
3119 return NULL;
3120 }
3121
3122 ast_dsp_set_features(session->dsp, dsp_features);
3123 }
3124
3125 session->endpoint = ao2_bump(endpoint);
3126
3127 if (rdata) {
3128 /*
3129 * We must continue using the serializer that the original
3130 * INVITE came in on for the dialog. There may be
3131 * retransmissions already enqueued in the original
3132 * serializer that can result in reentrancy and message
3133 * sequencing problems.
3134 */
3135 session->serializer = ast_sip_get_distributor_serializer(rdata);
3136 } else {
3137 /*
3138 * This is an outgoing session, so we can just choose a serializer
3139 * from the distributor pool based on the dialog.
3140 */
3142 }
3143 if (!session->serializer) {
3144 return NULL;
3145 }
3148
3149 /* When a PJSIP INVITE session is created it is created with a reference
3150 * count of 1, with that reference being managed by the underlying state
3151 * of the INVITE session itself. When the INVITE session transitions to
3152 * a DISCONNECTED state that reference is released. This means we can not
3153 * rely on that reference to ensure the INVITE session remains for the
3154 * lifetime of our session. To ensure it does we add our own reference
3155 * and release it when our own session goes away, ensuring that the INVITE
3156 * session remains for the lifetime of session.
3157 */
3158
3159#ifdef HAVE_PJSIP_INV_SESSION_REF
3160 if (pjsip_inv_add_ref(inv_session) != PJ_SUCCESS) {
3161 ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
3162 return NULL;
3163 }
3164#endif
3165
3166 pjsip_dlg_inc_session(inv_session->dlg, &session_module);
3167 inv_session->mod_data[session_module.id] = ao2_bump(session);
3168 session->contact = ao2_bump(contact);
3169 session->inv_session = inv_session;
3170
3171 session->dtmf = endpoint->dtmf;
3172 session->moh_passthrough = endpoint->moh_passthrough;
3173
3175 /* Release the ref held by session->inv_session */
3176 ao2_ref(session, -1);
3177 return NULL;
3178 }
3179
3180 session->authentication_challenge_count = 0;
3181
3182 /* Fire session begin handlers */
3184
3185 /* Avoid unnecessary ref manipulation to return a session */
3186 ret_session = session;
3187 session = NULL;
3188 return ret_session;
3189}
3190
3195
3200
3201/*!
3202 * \internal
3203 * \brief Handle initial INVITE challenge response message.
3204 * \since 13.5.0
3205 *
3206 * \param rdata PJSIP receive response message data.
3207 *
3208 * \retval PJ_FALSE Did not handle message.
3209 * \retval PJ_TRUE Handled message.
3210 */
3211static pj_bool_t outbound_invite_auth(pjsip_rx_data *rdata)
3212{
3213 pjsip_transaction *tsx;
3214 pjsip_dialog *dlg;
3215 pjsip_inv_session *inv;
3216 pjsip_tx_data *tdata;
3217 struct ast_sip_session *session;
3218
3219 if (rdata->msg_info.msg->line.status.code != 401
3220 && rdata->msg_info.msg->line.status.code != 407) {
3221 /* Doesn't pertain to us. Move on */
3222 return PJ_FALSE;
3223 }
3224
3225 tsx = pjsip_rdata_get_tsx(rdata);
3226 dlg = pjsip_rdata_get_dlg(rdata);
3227 if (!dlg || !tsx) {
3228 return PJ_FALSE;
3229 }
3230
3231 if (tsx->method.id != PJSIP_INVITE_METHOD) {
3232 /* Not an INVITE that needs authentication */
3233 return PJ_FALSE;
3234 }
3235
3236 inv = pjsip_dlg_get_inv_session(dlg);
3237 session = inv->mod_data[session_module.id];
3238
3239 if (PJSIP_INV_STATE_CONFIRMED <= inv->state) {
3240 /*
3241 * We cannot handle reINVITE authentication at this
3242 * time because the reINVITE transaction is still in
3243 * progress.
3244 */
3245 ast_debug(3, "%s: A reINVITE is being challenged\n", ast_sip_session_get_name(session));
3246 return PJ_FALSE;
3247 }
3248 ast_debug(3, "%s: Initial INVITE is being challenged.\n", ast_sip_session_get_name(session));
3249
3250 if (++session->authentication_challenge_count > MAX_RX_CHALLENGES) {
3251 ast_debug(3, "%s: Initial INVITE reached maximum number of auth attempts.\n", ast_sip_session_get_name(session));
3252 return PJ_FALSE;
3253 }
3254
3255 if (ast_sip_create_request_with_auth(&session->endpoint->outbound_auths, rdata,
3256 tsx->last_tx, &tdata)) {
3257 return PJ_FALSE;
3258 }
3259
3260 /*
3261 * Restart the outgoing initial INVITE transaction to deal
3262 * with authentication.
3263 */
3264 pjsip_inv_uac_restart(inv, PJ_FALSE);
3265
3267 return PJ_TRUE;
3268}
3269
3270static pjsip_module outbound_invite_auth_module = {
3271 .name = {"Outbound INVITE Auth", 20},
3272 .priority = PJSIP_MOD_PRIORITY_DIALOG_USAGE,
3273 .on_rx_response = outbound_invite_auth,
3274};
3275
3276/*!
3277 * \internal
3278 * \brief Setup outbound initial INVITE authentication.
3279 * \since 13.5.0
3280 *
3281 * \param dlg PJSIP dialog to attach outbound authentication.
3282 *
3283 * \retval 0 on success.
3284 * \retval -1 on error.
3285 */
3286static int setup_outbound_invite_auth(pjsip_dialog *dlg)
3287{
3288 pj_status_t status;
3289
3290 ++dlg->sess_count;
3291 status = pjsip_dlg_add_usage(dlg, &outbound_invite_auth_module, NULL);
3292 --dlg->sess_count;
3293
3294 return status != PJ_SUCCESS ? -1 : 0;
3295}
3296
3298 struct ast_sip_contact *contact, const char *location, const char *request_user,
3299 struct ast_stream_topology *req_topology)
3300{
3301 const char *uri = NULL;
3302 RAII_VAR(struct ast_sip_aor *, found_aor, NULL, ao2_cleanup);
3303 RAII_VAR(struct ast_sip_contact *, found_contact, NULL, ao2_cleanup);
3304 pjsip_timer_setting timer;
3305 pjsip_dialog *dlg;
3306 struct pjsip_inv_session *inv_session;
3308 struct ast_sip_session *ret_session;
3309 SCOPE_ENTER(1, "%s %s Topology: %s\n", ast_sorcery_object_get_id(endpoint), request_user,
3310 ast_str_tmp(256, ast_stream_topology_to_str(req_topology, &STR_TMP)));
3311
3313 request_user, req_topology)) {
3314 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Session creation blocked by supplement\n",
3316 }
3317
3318 /* If no location has been provided use the AOR list from the endpoint itself */
3319 if (location || !contact) {
3320 location = S_OR(location, endpoint->aors);
3321
3323 &found_aor, &found_contact);
3324 if (!found_contact || ast_strlen_zero(found_contact->uri)) {
3325 uri = location;
3326 } else {
3327 uri = found_contact->uri;
3328 }
3329 } else {
3330 uri = contact->uri;
3331 }
3332
3333 /* If we still have no URI to dial fail to create the session */
3334 if (ast_strlen_zero(uri)) {
3335 ast_log(LOG_ERROR, "Endpoint '%s': No URI available. Is endpoint registered?\n",
3337 SCOPE_EXIT_RTN_VALUE(NULL, "No URI\n");
3338 }
3339
3340 if (!(dlg = ast_sip_create_dialog_uac(endpoint, uri, request_user))) {
3341 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create dialog\n");
3342 }
3343
3344 if (setup_outbound_invite_auth(dlg)) {
3345 pjsip_dlg_terminate(dlg);
3346 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't setup auth\n");
3347 }
3348
3349 if (pjsip_inv_create_uac(dlg, NULL, endpoint->extensions.flags, &inv_session) != PJ_SUCCESS) {
3350 pjsip_dlg_terminate(dlg);
3351 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create uac\n");
3352 }
3353#if defined(HAVE_PJSIP_REPLACE_MEDIA_STREAM) || defined(PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE)
3354 inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
3355#endif
3356
3357 pjsip_timer_setting_default(&timer);
3359 timer.sess_expires = endpoint->extensions.timer.sess_expires;
3360 pjsip_timer_init_session(inv_session, &timer);
3361
3362 session = ast_sip_session_alloc(endpoint, found_contact ? found_contact : contact,
3363 inv_session, NULL);
3364 if (!session) {
3365 pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
3366 return NULL;
3367 }
3368 session->aor = ao2_bump(found_aor);
3369 session->call_direction = AST_SIP_SESSION_OUTGOING_CALL;
3370
3372
3373 if (ast_stream_topology_get_count(req_topology) > 0) {
3374 /* get joint caps between req_topology and endpoint topology */
3375 int i;
3376
3377 for (i = 0; i < ast_stream_topology_get_count(req_topology); ++i) {
3378 struct ast_stream *req_stream;
3379 struct ast_stream *clone_stream;
3380
3381 req_stream = ast_stream_topology_get_stream(req_topology, i);
3382
3384 continue;
3385 }
3386
3387 clone_stream = ast_sip_session_create_joint_call_stream(session, req_stream);
3388 if (!clone_stream || ast_stream_get_format_count(clone_stream) == 0) {
3389 ast_stream_free(clone_stream);
3390 continue;
3391 }
3392
3393 if (!session->pending_media_state->topology) {
3394 session->pending_media_state->topology = ast_stream_topology_alloc();
3395 if (!session->pending_media_state->topology) {
3396 pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
3397 ao2_ref(session, -1);
3398 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create topology\n");
3399 }
3400 }
3401
3402 if (ast_stream_topology_append_stream(session->pending_media_state->topology, clone_stream) < 0) {
3403 ast_stream_free(clone_stream);
3404 continue;
3405 }
3406 }
3407 }
3408
3409 if (!session->pending_media_state->topology) {
3410 /* Use the configured topology on the endpoint as the pending one */
3411 session->pending_media_state->topology = ast_stream_topology_clone(endpoint->media.topology);
3412 if (!session->pending_media_state->topology) {
3413 pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
3414 ao2_ref(session, -1);
3415 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't clone topology\n");
3416 }
3417 }
3418
3419 if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
3420 pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
3421 /* Since we are not notifying ourselves that the INVITE session is being terminated
3422 * we need to manually drop its reference to session
3423 */
3424 ao2_ref(session, -1);
3425 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't add usage\n");
3426 }
3427
3428 /* Avoid unnecessary ref manipulation to return a session */
3429 ret_session = session;
3430 session = NULL;
3431 SCOPE_EXIT_RTN_VALUE(ret_session);
3432}
3433
3434static int session_end(void *vsession);
3435static int session_end_completion(void *vsession);
3436
3438{
3439 pj_status_t status;
3440 pjsip_tx_data *packet = NULL;
3441 SCOPE_ENTER(1, "%s Response %d\n", ast_sip_session_get_name(session), response);
3442
3443 if (session->defer_terminate) {
3444 session->terminate_while_deferred = 1;
3445 SCOPE_EXIT_RTN("Deferred\n");
3446 }
3447
3448 if (!response) {
3449 response = 603;
3450 }
3451
3452 /* The media sessions need to exist for the lifetime of the underlying channel
3453 * to ensure that anything (such as bridge_native_rtp) has access to them as
3454 * appropriate. Since ast_sip_session_terminate is called by chan_pjsip and other
3455 * places when the session is to be terminated we terminate any existing
3456 * media sessions here.
3457 */
3458 ast_sip_session_media_stats_save(session, session->active_media_state);
3459 SWAP(session->active_media_state, session->pending_media_state);
3460 ast_sip_session_media_state_reset(session->pending_media_state);
3461
3462 switch (session->inv_session->state) {
3463 case PJSIP_INV_STATE_NULL:
3464 if (!session->inv_session->invite_tsx) {
3465 /*
3466 * Normally, it's pjproject's transaction cleanup that ultimately causes the
3467 * final session reference to be released but if both STATE and invite_tsx are NULL,
3468 * we never created a transaction in the first place. In this case, we need to
3469 * do the cleanup ourselves.
3470 */
3471 /* Transfer the inv_session session reference to the session_end_task */
3472 session->inv_session->mod_data[session_module.id] = NULL;
3473 pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
3475 /*
3476 * session_end_completion will cleanup the final session reference unless
3477 * ast_sip_session_terminate's caller is holding one.
3478 */
3480 } else {
3481 pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
3482 }
3483 break;
3484 case PJSIP_INV_STATE_CONFIRMED:
3485 if (session->inv_session->invite_tsx) {
3486 ast_debug(3, "%s: Delay sending BYE because of outstanding transaction...\n",
3488 /*
3489 * If this is delayed the only thing that will happen is a BYE request, so
3490 * no response code needs to be stored. Queue the BYE as before, then arm
3491 * a transaction timeout so a malformed/lost final re-INVITE response
3492 * cannot leave the session and RTP state referenced forever.
3493 */
3495 ast_log(LOG_ERROR, "%s: Unable to delay BYE request\n",
3498 session->terminate_on_invite_timeout = 1;
3499 }
3500 break;
3501 }
3502 /* Fall through */
3503 default:
3504 status = pjsip_inv_end_session(session->inv_session, response, NULL, &packet);
3505 if (status == PJ_SUCCESS && packet) {
3506 /* Flush any delayed requests so they cannot overlap this transaction. */
3508
3509 if (packet->msg->type == PJSIP_RESPONSE_MSG) {
3511 } else {
3513 }
3514 }
3515 break;
3516 }
3518}
3519
3520static int session_termination_task(void *data)
3521{
3522 struct ast_sip_session *session = data;
3523
3524 if (session->defer_terminate) {
3525 session->defer_terminate = 0;
3526 if (session->inv_session) {
3528 }
3529 }
3530
3531 ao2_ref(session, -1);
3532 return 0;
3533}
3534
3535static void session_termination_cb(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry)
3536{
3537 struct ast_sip_session *session = entry->user_data;
3538
3541 }
3542}
3543
3545{
3546 pj_time_val delay = { .sec = 60, };
3547 int res;
3548
3549 /* The session should not have an active deferred termination request. */
3550 ast_assert(!session->defer_terminate);
3551
3552 session->defer_terminate = 1;
3553
3554 session->defer_end = 1;
3555 session->ended_while_deferred = 0;
3556
3557 ao2_ref(session, +1);
3558 pj_timer_entry_init(&session->scheduled_termination, 0, session, session_termination_cb);
3559
3560 res = (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(),
3561 &session->scheduled_termination, &delay) != PJ_SUCCESS) ? -1 : 0;
3562 if (res) {
3563 session->defer_terminate = 0;
3564 ao2_ref(session, -1);
3565 }
3566 return res;
3567}
3568
3569/*!
3570 * \internal
3571 * \brief Stop the defer termination timer if it is still running.
3572 * \since 13.5.0
3573 *
3574 * \param session Which session to stop the timer.
3575 */
3577{
3578 if (pj_timer_heap_cancel_if_active(pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()),
3579 &session->scheduled_termination, session->scheduled_termination.id)) {
3580 ao2_ref(session, -1);
3581 }
3582}
3583
3585{
3586 if (!session->defer_terminate) {
3587 /* Already canceled or timer fired. */
3588 return;
3589 }
3590
3591 session->defer_terminate = 0;
3592
3593 if (session->terminate_while_deferred) {
3594 /* Complete the termination started by the upper layer. */
3596 }
3597
3598 /* Stop the termination timer if it is still running. */
3600}
3601
3603{
3604 if (!session->defer_end) {
3605 return;
3606 }
3607
3608 session->defer_end = 0;
3609
3610 if (session->ended_while_deferred) {
3611 /* Complete the session end started by the remote hangup. */
3612 ast_debug(3, "%s: Ending session after being deferred\n", ast_sip_session_get_name(session));
3613 session->ended_while_deferred = 0;
3615 }
3616}
3617
3619{
3620 pjsip_inv_session *inv_session = pjsip_dlg_get_inv_session(dlg);
3621 struct ast_sip_session *session;
3622
3623 if (!inv_session ||
3624 !(session = inv_session->mod_data[session_module.id])) {
3625 return NULL;
3626 }
3627
3628 ao2_ref(session, +1);
3629
3630 return session;
3631}
3632
3634{
3635 pjsip_inv_session *inv_session = session->inv_session;
3636
3637 if (!inv_session) {
3638 return NULL;
3639 }
3640
3641 return inv_session->dlg;
3642}
3643
3645{
3646 pjsip_inv_session *inv_session = session->inv_session;
3647
3648 if (!inv_session) {
3649 return PJSIP_INV_STATE_NULL;
3650 }
3651
3652 return inv_session->state;
3653}
3654
3655/*! \brief Fetch just the Caller ID number in order of PAI, RPID, From */
3656static int fetch_callerid_num(struct ast_sip_session *session, pjsip_rx_data *rdata, char *buf, size_t len)
3657{
3658 int res = -1;
3659 struct ast_party_id id;
3660
3661 ast_party_id_init(&id);
3662 if (!ast_sip_set_id_from_invite(rdata, &id, &session->endpoint->id.self, session->endpoint->id.trust_inbound)) {
3663 ast_copy_string(buf, id.number.str, len);
3664 res = 0;
3665 }
3666 ast_party_id_free(&id);
3667 return res;
3668}
3669
3671 /*! The extension was successfully found */
3673 /*! The extension specified in the RURI was not found */
3675 /*! The extension specified in the RURI was a partial match */
3677 /*! The RURI is of an unsupported scheme */
3679};
3680
3681/*!
3682 * \brief Determine where in the dialplan a call should go
3683 *
3684 * This uses the username in the request URI to try to match
3685 * an extension in the endpoint's configured context in order
3686 * to route the call.
3687 *
3688 * \param session The inbound SIP session
3689 * \param rdata The SIP INVITE
3690 */
3691static enum sip_get_destination_result get_destination(struct ast_sip_session *session, pjsip_rx_data *rdata)
3692{
3693 char cid_num[AST_CHANNEL_NAME];
3694 pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
3695 struct ast_features_pickup_config *pickup_cfg;
3696 const char *pickupexten;
3697
3698 if (!ast_sip_is_allowed_uri(ruri)) {
3700 }
3701
3702 ast_copy_pj_str(session->exten, ast_sip_pjsip_uri_get_username(ruri), sizeof(session->exten));
3703 if (ast_strlen_zero(session->exten)) {
3704 /* Some SIP devices send an empty extension for PLAR: this should map to s */
3705 ast_debug(1, "RURI contains no user portion: defaulting to extension 's'\n");
3706 ast_copy_string(session->exten, "s", sizeof(session->exten));
3707 }
3708
3709 /*
3710 * We may want to match in the dialplan without any user
3711 * options getting in the way.
3712 */
3714
3715 pickup_cfg = ast_get_chan_features_pickup_config(NULL); /* session->channel doesn't exist yet, using NULL */
3716 if (!pickup_cfg) {
3717 ast_log(LOG_ERROR, "%s: Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n",
3719 pickupexten = "";
3720 } else {
3721 pickupexten = ast_strdupa(pickup_cfg->pickupexten);
3722 ao2_ref(pickup_cfg, -1);
3723 }
3724
3725 fetch_callerid_num(session, rdata, cid_num, sizeof(cid_num));
3726
3727 /* If there's an overlap_context override specified, use that; otherwise, just use the endpoint's context */
3728
3729 if (!strcmp(session->exten, pickupexten) ||
3730 ast_exists_extension(NULL, S_OR(session->endpoint->overlap_context, session->endpoint->context), session->exten, 1, S_OR(cid_num, NULL))) {
3731 /*
3732 * Save off the INVITE Request-URI in case it is
3733 * needed: CHANNEL(pjsip,request_uri)
3734 */
3735 session->request_uri = pjsip_uri_clone(session->inv_session->pool, ruri);
3736
3738 }
3739
3740 /*
3741 * Check for partial match via overlap dialling (if enabled)
3742 */
3743 if (session->endpoint->allow_overlap && (
3744 !strncmp(session->exten, pickupexten, strlen(session->exten)) ||
3745 ast_canmatch_extension(NULL, S_OR(session->endpoint->overlap_context, session->endpoint->context), session->exten, 1, S_OR(cid_num, NULL)))) {
3746 /* Overlap partial match */
3748 }
3749
3751}
3752
3753/*!
