Add channel locking for channel snapshot creation.
[asterisk/asterisk.git] / channels / chan_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Joshua Colp <jcolp@digium.com>
7  *
8  * See http://www.asterisk.org for more information about
9  * the Asterisk project. Please do not directly contact
10  * any of the maintainers of this project for assistance;
11  * the project provides a web site, mailing lists and IRC
12  * channels for your use.
13  *
14  * This program is free software, distributed under the terms of
15  * the GNU General Public License Version 2. See the LICENSE file
16  * at the top of the source tree.
17  */
18
19 /*! \file
20  *
21  * \author Joshua Colp <jcolp@digium.com>
22  *
23  * \brief PSJIP SIP Channel Driver
24  *
25  * \ingroup channel_drivers
26  */
27
28 /*** MODULEINFO
29         <depend>pjproject</depend>
30         <depend>res_pjsip</depend>
31         <depend>res_pjsip_session</depend>
32         <support_level>core</support_level>
33  ***/
34
35 #include "asterisk.h"
36
37 #include <pjsip.h>
38 #include <pjsip_ua.h>
39 #include <pjlib.h>
40
41 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
42
43 #include "asterisk/lock.h"
44 #include "asterisk/channel.h"
45 #include "asterisk/module.h"
46 #include "asterisk/pbx.h"
47 #include "asterisk/rtp_engine.h"
48 #include "asterisk/acl.h"
49 #include "asterisk/callerid.h"
50 #include "asterisk/file.h"
51 #include "asterisk/cli.h"
52 #include "asterisk/app.h"
53 #include "asterisk/musiconhold.h"
54 #include "asterisk/causes.h"
55 #include "asterisk/taskprocessor.h"
56 #include "asterisk/dsp.h"
57 #include "asterisk/stasis_endpoints.h"
58 #include "asterisk/stasis_channels.h"
59 #include "asterisk/indications.h"
60
61 #include "asterisk/res_pjsip.h"
62 #include "asterisk/res_pjsip_session.h"
63
64 /*** DOCUMENTATION
65         <function name="PJSIP_DIAL_CONTACTS" language="en_US">
66                 <synopsis>
67                         Return a dial string for dialing all contacts on an AOR.
68                 </synopsis>
69                 <syntax>
70                         <parameter name="endpoint" required="true">
71                                 <para>Name of the endpoint</para>
72                         </parameter>
73                         <parameter name="aor" required="false">
74                                 <para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
75                         </parameter>
76                         <parameter name="request_user" required="false">
77                                 <para>Optional request user to use in the request URI</para>
78                         </parameter>
79                 </syntax>
80                 <description>
81                         <para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
82                 </description>
83         </function>
84         <function name="PJSIP_MEDIA_OFFER" language="en_US">
85                 <synopsis>
86                         Media and codec offerings to be set on an outbound SIP channel prior to dialing.
87                 </synopsis>
88                 <syntax>
89                         <parameter name="media" required="true">
90                                 <para>types of media offered</para>
91                         </parameter>
92                 </syntax>
93                 <description>
94                         <para>Returns the codecs offered based upon the media choice</para>
95                 </description>
96         </function>
97  ***/
98
99 static const char desc[] = "PJSIP Channel";
100 static const char channel_type[] = "PJSIP";
101
102 static unsigned int chan_idx;
103
104 /*!