3754 * \internal
3755 * \brief Process initial answer for an incoming invite
3756 *
3757 * This function should only be called during the setup, and handling of a
3758 * new incoming invite. Most, if not all of the time, this will be called
3759 * when an error occurs and we need to respond as such.
3760 *
3761 * When a SIP session termination code is given for the answer it's assumed
3762 * this call then will be the final bit of processing before ending session
3763 * setup. As such, we've been holding a lock, and a reference on the invite
3764 * session's dialog. So before returning this function removes that reference,
3765 * and unlocks the dialog.
3766 *
3767 * \param inv_session The session on which to answer
3768 * \param rdata The original request
3769 * \param answer_code The answer's numeric code
3770 * \param terminate_code The termination code if the answer fails
3771 * \param notify Whether or not to call on_state_changed
3772 *
3773 * \retval 0 if invite successfully answered, -1 if an error occurred
3774 */
3775static int new_invite_initial_answer(pjsip_inv_session *inv_session, pjsip_rx_data *rdata,
3776 int answer_code, int terminate_code, pj_bool_t notify)
3777{
3778 pjsip_tx_data *tdata = NULL;
3779 int res = 0;
3780
3781 if (inv_session->state != PJSIP_INV_STATE_DISCONNECTED) {
3782 if (pjsip_inv_initial_answer(
3783 inv_session, rdata, answer_code, NULL, NULL, &tdata) != PJ_SUCCESS) {
3784
3785 pjsip_inv_terminate(inv_session, terminate_code ? terminate_code : answer_code, notify);
3786 res = -1;
3787 } else {
3788 pjsip_inv_send_msg(inv_session, tdata);
3789 }
3790 }
3791
3792 if (answer_code >= 300) {
3793 /*
3794 * A session is ending. The dialog has a reference that needs to be
3795 * removed and holds a lock that needs to be unlocked before returning.
3796 */
3797 pjsip_dlg_dec_lock(inv_session->dlg);
3798 }
3799
3800 return res;
3801}
3802
3803/*!
3804 * \internal
3805 * \brief Create and initialize a pjsip invite session
3806 *
3807 * pjsip_inv_session adds, and maintains a reference to the dialog upon a successful
3808 * invite session creation until the session is destroyed. However, we'll wait to
3809 * remove the reference that was added for the dialog when it gets created since we're
3810 * not ready to unlock the dialog in this function.
3811 *
3812 * So, if this function successfully returns that means it returns with its newly
3813 * created, and associated dialog locked and with two references (i.e. dialog's
3814 * reference count should be 2).
3815 *
3816 * \param rdata The request that is starting the dialog
3817 * \param endpoint A pointer to the endpoint
3818 *
3819 * \return A pjsip invite session object
3820 * \retval NULL on error
3821 */
3822static pjsip_inv_session *pre_session_setup(pjsip_rx_data *rdata, const struct ast_sip_endpoint *endpoint)
3823{
3824 pjsip_tx_data *tdata;
3825 pjsip_dialog *dlg;
3826 pjsip_inv_session *inv_session;
3827 unsigned int options = endpoint->extensions.flags;
3828 const pj_str_t STR_100REL = { "100rel", 6};
3829 unsigned int i;
3830 pj_status_t dlg_status = PJ_EUNKNOWN;
3831
3832 /*
3833 * If 100rel is set to "peer_supported" on the endpoint and the peer indicated support for 100rel
3834 * in the Supported header, send 1xx responses reliably by adding PJSIP_INV_REQUIRE_100REL to pjsip_inv_options flags.
3835 */
3836 if (endpoint->rel100 == AST_SIP_100REL_PEER_SUPPORTED && rdata->msg_info.supported != NULL) {
3837 for (i = 0; i < rdata->msg_info.supported->count; ++i) {
3838 if (pj_stricmp(&rdata->msg_info.supported->values[i], &STR_100REL) == 0) {
3839 options |= PJSIP_INV_REQUIRE_100REL;
3840 break;
3841 }
3842 }
3843 }
3844
3845 if (pjsip_inv_verify_request(rdata, &options, NULL, NULL, ast_sip_get_pjsip_endpoint(), &tdata) != PJ_SUCCESS) {
3846 if (tdata) {
3847 if (pjsip_endpt_send_response2(ast_sip_get_pjsip_endpoint(), rdata, tdata, NULL, NULL) != PJ_SUCCESS) {
3848 pjsip_tx_data_dec_ref(tdata);
3849 }
3850 } else {
3851 pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
3852 }
3853 return NULL;
3854 }
3855
3856 dlg = ast_sip_create_dialog_uas_locked(endpoint, rdata, &dlg_status);
3857 if (!dlg) {
3858 if (dlg_status != PJ_EEXISTS) {
3859 pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
3860 }
3861 return NULL;
3862 }
3863
3864 /*
3865 * The returned dialog holds a lock and has a reference added. Any paths where the
3866 * dialog invite session is not returned must unlock the dialog and remove its reference.
3867 */
3868
3869 if (pjsip_inv_create_uas(dlg, rdata, NULL, options, &inv_session) != PJ_SUCCESS) {
3870 pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
3871 /*
3872 * The acquired dialog holds a lock, and a reference. Since the dialog is not
3873 * going to be returned here it must first be unlocked and de-referenced. This
3874 * must be done prior to calling dialog termination.
3875 */
3876 pjsip_dlg_dec_lock(dlg);
3877 pjsip_dlg_terminate(dlg);
3878 return NULL;
3879 }
3880
3881#if defined(HAVE_PJSIP_REPLACE_MEDIA_STREAM) || defined(PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE)
3882 inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
3883#endif
3884 if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
3885 /* Dialog's lock and a reference are removed in new_invite_initial_answer */
3886 new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE);
3887 /* Remove 2nd reference added at inv_session creation */
3888 pjsip_dlg_dec_session(inv_session->dlg, &session_module);
3889 return NULL;
3890 }
3891
3892 return inv_session;
3893}
3894
3896 /*! \brief Session created for the new INVITE */
3898
3899 /*! \brief INVITE request itself */
3900 pjsip_rx_data *rdata;
3901};
3902
3903static int check_sdp_content_type_supported(pjsip_media_type *content_type)
3904{
3905 pjsip_media_type app_sdp;
3906 pjsip_media_type_init2(&app_sdp, "application", "sdp");
3907
3908 if (!pjsip_media_type_cmp(content_type, &app_sdp, 0)) {
3909 return 1;
3910 }
3911
3912 return 0;
3913}
3914
3915static int check_content_disposition_in_multipart(pjsip_multipart_part *part)
3916{
3917 pjsip_hdr *hdr = part->hdr.next;
3918 static const pj_str_t str_handling_required = {"handling=required", 16};
3919
3920 while (hdr != &part->hdr) {
3921 if (hdr->type == PJSIP_H_OTHER) {
3922 pjsip_generic_string_hdr *generic_hdr = (pjsip_generic_string_hdr*)hdr;
3923
3924 if (!pj_stricmp2(&hdr->name, "Content-Disposition") &&
3925 pj_stristr(&generic_hdr->hvalue, &str_handling_required) &&
3926 !check_sdp_content_type_supported(&part->body->content_type)) {
3927 return 1;
3928 }
3929 }
3930 hdr = hdr->next;
3931 }
3932
3933 return 0;
3934}
3935
3936/**
3937 * if there is required media we don't understand, return 1
3938 */
3939static int check_content_disposition(pjsip_rx_data *rdata)
3940{
3941 pjsip_msg_body *body = rdata->msg_info.msg->body;
3942 pjsip_ctype_hdr *ctype_hdr = rdata->msg_info.ctype;
3943
3944 if (body && ctype_hdr &&
3947 pjsip_multipart_part *part = pjsip_multipart_get_first_part(body);
3948 while (part != NULL) {
3950 return 1;
3951 }
3952 part = pjsip_multipart_get_next_part(body, part);
3953 }
3954 }
3955 return 0;
3956}
3957
3958static int new_invite(struct new_invite *invite)
3959{
3960 pjsip_tx_data *tdata = NULL;
3961 pjsip_timer_setting timer;
3962 pjsip_rdata_sdp_info *sdp_info;
3963 pjmedia_sdp_session *local = NULL;
3964 char buffer[AST_SOCKADDR_BUFLEN];
3965 SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(invite->session));
3966
3967
3968 /* From this point on, any calls to pjsip_inv_terminate have the last argument as PJ_TRUE
3969 * so that we will be notified so we can destroy the session properly
3970 */
3971
3972 if (invite->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
3973 ast_trace_log(-1, LOG_ERROR, "%s: Session already DISCONNECTED [reason=%d (%s)]\n",
3975 invite->session->inv_session->cause,
3976 pjsip_get_status_text(invite->session->inv_session->cause)->ptr);
3978 }
3979
3980 switch (get_destination(invite->session, invite->rdata)) {
3982 /* Things worked. Keep going */
3983 break;
3985 ast_trace(-1, "%s: Call (%s:%s) to extension '%s' - unsupported uri\n",
3987 invite->rdata->tp_info.transport->type_name,
3988 pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
3989 invite->session->exten);
3990 if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 416, NULL, NULL, &tdata) == PJ_SUCCESS) {
3991 ast_sip_session_send_response(invite->session, tdata);
3992 } else {
3993 pjsip_inv_terminate(invite->session->inv_session, 416, PJ_TRUE);
3994 }
3995 goto end;
3997 ast_trace(-1, "%s: Call (%s:%s) to extension '%s' - partial match\n",
3999 invite->rdata->tp_info.transport->type_name,
4000 pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
4001 invite->session->exten);
4002
4003 if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 484, NULL, NULL, &tdata) == PJ_SUCCESS) {
4004 ast_sip_session_send_response(invite->session, tdata);
4005 } else {
4006 pjsip_inv_terminate(invite->session->inv_session, 484, PJ_TRUE);
4007 }
4008 goto end;
4010 default:
4011 ast_trace_log(-1, LOG_NOTICE, "%s: Call (%s:%s) to extension '%s' rejected because extension not found in context '%s'.\n",
4013 invite->rdata->tp_info.transport->type_name,
4014 pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
4015 invite->session->exten,
4016 invite->session->endpoint->context);
4017
4018 if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 404, NULL, NULL, &tdata) == PJ_SUCCESS) {
4019 ast_sip_session_send_response(invite->session, tdata);
4020 } else {
4021 pjsip_inv_terminate(invite->session->inv_session, 404, PJ_TRUE);
4022 }
4023 goto end;
4024 };
4025
4026 if (check_content_disposition(invite->rdata)) {
4027 if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 415, NULL, NULL, &tdata) == PJ_SUCCESS) {
4028 ast_sip_session_send_response(invite->session, tdata);
4029 } else {
4030 pjsip_inv_terminate(invite->session->inv_session, 415, PJ_TRUE);
4031 }
4032 goto end;
4033 }
4034
4035 pjsip_timer_setting_default(&timer);
4036 timer.min_se = invite->session->endpoint->extensions.timer.min_se;
4037 timer.sess_expires = invite->session->endpoint->extensions.timer.sess_expires;
4038 pjsip_timer_init_session(invite->session->inv_session, &timer);
4039
4040 /*
4041 * At this point, we've verified what we can that won't take awhile,
4042 * so let's go ahead and send a 100 Trying out to stop any
4043 * retransmissions.
4044 */
4045 if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 100, NULL, NULL, &tdata) != PJ_SUCCESS) {
4046 if (tdata) {
4047 pjsip_inv_send_msg(invite->session->inv_session, tdata);
4048 } else {
4049 pjsip_inv_terminate(invite->session->inv_session, 500, PJ_TRUE);
4050 }
4051 goto end;
4052 }
4053
4054 ast_trace(-1, "%s: Call (%s:%s) to extension '%s' sending 100 Trying\n",
4056 invite->rdata->tp_info.transport->type_name,
4057 pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
4058 invite->session->exten);
4059 ast_sip_session_send_response(invite->session, tdata);
4060
4061 sdp_info = pjsip_rdata_get_sdp_info(invite->rdata);
4062 if (sdp_info && (sdp_info->sdp_err == PJ_SUCCESS) && sdp_info->sdp) {
4063 if (handle_incoming_sdp(invite->session, sdp_info->sdp)) {
4064 tdata = NULL;
4065 if (pjsip_inv_end_session(invite->session->inv_session, 488, NULL, &tdata) == PJ_SUCCESS
4066 && tdata) {
4067 ast_sip_session_send_response(invite->session, tdata);
4068 }
4069 goto end;
4070 }
4071 /* We are creating a local SDP which is an answer to their offer */
4072 local = create_local_sdp(invite->session->inv_session, invite->session, sdp_info->sdp, 0);
4073 } else {
4074 /* We are creating a local SDP which is an offer */
4075 local = create_local_sdp(invite->session->inv_session, invite->session, NULL, 0);
4076 }
4077
4078 /* If we were unable to create a local SDP terminate the session early, it won't go anywhere */
4079 if (!local) {
4080 tdata = NULL;
4081 if (pjsip_inv_end_session(invite->session->inv_session, 500, NULL, &tdata) == PJ_SUCCESS
4082 && tdata) {
4083 ast_sip_session_send_response(invite->session, tdata);
4084 }
4085 goto end;
4086 }
4087
4088 pjsip_inv_set_local_sdp(invite->session->inv_session, local);
4089 pjmedia_sdp_neg_set_prefer_remote_codec_order(invite->session->inv_session->neg, PJ_FALSE);
4090#ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
4091 if (!invite->session->endpoint->preferred_codec_only) {
4092 pjmedia_sdp_neg_set_answer_multiple_codecs(invite->session->inv_session->neg, PJ_TRUE);
4093 }
4094#endif
4095
4096 handle_incoming_request(invite->session, invite->rdata);
4097
4098end:
4100}
4101
4102static void handle_new_invite_request(pjsip_rx_data *rdata)
4103{
4106 pjsip_inv_session *inv_session = NULL;
4107 struct ast_sip_session *session;
4108 struct new_invite invite;
4109 char *req_uri = TRACE_ATLEAST(1) ? ast_alloca(256) : "";
4110 int res = TRACE_ATLEAST(1) ? pjsip_uri_print(PJSIP_URI_IN_REQ_URI, rdata->msg_info.msg->line.req.uri, req_uri, 256) : 0;
4111 SCOPE_ENTER(1, "Request: %s\n", res ? req_uri : "");
4112
4113 ast_assert(endpoint != NULL);
4114
4115 inv_session = pre_session_setup(rdata, endpoint);
4116 if (!inv_session) {
4117 /* pre_session_setup() returns a response on failure */
4118 SCOPE_EXIT_RTN("Failure in pre session setup\n");
4119 }
4120
4121 /*
4122 * Upon a successful pre_session_setup the associated dialog is returned locked
4123 * and with an added reference. Well actually two references. One added when the
4124 * dialog itself was created, and another added when the pjsip invite session was
4125 * created and the dialog was added to it.
4126 *
4127 * In order to ensure the dialog's, and any of its internal attributes, lifetimes
4128 * we'll hold the lock and maintain the reference throughout the entire new invite
4129 * handling process. See ast_sip_create_dialog_uas_locked for more details but,
4130 * basically we do this to make sure a transport failure does not destroy the dialog
4131 * and/or transaction out from underneath us between pjsip calls. Alternatively, we
4132 * could probably release the lock if we needed to, but then we'd have to re-lock and
4133 * check the dialog and transaction prior to every pjsip call.
4134 *
4135 * That means any off nominal/failure paths in this function must remove the associated
4136 * dialog reference added at dialog creation, and remove the lock. As well the
4137 * referenced pjsip invite session must be "cleaned up", which should also then
4138 * remove its reference to the dialog at that time.
4139 *
4140 * Nominally we'll unlock the dialog, and release the reference when all new invite
4141 * process handling has successfully completed.
4142 */
4143
4144 session = ast_sip_session_alloc(endpoint, NULL, inv_session, rdata);
4145 if (!session) {
4146 /* Dialog's lock and reference are removed in new_invite_initial_answer */
4147 if (!new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE)) {
4148 /* Terminate the session if it wasn't done in the answer */
4149 pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
4150 }
4151 SCOPE_EXIT_RTN("Couldn't create session\n");
4152 }
4153 session->call_direction = AST_SIP_SESSION_INCOMING_CALL;
4154
4155 /*
4156 * The current thread is supposed be the session serializer to prevent
4157 * any initial INVITE retransmissions from trying to setup the same
4158 * call again.
4159 */
4161
4162 invite.session = session;
4163 invite.rdata = rdata;
4164 new_invite(&invite);
4165
4166 /*
4167 * The dialog lock and reference added at dialog creation time must be
4168 * maintained throughout the new invite process. Since we're pretty much
4169 * done at this point with things it's safe to go ahead and remove the lock
4170 * and the reference here. See ast_sip_create_dialog_uas_locked for more info.
4171 *
4172 * Note, any future functionality added that does work using the dialog must
4173 * be done before this.
4174 */
4175 pjsip_dlg_dec_lock(inv_session->dlg);
4176
4177 SCOPE_EXIT("Request: %s Session: %s\n", req_uri, ast_sip_session_get_name(session));
4178 ao2_ref(session, -1);
4179}
4180
4181static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
4182{
4183 pj_str_t method;
4184
4185 if (ast_strlen_zero(supplement_method)) {
4186 return PJ_TRUE;
4187 }
4188
4189 pj_cstr(&method, supplement_method);
4190
4191 return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
4192}
4193
4194static pj_bool_t has_supplement(const struct ast_sip_session *session, const pjsip_rx_data *rdata)
4195{
4196 struct ast_sip_session_supplement *supplement;
4197 struct pjsip_method *method = &rdata->msg_info.msg->line.req.method;
4198
4199 if (!session) {
4200 return PJ_FALSE;
4201 }
4202
4203 AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
4204 if (does_method_match(&method->name, supplement->method)) {
4205 return PJ_TRUE;
4206 }
4207 }
4208 return PJ_FALSE;
4209}
4210
4211/*!
4212 * \internal
4213 * Added for debugging purposes
4214 */
4215static void session_on_tsx_state(pjsip_transaction *tsx, pjsip_event *e)
4216{
4217
4218 pjsip_dialog *dlg = pjsip_tsx_get_dlg(tsx);
4219 pjsip_inv_session *inv_session = (dlg ? pjsip_dlg_get_inv_session(dlg) : NULL);
4220 struct ast_sip_session *session = (inv_session ? inv_session->mod_data[session_module.id] : NULL);
4221 SCOPE_ENTER(1, "%s TSX State: %s Inv State: %s\n", ast_sip_session_get_name(session),
4222 pjsip_tsx_state_str(tsx->state), inv_session ? pjsip_inv_state_name(inv_session->state) : "unknown");
4223
4224 if (session) {
4225 ast_trace(2, "Topology: Pending: %s Active: %s\n",
4226 ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)),
4227 ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
4228 }
4229
4231}
4232
4233/*!
4234 * \internal
4235 * Added for debugging purposes
4236 */
4237static pj_bool_t session_on_rx_response(pjsip_rx_data *rdata)
4238{
4239
4240 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
4241 pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
4242 pjsip_inv_session *inv_session = dlg ? pjsip_dlg_get_inv_session(dlg) : NULL;
4243 struct ast_sip_session *session = (inv_session ? inv_session->mod_data[session_module.id] : NULL);
4244 SCOPE_ENTER(1, "%s Method: %.*s Status: %d\n", ast_sip_session_get_name(session),
4245 (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
4246
4247 SCOPE_EXIT_RTN_VALUE(PJ_FALSE);
4248}
4249
4250/*!
4251 * \brief Called when a new SIP request comes into PJSIP
4252 *
4253 * This function is called under two circumstances
4254 * 1) An out-of-dialog request is received by PJSIP
4255 * 2) An in-dialog request that the inv_session layer does not
4256 * handle is received (such as an in-dialog INFO)
4257 *
4258 * Except for INVITEs, there is very little we actually do in this function
4259 * 1) For requests we don't handle, we return PJ_FALSE
4260 * 2) For new INVITEs, handle them now to prevent retransmissions from
4261 * trying to setup the same call again.
4262 * 3) For in-dialog requests we handle, we process them in the
4263 * .on_state_changed = session_inv_on_state_changed or
4264 * .on_tsx_state_changed = session_inv_on_tsx_state_changed
4265 * callbacks instead.