105  * \brief Positions of various media
106  */
107 enum sip_session_media_position {
108         /*! \brief First is audio */
109         SIP_MEDIA_AUDIO = 0,
110         /*! \brief Second is video */
111         SIP_MEDIA_VIDEO,
112         /*! \brief Last is the size for media details */
113         SIP_MEDIA_SIZE,
114 };
115
116 struct chan_pjsip_pvt {
117         struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
118 };
119
120 static void chan_pjsip_pvt_dtor(void *obj)
121 {
122         struct chan_pjsip_pvt *pvt = obj;
123         int i;
124
125         for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
126                 ao2_cleanup(pvt->media[i]);
127                 pvt->media[i] = NULL;
128         }
129 }
130
131 /* \brief Asterisk core interaction functions */
132 static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
133 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
134 static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
135 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
136 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
137 static int chan_pjsip_hangup(struct ast_channel *ast);
138 static int chan_pjsip_answer(struct ast_channel *ast);
139 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast);
140 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
141 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
142 static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
143 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
144 static int chan_pjsip_devicestate(const char *data);
145 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
146
147 /*! \brief PBX interface structure for channel registration */
148 static struct ast_channel_tech chan_pjsip_tech = {
149         .type = channel_type,
150         .description = "PJSIP Channel Driver",
151         .requester = chan_pjsip_request,
152         .send_text = chan_pjsip_sendtext,
153         .send_digit_begin = chan_pjsip_digit_begin,
154         .send_digit_end = chan_pjsip_digit_end,
155         .call = chan_pjsip_call,
156         .hangup = chan_pjsip_hangup,
157         .answer = chan_pjsip_answer,
158         .read = chan_pjsip_read,
159         .write = chan_pjsip_write,
160         .write_video = chan_pjsip_write,
161         .exception = chan_pjsip_read,
162         .indicate = chan_pjsip_indicate,
163         .transfer = chan_pjsip_transfer,
164         .fixup = chan_pjsip_fixup,
165         .devicestate = chan_pjsip_devicestate,
166         .queryoption = chan_pjsip_queryoption,
167         .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
168 };
169
170 /*! \brief SIP session interaction functions */
171 static void chan_pjsip_session_begin(struct ast_sip_session *session);
172 static void chan_pjsip_session_end(struct ast_sip_session *session);
173 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
174 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
175
176 /*! \brief SIP session supplement structure */
177 static struct ast_sip_session_supplement chan_pjsip_supplement = {
178         .method = "INVITE",
179         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
180         .session_begin = chan_pjsip_session_begin,
181         .session_end = chan_pjsip_session_end,
182         .incoming_request = chan_pjsip_incoming_request,
183         .incoming_response = chan_pjsip_incoming_response,
184 };
185
186 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
187
188 static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
189         .method = "ACK",
190         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
191         .incoming_request = chan_pjsip_incoming_ack,
192 };
193
194 /*! \brief Dialplan function for constructing a dial string for calling all contacts */
195 static int chan_pjsip_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
196 {
197         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
198         RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
199         const char *aor_name;
200         char *rest;
201
202         AST_DECLARE_APP_ARGS(args,
203                 AST_APP_ARG(endpoint_name);
204                 AST_APP_ARG(aor_name);
205                 AST_APP_ARG(request_user);
206         );
207
208         AST_STANDARD_APP_ARGS(args, data);
209
210         if (ast_strlen_zero(args.endpoint_name)) {
211                 ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
212                 return -1;
213         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
214                 ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
215                 return -1;
216         }
217
218         aor_name = S_OR(args.aor_name, endpoint->aors);
219
220         if (ast_strlen_zero(aor_name)) {
221                 ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
222                 return -1;
223         } else if (!(dial = ast_str_create(len))) {
224                 ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
225                 return -1;
226         } else if (!(rest = ast_strdupa(aor_name))) {
227                 ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
228                 return -1;
229         }
230
231         while ((aor_name = strsep(&rest, ","))) {
232                 RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
233                 RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
234                 struct ao2_iterator it_contacts;
235                 struct ast_sip_contact *contact;
236
237                 if (!aor) {
238                         /* If the AOR provided is not found skip it, there may be more */
239                         continue;
240                 } else if (!(contacts = ast_sip_location_retrieve_aor_contacts(aor))) {
241                         /* No contacts are available, skip it as well */
242                         continue;
243                 } else if (!ao2_container_count(contacts)) {
244                         /* We were given a container but no contacts are in it... */
245                         continue;
246                 }
247
248                 it_contacts = ao2_iterator_init(contacts, 0);
249                 for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
250                         ast_str_append(&dial, -1, "PJSIP/");
251
252                         if (!ast_strlen_zero(args.request_user)) {
253                                 ast_str_append(&dial, -1, "%s@", args.request_user);
254                         }
255                         ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
256                 }
257                 ao2_iterator_destroy(&it_contacts);
258         }
259
260         /* Trim the '&' at the end off */
261         ast_str_truncate(dial, ast_str_strlen(dial) - 1);
262
263         ast_copy_string(buf, ast_str_buffer(dial), len);
264
265         return 0;
266 }
267
268 static struct ast_custom_function chan_pjsip_dial_contacts_function = {
269         .name = "PJSIP_DIAL_CONTACTS",
270         .read = chan_pjsip_dial_contacts,
271 };
272
273 static int media_offer_read_av(struct ast_sip_session *session, char *buf,
274                                size_t len, enum ast_format_type media_type)
275 {
276         int i, size = 0;
277         struct ast_format fmt;
278         const char *name;
279
280         for (i = 0; ast_codec_pref_index(&session->override_prefs, i, &fmt); ++i) {
281                 if (AST_FORMAT_GET_TYPE(fmt.id) != media_type) {
282                         continue;
283                 }
284
285                 name = ast_getformatname(&fmt);
286
287                 if (ast_strlen_zero(name)) {
288                         ast_log(LOG_WARNING, "PJSIP_MEDIA_OFFER unrecognized format %s\n", name);
289                         continue;
290                 }
291
292                 /* add one since we'll include a comma */
293                 size = strlen(name) + 1;
294                 len -= size;
295                 if ((len) < 0) {
296                         break;
297                 }
298
299                 /* no reason to use strncat here since we have already ensured buf has
300                    enough space, so strcat can be safely used */
301                 strcat(buf, name);
302                 strcat(buf, ",");
303         }
304
305         if (size) {
306                 /* remove the extra comma */
307                 buf[strlen(buf) - 1] = '\0';
308         }
309         return 0;
310 }
311
312 struct media_offer_data {
313         struct ast_sip_session *session;
314         enum ast_format_type media_type;
315         const char *value;
316 };
317
318 static int media_offer_write_av(void *obj)
319 {
320         struct media_offer_data *data = obj;
321         int i;
322         struct ast_format fmt;
323         /* remove all of the given media type first */
324         for (i = 0; ast_codec_pref_index(&data->session->override_prefs, i, &fmt); ++i) {
325                 if (AST_FORMAT_GET_TYPE(fmt.id) == data->media_type) {
326                         ast_codec_pref_remove(&data->session->override_prefs, &fmt);
327                 }
328         }
329         ast_format_cap_remove_bytype(data->session->req_caps, data->media_type);
330         ast_parse_allow_disallow(&data->session->override_prefs, data->session->req_caps, data->value, 1);
331
332         return 0;
333 }
334
335 static int media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
336 {
337         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
338
339         if (!strcmp(data, "audio")) {
340                 return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_AUDIO);
341         } else if (!strcmp(data, "video")) {
342                 return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_VIDEO);
343         }
344
345         return 0;
346 }
347
348 static int media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
349 {
350         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
351
352         struct media_offer_data mdata = {
353                 .session = channel->session,
354                 .value = value
355         };
356
357         if (!strcmp(data, "audio")) {
358                 mdata.media_type = AST_FORMAT_TYPE_AUDIO;
359         } else if (!strcmp(data, "video")) {
360                 mdata.media_type = AST_FORMAT_TYPE_VIDEO;
361         }
362
363         return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
364 }
365
366 static struct ast_custom_function media_offer_function = {
367         .name = "PJSIP_MEDIA_OFFER",
368         .read = media_offer_read,
369         .write = media_offer_write
370 };
371
372 /*! \brief Function called by RTP engine to get local audio RTP peer */
373 static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
374 {
375         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
376         struct chan_pjsip_pvt *pvt = channel->pvt;
377         struct ast_sip_endpoint *endpoint;
378
379         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
380                 return AST_RTP_GLUE_RESULT_FORBID;
381         }
382
383         endpoint = channel->session->endpoint;
384
385         *instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
386         ao2_ref(*instance, +1);
387
388         ast_assert(endpoint != NULL);
389         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
390                 return AST_RTP_GLUE_RESULT_FORBID;
391         }
392
393         if (endpoint->media.direct_media.enabled) {
394                 return AST_RTP_GLUE_RESULT_REMOTE;
395         }
396
397         return AST_RTP_GLUE_RESULT_LOCAL;
398 }
399
400 /*! \brief Function called by RTP engine to get local video RTP peer */
401 static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
402 {
403         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
404         struct chan_pjsip_pvt *pvt = channel->pvt;
405         struct ast_sip_endpoint *endpoint;
406