4266 */
4267static pj_bool_t session_on_rx_request(pjsip_rx_data *rdata)
4268{
4269 pj_status_t handled = PJ_FALSE;
4270 struct pjsip_request_line req = rdata->msg_info.msg->line.req;
4271 pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
4272 pjsip_inv_session *inv_session = (dlg ? pjsip_dlg_get_inv_session(dlg) : NULL);
4273 struct ast_sip_session *session = (inv_session ? inv_session->mod_data[session_module.id] : NULL);
4274 char *req_uri = TRACE_ATLEAST(1) ? ast_alloca(256) : "";
4275 int res = TRACE_ATLEAST(1) ? pjsip_uri_print(PJSIP_URI_IN_REQ_URI, rdata->msg_info.msg->line.req.uri, req_uri, 256) : 0;
4276 SCOPE_ENTER(1, "%s Request: %.*s %s\n", ast_sip_session_get_name(session),
4277 (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name), res ? req_uri : "");
4278
4279 switch (req.method.id) {
4280 case PJSIP_INVITE_METHOD:
4281 if (dlg) {
4282 ast_log(LOG_WARNING, "on_rx_request called for INVITE in mid-dialog?\n");
4283 break;
4284 }
4285 handled = PJ_TRUE;
4287 break;
4288 default:
4289 /* Handle other in-dialog methods if their supplements have been registered */
4290 handled = dlg && (inv_session = pjsip_dlg_get_inv_session(dlg)) &&
4291 has_supplement(inv_session->mod_data[session_module.id], rdata);
4292 break;
4293 }
4294
4295 SCOPE_EXIT_RTN_VALUE(handled, "%s Handled request %.*s %s ? %s\n", ast_sip_session_get_name(session),
4296 (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name), req_uri,
4297 handled == PJ_TRUE ? "yes" : "no");
4298}
4299
4300
4301static pj_bool_t session_on_tx_response(pjsip_tx_data *tdata)
4302{
4303 pjsip_dialog *dlg = pjsip_tdata_get_dlg(tdata);
4305 if (session) {
4307 }
4308
4309 return PJ_SUCCESS;
4310}
4311
4312static void resend_reinvite(pj_timer_heap_t *timer, pj_timer_entry *entry)
4313{
4314 struct ast_sip_session *session = entry->user_data;
4315
4316 ast_debug(3, "%s: re-INVITE collision timer expired.\n",
4318
4319 if (AST_LIST_EMPTY(&session->delayed_requests)) {
4320 /* No delayed request pending, so just return */
4321 ao2_ref(session, -1);
4322 return;
4323 }
4325 /*
4326 * Uh oh. We now have nothing in the foreseeable future
4327 * to trigger sending the delayed requests.
4328 */
4329 ao2_ref(session, -1);
4330 }
4331}
4332
4334{
4335 pjsip_inv_session *inv = session->inv_session;
4336 pj_time_val tv;
4337 struct ast_sip_session_media_state *pending_media_state = NULL;
4338 struct ast_sip_session_media_state *active_media_state = NULL;
4339 const char *session_name = ast_sip_session_get_name(session);
4340 int use_pending = 0;
4341 int use_active = 0;
4342
4343 SCOPE_ENTER(3, "%s\n", session_name);
4344
4345 /*
4346 * If the two media state topologies are the same this means that the session refresh request
4347 * did not specify a desired topology, so it does not care. If that is the case we don't even
4348 * pass one in here resulting in the current topology being used. It's possible though that
4349 * either one of the topologies could be NULL so we have to test for that before we check for
4350 * equality.
4351 */
4352
4353 /* We only want to clone a media state if its topology is not null */
4354 use_pending = session->pending_media_state->topology != NULL;
4355 use_active = session->active_media_state->topology != NULL;
4356
4357 /*
4358 * If both media states have topologies, we can test for equality. If they're equal we're not going to
4359 * clone either states.
4360 */
4361 if (use_pending && use_active && ast_stream_topology_equal(session->active_media_state->topology, session->pending_media_state->topology)) {
4362 use_pending = 0;
4363 use_active = 0;
4364 }
4365
4366 if (use_pending) {
4367 pending_media_state = ast_sip_session_media_state_clone(session->pending_media_state);
4368 if (!pending_media_state) {
4369 SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Failed to clone pending media state\n", session_name);
4370 }
4371 }
4372
4373 if (use_active) {
4374 active_media_state = ast_sip_session_media_state_clone(session->active_media_state);
4375 if (!active_media_state) {
4376 ast_sip_session_media_state_free(pending_media_state);
4377 SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Failed to clone active media state\n", session_name);
4378 }
4379 }
4380
4381 if (delay_request(session, NULL, NULL, on_response, 1, DELAYED_METHOD_INVITE, pending_media_state,
4382 active_media_state, 1)) {
4383 ast_sip_session_media_state_free(pending_media_state);
4384 ast_sip_session_media_state_free(active_media_state);
4385 SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Failed to add delayed request\n", session_name);
4386 }
4387
4388 if (pj_timer_entry_running(&session->rescheduled_reinvite)) {
4389 /* Timer already running. Something weird is going on. */
4390 SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: re-INVITE collision while timer running!!!\n", session_name);
4391 }
4392
4393 tv.sec = 0;
4394 if (inv->role == PJSIP_ROLE_UAC) {
4395 tv.msec = 2100 + ast_random() % 2000;
4396 } else {
4397 tv.msec = ast_random() % 2000;
4398 }
4399 pj_timer_entry_init(&session->rescheduled_reinvite, 0, session, resend_reinvite);
4400
4401 ao2_ref(session, +1);
4402 if (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(),
4403 &session->rescheduled_reinvite, &tv) != PJ_SUCCESS) {
4404 ao2_ref(session, -1);
4405 SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Couldn't schedule timer\n", session_name);
4406 }
4407
4409}
4410
4411static void __print_debug_details(const char *function, pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
4412{
4413 int id = session_module.id;
4414 struct ast_sip_session *session = NULL;
4415
4416 if (!DEBUG_ATLEAST(5)) {
4417 /* Debug not spamy enough */
4418 return;
4419 }
4420
4421 ast_log(LOG_DEBUG, "Function %s called on event %s\n",
4422 function, pjsip_event_str(e->type));
4423 if (!inv) {
4424 ast_log(LOG_DEBUG, "Transaction %p does not belong to an inv_session?\n", tsx);
4425 ast_log(LOG_DEBUG, "The transaction state is %s\n",
4426 pjsip_tsx_state_str(tsx->state));
4427 return;
4428 }
4429 if (id > -1) {
4430 session = inv->mod_data[session_module.id];
4431 }
4432 if (!session) {
4433 ast_log(LOG_DEBUG, "inv_session %p has no ast session\n", inv);
4434 } else {
4435 ast_log(LOG_DEBUG, "The state change pertains to the endpoint '%s(%s)'\n",
4437 session->channel ? ast_channel_name(session->channel) : "");
4438 }
4439 if (inv->invite_tsx) {
4440 ast_log(LOG_DEBUG, "The inv session still has an invite_tsx (%p)\n",
4441 inv->invite_tsx);
4442 } else {
4443 ast_log(LOG_DEBUG, "The inv session does NOT have an invite_tsx\n");
4444 }
4445 if (tsx) {
4446 ast_log(LOG_DEBUG, "The %s %.*s transaction involved in this state change is %p\n",
4447 pjsip_role_name(tsx->role),
4448 (int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
4449 tsx);
4450 ast_log(LOG_DEBUG, "The current transaction state is %s\n",
4451 pjsip_tsx_state_str(tsx->state));
4452 ast_log(LOG_DEBUG, "The transaction state change event is %s\n",
4453 pjsip_event_str(e->body.tsx_state.type));
4454 } else {
4455 ast_log(LOG_DEBUG, "There is no transaction involved in this state change\n");
4456 }
4457 ast_log(LOG_DEBUG, "The current inv state is %s\n", pjsip_inv_state_name(inv->state));
4458}
4459
4460#define print_debug_details(inv, tsx, e) __print_debug_details(__PRETTY_FUNCTION__, (inv), (tsx), (e))
4461
4462static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
4463{
4464 struct ast_sip_session_supplement *supplement;
4465 struct pjsip_request_line req = rdata->msg_info.msg->line.req;
4466 SCOPE_ENTER(3, "%s: Method is %.*s\n", ast_sip_session_get_name(session), (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));
4467
4468 AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
4469 if (supplement->incoming_request && does_method_match(&req.method.name, supplement->method)) {
4470 if (supplement->incoming_request(session, rdata)) {
4471 break;
4472 }
4473 }
4474 }
4475
4477}
4478
4480{
4481 struct ast_sip_session_supplement *iter;
4482
4483 AST_LIST_TRAVERSE(&session->supplements, iter, next) {
4484 if (iter->session_begin) {
4485 iter->session_begin(session);
4486 }
4487 }
4488}
4489
4491{
4492 struct ast_sip_session_supplement *iter;
4493
4494 AST_LIST_TRAVERSE(&session->supplements, iter, next) {
4495 if (iter->session_destroy) {
4496 iter->session_destroy(session);
4497 }
4498 }
4499}
4500
4502{
4503 struct ast_sip_session_supplement *iter;
4504
4505 /* Session is dead. Notify the supplements. */
4506 AST_LIST_TRAVERSE(&session->supplements, iter, next) {
4507 if (iter->session_end) {
4508 iter->session_end(session);
4509 }
4510 }
4511}
4512
4513static void handle_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata,
4515{
4516 struct ast_sip_session_supplement *supplement;
4517 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
4518 SCOPE_ENTER(3, "%s: Response is %d %.*s\n", ast_sip_session_get_name(session),
4519 status.code, (int) pj_strlen(&status.reason), pj_strbuf(&status.reason));
4520
4521 AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
4522 if (!(supplement->response_priority & response_priority)) {
4523 continue;
4524 }
4525 if (supplement->incoming_response && does_method_match(&rdata->msg_info.cseq->method.name, supplement->method)) {
4526 supplement->incoming_response(session, rdata);
4527 }
4528 }
4529
4531}
4532
4533static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata,
4534 enum ast_sip_session_response_priority response_priority)
4535{
4536 if (rdata->msg_info.msg->type == PJSIP_REQUEST_MSG) {
4538 } else {
4539 handle_incoming_response(session, rdata, response_priority);
4540 }
4541
4542 return 0;
4543}
4544
4545static void handle_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
4546{
4547 struct ast_sip_session_supplement *supplement;
4548 struct pjsip_request_line req = tdata->msg->line.req;
4549 SCOPE_ENTER(3, "%s: Method is %.*s\n", ast_sip_session_get_name(session),
4550 (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));
4551
4552 ast_sip_message_apply_transport(session->endpoint->transport, tdata);
4553
4554 AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
4555 if (supplement->outgoing_request && does_method_match(&req.method.name, supplement->method)) {
4556 supplement->outgoing_request(session, tdata);
4557 }
4558 }
4560}
4561
4562static void handle_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
4563{
4564 struct ast_sip_session_supplement *supplement;
4565 struct pjsip_status_line status = tdata->msg->line.status;
4566 pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
4567 SCOPE_ENTER(3, "%s: Method is %.*s, Response is %d %.*s\n", ast_sip_session_get_name(session),
4568 (int) pj_strlen(&cseq->method.name),
4569 pj_strbuf(&cseq->method.name), status.code, (int) pj_strlen(&status.reason),
4570 pj_strbuf(&status.reason));
4571
4572
4573 if (!cseq) {
4574 SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Cannot send response due to missing sequence header",
4576 }
4577
4578 ast_sip_message_apply_transport(session->endpoint->transport, tdata);
4579
4580 AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
4581 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
4582 supplement->outgoing_response(session, tdata);
4583 }
4584 }
4585
4587}
4588
4589static int session_end(void *vsession)
4590{
4591 struct ast_sip_session *session = vsession;
4592
4593 /* Stop the scheduled termination */
4595
4596 /* Session is dead. Notify the supplements. */
4598
4599 return 0;
4600}
4601
4602/*!
4603 * \internal
4604 * \brief Complete ending session activities.
4605 * \since 13.5.0
4606 *
4607 * \param vsession Which session to complete stopping.
4608 *
4609 * \retval 0 on success.
4610 * \retval -1 on error.
4611 */
4612static int session_end_completion(void *vsession)
4613{
4614 struct ast_sip_session *session = vsession;
4615
4616 ast_sip_dialog_set_serializer(session->inv_session->dlg, NULL);
4617 ast_sip_dialog_set_endpoint(session->inv_session->dlg, NULL);
4618
4619 /* Now we can release the ref that was held by session->inv_session */
4621 return 0;
4622}
4623
4624static int check_request_status(pjsip_inv_session *inv, pjsip_event *e)
4625{
4626 struct ast_sip_session *session = inv->mod_data[session_module.id];
4627 pjsip_transaction *tsx = e->body.tsx_state.tsx;
4628
4629 if (inv->state == PJSIP_INV_STATE_DISCONNECTED && inv->cancelling) {
4630 return 0;
4631 }
4632
4633 if (tsx->status_code != 503 && tsx->status_code != 408) {
4634 return 0;
4635 }
4636
4637 if (!ast_sip_failover_request(tsx->last_tx)) {
4638 return 0;
4639 }
4640
4641 pjsip_inv_uac_restart(inv, PJ_FALSE);
4642 /*
4643 * Bump the ref since it will be on a new transaction and
4644 * we don't want it to go away along with the old transaction.
4645 */
4646 pjsip_tx_data_add_ref(tsx->last_tx);
4648 return 1;
4649}
4650
4651static void handle_incoming_before_media(pjsip_inv_session *inv,
4652 struct ast_sip_session *session, pjsip_rx_data *rdata)
4653{
4654 pjsip_msg *msg;
4655 ast_debug(3, "%s: Received %s\n", ast_sip_session_get_name(session), rdata->msg_info.msg->type == PJSIP_REQUEST_MSG ?
4656 "request" : "response");
4657
4658
4660 msg = rdata->msg_info.msg;
4661 if (msg->type == PJSIP_REQUEST_MSG
4662 && msg->line.req.method.id == PJSIP_ACK_METHOD
4663 && pjmedia_sdp_neg_get_state(inv->neg) != PJMEDIA_SDP_NEG_STATE_DONE) {
4664 pjsip_tx_data *tdata;
4665
4666 /*
4667 * SDP negotiation failed on an incoming call that delayed
4668 * negotiation and then gave us an invalid SDP answer. We
4669 * need to send a BYE to end the call because of the invalid
4670 * SDP answer.
4671 */
4672 ast_debug(1,
4673 "%s: Ending session due to incomplete SDP negotiation. %s\n",
4675 pjsip_rx_data_get_info(rdata));
4676 if (pjsip_inv_end_session(inv, 400, NULL, &tdata) == PJ_SUCCESS
4677 && tdata) {
4679 }
4680 }
4681}
4682
4683static void session_inv_on_state_changed(pjsip_inv_session *inv, pjsip_event *e)
4684{
4685 pjsip_event_id_e type;
4686 struct ast_sip_session *session = inv->mod_data[session_module.id];
4687 SCOPE_ENTER(1, "%s Event: %s Inv State: %s\n", ast_sip_session_get_name(session),
4688 pjsip_event_str(e->type), pjsip_inv_state_name(inv->state));
4689
4690 if (ast_shutdown_final()) {
4691 SCOPE_EXIT_RTN("Shutting down\n");
4692 }
4693
4694 if (e) {
4695 print_debug_details(inv, NULL, e);
4696 type = e->type;
4697 } else {
4698 type = PJSIP_EVENT_UNKNOWN;
4699 }
4700
4701 session = inv->mod_data[session_module.id];
4702 if (!session) {
4703 SCOPE_EXIT_RTN("No session\n");
4704 }
4705
4706 switch(type) {
4707 case PJSIP_EVENT_TX_MSG:
4708 break;
4709 case PJSIP_EVENT_RX_MSG:
4710 handle_incoming_before_media(inv, session, e->body.rx_msg.rdata);
4711 break;
4712 case PJSIP_EVENT_TSX_STATE:
4713 ast_debug(3, "%s: Source of transaction state change is %s\n", ast_sip_session_get_name(session),
4714 pjsip_event_str(e->body.tsx_state.type));
4715 /* Transaction state changes are prompted by some other underlying event. */
4716 switch(e->body.tsx_state.type) {
4717 case PJSIP_EVENT_TX_MSG:
4718 break;
4719 case PJSIP_EVENT_RX_MSG:
4720 if (!check_request_status(inv, e)) {
4721 handle_incoming_before_media(inv, session, e->body.tsx_state.src.rdata);
4722 }
4723 break;
4724 case PJSIP_EVENT_TRANSPORT_ERROR:
4725 case PJSIP_EVENT_TIMER:
4726 /*
4727 * Check the request status on transport error or timeout. A transport
4728 * error can occur when a TCP socket closes and that can be the result
4729 * of a 503. Also we may need to failover on a timeout (408).
4730 */
4731 check_request_status(inv, e);
4732 break;
4733 case PJSIP_EVENT_USER:
4734 case PJSIP_EVENT_UNKNOWN:
4735 case PJSIP_EVENT_TSX_STATE:
4736 /* Inception? */
4737 break;
4738 }
4739 break;
4740 case PJSIP_EVENT_TRANSPORT_ERROR:
4741 case PJSIP_EVENT_TIMER:
4742 case PJSIP_EVENT_UNKNOWN:
4743 case PJSIP_EVENT_USER:
4744 default:
4745 break;
4746 }
4747
4748 if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
4749 if (session->defer_end) {
4750 ast_debug(3, "%s: Deferring session end\n", ast_sip_session_get_name(session));
4751 session->ended_while_deferred = 1;
4752 SCOPE_EXIT_RTN("Deferring\n");
4753 }
4754
4755 if (ast_sip_push_task(session->serializer, session_end, session)) {
4756 /* Do it anyway even though this is not the right thread. */
4758 }
4759 }
4760
4762}
4763
4764static void session_inv_on_new_session(pjsip_inv_session *inv, pjsip_event *e)
4765{
4766 /* XXX STUB */
4767}
4768
4769static int session_end_if_disconnected(int id, pjsip_inv_session *inv)
4770{
4771 struct ast_sip_session *session;
4772
4773 if (inv->state != PJSIP_INV_STATE_DISCONNECTED) {
4774 return 0;
4775 }
4776
4777 /*
4778 * We are locking because ast_sip_dialog_get_session() needs
4779 * the dialog locked to get the session by other threads.
4780 */
4781 pjsip_dlg_inc_lock(inv->dlg);
4782 session = inv->mod_data[id];
4783 inv->mod_data[id] = NULL;
4784 pjsip_dlg_dec_lock(inv->dlg);
4785
4786 /*
4787 * Pass the session ref held by session->inv_session to
4788 * session_end_completion().
4789 */
4790 if (session
4792 /* Do it anyway even though this is not the right thread. */
4794 }
4795
4796 return 1;
4797}
4798
4799static void session_inv_on_tsx_state_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
4800{
4802 int id = session_module.id;
4803 pjsip_tx_data *tdata;
4804 struct ast_sip_session *session = inv->mod_data[session_module.id];
4805 SCOPE_ENTER(1, "%s TSX State: %s Inv State: %s\n", ast_sip_session_get_name(session),
4806 pjsip_tsx_state_str(tsx->state), pjsip_inv_state_name(inv->state));
4807
4808 if (ast_shutdown_final()) {
4809 SCOPE_EXIT_RTN("Shutting down\n");
4810 }
4811
4812 session = inv->mod_data[id];
4813
4814 print_debug_details(inv, tsx, e);
4815 if (!session) {
4816 /* The session has ended. Ignore the transaction change. */
4817 SCOPE_EXIT_RTN("Session ended\n");
4818 }
4819
4820 /*
4821 * If the session is disconnected really nothing else to do unless currently transacting
4822 * a BYE. If a BYE then hold off destruction until the transaction timeout occurs. This
4823 * has to be done for BYEs because sometimes the dialog can be in a disconnected
4824 * state but the BYE request transaction has not yet completed.
4825 */
4826 if (tsx->method.id != PJSIP_BYE_METHOD && session_end_if_disconnected(id, inv)) {
4827 SCOPE_EXIT_RTN("Disconnected\n");
4828 }
4829
4830 switch (e->body.tsx_state.type) {
4831 case PJSIP_EVENT_TX_MSG:
4832 /* When we create an outgoing request, we do not have access to the transaction that
4833 * is created. Instead, We have to place transaction-specific data in the tdata. Here,
4834 * we transfer the data into the transaction. This way, when we receive a response, we
4835 * can dig this data out again
4836 */
4837 tsx->mod_data[id] = e->body.tsx_state.src.tdata->mod_data[id];
4838 break;
4839 case PJSIP_EVENT_RX_MSG:
4840 cb = ast_sip_mod_data_get(tsx->mod_data, id, MOD_DATA_ON_RESPONSE);
4841 /* As the PJSIP invite session implementation responds with a 200 OK before we have a
4842 * chance to be invoked session supplements for BYE requests actually end up executing
4843 * in the invite session state callback as well. To prevent session supplements from
4844 * running on the BYE request again we explicitly squash invocation of them here.