407         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
408                 return AST_RTP_GLUE_RESULT_FORBID;
409         }
410
411         endpoint = channel->session->endpoint;
412
413         *instance = pvt->media[SIP_MEDIA_VIDEO]->rtp;
414         ao2_ref(*instance, +1);
415
416         ast_assert(endpoint != NULL);
417         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
418                 return AST_RTP_GLUE_RESULT_FORBID;
419         }
420
421         return AST_RTP_GLUE_RESULT_LOCAL;
422 }
423
424 /*! \brief Function called by RTP engine to get peer capabilities */
425 static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
426 {
427         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
428
429         ast_format_cap_copy(result, channel->session->endpoint->media.codecs);
430 }
431
432 static int send_direct_media_request(void *data)
433 {
434         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
435
436         return ast_sip_session_refresh(session, NULL, NULL, NULL,
437                         session->endpoint->media.direct_media.method, 1);
438 }
439
440 static struct ast_datastore_info direct_media_mitigation_info = { };
441
442 static int direct_media_mitigate_glare(struct ast_sip_session *session)
443 {
444         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
445
446         if (session->endpoint->media.direct_media.glare_mitigation ==
447                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
448                 return 0;
449         }
450
451         datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
452         if (!datastore) {
453                 return 0;
454         }
455
456         /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
457         ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
458
459         if ((session->endpoint->media.direct_media.glare_mitigation ==
460                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
461                         session->inv_session->role == PJSIP_ROLE_UAC) ||
462                         (session->endpoint->media.direct_media.glare_mitigation ==
463                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
464                         session->inv_session->role == PJSIP_ROLE_UAS)) {
465                 return 1;
466         }
467
468         return 0;
469 }
470
471 static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
472                 struct ast_sip_session_media *media, int rtcp_fd)
473 {
474         int changed = 0;
475
476         if (rtp) {
477                 changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
478                 if (media->rtp) {
479                         ast_channel_set_fd(chan, rtcp_fd, -1);
480                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
481                 }
482         } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
483                 ast_sockaddr_setnull(&media->direct_media_addr);
484                 changed = 1;
485                 if (media->rtp) {
486                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
487                         ast_channel_set_fd(chan, rtcp_fd, ast_rtp_instance_fd(media->rtp, 1));
488                 }
489         }
490
491         return changed;
492 }
493
494 /*! \brief Function called by RTP engine to change where the remote party should send media */
495 static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
496                 struct ast_rtp_instance *rtp,
497                 struct ast_rtp_instance *vrtp,
498                 struct ast_rtp_instance *tpeer,
499                 const struct ast_format_cap *cap,
500                 int nat_active)
501 {
502         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
503         struct chan_pjsip_pvt *pvt = channel->pvt;
504         struct ast_sip_session *session = channel->session;
505         int changed = 0;
506         struct ast_channel *bridge_peer;
507
508         /* Don't try to do any direct media shenanigans on early bridges */
509         bridge_peer = ast_channel_bridge_peer(chan);
510         if ((rtp || vrtp || tpeer) && !bridge_peer) {
511                 return 0;
512         }
513         ast_channel_cleanup(bridge_peer);
514
515         if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
516                 return 0;
517         }
518
519         if (pvt->media[SIP_MEDIA_AUDIO]) {
520                 changed |= check_for_rtp_changes(chan, rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
521         }
522         if (pvt->media[SIP_MEDIA_VIDEO]) {
523                 changed |= check_for_rtp_changes(chan, vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
524         }
525
526         if (direct_media_mitigate_glare(session)) {
527                 return 0;
528         }
529
530         if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(session->direct_media_cap, cap)) {
531                 ast_format_cap_copy(session->direct_media_cap, cap);
532                 changed = 1;
533         }
534
535         if (changed) {
536                 ao2_ref(session, +1);
537
538
539                 if (ast_sip_push_task(session->serializer, send_direct_media_request, session)) {
540                         ao2_cleanup(session);
541                 }
542         }
543
544         return 0;
545 }
546
547 /*! \brief Local glue for interacting with the RTP engine core */
548 static struct ast_rtp_glue chan_pjsip_rtp_glue = {
549         .type = "PJSIP",
550         .get_rtp_info = chan_pjsip_get_rtp_peer,
551         .get_vrtp_info = chan_pjsip_get_vrtp_peer,
552         .get_codec = chan_pjsip_get_codec,
553         .update_peer = chan_pjsip_set_rtp_peer,
554 };
555
556 /*! \brief Function called to create a new PJSIP Asterisk channel */
557 static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const char *linkedid, const char *cid_name)
558 {
559         struct ast_channel *chan;
560         struct ast_format fmt;
561         RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
562         struct ast_sip_channel_pvt *channel;
563
564         if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
565                 return NULL;
566         }
567
568         if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "PJSIP/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
569                 ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
570                 return NULL;
571         }
572
573         ast_channel_tech_set(chan, &chan_pjsip_tech);
574
575         if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
576                 ast_hangup(chan);
577                 return NULL;
578         }
579
580         /* If res_pjsip_session is ever updated to create/destroy ast_sip_session_media
581          * during a call such as if multiple same-type stream support is introduced,
582          * these will need to be recaptured as well */
583         pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
584         pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
585
586         ast_channel_lock(chan);
587         ast_channel_stage_snapshot(chan);
588
589         ast_channel_tech_pvt_set(chan, channel);
590         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
591                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
592         }
593         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
594                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
595         }
596
597         if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->media.codecs)) {
598                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->media.codecs);
599         } else {
600                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->req_caps);
601         }
602
603         ast_codec_choose(&session->endpoint->media.prefs, ast_channel_nativeformats(chan), 1, &fmt);
604         ast_format_copy(ast_channel_writeformat(chan), &fmt);
605         ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
606         ast_format_copy(ast_channel_readformat(chan), &fmt);
607         ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
608
609         if (state == AST_STATE_RING) {
610                 ast_channel_rings_set(chan, 1);
611         }
612
613         ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
614
615         ast_channel_context_set(chan, session->endpoint->context);
616         ast_channel_exten_set(chan, S_OR(exten, "s"));
617         ast_channel_priority_set(chan, 1);
618
619         ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
620         ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);
621
622         ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
623         ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);
624
625         if (!ast_strlen_zero(session->endpoint->language)) {
626                 ast_channel_language_set(chan, session->endpoint->language);
627         }
628
629         if (!ast_strlen_zero(session->endpoint->zone)) {
630                 struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
631                 if (!zone) {
632                         ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
633                 }
634                 ast_channel_zone_set(chan, zone);
635         }
636
637         ast_channel_stage_snapshot_done(chan);
638         ast_channel_unlock(chan);
639
640         ast_endpoint_add_channel(session->endpoint->persistent, chan);
641
642         return chan;
643 }
644
645 static int answer(void *data)
646 {
647         pj_status_t status = PJ_SUCCESS;
648         pjsip_tx_data *packet;
649         struct ast_sip_session *session = data;
650
651         pjsip_dlg_inc_lock(session->inv_session->dlg);
652         if (session->inv_session->invite_tsx) {
653                 status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
654         }
655         pjsip_dlg_dec_lock(session->inv_session->dlg);
656
657         if (status == PJ_SUCCESS && packet) {
658                 ast_sip_session_send_response(session, packet);
659         }
660
661         ao2_ref(session, -1);
662
663         return (status == PJ_SUCCESS) ? 0 : -1;
664 }
665
666 /*! \brief Function called by core when we should answer a PJSIP session */
667 static int chan_pjsip_answer(struct ast_channel *ast)
668 {
669         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
670
671         if (ast_channel_state(ast) == AST_STATE_UP) {
672                 return 0;
673         }
674
675         ast_setstate(ast, AST_STATE_UP);
676
677         ao2_ref(channel->session, +1);
678         if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
679                 ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
680                 ao2_cleanup(channel->session);
681                 return -1;
682         }
683
684         return 0;
685 }
686
687 /*! \brief Internal helper function called when CNG tone is detected */
688 static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
689 {
690         const char *target_context;
691         int exists;
692
693         /* If we only needed this DSP for fax detection purposes we can just drop it now */
694         if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
695                 ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
696         } else {
697                 ast_dsp_free(session->dsp);
698                 session->dsp = NULL;
699         }
700
701         /* If already executing in the fax extension don't do anything */
702         if (!strcmp(ast_channel_exten(session->channel), "fax")) {
703                 return f;
704         }
705
706         target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
707
708         /* We need to unlock the channel here because ast_exists_extension has the
709          * potential to start and stop an autoservice on the channel. Such action
710          * is prone to deadlock if the channel is locked.