4845 */
4846 if ((e->body.tsx_state.src.rdata->msg_info.msg->type != PJSIP_REQUEST_MSG) ||
4847 (tsx->method.id != PJSIP_BYE_METHOD)) {
4848 handle_incoming(session, e->body.tsx_state.src.rdata,
4850 }
4851 if (tsx->method.id == PJSIP_INVITE_METHOD) {
4852 if (tsx->role == PJSIP_ROLE_UAC) {
4853 if (tsx->state == PJSIP_TSX_STATE_COMPLETED) {
4854 /* This means we got a non 2XX final response to our outgoing INVITE */
4855 if (tsx->status_code == PJSIP_SC_REQUEST_PENDING) {
4857 SCOPE_EXIT_RTN("Non 2XX final response\n");
4858 }
4859 if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
4860 ast_debug(1, "%s: reINVITE received final response code %d\n",
4862 tsx->status_code);
4863 if ((tsx->status_code == 401 || tsx->status_code == 407
4864 || (session->endpoint->security_negotiation && tsx->status_code == 494))
4865 && ++session->authentication_challenge_count < MAX_RX_CHALLENGES
4867 &session->endpoint->outbound_auths,
4868 e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
4869 /* Send authed reINVITE */
4871 SCOPE_EXIT_RTN("Sending authed reinvite\n");
4872 }
4873 /* Per RFC3261 14.1 a response to a re-INVITE should only terminate
4874 * the dialog if a 481 or 408 occurs. All other responses should leave
4875 * the dialog untouched.
4876 */
4877 if (tsx->status_code == 481 || tsx->status_code == 408) {
4878 if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS
4879 && tdata) {
4881 }
4882 }
4883 }
4884 } else if (tsx->state == PJSIP_TSX_STATE_TERMINATED) {
4885 if (!inv->cancelling
4886 && inv->role == PJSIP_ROLE_UAC
4887 && inv->state == PJSIP_INV_STATE_CONFIRMED
4888 && pjmedia_sdp_neg_was_answer_remote(inv->neg)
4889 && pjmedia_sdp_neg_get_state(inv->neg) == PJMEDIA_SDP_NEG_STATE_DONE
4891 ) {
4892 /*
4893 * We didn't send a CANCEL but the UAS sent us the 200 OK with an invalid or unacceptable codec SDP.
4894 * In this case the SDP negotiation is incomplete and PJPROJECT has already sent the ACK.
4895 * So, we send the BYE with 503 status code here. And the actual hangup cause code is already set
4896 * to AST_CAUSE_BEARERCAPABILITY_NOTAVAIL by the session_inv_on_media_update(), setting the 503
4897 * status code doesn't affect to hangup cause code.
4898 */
4899 ast_debug(1, "Endpoint '%s(%s)': Ending session due to 200 OK with incomplete SDP negotiation. %s\n",
4901 session->channel ? ast_channel_name(session->channel) : "",
4902 pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
4903 pjsip_inv_end_session(session->inv_session, 503, NULL, &tdata);
4904 SCOPE_EXIT_RTN("Incomplete SDP negotiation\n");
4905 }
4906
4907 if (inv->cancelling && tsx->status_code == PJSIP_SC_OK) {
4908 int sdp_negotiation_done =
4909 pjmedia_sdp_neg_get_state(inv->neg) == PJMEDIA_SDP_NEG_STATE_DONE;
4910
4911 /*
4912 * We can get here for the following reasons.
4913 *
4914 * 1) The race condition detailed in RFC5407 section 3.1.2.
4915 * We sent a CANCEL at the same time that the UAS sent us a
4916 * 200 OK with a valid SDP for the original INVITE. As a
4917 * result, we have now received a 200 OK for a cancelled
4918 * call and the SDP negotiation is complete. We need to
4919 * immediately send a BYE to end the dialog.
4920 *
4921 * 2) We sent a CANCEL and hit the race condition but the
4922 * UAS sent us an invalid SDP with the 200 OK. In this case
4923 * the SDP negotiation is incomplete and PJPROJECT has
4924 * already sent the BYE for us because of the invalid SDP.
4925 */
4926 ast_test_suite_event_notify("PJSIP_SESSION_CANCELED",
4927 "Endpoint: %s\r\n"
4928 "Channel: %s\r\n"
4929 "Message: %s\r\n"
4930 "SDP: %s",
4932 session->channel ? ast_channel_name(session->channel) : "",
4933 pjsip_rx_data_get_info(e->body.tsx_state.src.rdata),
4934 sdp_negotiation_done ? "complete" : "incomplete");
4935 if (!sdp_negotiation_done) {
4936 ast_debug(1, "%s: Incomplete SDP negotiation cancelled session. %s\n",
4938 pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
4939 } else if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS
4940 && tdata) {
4941 ast_debug(1, "%s: Ending session due to RFC5407 race condition. %s\n",
4943 pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
4945 }
4946 }
4947 }
4948 }
4949 } else {
4950 /* All other methods */
4951 if (tsx->role == PJSIP_ROLE_UAC) {
4952 if (tsx->state == PJSIP_TSX_STATE_COMPLETED) {
4953 /* This means we got a final response to our outgoing method */
4954 ast_debug(1, "%s: %.*s received final response code %d\n",
4956 (int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
4957 tsx->status_code);
4958 if ((tsx->status_code == 401 || tsx->status_code == 407 || tsx->status_code == 494)
4959 && ++session->authentication_challenge_count < MAX_RX_CHALLENGES
4961 &session->endpoint->outbound_auths,
4962 e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
4963 /* Send authed version of the method */
4965 SCOPE_EXIT_RTN("Sending authed %.*s\n",
4966 (int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name));
4967 }
4968 }
4969 }
4970 }
4971 if (cb) {
4972 cb(session, e->body.tsx_state.src.rdata);
4973 }
4974 break;
4975 case PJSIP_EVENT_TRANSPORT_ERROR:
4976 case PJSIP_EVENT_TIMER:
4977 /*
4978 * The timer event is run by the pjsip monitor thread and not
4979 * by the session serializer.
4980 */
4981 if (session_end_if_disconnected(id, inv)) {
4982 SCOPE_EXIT_RTN("Disconnected\n");
4983 }
4984 break;
4985 case PJSIP_EVENT_USER:
4986 case PJSIP_EVENT_UNKNOWN:
4987 case PJSIP_EVENT_TSX_STATE:
4988 /* Inception? */
4989 break;
4990 }
4991
4992 if (session->terminate_on_invite_timeout && uac_invite_tsx_terminates_dialog(tsx)) {
4993 /*
4994 * PJPROJECT already considers this dialog terminated; the delayed BYE is
4995 * obsolete and still owns media state that must be released.
4996 */
4997 ast_debug(3, "%s: Flushing delayed requests because outstanding INVITE terminated dialog\n",
5000 session->terminate_on_invite_timeout = 0;
5001 }
5002
5003 if (AST_LIST_EMPTY(&session->delayed_requests)) {
5004 /* No delayed request pending, so just return */
5005 SCOPE_EXIT_RTN("Nothing delayed\n");
5006 }
5007
5008 if (tsx->method.id == PJSIP_INVITE_METHOD) {
5009 if (tsx->state == PJSIP_TSX_STATE_PROCEEDING) {
5010 ast_debug(3, "%s: INVITE delay check. tsx-state:%s\n",
5012 pjsip_tsx_state_str(tsx->state));
5014 } else if (tsx->state == PJSIP_TSX_STATE_TERMINATED) {
5015 /*
5016 * Terminated INVITE transactions always should result in
5017 * queuing delayed requests, no matter what event caused
5018 * the transaction to terminate.
5019 */
5020 ast_debug(3, "%s: INVITE delay check. tsx-state:%s\n",
5022 pjsip_tsx_state_str(tsx->state));
5024 }
5025 } else if (tsx->role == PJSIP_ROLE_UAC
5026 && tsx->state == PJSIP_TSX_STATE_COMPLETED
5027 && !pj_strcmp2(&tsx->method.name, "UPDATE")) {
5028 ast_debug(3, "%s: UPDATE delay check. tsx-state:%s\n",
5030 pjsip_tsx_state_str(tsx->state));
5032 }
5033
5035}
5036
5037static int add_sdp_streams(struct ast_sip_session_media *session_media,
5038 struct ast_sip_session *session, pjmedia_sdp_session *answer,
5039 const struct pjmedia_sdp_session *remote,
5040 struct ast_stream *stream)
5041{
5042 struct ast_sip_session_sdp_handler *handler = session_media->handler;
5043 RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
5044 int res = 0;
5045 SCOPE_ENTER(1, "%s Stream: %s\n", ast_sip_session_get_name(session),
5046 ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
5047
5048 if (handler) {
5049 /* if an already assigned handler reports a catastrophic error, fail */
5050 res = handler->create_outgoing_sdp_stream(session, session_media, answer, remote, stream);
5051 if (res < 0) {
5052 SCOPE_EXIT_RTN_VALUE(-1, "Coudn't create sdp stream\n");
5053 }
5054 SCOPE_EXIT_RTN_VALUE(0, "Had handler\n");
5055 }
5056
5057 handler_list = ao2_find(sdp_handlers, ast_codec_media_type2str(session_media->type), OBJ_KEY);
5058 if (!handler_list) {
5059 SCOPE_EXIT_RTN_VALUE(0, "No handlers\n");
5060 }
5061
5062 /* no handler for this stream type and we have a list to search */
5063 AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
5064 if (handler == session_media->handler) {
5065 continue;
5066 }
5067 res = handler->create_outgoing_sdp_stream(session, session_media, answer, remote, stream);
5068 if (res < 0) {
5069 /* catastrophic error */
5070 SCOPE_EXIT_RTN_VALUE(-1, "Coudn't create sdp stream\n");
5071 }
5072 if (res > 0) {
5073 /* Handled by this handler. Move to the next stream */
5074 session_media_set_handler(session_media, handler);
5075 SCOPE_EXIT_RTN_VALUE(0, "Handled\n");
5076 }
5077 }
5078
5079 /* streams that weren't handled won't be included in generated outbound SDP */
5080 SCOPE_EXIT_RTN_VALUE(0, "Not handled\n");
5081}
5082
5083/*! \brief Bundle group building structure */
5085 /*! \brief The media identifiers in this bundle group */
5086 char *mids[PJMEDIA_MAX_SDP_MEDIA];
5087 /*! \brief SDP attribute string */
5089};
5090
5091static int add_bundle_groups(struct ast_sip_session *session, pj_pool_t *pool, pjmedia_sdp_session *answer)
5092{
5093 pj_str_t stmp;
5094 pjmedia_sdp_attr *attr;
5095 struct sip_session_media_bundle_group bundle_groups[PJMEDIA_MAX_SDP_MEDIA];
5096 int index, mid_id;
5097 struct sip_session_media_bundle_group *bundle_group;
5098
5099 if (session->endpoint->media.webrtc) {
5100 attr = pjmedia_sdp_attr_create(pool, "msid-semantic", pj_cstr(&stmp, "WMS *"));
5101 pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
5102 }
5103
5104 if (!session->endpoint->media.bundle) {
5105 return 0;
5106 }
5107
5108 memset(bundle_groups, 0, sizeof(bundle_groups));
5109
5110 /* Build the bundle group layout so we can then add it to the SDP */
5111 for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
5112 struct ast_sip_session_media *session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
5113
5114 /* If this stream is not part of a bundle group we can't add it */
5115 if (session_media->bundle_group == -1) {
5116 continue;
5117 }
5118
5119 bundle_group = &bundle_groups[session_media->bundle_group];
5120
5121 /* If this is the first mid then we need to allocate the attribute string and place BUNDLE in front */
5122 if (!bundle_group->mids[0]) {
5123 bundle_group->mids[0] = session_media->mid;
5124 bundle_group->attr_string = ast_str_create(64);
5125 if (!bundle_group->attr_string) {
5126 continue;
5127 }
5128
5129 ast_str_set(&bundle_group->attr_string, 0, "BUNDLE %s", session_media->mid);
5130 continue;
5131 }
5132
5133 for (mid_id = 1; mid_id < PJMEDIA_MAX_SDP_MEDIA; ++mid_id) {
5134 if (!bundle_group->mids[mid_id]) {
5135 bundle_group->mids[mid_id] = session_media->mid;
5136 ast_str_append(&bundle_group->attr_string, 0, " %s", session_media->mid);
5137 break;
5138 } else if (!strcmp(bundle_group->mids[mid_id], session_media->mid)) {
5139 break;
5140 }
5141 }
5142 }
5143
5144 /* Add all bundle groups that have mids to the SDP */
5145 for (index = 0; index < PJMEDIA_MAX_SDP_MEDIA; ++index) {
5146 bundle_group = &bundle_groups[index];
5147
5148 if (!bundle_group->attr_string) {
5149 continue;
5150 }
5151
5152 attr = pjmedia_sdp_attr_create(pool, "group", pj_cstr(&stmp, ast_str_buffer(bundle_group->attr_string)));
5153 pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
5154
5155 ast_free(bundle_group->attr_string);
5156 }
5157
5158 return 0;
5159}
5160
5161static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer, const unsigned int ignore_active_stream_topology)
5162{
5163 static const pj_str_t STR_IN = { "IN", 2 };
5164 static const pj_str_t STR_IP4 = { "IP4", 3 };
5165 static const pj_str_t STR_IP6 = { "IP6", 3 };
5166 pjmedia_sdp_session *local;
5167 int i;
5168 int stream;
5170
5171 if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
5172 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "%s: Failed to create session SDP. Session has been already disconnected\n",
5174 }
5175
5176 if (!inv->pool_prov || !(local = PJ_POOL_ZALLOC_T(inv->pool_prov, pjmedia_sdp_session))) {
5177 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "%s: Pool allocation failure\n", ast_sip_session_get_name(session));
5178 }
5179
5180 if (!offer) {
5181 local->origin.version = local->origin.id = (pj_uint32_t)(ast_random());
5182 } else {
5183 local->origin.version = offer->origin.version + 1;
5184 local->origin.id = offer->origin.id;
5185 }
5186
5187 pj_strdup2(inv->pool_prov, &local->origin.user, session->endpoint->media.sdpowner);
5188 pj_strdup2(inv->pool_prov, &local->name, session->endpoint->media.sdpsession);
5189
5190 if (!session->pending_media_state->topology || !ast_stream_topology_get_count(session->pending_media_state->topology)) {
5191 /* We've encountered a situation where we have been told to create a local SDP but noone has given us any indication
5192 * of what kind of stream topology they would like. We try to not alter the current state of the SDP negotiation
5193 * by using what is currently negotiated. If this is unavailable we fall back to what is configured on the endpoint.
5194 * We will also do this if wanted by the ignore_active_stream_topology flag.
5195 */
5196 ast_trace(-1, "no information about stream topology received\n");
5197 ast_stream_topology_free(session->pending_media_state->topology);
5198 if (session->active_media_state->topology && !ignore_active_stream_topology) {
5199 ast_trace(-1, "using existing topology\n");
5200 session->pending_media_state->topology = ast_stream_topology_clone(session->active_media_state->topology);
5201 } else {
5202 if (ignore_active_stream_topology) {
5203 ast_trace(-1, "fall back to endpoint configuration - ignore active stream topolog\n");
5204 } else {
5205 ast_trace(-1, "fall back to endpoint configuration\n");
5206 }
5207 session->pending_media_state->topology = ast_stream_topology_clone(session->endpoint->media.topology);
5208 }
5209 if (!session->pending_media_state->topology) {
5210 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "%s: No pending media state topology\n", ast_sip_session_get_name(session));
5211 }
5212 }
5213
5214 ast_trace(-1, "%s: Processing streams\n", ast_sip_session_get_name(session));
5215
5216 for (i = 0; i < ast_stream_topology_get_count(session->pending_media_state->topology); ++i) {
5217 struct ast_sip_session_media *session_media;
5218 struct ast_stream *stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
5219 unsigned int streams = local->media_count;
5220 SCOPE_ENTER(4, "%s: Processing stream %s\n", ast_sip_session_get_name(session),
5221 ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
5222
5223 /* This code does not enforce any maximum stream count limitations as that is done on either
5224 * the handling of an incoming SDP offer or on the handling of a session refresh.
5225 */
5226
5227 session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_stream_get_type(stream), i);
5228 if (!session_media) {
5229 local = NULL;
5230 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't alloc/add session media for stream %s\n",
5232 }
5233
5234 if (add_sdp_streams(session_media, session, local, offer, stream)) {
5235 local = NULL;
5236 SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't add sdp streams for stream %s\n",
5238 }
5239
5240 /* If a stream was actually added then add any additional details */
5241 if (streams != local->media_count) {
5242 pjmedia_sdp_media *media = local->media[streams];
5243 pj_str_t stmp;
5244 pjmedia_sdp_attr *attr;
5245
5246 /* Add the media identifier if present */
5247 if (!ast_strlen_zero(session_media->mid)) {
5248 attr = pjmedia_sdp_attr_create(inv->pool_prov, "mid", pj_cstr(&stmp, session_media->mid));
5249 pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
5250 }
5251
5252 ast_trace(-1, "%s: Stream %s added%s%s\n", ast_sip_session_get_name(session),
5253 ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)),
5254 S_COR(!ast_strlen_zero(session_media->mid), " with mid ", ""), S_OR(session_media->mid, ""));
5255
5256 }
5257
5258 /* Ensure that we never exceed the maximum number of streams PJMEDIA will allow. */
5259 if (local->media_count == PJMEDIA_MAX_SDP_MEDIA) {
5260 SCOPE_EXIT_EXPR(break, "%s: Stream %s exceeded max pjmedia count of %d\n",
5262 PJMEDIA_MAX_SDP_MEDIA);
5263 }
5264
5265 SCOPE_EXIT("%s: Done with %s\n", ast_sip_session_get_name(session),
5266 ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
5267
5268 }
5269
5270 /* Add any bundle groups that are present on the media state */
5271 ast_trace(-1, "%s: Adding bundle groups (if available)\n", ast_sip_session_get_name(session));
5272 if (add_bundle_groups(session, inv->pool_prov, local)) {
5273 SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "%s: Couldn't add bundle groups\n", ast_sip_session_get_name(session));
5274 }
5275
5276 /* Use the connection details of an available media if possible for SDP level */
5277 ast_trace(-1, "%s: Copying connection details\n", ast_sip_session_get_name(session));
5278
5279 for (stream = 0; stream < local->media_count; stream++) {
5280 SCOPE_ENTER(4, "%s: Processing media %d\n", ast_sip_session_get_name(session), stream);
5281 if (!local->media[stream]->conn) {
5282 SCOPE_EXIT_EXPR(continue, "%s: Media %d has no connection info\n", ast_sip_session_get_name(session), stream);
5283 }
5284
5285 if (local->conn) {
5286 if (!pj_strcmp(&local->conn->net_type, &local->media[stream]->conn->net_type) &&
5287 !pj_strcmp(&local->conn->addr_type, &local->media[stream]->conn->addr_type) &&
5288 !pj_strcmp(&local->conn->addr, &local->media[stream]->conn->addr)) {
5289 local->media[stream]->conn = NULL;
5290 }
5291 SCOPE_EXIT_EXPR(continue, "%s: Media %d has good existing connection info\n", ast_sip_session_get_name(session), stream);
5292 }
5293
5294 /* This stream's connection info will serve as the connection details for SDP level */
5295 local->conn = local->media[stream]->conn;
5296 local->media[stream]->conn = NULL;
5297
5298 SCOPE_EXIT_EXPR(continue, "%s: Media %d reset\n", ast_sip_session_get_name(session), stream);
5299 }
5300
5301 /* If no SDP level connection details are present then create some */
5302 if (!local->conn) {
5303 ast_trace(-1, "%s: Creating connection details\n", ast_sip_session_get_name(session));
5304
5305 local->conn = pj_pool_zalloc(inv->pool_prov, sizeof(struct pjmedia_sdp_conn));
5306 local->conn->net_type = STR_IN;
5307 local->conn->addr_type = session->endpoint->media.rtp.ipv6 ? STR_IP6 : STR_IP4;
5308
5309 if (!ast_strlen_zero(session->endpoint->media.address)) {
5310 pj_strdup2(inv->pool_prov, &local->conn->addr, session->endpoint->media.address);
5311 } else {
5312 pj_strdup2(inv->pool_prov, &local->conn->addr, ast_sip_get_host_ip_string(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET()));
5313 }
5314 }
5315
5316 pj_strassign(&local->origin.net_type, &local->conn->net_type);
5317 pj_strassign(&local->origin.addr_type, &local->conn->addr_type);
5318 pj_strassign(&local->origin.addr, &local->conn->addr);
5319
5320end:
5322}
5323
5324static void session_inv_on_rx_offer(pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
5325{
5326 struct ast_sip_session *session = inv->mod_data[session_module.id];
5327 pjmedia_sdp_session *answer;
5329
5330 if (ast_shutdown_final()) {
5331 SCOPE_EXIT_RTN("%s: Shutdown in progress\n", ast_sip_session_get_name(session));
5332 }
5333
5334 session = inv->mod_data[session_module.id];
5335 if (handle_incoming_sdp(session, offer)) {
5336 ast_sip_session_media_state_reset(session->pending_media_state);
5337 SCOPE_EXIT_RTN("%s: handle_incoming_sdp failed\n", ast_sip_session_get_name(session));
5338 }
5339
5340 if ((answer = create_local_sdp(inv, session, offer, 0))) {
5341 pjsip_inv_set_sdp_answer(inv, answer);
5342 SCOPE_EXIT_RTN("%s: Set SDP answer\n", ast_sip_session_get_name(session));
5343 }
5344 SCOPE_EXIT_RTN("%s: create_local_sdp failed\n", ast_sip_session_get_name(session));
5345}
5346
5347static void session_inv_on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
5348{
5349 struct ast_sip_session *session = inv->mod_data[session_module.id];
5350 const pjmedia_sdp_session *previous_sdp = NULL;
5351 pjmedia_sdp_session *offer;
5352 int i;
5353 unsigned int ignore_active_stream_topology = 0;
5354
5355 /* We allow PJSIP to produce an SDP if no channel is present. This may result
5356 * in an incorrect SDP occurring, but if no channel is present then we are in
5357 * the midst of a BYE and are hanging up. This ensures that all the code to
5358 * produce an SDP doesn't need to worry about a channel being present or not,
5359 * just in case.
5360 */
5362 if (!session->channel) {
5363 SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
5364 }
5365
5366 /* Some devices send a re-INVITE offer with empty SDP. Asterisk by default return
5367 * an answer with the current used codecs, which is not strictly compliant to RFC
5368 * 3261 (SHOULD requirement). So we detect this condition and include all
5369 * configured codecs in the answer if the workaround is activated. The actual
5370 * logic is in the create_local_sdp function. We can't detect here that we have
5371 * no SDP body in the INVITE, as we don't have access to the message.
5372 */
5373 if (inv->invite_tsx && inv->state == PJSIP_INV_STATE_CONFIRMED
5374 && inv->invite_tsx->method.id == PJSIP_INVITE_METHOD) {
5375 ast_trace(-1, "re-INVITE\n");
5376 if (inv->invite_tsx->role == PJSIP_ROLE_UAS
5378 ast_trace(-1, "UAS role, include all codecs in the answer on empty SDP\n");
5379 ignore_active_stream_topology = 1;
5380 }
5381 }
5382
5383 if (inv->neg) {
5384 if (pjmedia_sdp_neg_was_answer_remote(inv->neg)) {
5385 pjmedia_sdp_neg_get_active_remote(inv->neg, &previous_sdp);
5386 } else {
5387 pjmedia_sdp_neg_get_active_local(inv->neg, &previous_sdp);
5388 }
5389 }
5390
5391 if (ignore_active_stream_topology) {
5392 offer = create_local_sdp(inv, session, NULL, 1);
5393 } else {
5394 offer = create_local_sdp(inv, session, previous_sdp, 0);
5395 }
5396 if (!offer) {
5397 SCOPE_EXIT_RTN("%s: create offer failed\n", ast_sip_session_get_name(session));
5398 }
5399
5400 ast_queue_unhold(session->channel);
5401
5402 /*
5403 * Some devices indicate hold with deferred SDP reinvites (i.e. no SDP in the reinvite).
5404 * When hold is initially indicated, we
5405 * - Receive an INVITE with no SDP
5406 * - Send a 200 OK with SDP, indicating sendrecv in the media streams
5407 * - Receive an ACK with SDP, indicating sendonly in the media streams
5408 *
5409 * At this point, the pjmedia negotiator saves the state of the media direction so that
5410 * if we are to send any offers, we'll offer recvonly in the media streams. This is
5411 * problematic if the device is attempting to unhold, though. If the device unholds
5412 * by sending a reinvite with no SDP, then we will respond with a 200 OK with recvonly.
5413 * According to RFC 3264, if an offerer offers recvonly, then the answerer MUST respond
5414 * with sendonly or inactive. The result of this is that the stream is not off hold.
5415 *
5416 * Therefore, in this case, when we receive a reinvite while the stream is on hold, we
5417 * need to be sure to offer sendrecv. This way, the answerer can respond with sendrecv
5418 * in order to get the stream off hold. If this is actually a different purpose reinvite
5419 * (like a session timer refresh), then the answerer can respond to our sendrecv with
5420 * sendonly, keeping the stream on hold.
5421 */
5422 for (i = 0; i < offer->media_count; ++i) {
5423 pjmedia_sdp_media *m = offer->media[i];
5424 pjmedia_sdp_attr *recvonly;
5425 pjmedia_sdp_attr *inactive;
5426 pjmedia_sdp_attr *sendonly;
5427
5428 recvonly = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "recvonly", NULL);
5429 inactive = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "inactive", NULL);
5430 sendonly = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "sendonly", NULL);
5431 if (recvonly || inactive || sendonly) {
5432 pjmedia_sdp_attr *to_remove = recvonly ?: inactive ?: sendonly;
5433 pjmedia_sdp_attr *sendrecv;
5434
5435 pjmedia_sdp_attr_remove(&m->attr_count, m->attr, to_remove);
5436
5437 sendrecv = pjmedia_sdp_attr_create(session->inv_session->pool, "sendrecv", NULL);
5438 pjmedia_sdp_media_add_attr(m, sendrecv);
5439 }
5440 }
5441
5442 *p_offer = offer;
5443 SCOPE_EXIT_RTN("%s: offer created\n", ast_sip_session_get_name(session));
5444}
5445
5446static void session_inv_on_media_update(pjsip_inv_session *inv, pj_status_t status)
5447{
5448 struct ast_sip_session *session = inv->mod_data[session_module.id];
5449 const pjmedia_sdp_session *local, *remote;
5451
5452 if (ast_shutdown_final()) {
5453 SCOPE_EXIT_RTN("%s: Shutdown in progress\n", ast_sip_session_get_name(session));
5454 }
5455
5456 session = inv->mod_data[session_module.id];
5457 if (!session || !session->channel) {
5458 /*
5459 * If we don't have a session or channel then we really
5460 * don't care about media updates.
5461 * Just ignore
5462 */
5463 SCOPE_EXIT_RTN("%s: No channel or session\n", ast_sip_session_get_name(session));
5464 }
5465
5466 if (session->endpoint) {
5467 int bail = 0;
5468
5469 /*
5470 * If following_fork is set, then this is probably the result of a
5471 * forked INVITE and SDP asnwers coming from the different fork UAS
5472 * destinations. In this case updated_sdp_answer will also be set.
5473 *
5474 * If only updated_sdp_answer is set, then this is the non-forking
5475 * scenario where the same UAS just needs to change something like
5476 * the media port.
5477 */
5478
5479 if (inv->following_fork) {
5480 if (session->endpoint->media.rtp.follow_early_media_fork) {
5481 ast_trace(-1, "%s: Following early media fork with different To tags\n", ast_sip_session_get_name(session));
5482 } else {
5483 ast_trace(-1, "%s: Not following early media fork with different To tags\n", ast_sip_session_get_name(session));
5484 bail = 1;
5485 }
5486 }
5487#ifdef HAVE_PJSIP_INV_ACCEPT_MULTIPLE_SDP_ANSWERS
5488 else if (inv->updated_sdp_answer) {
5489 if (session->endpoint->media.rtp.accept_multiple_sdp_answers) {
5490 ast_trace(-1, "%s: Accepting updated SDP with same To tag\n", ast_sip_session_get_name(session));
5491 } else {
5492 ast_trace(-1, "%s: Ignoring updated SDP answer with same To tag\n", ast_sip_session_get_name(session));
5493 bail = 1;
5494 }
5495 }
5496#endif
5497 if (bail) {
5499 }
5500 }
5501
5502 if ((status != PJ_SUCCESS) || (pjmedia_sdp_neg_get_active_local(inv->neg, &local) != PJ_SUCCESS) ||
5503 (pjmedia_sdp_neg_get_active_remote(inv->neg, &remote) != PJ_SUCCESS)) {
5505 ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
5506 ast_queue_hangup(session->channel);
5507 SCOPE_EXIT_RTN("%s: Couldn't get active or local or remote negotiator. Hanging up\n", ast_sip_session_get_name(session));
5508 }
5509
5510 if (handle_negotiated_sdp(session, local, remote)) {
5511 ast_sip_session_media_state_reset(session->pending_media_state);
5512 SCOPE_EXIT_RTN("%s: handle_negotiated_sdp failed. Resetting pending media state\n", ast_sip_session_get_name(session));
5513 }
5515}
5516
5517static pjsip_redirect_op session_inv_on_redirected(pjsip_inv_session *inv, const pjsip_uri *target, const pjsip_event *e)
5518{
5519 struct ast_sip_session *session;
5520 const pjsip_sip_uri *uri;
5521
5522 if (ast_shutdown_final()) {
5523 return PJSIP_REDIRECT_STOP;
5524 }
5525
5526 session = inv->mod_data[session_module.id];
5527 if (!session || !session->channel) {
5528 return PJSIP_REDIRECT_STOP;
5529 }
5530
5531 if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_PJSIP) {
5532 return PJSIP_REDIRECT_ACCEPT;
5533 }
5534
5535 if (!PJSIP_URI_SCHEME_IS_SIP(target) && !PJSIP_URI_SCHEME_IS_SIPS(target)) {
5536 return PJSIP_REDIRECT_STOP;
5537 }
5538
5540
5541 uri = pjsip_uri_get_uri(target);
5542
5543 if (session->endpoint->redirect_method == AST_SIP_REDIRECT_USER) {
5545
5546 ast_copy_pj_str(exten, &uri->user, sizeof(exten));
5547
5548 /*
5549 * We may want to match in the dialplan without any user
5550 * options getting in the way.
5551 */
5553
5554 ast_channel_call_forward_set(session->channel, exten);
5555 } else if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_CORE) {
5556 char target_uri[PJSIP_MAX_URL_SIZE];
5557 /* PJSIP/ + endpoint length + / + max URL size */
5558 char forward[8 + strlen(ast_sorcery_object_get_id(session->endpoint)) + PJSIP_MAX_URL_SIZE];
5559
5560 pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri, target_uri, sizeof(target_uri));
5561 sprintf(forward, "PJSIP/%s/%s", ast_sorcery_object_get_id(session->endpoint), target_uri);
5562 ast_channel_call_forward_set(session->channel, forward);
5563 }
5564
5565 return PJSIP_REDIRECT_STOP;
5566}
5567
5568static pjsip_inv_callback inv_callback = {
5569 .on_state_changed = session_inv_on_state_changed,
5570 .on_new_session = session_inv_on_new_session,
5571 .on_tsx_state_changed = session_inv_on_tsx_state_changed,
5572 .on_rx_offer = session_inv_on_rx_offer,
5573 .on_create_offer = session_inv_on_create_offer,
5574 .on_media_update = session_inv_on_media_update,
5575 .on_redirected = session_inv_on_redirected,
5576};
5577
5578/*! \brief Hook for modifying outgoing messages with SDP to contain the proper address information */
5579static void session_outgoing_nat_hook(pjsip_tx_data *tdata, struct ast_sip_transport *transport)
5580{
5582 pjsip_sdp_info *sdp_info;
5583 pjmedia_sdp_session *sdp;
5584 pjsip_dialog *dlg = pjsip_tdata_get_dlg(tdata);
5586 int stream;
5587
5588 /*
5589 * If there's no transport_state or body, just return.
5590 */
5591 if (ast_strlen_zero(transport->external_media_address) || !transport_state || !tdata->msg->body) {
5592 return;
5593 }
5594
5595 sdp_info = pjsip_get_sdp_info(tdata->pool, tdata->msg->body, NULL, &pjsip_media_type_application_sdp);
5596 if (sdp_info->sdp_err != PJ_SUCCESS || !sdp_info->sdp) {
5597 return;
5598 }
5599 sdp = sdp_info->sdp;
5600
5601 if (sdp->conn) {
5602 char host[NI_MAXHOST];
5603 struct ast_sockaddr our_sdp_addr = { { 0, } };
5604
5605 ast_copy_pj_str(host, &sdp->conn->addr, sizeof(host));
5606 ast_sockaddr_parse(&our_sdp_addr, host, PARSE_PORT_FORBID);
5607
5608 /* Reversed check here. We don't check the remote
5609 * endpoint being in our local net, but whether our
5610 * outgoing session IP is local. If it is, we'll do
5611 * rewriting. No localnet configured? Always rewrite. */
5612 if (ast_sip_transport_is_local(transport_state, &our_sdp_addr) || !transport_state->localnet) {
5613 ast_debug(5, "%s: Setting external media address to %s\n", ast_sip_session_get_name(session),
5614 ast_sockaddr_stringify_addr_remote(&transport_state->external_media_address));
5615 pj_strdup2(tdata->pool, &sdp->conn->addr, ast_sockaddr_stringify_addr_remote(&transport_state->external_media_address));
5616 pj_strassign(&sdp->origin.addr, &sdp->conn->addr);
5617 }
5618 }
5619
5620 for (stream = 0; stream < sdp->media_count; ++stream) {
5621 /* See if there are registered handlers for this media stream type */
5622 char media[20];
5624 RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
5625
5626 /* We need a null-terminated version of the media string */
5627 ast_copy_pj_str(media, &sdp->media[stream]->desc.media, sizeof(media));
5628
5629 handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
5630 if (!handler_list) {
5631 ast_debug(4, "%s: No registered SDP handlers for media type '%s'\n", ast_sip_session_get_name(session),
5632 media);
5633 continue;
5634 }
5635 AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
5636 if (handler->change_outgoing_sdp_stream_media_address) {
5637 handler->change_outgoing_sdp_stream_media_address(tdata, sdp->media[stream], transport);
5638 }
5639 }
5640 }
5641
5642}
5643
5644#ifdef TEST_FRAMEWORK
5645
5646static struct ast_stream *test_stream_alloc(const char *name, enum ast_media_type type, enum ast_stream_state state)
5647{
5648 struct ast_stream *stream;
5649
5650 stream = ast_stream_alloc(name, type);
5651 if (!stream) {
5652 return NULL;
5653 }
5654 ast_stream_set_state(stream, state);
5655
5656 return stream;
5657}
5658
5659static struct ast_sip_session_media *test_media_add(
5660 struct ast_sip_session_media_state *media_state, const char *name, enum ast_media_type type,
5661 enum ast_stream_state state, int position)
5662{
5663 struct ast_sip_session_media *session_media = NULL;
5664 struct ast_stream *stream = NULL;
5665
5666 stream = test_stream_alloc(name, type, state);
5667 if (!stream) {
5668 return NULL;
5669 }
5670
5671 if (position >= 0 && position < ast_stream_topology_get_count(media_state->topology)) {
5672 ast_stream_topology_set_stream(media_state->topology, position, stream);
5673 } else {
5674 position = ast_stream_topology_append_stream(media_state->topology, stream);
5675 }
5676
5677 session_media = ao2_alloc_options(sizeof(*session_media), session_media_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
5678 if (!session_media) {
5679 return NULL;
5680 }
5681
5682 session_media->keepalive_sched_id = -1;
5683 session_media->timeout_sched_id = -1;
5684 session_media->type = type;
5685 session_media->stream_num = position;
5686 session_media->bundle_group = -1;
5687 strcpy(session_media->label, name);
5688
5689 if (AST_VECTOR_REPLACE(&media_state->sessions, position, session_media)) {
5690 ao2_ref(session_media, -1);
5691
5692 return NULL;
5693 }
5694
5695 /* If this stream will be active in some way and it is the first of this type then consider this the default media session to match */
5697 media_state->default_session[type] = session_media;
5698 }
5699
5700 return session_media;
5701}
5702
5703static int test_is_media_session_equal(struct ast_sip_session_media *left, struct ast_sip_session_media *right)
5704{
5705 if (left == right) {
5706 return 1;
5707 }
5708
5709 if (!left) {
5710 return 1;
5711 }
5712
5713 if (!right) {
5714 return 0;
5715 }
5716 return memcmp(left, right, sizeof(*left)) == 0;
5717}
5718
5719static int test_is_media_state_equal(struct ast_sip_session_media_state *left, struct ast_sip_session_media_state *right,
5720 int assert_on_failure)
5721{
5722 int i;
5723 SCOPE_ENTER(2);
5724
5725 if (left == right) {
5726 SCOPE_EXIT_RTN_VALUE(1, "equal\n");
5727 }
5728
5729 if (!(left && right)) {
5730 ast_assert(!assert_on_failure);
5731 SCOPE_EXIT_RTN_VALUE(0, "one is null: left: %p right: %p\n", left, right);
5732 }
5733
5734 if (!ast_stream_topology_equal(left->topology, right->topology)) {
5735 ast_assert(!assert_on_failure);
5736 SCOPE_EXIT_RTN_VALUE(0, "topologies differ\n");
5737 }
5738 if (AST_VECTOR_SIZE(&left->sessions) != AST_VECTOR_SIZE(&right->sessions)) {
5739 ast_assert(!assert_on_failure);
5740 SCOPE_EXIT_RTN_VALUE(0, "session vector sizes different: left %zu != right %zu\n",
5741 AST_VECTOR_SIZE(&left->sessions),
5742 AST_VECTOR_SIZE(&right->sessions));
5743 }
5745 ast_assert(!assert_on_failure);
5746 SCOPE_EXIT_RTN_VALUE(0, "read_callback vector sizes different: left %zu != right %zu\n",
5749 }
5750
5751 for (i = 0; i < AST_VECTOR_SIZE(&left->sessions) ; i++) {
5752 if (!test_is_media_session_equal(AST_VECTOR_GET(&left->sessions, i), AST_VECTOR_GET(&right->sessions, i))) {
5753 ast_assert(!assert_on_failure);
5754 SCOPE_EXIT_RTN_VALUE(0, "Media session %d different\n", i);
5755 }
5756 }
5757
5758 for (i = 0; i < AST_VECTOR_SIZE(&left->read_callbacks) ; i++) {
5759 if (memcmp(AST_VECTOR_GET_ADDR(&left->read_callbacks, i),
5761 sizeof(struct ast_sip_session_media_read_callback_state)) != 0) {
5762 ast_assert(!assert_on_failure);
5763 SCOPE_EXIT_RTN_VALUE(0, "read_callback %d different\n", i);
5764 }
5765 }
5766
5767 for (i = 0; i < AST_MEDIA_TYPE_END; i++) {
5768 if (!(left->default_session[i] && right->default_session[i])) {
5769 continue;
5770 }
5771 if (!left->default_session[i] || !right->default_session[i]
5772 || left->default_session[i]->stream_num != right->default_session[i]->stream_num) {
5773 ast_assert(!assert_on_failure);
5774 SCOPE_EXIT_RTN_VALUE(0, "Default media session %d different. Left: %s Right: %s\n", i,
5775 left->default_session[i] ? left->default_session[i]->label : "null",
5776 right->default_session[i] ? right->default_session[i]->label : "null");
5777 }
5778 }
5779
5780 SCOPE_EXIT_RTN_VALUE(1, "equal\n");
5781}
5782
5783AST_TEST_DEFINE(test_resolve_refresh_media_states)
5784{
5785#define FREE_STATE() \
5786({ \
5787 ast_sip_session_media_state_free(new_pending_state); \
5788 new_pending_state = NULL; \
5789 ast_sip_session_media_state_free(delayed_pending_state); \
5790 delayed_pending_state = NULL; \
5791 ast_sip_session_media_state_free(delayed_active_state); \
5792 delayed_active_state = NULL; \
5793 ast_sip_session_media_state_free(current_active_state); \
5794 current_active_state = NULL; \
5795 ast_sip_session_media_state_free(expected_pending_state); \
5796 expected_pending_state = NULL; \
5797})
5798
5799#define RESET_STATE(__num) \
5800({ \
5801 testnum=__num; \
5802 ast_trace(-1, "Test %d\n", testnum); \
5803 test_failed = 0; \
5804 delayed_pending_state = ast_sip_session_media_state_alloc(); \
5805 delayed_pending_state->topology = ast_stream_topology_alloc(); \
5806 delayed_active_state = ast_sip_session_media_state_alloc(); \
5807 delayed_active_state->topology = ast_stream_topology_alloc(); \
5808 current_active_state = ast_sip_session_media_state_alloc(); \
5809 current_active_state->topology = ast_stream_topology_alloc(); \
5810 expected_pending_state = ast_sip_session_media_state_alloc(); \
5811 expected_pending_state->topology = ast_stream_topology_alloc(); \
5812})
5813
5814#define CHECKER() \
5815({ \
5816 new_pending_state = resolve_refresh_media_states("unittest", delayed_pending_state, delayed_active_state, current_active_state, 1); \
5817 if (!test_is_media_state_equal(new_pending_state, expected_pending_state, 0)) { \
5818 res = AST_TEST_FAIL; \
5819 test_failed = 1; \
5820 ast_test_status_update(test, "da: %s\n", ast_str_tmp(256, ast_stream_topology_to_str(delayed_active_state->topology, &STR_TMP))); \
5821 ast_test_status_update(test, "dp: %s\n", ast_str_tmp(256, ast_stream_topology_to_str(delayed_pending_state->topology, &STR_TMP))); \
5822 ast_test_status_update(test, "ca: %s\n", ast_str_tmp(256, ast_stream_topology_to_str(current_active_state->topology, &STR_TMP))); \
5823 ast_test_status_update(test, "ep: %s\n", ast_str_tmp(256, ast_stream_topology_to_str(expected_pending_state->topology, &STR_TMP))); \
5824 ast_test_status_update(test, "np: %s\n", ast_str_tmp(256, ast_stream_topology_to_str(new_pending_state->topology, &STR_TMP))); \
5825 } \
5826 ast_test_status_update(test, "Test %d %s\n", testnum, test_failed ? "FAILED" : "passed"); \
5827 ast_trace(-1, "Test %d %s\n", testnum, test_failed ? "FAILED" : "passed"); \
5828 test_failed = 0; \
5829 FREE_STATE(); \
5830})
5831
5832
5833 struct ast_sip_session_media_state * delayed_pending_state = NULL;
5834 struct ast_sip_session_media_state * delayed_active_state = NULL;
5835 struct ast_sip_session_media_state * current_active_state = NULL;
5836 struct ast_sip_session_media_state * new_pending_state = NULL;
5837 struct ast_sip_session_media_state * expected_pending_state = NULL;
5839 int test_failed = 0;
5840 int testnum = 0;
5841 SCOPE_ENTER(1);
5842
5843 switch (cmd) {
5844 case TEST_INIT:
5845 info->name = "merge_refresh_topologies";
5846 info->category = "/res/res_pjsip_session/";
5847 info->summary = "Test merging of delayed request topologies";
5848 info->description = "Test merging of delayed request topologies";
5850 case TEST_EXECUTE:
5851 break;
5852 }
5853
5854 RESET_STATE(1);
5855 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5856 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5857 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5858
5859 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5860 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5861 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5862 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5863
5864 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5865 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5866 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5867
5868 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5869 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5870 test_media_add(expected_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5871 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5872 CHECKER();
5873
5874 RESET_STATE(2);
5875 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5876 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5877 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5878
5879 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5880 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5881 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5882 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5883
5884 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5885 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5886 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5887
5888 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5889 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5890 test_media_add(expected_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5891 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5892 CHECKER();
5893
5894 RESET_STATE(3);
5895 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5896 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5897 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5898
5899 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5900 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5901 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5902 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5903
5904 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5905 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5906 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5907 test_media_add(current_active_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5908 test_media_add(current_active_state, "myvideo5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5909
5910 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5911 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5912 test_media_add(expected_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5913 test_media_add(expected_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5914 test_media_add(expected_pending_state, "myvideo5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5915 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5916 CHECKER();
5917
5918 RESET_STATE(4);
5919 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5920 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5921 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5922
5923 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5924 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5925 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5926 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5927
5928 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5929 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5930 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
5931
5932 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5933 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5934 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5935 CHECKER();
5936
5937 RESET_STATE(5);
5938 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5939 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5940 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5941
5942 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5943 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5944 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
5945
5946 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5947 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5948 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
5949
5950 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5951 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5952 test_media_add(expected_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
5953 CHECKER();
5954
5955 RESET_STATE(6);
5956 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5957 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5958 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5959
5960 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5961 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5962 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5963 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5964
5965 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5966 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5967 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
5968 test_media_add(current_active_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5969
5970 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5971 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5972 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5973 test_media_add(expected_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5974 CHECKER();
5975
5976 RESET_STATE(7);
5977 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5978 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5979 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5980
5981 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5982 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5983 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5984 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5985 test_media_add(delayed_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5986
5987 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5988 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5989 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5990 test_media_add(current_active_state, "myvideo5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5991 test_media_add(current_active_state, "myvideo6", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5992
5993 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
5994 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5995 test_media_add(expected_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5996 test_media_add(expected_pending_state, "myvideo5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5997 test_media_add(expected_pending_state, "myvideo6", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5998 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
5999 test_media_add(expected_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6000 CHECKER();
6001
6002 RESET_STATE(8);
6003 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6004 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6005 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6006
6007 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6008 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6009 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6010 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6011 test_media_add(delayed_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6012
6013 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6014 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6015 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
6016
6017 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6018 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6019 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6020 test_media_add(expected_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6021 CHECKER();
6022
6023 RESET_STATE(9);
6024 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6025 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6026 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6027
6028 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6029 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6030 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6031 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6032 test_media_add(delayed_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6033
6034 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6035 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
6036 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
6037
6038 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6039 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6040 test_media_add(expected_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6041 CHECKER();
6042
6043 RESET_STATE(10);
6044 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6045 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6046 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6047
6048 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6049 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
6050 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
6051
6052 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6053 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6054 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6055 test_media_add(current_active_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6056
6057 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6058 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
6059 test_media_add(expected_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_REMOVED, -1);
6060 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6061 CHECKER();
6062
6063 RESET_STATE(11);
6064 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6065 test_media_add(delayed_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6066 test_media_add(delayed_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6067 test_media_add(delayed_active_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6068
6069 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6070 test_media_add(delayed_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6071 test_media_add(delayed_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6072 test_media_add(delayed_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6073
6074 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6075 test_media_add(current_active_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6076 test_media_add(current_active_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6077 test_media_add(current_active_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6078
6079 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6080 test_media_add(expected_pending_state, "myvideo1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6081 test_media_add(expected_pending_state, "myvideo2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6082 test_media_add(expected_pending_state, "myvideo4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6083 test_media_add(expected_pending_state, "myvideo3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6084 CHECKER();
6085
6086 RESET_STATE(12);
6087 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6088 test_media_add(delayed_active_state, "292-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6089 test_media_add(delayed_active_state, "296-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6090
6091 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6092 test_media_add(delayed_pending_state, "292-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6093 test_media_add(delayed_pending_state, "296-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6094 test_media_add(delayed_pending_state, "297-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6095 test_media_add(delayed_pending_state, "294-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6096
6097 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6098 test_media_add(current_active_state, "292-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6099 test_media_add(current_active_state, "296-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6100 test_media_add(current_active_state, "290-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6101 test_media_add(current_active_state, "297-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6102
6103 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6104 test_media_add(expected_pending_state, "292-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6105 test_media_add(expected_pending_state, "296-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6106 test_media_add(expected_pending_state, "290-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6107 test_media_add(expected_pending_state, "297-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6108 test_media_add(expected_pending_state, "294-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6109 CHECKER();
6110
6111 RESET_STATE(13);
6112 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6113 test_media_add(delayed_active_state, "293-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6114 test_media_add(delayed_active_state, "292-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6115 test_media_add(delayed_active_state, "294-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6116 test_media_add(delayed_active_state, "295-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6117 test_media_add(delayed_active_state, "296-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6118
6119 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6120 test_media_add(delayed_pending_state, "293-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6121 test_media_add(delayed_pending_state, "292-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6122 test_media_add(delayed_pending_state, "294-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6123 test_media_add(delayed_pending_state, "295-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6124 test_media_add(delayed_pending_state, "296-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6125 test_media_add(delayed_pending_state, "298-7", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6126
6127 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6128 test_media_add(current_active_state, "293-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6129 test_media_add(current_active_state, "292-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6130 test_media_add(current_active_state, "294-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6131 test_media_add(current_active_state, "295-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6132 test_media_add(current_active_state, "296-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6133 test_media_add(current_active_state, "290-6", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6134
6135 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6136 test_media_add(expected_pending_state, "293-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6137 test_media_add(expected_pending_state, "292-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6138 test_media_add(expected_pending_state, "294-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6139 test_media_add(expected_pending_state, "295-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6140 test_media_add(expected_pending_state, "296-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6141 test_media_add(expected_pending_state, "290-6", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6142 test_media_add(expected_pending_state, "298-7", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6143 CHECKER();
6144
6145 RESET_STATE(14);
6146 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6147 test_media_add(delayed_active_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6148 test_media_add(delayed_active_state, "297-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6149
6150 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6151 test_media_add(delayed_pending_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6152 test_media_add(delayed_pending_state, "294-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6153 test_media_add(delayed_pending_state, "295-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6154
6155 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6156 test_media_add(current_active_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6157 test_media_add(current_active_state, "297-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6158 test_media_add(current_active_state, "291-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6159 test_media_add(current_active_state, "294-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6160
6161 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6162 test_media_add(expected_pending_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6163 test_media_add(expected_pending_state, "297-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6164 test_media_add(expected_pending_state, "291-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6165 test_media_add(expected_pending_state, "294-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6166 test_media_add(expected_pending_state, "295-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6167 CHECKER();
6168
6169 RESET_STATE(15);
6170 test_media_add(delayed_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6171 test_media_add(delayed_active_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6172 test_media_add(delayed_active_state, "297-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6173
6174 test_media_add(delayed_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6175 test_media_add(delayed_pending_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDONLY, -1);
6176 test_media_add(delayed_pending_state, "294-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6177 test_media_add(delayed_pending_state, "295-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6178
6179 test_media_add(current_active_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6180 test_media_add(current_active_state, "297-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6181 test_media_add(current_active_state, "291-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6182 test_media_add(current_active_state, "294-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6183 test_media_add(current_active_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6184
6185 test_media_add(expected_pending_state, "audio", AST_MEDIA_TYPE_AUDIO, AST_STREAM_STATE_SENDRECV, -1);
6186 test_media_add(expected_pending_state, "297-2", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6187 test_media_add(expected_pending_state, "291-3", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6188 test_media_add(expected_pending_state, "294-4", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6189 test_media_add(expected_pending_state, "298-1", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDONLY, -1);
6190 test_media_add(expected_pending_state, "295-5", AST_MEDIA_TYPE_VIDEO, AST_STREAM_STATE_SENDRECV, -1);
6191 CHECKER();
6192
6194}
6195#endif /* TEST_FRAMEWORK */
6196
6197static int load_module(void)
6198{
6199 pjsip_endpoint *endpt;
6200
6203 }
6204 if (!(nat_hook = ast_sorcery_alloc(ast_sip_get_sorcery(), "nat_hook", NULL))) {
6206 }
6211 if (!sdp_handlers) {
6213 }
6215 pjsip_inv_usage_init(endpt, &inv_callback);
6216 pjsip_100rel_init_module(endpt);
6217 pjsip_timer_init_module(endpt);
6220 }
6223
6225
6227#ifdef TEST_FRAMEWORK
6228 AST_TEST_REGISTER(test_resolve_refresh_media_states);
6229#endif
6231}
6232
6233static int unload_module(void)
6234{
6236
6237#ifdef TEST_FRAMEWORK
6238 AST_TEST_UNREGISTER(test_resolve_refresh_media_states);
6239#endif
6246 return 0;
6247}
6248
6250 .support_level = AST_MODULE_SUPPORT_CORE,
6251 .load = load_module,
6252 .unload = unload_module,
6253 .load_pri = AST_MODPRI_APP_DEPEND,
6254 .requires = "res_pjsip",
Access Control of various sorts.
jack_status_t status
Definition app_jack.c:149
const char * str
Definition app_jack.c:150
enum queue_result id
Definition app_queue.c:1790
ast_mutex_t lock
Definition app_sla.c:337
char * strsep(char **str, const char *delims)
Asterisk main include file. File version handling, generic pbx functions.
int ast_shutdown_final(void)
Definition asterisk.c:1884
static struct ast_mansession session
#define ast_alloca(size)
call __builtin_alloca to ensure we get gcc builtin semantics
Definition astmm.h:288
#define ast_free(a)
Definition astmm.h:180
#define ast_strdup(str)
A wrapper for strdup()
Definition astmm.h:241
#define ast_strdupa(s)
duplicate a string in memory from the stack
Definition astmm.h:298
#define ast_asprintf(ret, fmt,...)
A wrapper for asprintf()
Definition astmm.h:267
#define ast_calloc(num, len)
A wrapper for calloc()
Definition astmm.h:202
#define ast_log
Definition astobj2.c:42
#define ao2_link(container, obj)
Add an object to a container.
Definition astobj2.h:1532
@ CMP_MATCH
Definition astobj2.h:1027
@ CMP_STOP
Definition astobj2.h:1028
#define OBJ_KEY
Definition astobj2.h:1151
@ AO2_ALLOC_OPT_LOCK_NOLOCK
Definition astobj2.h:367
@ AO2_ALLOC_OPT_LOCK_MUTEX
Definition astobj2.h:363
#define ao2_callback(c, flags, cb_fn, arg)
ao2_callback() is a generic function that applies cb_fn() to all objects in a container,...
Definition astobj2.h:1693
#define ao2_cleanup(obj)
Definition astobj2.h:1934
#define ao2_callback_data(container, flags, cb_fn, arg, data)
Definition astobj2.h:1723
#define ao2_find(container, arg, flags)
Definition astobj2.h:1736
#define ao2_ref(o, delta)
Reference/unreference an object and return the old refcount.
Definition astobj2.h:459
#define ao2_alloc_options(data_size, destructor_fn, options)
Definition astobj2.h:404
#define ao2_bump(obj)
Bump refcount on an AO2 object by one, returning the object.
Definition astobj2.h:480
@ OBJ_NODATA
Definition astobj2.h:1044
@ OBJ_UNLINK
Definition astobj2.h:1039
#define ao2_alloc(data_size, destructor_fn)
Definition astobj2.h:409
#define ao2_container_alloc_hash(ao2_options, container_options, n_buckets, hash_fn, sort_fn, cmp_fn)
Allocate and initialize a hash container with the desired number of buckets.
Definition astobj2.h:1303
CallerID (and other GR30) management and generation Includes code and algorithms from the Zapata libr...
#define AST_PRES_ALLOWED
Definition callerid.h:432
#define AST_PRES_RESTRICTION
Definition callerid.h:431
Internal Asterisk hangup causes.
#define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL
Definition causes.h:130
static struct ast_timer * timer
Definition chan_iax2.c:401
static const char type[]
static int answer(void *data)
Definition chan_pjsip.c:783
static struct unistimsession * sessions
const char * ast_channel_name(const struct ast_channel *chan)
#define AST_EXTENDED_FDS
Definition channel.h:197
struct ast_stream_topology * ast_channel_set_stream_topology(struct ast_channel *chan, struct ast_stream_topology *topology)
Set the topology of streams on a channel.
void ast_party_id_init(struct ast_party_id *init)
Initialize the given party id structure.
Definition channel.c:1744
int ast_queue_hangup(struct ast_channel *chan)
Queue a hangup frame.
Definition channel.c:1182
int ast_party_id_presentation(const struct ast_party_id *id)
Determine the overall presentation value for the given party.
Definition channel.c:1808
#define ast_channel_lock(chan)
Definition channel.h:2983
void ast_party_id_free(struct ast_party_id *doomed)
Destroy the party id contents.
Definition channel.c:1798
int ast_queue_frame(struct ast_channel *chan, struct ast_frame *f)
Queue one or more frames to a channel's frame queue.
Definition channel.c:1171
void ast_channel_internal_fd_set(struct ast_channel *chan, int which, int value)
int ast_channel_hangupcause(const struct ast_channel *chan)
void ast_set_hangupsource(struct ast_channel *chan, const char *source, int force)
Set the source of the hangup in this channel and it's bridge.
Definition channel.c:2498
int ast_queue_unhold(struct ast_channel *chan)
Queue an unhold frame.
Definition channel.c:1274
#define AST_CHANNEL_NAME
Definition channel.h:173
int ast_channel_stream_topology_changed_externally(struct ast_channel *chan)
Provide notice from a channel that the topology has changed on it as a result of the remote party ren...
Definition channel.c:11188
void ast_party_id_copy(struct ast_party_id *dest, const struct ast_party_id *src)
Copy the source party id information to the destination party id.
Definition channel.c:1752
struct ast_party_id ast_channel_connected_effective_id(struct ast_channel *chan)
void ast_channel_internal_fd_clear(struct ast_channel *chan, int which)
void ast_channel_hangupcause_set(struct ast_channel *chan, int value)
#define ast_channel_unlock(chan)
Definition channel.h:2984
#define AST_MAX_EXTENSION
Definition channel.h:134
static struct ast_channel * callback(struct ast_channelstorage_instance *driver, ao2_callback_data_fn *cb_fn, void *arg, void *data, int ao2_flags, int rdlock)
ast_media_type
Types of media.
Definition codec.h:30
@ AST_MEDIA_TYPE_AUDIO
Definition codec.h:32
@ AST_MEDIA_TYPE_UNKNOWN
Definition codec.h:31
@ AST_MEDIA_TYPE_VIDEO
Definition codec.h:33
@ AST_MEDIA_TYPE_END
Definition codec.h:36
@ AST_MEDIA_TYPE_IMAGE
Definition codec.h:34
@ AST_MEDIA_TYPE_TEXT
Definition codec.h:35
enum ast_media_type ast_media_type_from_str(const char *media_type_str)
Conversion function to take a media string and convert it to a media type.
Definition codec.c:364
const char * ast_codec_media_type2str(enum ast_media_type type)
Conversion function to take a media type and turn it into a string.
Definition codec.c:348
#define SENTINEL
Definition compiler.h:87
Asterisk datastore objects.
Convenient Signal Processing routines.
void ast_dsp_free(struct ast_dsp *dsp)
Definition dsp.c:1968
#define DSP_FEATURE_DIGIT_DETECT
Definition dsp.h:28
#define DSP_FEATURE_FAX_DETECT
Definition dsp.h:29
void ast_dsp_set_features(struct ast_dsp *dsp, int features)
Select feature set.
Definition dsp.c:1953
struct ast_dsp * ast_dsp_new(void)
Allocates a new dsp, assumes 8khz for internal sample rate.
Definition dsp.c:1943
char * end
Definition eagi_proxy.c:73
char buf[BUFSIZE]
Definition eagi_proxy.c:66
int ast_format_cap_get_compatible(const struct ast_format_cap *cap1, const struct ast_format_cap *cap2, struct ast_format_cap *result)
Find the compatible formats between two capabilities structures.
Definition format_cap.c:636
@ AST_FORMAT_CAP_FLAG_DEFAULT
Definition format_cap.h:38
#define ast_format_cap_alloc(flags)
Allocate a new ast_format_cap structure.
Definition format_cap.h:49
size_t ast_format_cap_count(const struct ast_format_cap *cap)
Get the number of formats present within the capabilities structure.
Definition format_cap.c:403
static const char name[]
Definition format_mp3.c:68
static int len(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t buflen)
#define SCOPE_EXIT_LOG_EXPR(__expr, __log_level,...)
#define SCOPE_EXIT_RTN(...)
#define TRACE_ATLEAST(level)
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
#define ast_trace_log(__level, __log_level,...)
#define SCOPE_EXIT_LOG_RTN_VALUE(__value, __log_level,...)
#define SCOPE_ENTER(level,...)
#define SCOPE_EXIT_EXPR(__expr,...)
#define SCOPE_EXIT(...)
#define SCOPE_EXIT_LOG_RTN(__log_level,...)
#define ast_trace(level,...)
struct ast_taskprocessor * ast_sip_get_distributor_serializer(pjsip_rx_data *rdata)
Determine the distributor serializer for the SIP message.
#define ast_sip_push_task(serializer, sip_task, task_data)
Definition res_pjsip.h:2126
void ast_sip_dialog_set_endpoint(pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
Set an endpoint on a SIP dialog so in-dialog requests do not undergo endpoint lookup.
void ast_sip_dialog_set_serializer(pjsip_dialog *dlg, struct ast_taskprocessor *serializer)
Set a serializer on a SIP dialog so requests and responses are automatically serialized.
struct ast_taskprocessor * ast_sip_get_distributor_serializer_dialog(pjsip_dialog *dlg)
Determine the distributor serializer for the SIP dialog.
static int session_count
Definition http.c:110
struct ast_features_pickup_config * ast_get_chan_features_pickup_config(struct ast_channel *chan)
Get the pickup configuration options for a channel.
@ AST_FRAME_CONTROL
@ AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED
struct ast_frame ast_null_frame
Definition main/frame.c:79
Support for logging to various files, console and syslog Configuration in file logger....
#define DEBUG_ATLEAST(level)
#define ast_debug(level,...)
Log a DEBUG message.
#define LOG_DEBUG
#define LOG_ERROR
#define LOG_NOTICE
#define LOG_WARNING
#define AST_LIST_HEAD_INIT_NOLOCK(head)
Initializes a list head structure.
#define AST_LIST_HEAD_NOLOCK(name, type)
Defines a structure to be used to hold a list of specified type (with no lock).
#define AST_LIST_TRAVERSE(head, var, field)
Loops over (traverses) the entries in a list.
#define AST_LIST_HEAD_DESTROY(head)
Destroys a list head structure.
#define AST_LIST_EMPTY(head)
Checks whether the specified list contains any entries.
#define AST_LIST_INSERT_TAIL(head, elm, field)
Appends a list entry to the tail of a list.
#define AST_LIST_ENTRY(type)
Declare a forward link structure inside a list entry.
#define AST_LIST_HEAD_INIT(head)
Initializes a list head structure.
#define AST_LIST_TRAVERSE_SAFE_END
Closes a safe loop traversal block.
#define AST_LIST_INSERT_HEAD(head, elm, field)
Inserts a list entry at the head of a list.
#define AST_LIST_TRAVERSE_SAFE_BEGIN(head, var, field)
Loops safely over (traverses) the entries in a list.
#define AST_LIST_REMOVE_CURRENT(field)
Removes the current entry from a list during a traversal.
#define AST_LIST_REMOVE_HEAD(head, field)
Removes and returns the head entry from a list.
Asterisk locking-related definitions:
#define SCOPED_AO2LOCK(varname, obj)
scoped lock specialization for ao2 mutexes.
Definition lock.h:611
Asterisk module definitions.
@ AST_MODFLAG_LOAD_ORDER
Definition module.h:331
@ AST_MODFLAG_GLOBAL_SYMBOLS
Definition module.h:330
#define ast_module_shutdown_ref(mod)
Prevent unload of the module before shutdown.
Definition module.h:478
#define AST_MODULE_INFO(keystr, flags_to_set, desc, fields...)
Definition module.h:557
@ AST_MODPRI_APP_DEPEND
Definition module.h:342
@ AST_MODULE_SUPPORT_CORE
Definition module.h:121
#define ASTERISK_GPL_KEY
The text the key() function should return.
Definition module.h:46
@ AST_MODULE_LOAD_SUCCESS
Definition module.h:70
@ AST_MODULE_LOAD_DECLINE
Module has failed to load, may be in an inconsistent state.
Definition module.h:78
int ast_sockaddr_parse(struct ast_sockaddr *addr, const char *str, int flags)
Parse an IPv4 or IPv6 address string.
Definition netsock2.c:230
#define AST_SOCKADDR_BUFLEN
Definition netsock2.h:46
static char * ast_sockaddr_stringify_addr_remote(const struct ast_sockaddr *addr)
Wrapper around ast_sockaddr_stringify_fmt() to return an address only.
Definition netsock2.h:313
Core PBX routines and definitions.
const char * pbx_builtin_getvar_helper(struct ast_channel *chan, const char *name)
Return a pointer to the value of the corresponding channel variable.
int ast_exists_extension(struct ast_channel *c, const char *context, const char *exten, int priority, const char *callerid)
Determine whether an extension exists.
Definition pbx.c:2735
int ast_canmatch_extension(struct ast_channel *c, const char *context, const char *exten, int priority, const char *callerid)
Looks for a valid matching extension.
Definition pbx.c:2750
Call Pickup API.
void pjsip_reason_header_unload(void)
void pjsip_reason_header_load(void)
static const pjsip_method message_method
Definition res_pjsip.c:1273
const char * method
Definition res_pjsip.c:1277
void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri)
Add 'user=phone' parameter to URI if enabled and user is a phone number.
Definition res_pjsip.c:928
const pj_str_t * ast_sip_pjsip_uri_get_username(pjsip_uri *uri)
Get the user portion of the pjsip_uri.
Definition res_pjsip.c:3452
void ast_sip_unregister_service(pjsip_module *module)
Definition res_pjsip.c:127
const char * ast_sip_get_host_ip_string(int af)
Retrieve the local host address in string form.
Definition res_pjsip.c:2468
@ AST_SIP_REDIRECT_URI_CORE
Definition res_pjsip.h:750
@ AST_SIP_REDIRECT_URI_PJSIP
Definition res_pjsip.h:752
@ AST_SIP_REDIRECT_USER
Definition res_pjsip.h:748
int ast_sip_register_service(pjsip_module *module)
Register a SIP service in Asterisk.
Definition res_pjsip.c:111
int ast_sip_is_allowed_uri(pjsip_uri *uri)
Check whether a pjsip_uri is allowed or not.
Definition res_pjsip.c:3447
pjsip_dialog * ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *aor_name, const char *request_user)
General purpose method for creating a UAC dialog with an endpoint.
Definition res_pjsip.c:962
int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge, pjsip_tx_data *tdata, pjsip_tx_data **new_request)
Create a response to an authentication challenge.
Definition res_pjsip.c:208
pjsip_media_type pjsip_media_type_application_sdp
Definition res_pjsip.c:3911
void ast_sip_message_apply_transport(const char *transport_name, pjsip_tx_data *tdata)
Apply the configuration for a transport to an outgoing message.
int ast_sip_set_id_from_invite(struct pjsip_rx_data *rdata, struct ast_party_id *id, struct ast_party_id *default_id, int trust_inbound)
Set the ID from an INVITE.
Definition res_pjsip.c:2802
pjsip_endpoint * ast_sip_get_pjsip_endpoint(void)
Get a pointer to the PJSIP endpoint.
Definition res_pjsip.c:518
#define AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(str)
Truncate the URI user field options string if enabled.
Definition res_pjsip.h:3561
struct ast_sip_endpoint * ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata)
Get the looked-up endpoint on an out-of dialog request or response.
void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
Copy a pj_str_t into a standard character buffer.
Definition res_pjsip.c:2176
pjsip_dialog * ast_sip_create_dialog_uas_locked(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pj_status_t *status)
General purpose method for creating a UAS dialog with an endpoint.
Definition res_pjsip.c:1190
int ast_sip_is_media_type_in(pjsip_media_type *a,...) attribute_sentinel
Check if a media type is in a list of others.
Definition res_pjsip.c:2203
unsigned int ast_sip_get_use_callerid_contact(void)
Retrieve the global setting 'use_callerid_contact'.
@ AST_SIP_CONTACT_FILTER_REACHABLE
Return only reachable or unknown contacts.
Definition res_pjsip.h:1443
#define ast_sip_mod_data_set(pool, mod_data, id, key, val)
Utilizing a mod_data array for a given id, set the value associated with the given key.
Definition res_pjsip.h:3156
@ AST_SIP_DTMF_AUTO
Definition res_pjsip.h:558
@ AST_SIP_DTMF_INBAND
Definition res_pjsip.h:554
#define ast_sip_mod_data_get(mod_data, id, key)
Using the dictionary stored in mod_data array at a given id, retrieve the value associated with the g...
Definition res_pjsip.h:3124
pjsip_media_type pjsip_media_type_multipart_mixed
Definition res_pjsip.c:3913
void ast_sip_location_retrieve_contact_and_aor_from_list_filtered(const char *aor_list, unsigned int flags, struct ast_sip_aor **aor, struct ast_sip_contact **contact)
Retrieve the first bound contact AND the AOR chosen from a list of AORs and filter based on flags.
Definition location.c:272
struct ast_sip_transport_state * ast_sip_get_transport_state(const char *transport_id)
Retrieve transport state.
pjsip_media_type pjsip_media_type_multipart_alternative
Definition res_pjsip.c:3912
struct ast_sorcery * ast_sip_get_sorcery(void)
Get a pointer to the SIP sorcery structure.
#define MAX_RX_CHALLENGES
Definition res_pjsip.h:108
unsigned int ast_sip_get_all_codecs_on_empty_reinvite(void)
Retrieve the system setting 'all_codecs_on_empty_reinvite'.
@ AST_SIP_100REL_PEER_SUPPORTED
Definition res_pjsip.h:539
#define ast_sip_transport_is_local(transport_state, addr)
Definition res_pjsip.h:213
int ast_sip_failover_request(pjsip_tx_data *tdata)
Set a request to use the next value in the list of resolved addresses.
Definition res_pjsip.c:1814
ast_sip_session_refresh_method
Definition res_pjsip.h:715
@ AST_SIP_SESSION_REFRESH_METHOD_UPDATE
Definition res_pjsip.h:719
@ AST_SIP_SESSION_REFRESH_METHOD_INVITE
Definition res_pjsip.h:717
void ast_sip_modify_id_header(pj_pool_t *pool, pjsip_fromto_hdr *id_hdr, const struct ast_party_id *id)
Set name and number information on an identity header.
Definition res_pjsip.c:2825
static void session_inv_on_media_update(pjsip_inv_session *inv, pj_status_t status)
#define print_debug_details(inv, tsx, e)
static int remove_handler(void *obj, void *arg, void *data, int flags)
static int check_content_disposition(pjsip_rx_data *rdata)
static int invite_collision_timeout(void *vsession)
static void flush_delayed_requests(struct ast_sip_session *session)
static int set_mid_and_bundle_group(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
struct ast_sip_session * ast_sip_dialog_get_session(pjsip_dialog *dlg)
Retrieves a session from a dialog.
int ast_sip_session_regenerate_answer(struct ast_sip_session *session, ast_sip_session_sdp_creation_cb on_sdp_creation)
Regenerate SDP Answer.
static int handle_negotiated_sdp_session_media(struct ast_sip_session_media *session_media, struct ast_sip_session *session, const pjmedia_sdp_session *local, const pjmedia_sdp_session *remote, int index, struct ast_stream *asterisk_stream)
static int handle_incoming_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
static int session_end_completion(void *vsession)
static int handle_negotiated_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *local, const pjmedia_sdp_session *remote)
void ast_sip_session_unregister_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type)
Unregister an SDP handler.
static pj_bool_t session_reinvite_on_rx_request(pjsip_rx_data *rdata)
static pjsip_module session_module
int ast_sip_can_present_connected_id(const struct ast_sip_session *session, const struct ast_party_id *id)
Determines if the Connected Line info can be presented for this session.
struct ast_sip_session * ast_sip_session_alloc(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, pjsip_inv_session *inv_session, pjsip_rx_data *rdata)
Allocate a new SIP session.
static int media_stats_local_ssrc_cmp(const struct ast_rtp_instance_stats *vec_elem, const struct ast_rtp_instance_stats *srch)
static int check_content_disposition_in_multipart(pjsip_multipart_part *part)
#define GET_STREAM_NAME_SAFE(_stream)
static pjsip_redirect_op session_inv_on_redirected(pjsip_inv_session *inv, const pjsip_uri *target, const pjsip_event *e)
static void handle_incoming_before_media(pjsip_inv_session *inv, struct ast_sip_session *session, pjsip_rx_data *rdata)
int ast_sip_session_register_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type)
Register an SDP handler.
static int session_termination_task(void *data)
static void handle_session_begin(struct ast_sip_session *session)
void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
Send a SIP request.
#define DATASTORE_BUCKETS
static int delay_request(struct ast_sip_session *session, ast_sip_session_request_creation_cb on_request, ast_sip_session_sdp_creation_cb on_sdp_creation, ast_sip_session_response_cb on_response, int generate_new_sdp, enum delayed_method method, struct ast_sip_session_media_state *pending_media_state, struct ast_sip_session_media_state *active_media_state, int queue_head)
static pj_bool_t outbound_invite_auth(pjsip_rx_data *rdata)
static void handle_session_destroy(struct ast_sip_session *session)
static pj_bool_t session_on_rx_request(pjsip_rx_data *rdata)
Called when a new SIP request comes into PJSIP.
pjsip_inv_state ast_sip_session_get_pjsip_inv_state(const struct ast_sip_session *session)
Retrieves the pjsip_inv_state from a session.
static int sdp_handler_list_cmp(void *obj, void *arg, int flags)
static pjsip_inv_session * pre_session_setup(pjsip_rx_data *rdata, const struct ast_sip_endpoint *endpoint)
static int sdp_requires_deferral(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
Determine whether the SDP provided requires deferral of negotiating or not.
static int set_outstanding_invite_timeout(struct ast_sip_session *session)
static int setup_outbound_invite_auth(pjsip_dialog *dlg)
static int uac_invite_tsx_terminates_dialog(pjsip_transaction *tsx)
int ast_sip_session_is_pending_stream_default(const struct ast_sip_session *session, const struct ast_stream *stream)
Determines if a provided pending stream will be the default stream or not.
static void handle_new_invite_request(pjsip_rx_data *rdata)
static void session_media_set_handler(struct ast_sip_session_media *session_media, struct ast_sip_session_sdp_handler *handler)
Set an SDP stream handler for a corresponding session media.
#define STREAM_REMOVED(_stream)
static int add_sdp_streams(struct ast_sip_session_media *session_media, struct ast_sip_session *session, pjmedia_sdp_session *answer, const struct pjmedia_sdp_session *remote, struct ast_stream *stream)
void ast_sip_session_defer_termination_cancel(struct ast_sip_session *session)
Cancel a pending deferred termination.
struct ast_datastore * ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name)
Retrieve a session datastore.
static pjsip_module session_reinvite_module
@ DELAYED_METHOD_INVITE
@ DELAYED_METHOD_BYE
@ DELAYED_METHOD_UPDATE
#define DEFAULT_NUM_SESSION_MEDIA
struct ast_sip_session_media_state * ast_sip_session_media_state_clone(const struct ast_sip_session_media_state *media_state)
Clone a media state.
static void session_inv_on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
static enum sip_get_destination_result get_destination(struct ast_sip_session *session, pjsip_rx_data *rdata)
Determine where in the dialplan a call should go.
struct ast_sip_session_media_state * ast_sip_session_media_state_alloc(void)
Allocate a session media state structure.
#define STATE_REMOVED(_stream_state)
#define SDP_HANDLER_BUCKETS
int ast_sip_session_media_add_read_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media, int fd, ast_sip_session_media_read_cb callback)
Set a read callback for a media session with a specific file descriptor.
static void session_destructor(void *obj)
int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore)
Add a datastore to a SIP session.
static const char * delayed_method2str(enum delayed_method method)
void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata, ast_sip_session_response_cb on_response)
Send a SIP request and get called back when a response is received.
static void handle_session_end(struct ast_sip_session *session)
static int invite_terminated(void *vsession)
static void __print_debug_details(const char *function, pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
static pjsip_module outbound_invite_auth_module
static void set_from_header(struct ast_sip_session *session)
static pjsip_inv_callback inv_callback
#define STATE_NONE(_stream_state)
static struct ast_sip_nat_hook * nat_hook
NAT hook for modifying outgoing messages with SDP.
static int add_bundle_groups(struct ast_sip_session *session, pj_pool_t *pool, pjmedia_sdp_session *answer)
static void session_termination_cb(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry)
static void session_media_dtor(void *obj)
static int send_delayed_request(struct ast_sip_session *session, struct ast_sip_session_delayed_request *delay)
void ast_sip_session_remove_datastore(struct ast_sip_session *session, const char *name)
Remove a session datastore from the session.
struct ast_sip_session_media * ast_sip_session_media_get_transport(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
Retrieve the underlying media session that is acting as transport for a media session.
pjsip_dialog * ast_sip_session_get_dialog(const struct ast_sip_session *session)
Retrieves a dialog from a session.
#define GET_STREAM_SAFE(_topology, _i)
void ast_sip_session_media_state_free(struct ast_sip_session_media_state *media_state)
Free a session media state structure.
static void sip_session_defer_termination_stop_timer(struct ast_sip_session *session)
static pj_status_t session_on_tx_response(pjsip_tx_data *tdata)
static void sip_channel_destroy(void *obj)
Destructor for SIP channel.
#define MOD_DATA_ON_RESPONSE
static int is_media_state_valid(const char *session_name, struct ast_sip_session_media_state *state)
static int get_mid_bundle_group(const pjmedia_sdp_session *sdp, const char *mid)
static int sdp_handler_list_hash(const void *obj, int flags)
static int sip_session_refresh(struct ast_sip_session *session, ast_sip_session_request_creation_cb on_request_creation, ast_sip_session_sdp_creation_cb on_sdp_creation, ast_sip_session_response_cb on_response, enum ast_sip_session_refresh_method method, int generate_new_sdp, struct ast_sip_session_media_state *pending_media_state, struct ast_sip_session_media_state *active_media_state, int queued)
static struct ao2_container * sdp_handlers
Registered SDP stream handlers.
struct ast_datastore * ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid)
Alternative for ast_datastore_alloc()
void ast_sip_session_end_if_deferred(struct ast_sip_session *session)
End the session if it had been previously deferred.
int ast_sip_session_create_invite(struct ast_sip_session *session, pjsip_tx_data **tdata)
Creates an INVITE request.
static struct ast_sip_session_media_state * internal_sip_session_media_state_alloc(size_t sessions, size_t read_callbacks)
int ast_sip_session_defer_termination(struct ast_sip_session *session)
Defer local termination of a session until remote side terminates, or an amount of time passes.
static struct ast_sip_session_media_state * resolve_refresh_media_states(const char *session_name, struct ast_sip_session_media_state *delayed_pending_state, struct ast_sip_session_media_state *delayed_active_state, struct ast_sip_session_media_state *current_active_state, int run_post_validation)
static void session_inv_on_tsx_state_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
static int invite_proceeding(void *vsession)
static void session_inv_on_state_changed(pjsip_inv_session *inv, pjsip_event *e)
static void session_inv_on_rx_offer(pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
struct ast_sip_session_media * ast_sip_session_media_state_add(struct ast_sip_session *session, struct ast_sip_session_media_state *media_state, enum ast_media_type type, int position)
Allocate an ast_session_media and add it to the media state's vector.
static int check_request_status(pjsip_inv_session *inv, pjsip_event *e)
sip_get_destination_result
@ SIP_GET_DEST_EXTEN_FOUND
@ SIP_GET_DEST_EXTEN_PARTIAL
@ SIP_GET_DEST_UNSUPPORTED_URI
@ SIP_GET_DEST_EXTEN_NOT_FOUND
static void set_remote_mslabel_and_stream_group(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream, struct ast_stream *asterisk_stream)
struct ast_sip_channel_pvt * ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_session *session)
Allocate a new SIP channel pvt structure.
static void delayed_request_free(struct ast_sip_session_delayed_request *delay)
static void session_inv_on_new_session(pjsip_inv_session *inv, pjsip_event *e)
static void session_on_tsx_state(pjsip_transaction *tsx, pjsip_event *e)
static void remove_stream_from_bundle(struct ast_sip_session_media *session_media, struct ast_stream *stream)
static int load_module(void)
void ast_sip_session_unsuspend(struct ast_sip_session *session)
Request the session serializer be unsuspended.
void ast_sip_session_terminate(struct ast_sip_session *session, int response)
Terminate a session and, if possible, send the provided response code.
static int datastore_cmp(void *obj, void *arg, int flags)
static pjmedia_sdp_session * generate_session_refresh_sdp(struct ast_sip_session *session)
static int check_sdp_content_type_supported(pjsip_media_type *content_type)
static void handle_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
static struct ast_sip_session_delayed_request * delayed_request_alloc(enum delayed_method method, ast_sip_session_request_creation_cb on_request_creation, ast_sip_session_sdp_creation_cb on_sdp_creation, ast_sip_session_response_cb on_response, int generate_new_sdp, struct ast_sip_session_media_state *pending_media_state, struct ast_sip_session_media_state *active_media_state)
static int update_completed(void *vsession)
static void handle_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata, enum ast_sip_session_response_priority response_priority)
static int unload_module(void)
static int session_end(void *vsession)
static int fetch_callerid_num(struct ast_sip_session *session, pjsip_rx_data *rdata, char *buf, size_t len)
Fetch just the Caller ID number in order of PAI, RPID, From.
static int new_invite_initial_answer(pjsip_inv_session *inv_session, pjsip_rx_data *rdata, int answer_code, int terminate_code, pj_bool_t notify)
static int is_stream_limitation_reached(enum ast_media_type type, const struct ast_sip_endpoint *endpoint, int *type_streams)
static void session_datastore_destroy(void *obj)
void ast_sip_session_resume_reinvite(struct ast_sip_session *session)
Resumes processing of a deferred incoming re-invite.
static pj_bool_t has_supplement(const struct ast_sip_session *session, const pjsip_rx_data *rdata)
static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
void ast_sip_session_media_state_reset(struct ast_sip_session_media_state *media_state)
Reset a media state to a clean state.
static struct pjmedia_sdp_session * create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer, const unsigned int ignore_active_stream_topology)
static pj_bool_t session_on_rx_response(pjsip_rx_data *rdata)
static void handle_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
static void check_delayed_requests(struct ast_sip_session *session, int(*cb)(void *vsession))
int ast_sip_session_refresh(struct ast_sip_session *session, ast_sip_session_request_creation_cb on_request_creation, ast_sip_session_sdp_creation_cb on_sdp_creation, ast_sip_session_response_cb on_response, enum ast_sip_session_refresh_method method, int generate_new_sdp, struct ast_sip_session_media_state *media_state)
Send a reinvite or UPDATE on a session.
static int datastore_hash(const void *obj, int flags)
struct ast_sip_session * ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, const char *location, const char *request_user, struct ast_stream_topology *req_topology)
Create a new outgoing SIP session.
int ast_sip_session_media_set_write_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media, ast_sip_session_media_write_cb callback)
Set a write callback for a media session.
static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata, enum ast_sip_session_response_priority response_priority)
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
Send a SIP response.
static void reschedule_reinvite(struct ast_sip_session *session, ast_sip_session_response_cb on_response)
static void resend_reinvite(pj_timer_heap_t *timer, pj_timer_entry *entry)
void ast_sip_session_suspend(struct ast_sip_session *session)
Request and wait for the session serializer to be suspended.
static void session_outgoing_nat_hook(pjsip_tx_data *tdata, struct ast_sip_transport *transport)
Hook for modifying outgoing messages with SDP to contain the proper address information.
void ast_sip_session_media_stats_save(struct ast_sip_session *sip_session, struct ast_sip_session_media_state *media_state)
Save a media stats.
static int stream_destroy(void *obj, void *arg, int flags)
#define GET_STREAM_STATE_SAFE(_stream)
static int session_end_if_disconnected(int id, pjsip_inv_session *inv)
int(* ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata)
int(* ast_sip_session_sdp_creation_cb)(struct ast_sip_session *session, pjmedia_sdp_session *sdp)
void ast_sip_session_remove_supplements(struct ast_sip_session *session)
Remove supplements from a SIP session.
int ast_sip_session_add_supplements(struct ast_sip_session *session)
Add supplements to a SIP session.
ast_sip_session_response_priority
Describes when a supplement should be called into on incoming responses.
@ AST_SIP_SESSION_BEFORE_REDIRECTING
@ AST_SIP_SESSION_AFTER_MEDIA
@ AST_SIP_SESSION_BEFORE_MEDIA
int(* ast_sip_session_response_cb)(struct ast_sip_session *session, pjsip_rx_data *rdata)
struct ast_frame *(* ast_sip_session_media_read_cb)(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
int(* ast_sip_session_media_write_cb)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, struct ast_frame *frame)
@ AST_SIP_SESSION_OUTGOING_CALL
@ AST_SIP_SESSION_INCOMING_CALL
int ast_sip_session_check_supplement_create(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, const char *location, const char *request_user, struct ast_stream_topology *req_topology)
Check registered supplements for permission to create an outgoing session.
ast_sip_session_sdp_stream_defer
@ AST_SIP_SESSION_SDP_DEFER_NEEDED
@ AST_SIP_SESSION_SDP_DEFER_NOT_HANDLED
@ AST_SIP_SESSION_SDP_DEFER_NOT_NEEDED
@ AST_SIP_SESSION_SDP_DEFER_ERROR
struct ast_stream * ast_sip_session_create_joint_call_stream(const struct ast_sip_session *session, struct ast_stream *remote)
Create a new stream of joint capabilities.
#define NULL
Definition resample.c:96
@ AST_RTP_INSTANCE_STAT_ALL
Definition rtp_engine.h:187
int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
Retrieve statistics about an RTP instance.
SRTP and SDP Security descriptions.
void ast_sdp_srtp_destroy(struct ast_sdp_srtp *srtp)
free a ast_sdp_srtp structure
Definition sdp_srtp.c:51
const char * ast_sorcery_object_get_id(const void *object)
Get the unique identifier of a sorcery object.
Definition sorcery.c:2381
int ast_sorcery_create(const struct ast_sorcery *sorcery, void *object)
Create and potentially persist an object using an available wizard.
Definition sorcery.c:2126
void * ast_sorcery_alloc(const struct ast_sorcery *sorcery, const char *type, const char *id)
Allocate an object.
Definition sorcery.c:1808
int ast_sorcery_delete(const struct ast_sorcery *sorcery, void *object)
Delete an object.
Definition sorcery.c:2302
Media Stream API.
struct ast_stream_topology * ast_stream_topology_alloc(void)
Create a stream topology.
Definition stream.c:652
const char * ast_stream_to_str(const struct ast_stream *stream, struct ast_str **buf)
Get a string representing the stream for debugging/display purposes.
Definition stream.c:337
struct ast_stream * ast_stream_alloc(const char *name, enum ast_media_type type)
Create a new media stream representation.
Definition stream.c:233
int ast_stream_topology_set_stream(struct ast_stream_topology *topology, unsigned int position, struct ast_stream *stream)
Set a specific position in a topology.
Definition stream.c:799
const char * ast_stream_get_metadata(const struct ast_stream *stream, const char *m_key)
Get a stream metadata value.
Definition stream.c:423
const char * ast_stream_get_name(const struct ast_stream *stream)
Get the name of a stream.
Definition stream.c:309
int ast_stream_set_metadata(struct ast_stream *stream, const char *m_key, const char *value)
Set a stream metadata value.
Definition stream.c:460
ast_stream_state
States that a stream may be in.
Definition stream.h:74
@ AST_STREAM_STATE_RECVONLY
Set when the stream is receiving media only.
Definition stream.h:90
@ AST_STREAM_STATE_END
Sentinel.
Definition stream.h:98
@ AST_STREAM_STATE_INACTIVE
Set when the stream is not sending OR receiving media.
Definition stream.h:94
@ AST_STREAM_STATE_REMOVED
Set when the stream has been removed/declined.
Definition stream.h:78
@ AST_STREAM_STATE_SENDRECV
Set when the stream is sending and receiving media.
Definition stream.h:82
@ AST_STREAM_STATE_SENDONLY
Set when the stream is sending media only.
Definition stream.h:86
const char * ast_stream_state2str(enum ast_stream_state state)
Convert the state of a stream into a string.
Definition stream.c:388
void ast_stream_set_state(struct ast_stream *stream, enum ast_stream_state state)
Set the state of a stream.
Definition stream.c:380
int ast_stream_topology_append_stream(struct ast_stream_topology *topology, struct ast_stream *stream)
Append a stream to the topology.
Definition stream.c:751
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition stream.c:939
struct ast_stream * ast_stream_clone(const struct ast_stream *stream, const char *name)
Create a deep clone of an existing stream.
Definition stream.c:257
struct ast_stream * ast_stream_topology_get_stream(const struct ast_stream_topology *topology, unsigned int position)
Get a specific stream from the topology.
Definition stream.c:791
int ast_stream_topology_get_count(const struct ast_stream_topology *topology)
Get the number of streams in a topology.
Definition stream.c:768
int ast_stream_get_format_count(const struct ast_stream *stream)
Get the count of the current negotiated formats of a stream.
Definition stream.c:358
enum ast_stream_state ast_stream_get_state(const struct ast_stream *stream)
Get the current state of a stream.
Definition stream.c:373
int ast_stream_topology_del_stream(struct ast_stream_topology *topology, unsigned int position)
Delete a specified stream from the given topology.
Definition stream.c:828
enum ast_media_type ast_stream_get_type(const struct ast_stream *stream)
Get the media type of a stream.
Definition stream.c:316
void ast_stream_topology_free(struct ast_stream_topology *topology)
Unreference and destroy a stream topology.
Definition stream.c:746
void ast_stream_set_formats(struct ast_stream *stream, struct ast_format_cap *caps)
Set the current negotiated formats of a stream.
Definition stream.c:365
void ast_stream_free(struct ast_stream *stream)
Destroy a media stream representation.
Definition stream.c:292
void ast_stream_set_group(struct ast_stream *stream, int group)
Set the stream group for a stream.
Definition stream.c:1087
const struct ast_format_cap * ast_stream_get_formats(const struct ast_stream *stream)
Get the current negotiated formats of a stream.
Definition stream.c:330
int ast_stream_topology_equal(const struct ast_stream_topology *left, const struct ast_stream_topology *right)
Compare two stream topologies to see if they are equal.
Definition stream.c:699
struct ast_stream_topology * ast_stream_topology_clone(const struct ast_stream_topology *topology)
Create a deep clone of an existing stream topology.
Definition stream.c:670
int ast_str_append(struct ast_str **buf, ssize_t max_len, const char *fmt,...)
Append to a thread local dynamic string.
Definition strings.h:1139
int ast_strings_equal(const char *str1, const char *str2)
Compare strings for equality checking for NULL.
Definition strings.c:238
#define S_OR(a, b)
returns the equivalent of logic or for strings: first one if not empty, otherwise second one.
Definition strings.h:80
static force_inline int attribute_pure ast_str_hash(const char *str)
Compute a hash value on a string.
Definition strings.h:1259
#define S_COR(a, b, c)
returns the equivalent of logic or for strings, with an additional boolean check: second one if not e...
Definition strings.h:87
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition strings.h:65
#define ast_str_tmp(init_len, __expr)
Provides a temporary ast_str and returns a copy of its buffer.
Definition strings.h:1189
#define ast_str_create(init_len)
Create a malloc'ed dynamic length string.
Definition strings.h:659
int ast_str_set(struct ast_str **buf, ssize_t max_len, const char *fmt,...)
Set a dynamic string using variable arguments.
Definition strings.h:1113
char *attribute_pure ast_str_buffer(const struct ast_str *buf)
Returns the string buffer within the ast_str buf.
Definition strings.h:761
void ast_copy_string(char *dst, const char *src, size_t size)
Size-limited null-terminating string copy.
Definition strings.h:425
Generic container type.
Structure for a data store type.
Definition datastore.h:31
void(* destroy)(void *data)
Definition datastore.h:34
Structure for a data store object.
Definition datastore.h:64
const struct ast_datastore_info * info
Definition datastore.h:67
const char * uid
Definition datastore.h:65
void * data
Definition datastore.h:66
Configuration relating to call pickup.
Format capabilities structure, holds formats + preference order + etc.
Definition format_cap.c:54
Data structure associated with a single frame of data.
struct ast_frame_subclass subclass
struct ast_module * self
Definition module.h:356
Information needed to identify an endpoint in a call.
Definition channel.h:340
struct ast_party_number number
Subscriber phone number.
Definition channel.h:344
unsigned char valid
TRUE if the number information is valid/present.
Definition channel.h:299
char * str
Subscriber phone number (Malloced)
Definition channel.h:293
unsigned int local_ssrc
Definition rtp_engine.h:452
A SIP address of record.
Definition res_pjsip.h:480
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
void * pvt
Pointer to channel specific implementation information, must be ao2 object.
Contact associated with an address of record.
Definition res_pjsip.h:392
const ast_string_field uri
Definition res_pjsip.h:414
struct ast_sip_timer_options timer
Definition res_pjsip.h:821
struct ast_stream_topology * topology
Definition res_pjsip.h:1027
An entity with which Asterisk communicates.
Definition res_pjsip.h:1067
struct ast_sip_endpoint_id_configuration id
Definition res_pjsip.h:1106
const ast_string_field aors
Definition res_pjsip.h:1096
unsigned int moh_passthrough
Definition res_pjsip.h:1140
unsigned int preferred_codec_only
Definition res_pjsip.h:1150
struct ast_sip_endpoint_extensions extensions
Definition res_pjsip.h:1098
const ast_string_field context
Definition res_pjsip.h:1096
struct ast_sip_endpoint_media_configuration media
Definition res_pjsip.h:1100
enum ast_sip_dtmf_mode dtmf
Definition res_pjsip.h:1116
enum ast_sip_100rel_mode rel100
Definition res_pjsip.h:1178
unsigned int faxdetect
Definition res_pjsip.h:1128
Structure for SIP nat hook information.
Definition res_pjsip.h:329
void(* outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport)
Definition res_pjsip.h:333
Structure used for sending delayed requests.
struct ast_sip_session_delayed_request * next
ast_sip_session_sdp_creation_cb on_sdp_creation
struct ast_sip_session_media_state * active_media_state
struct ast_sip_session_media_state * pending_media_state
ast_sip_session_request_creation_cb on_request_creation
ast_sip_session_response_cb on_response
Structure which contains read callback information.
ast_sip_session_media_read_cb read_callback
The callback to invoke.
Structure which contains media state information (streams, sessions)
struct ast_stream_topology * topology
The media stream topology.
struct ast_sip_session_media_state::@281 read_callbacks
Added read callbacks - these are whole structs and not pointers.
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
struct ast_sip_session_media_state::@280 sessions
Mapping of stream to media sessions.
A structure containing SIP session media information.
ast_sip_session_media_write_cb write_callback
The write callback when writing frames.
struct ast_sdp_srtp * srtp
Holds SRTP information.
char * stream_name
Stream name.
struct ast_sip_session_sdp_handler * handler
SDP handler that setup the RTP.
char label[AST_UUID_STR_LEN]
Track label.
unsigned int remote_ice
Does remote support ice.
enum ast_media_type type
Media type of this session media.
unsigned int remote_rtcp_mux
Does remote support rtcp_mux.
char * remote_label
Remote stream label.
int timeout_sched_id
Scheduler ID for RTP timeout.
int stream_num
The stream number to place into any resulting frames.
int bundle_group
The bundle group the stream belongs to.
struct ast_rtp_instance * rtp
RTP instance itself.
enum ast_sip_session_media_encryption encryption
What type of encryption is in use on this stream.
char * remote_mslabel
Remote media stream label.
unsigned int bundled
Whether this stream is currently bundled or not.
unsigned int changed
The underlying session has been changed in some fashion.
int keepalive_sched_id
Scheduler ID for RTP keepalive.
char * mid
Media identifier for this stream (may be shared across multiple streams)
A handler for SDPs in SIP sessions.
void(* stream_destroy)(struct ast_sip_session_media *session_media)
Destroy a session_media created by this handler.
enum ast_sip_session_sdp_stream_defer(* defer_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
Determine whether a stream requires that the re-invite be deferred. If a stream can not be immediatel...
struct ast_sip_session_sdp_handler * next
void(* stream_stop)(struct ast_sip_session_media *session_media)
Stop a session_media created by this handler but do not destroy resources.
A supplement to SIP message processing.
struct ast_module *const char * method
void(* session_begin)(struct ast_sip_session *session)
Notification that the session has begun This method will always be called from a SIP servant thread.
void(* incoming_response)(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
Called on an incoming SIP response This method is always called from a SIP servant thread.
void(* session_destroy)(struct ast_sip_session *session)
Notification that the session is being destroyed.
void(* session_end)(struct ast_sip_session *session)
Notification that the session has ended.
int(* incoming_request)(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
Called on incoming SIP request This method can indicate a failure in processing in its return....
struct ast_sip_session_supplement * next
enum ast_sip_session_response_priority response_priority
void(* outgoing_response)(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
Called on an outgoing SIP response This method is always called from a SIP servant thread.
void(* outgoing_request)(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
Called on an outgoing SIP request This method is always called from a SIP servant thread.
A structure describing a SIP session.
struct ast_sip_contact * contact
struct ast_sip_endpoint * endpoint
struct ast_sip_session::@284 media_stats
char exten[AST_MAX_EXTENSION]
struct ast_sip_session_media_state * active_media_state
struct ast_sip_session_media_state * pending_media_state
struct pjsip_inv_session * inv_session
unsigned int sess_expires
Definition res_pjsip.h:808
unsigned int min_se
Definition res_pjsip.h:806
Structure for SIP transport information.
Definition res_pjsip.h:117
Transport to bind to.
Definition res_pjsip.h:219
const ast_string_field external_media_address
Definition res_pjsip.h:241
Socket address structure.
Definition netsock2.h:97
Support for dynamic strings.
Definition strings.h:623
unsigned int position
The position of the stream in the topology.
Definition stream.c:90
pjsip_rx_data * rdata
INVITE request itself.
struct ast_sip_session * session
Session created for the new INVITE.
Number structure.
struct sdp_handler_list::@506 list
Bundle group building structure.
char * mids[PJMEDIA_MAX_SDP_MEDIA]
The media identifiers in this bundle group.
struct ast_str * attr_string
SDP attribute string.
int value
Definition syslog.c:37
int ast_taskpool_serializer_unsuspend(struct ast_taskprocessor *serializer)
Unsuspend a serializer, causing tasks to be executed.
Definition taskpool.c:1067
int ast_taskpool_serializer_suspend(struct ast_taskprocessor *serializer)
Suspend a serializer, causing tasks to be queued until unsuspended.
Definition taskpool.c:1018
An API for managing task processing threads that can be shared across modules.
void * ast_taskprocessor_unreference(struct ast_taskprocessor *tps)
Unreference the specified taskprocessor and its reference count will decrement.
int ast_taskprocessor_is_task(struct ast_taskprocessor *tps)
Am I the given taskprocessor's current task.
Test Framework API.
@ TEST_INIT
Definition test.h:200
@ TEST_EXECUTE
Definition test.h:201
#define AST_TEST_REGISTER(cb)
Definition test.h:127
#define AST_TEST_UNREGISTER(cb)
Definition test.h:128
#define ast_test_suite_event_notify(s, f,...)
Definition test.h:189
#define AST_TEST_DEFINE(hdr)
Definition test.h:126
ast_test_result_state
Definition test.h:193
@ AST_TEST_PASS
Definition test.h:195
@ AST_TEST_NOT_RUN
Definition test.h:194
static void handler(const char *name, int response_code, struct ast_variable *get_params, struct ast_variable *path_vars, struct ast_variable *headers, struct ast_json *body, struct ast_ari_response *response)
Definition test_ari.c:59
static struct test_options options
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition utils.h:981
#define ast_assert(a)
Definition utils.h:779
long int ast_random(void)
Definition utils.c:2346
#define SWAP(a, b)
Definition utils.h:256
#define MAX(a, b)
Definition utils.h:254
Universally unique identifier support.
#define AST_UUID_STR_LEN
Definition uuid.h:27
char * ast_uuid_generate_str(char *buf, size_t size)
Generate a UUID string.
Definition uuid.c:141
Vector container support.
#define AST_VECTOR_REPLACE(vec, idx, elem)
Replace an element at a specific position in a vector, growing the vector if needed.
Definition vector.h:295
#define AST_VECTOR_RESET(vec, cleanup)
Reset vector.
Definition vector.h:653
#define AST_VECTOR_ELEM_CLEANUP_NOOP(elem)
Vector element cleanup that does nothing.
Definition vector.h:599
#define AST_VECTOR_SIZE(vec)
Get the number of elements in a vector.
Definition vector.h:637
#define AST_VECTOR_FREE(vec)
Deallocates this vector.
Definition vector.h:185
#define AST_VECTOR_REMOVE_CMP_UNORDERED(vec, value, cmp, cleanup)
Remove an element from a vector that matches the given comparison.
Definition vector.h:516
#define AST_VECTOR_REMOVE(vec, idx, preserve_ordered)
Remove an element from a vector by index.
Definition vector.h:440
#define AST_VECTOR_INIT(vec, size)
Initialize a vector.
Definition vector.h:124
#define AST_VECTOR_APPEND(vec, elem)
Append an element to a vector, growing the vector if needed.
Definition vector.h:267
#define AST_VECTOR_GET(vec, idx)
Get an element from a vector.
Definition vector.h:708
#define AST_VECTOR_GET_ADDR(vec, idx)
Get an address of element in a vector.
Definition vector.h:696