711          */
712         ast_channel_unlock(session->channel);
713         exists = ast_exists_extension(session->channel, target_context, "fax", 1,
714                 S_COR(ast_channel_caller(session->channel)->id.number.valid,
715                         ast_channel_caller(session->channel)->id.number.str, NULL));
716         ast_channel_lock(session->channel);
717
718         if (exists) {
719                 ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
720                         ast_channel_name(session->channel));
721                 pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
722                 if (ast_async_goto(session->channel, target_context, "fax", 1)) {
723                         ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
724                                 ast_channel_name(session->channel), target_context);
725                 }
726                 ast_frfree(f);
727                 f = &ast_null_frame;
728         } else {
729                 ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
730                         ast_channel_name(session->channel), target_context);
731         }
732
733         return f;
734 }
735
736 /*! \brief Function called by core to read any waiting frames */
737 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
738 {
739         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
740         struct chan_pjsip_pvt *pvt = channel->pvt;
741         struct ast_frame *f;
742         struct ast_sip_session_media *media = NULL;
743         int rtcp = 0;
744         int fdno = ast_channel_fdno(ast);
745
746         switch (fdno) {
747         case 0:
748                 media = pvt->media[SIP_MEDIA_AUDIO];
749                 break;
750         case 1:
751                 media = pvt->media[SIP_MEDIA_AUDIO];
752                 rtcp = 1;
753                 break;
754         case 2:
755                 media = pvt->media[SIP_MEDIA_VIDEO];
756                 break;
757         case 3:
758                 media = pvt->media[SIP_MEDIA_VIDEO];
759                 rtcp = 1;
760                 break;
761         }
762
763         if (!media || !media->rtp) {
764                 return &ast_null_frame;
765         }
766
767         if (!(f = ast_rtp_instance_read(media->rtp, rtcp))) {
768                 return f;
769         }
770
771         if (f->frametype != AST_FRAME_VOICE) {
772                 return f;
773         }
774
775         if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
776                 ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
777                 ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
778                 ast_set_read_format(ast, ast_channel_readformat(ast));
779                 ast_set_write_format(ast, ast_channel_writeformat(ast));
780         }
781
782         if (channel->session->dsp) {
783                 f = ast_dsp_process(ast, channel->session->dsp, f);
784
785                 if (f && (f->frametype == AST_FRAME_DTMF)) {
786                         if (f->subclass.integer == 'f') {
787                                 ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
788                                 f = chan_pjsip_cng_tone_detected(channel->session, f);
789                         } else {
790                                 ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
791                                         ast_channel_name(ast));
792                         }
793                 }
794         }
795
796         return f;
797 }
798
799 /*! \brief Function called by core to write frames */
800 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
801 {
802         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
803         struct chan_pjsip_pvt *pvt = channel->pvt;
804         struct ast_sip_session_media *media;
805         int res = 0;
806
807         switch (frame->frametype) {
808         case AST_FRAME_VOICE:
809                 media = pvt->media[SIP_MEDIA_AUDIO];
810
811                 if (!media) {
812                         return 0;
813                 }
814                 if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
815                         char buf[256];
816
817                         ast_log(LOG_WARNING,
818                                 "Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
819                                 ast_getformatname(&frame->subclass.format),
820                                 ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
821                                 ast_getformatname(ast_channel_readformat(ast)),
822                                 ast_getformatname(ast_channel_writeformat(ast)));
823                         return 0;
824                 }
825                 if (media->rtp) {
826                         res = ast_rtp_instance_write(media->rtp, frame);
827                 }
828                 break;
829         case AST_FRAME_VIDEO:
830                 if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
831                         res = ast_rtp_instance_write(media->rtp, frame);
832                 }
833                 break;
834         case AST_FRAME_MODEM:
835                 break;
836         default:
837                 ast_log(LOG_WARNING, "Can't send %d type frames with PJSIP\n", frame->frametype);
838                 break;
839         }
840
841         return res;
842 }
843
844 struct fixup_data {
845         struct ast_sip_session *session;
846         struct ast_channel *chan;
847 };
848
849 static int fixup(void *data)
850 {
851         struct fixup_data *fix_data = data;
852         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
853         struct chan_pjsip_pvt *pvt = channel->pvt;
854
855         channel->session->channel = fix_data->chan;
856         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
857                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
858         }
859         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
860                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(fix_data->chan));
861         }
862
863         return 0;
864 }
865
866 /*! \brief Function called by core to change the underlying owner channel */
867 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
868 {
869         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
870         struct fixup_data fix_data;
871
872         fix_data.session = channel->session;
873         fix_data.chan = newchan;
874
875         if (channel->session->channel != oldchan) {
876                 return -1;
877         }
878
879         if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
880                 ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
881                 return -1;
882         }
883
884         return 0;
885 }
886
887 /*! \brief Function called to get the device state of an endpoint */
888 static int chan_pjsip_devicestate(const char *data)
889 {
890         RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
891         enum ast_device_state state = AST_DEVICE_UNKNOWN;
892         RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
893         RAII_VAR(struct stasis_cache *, cache, NULL, ao2_cleanup);
894         struct ast_devstate_aggregate aggregate;
895         int num, inuse = 0;
896
897         if (!endpoint) {
898                 return AST_DEVICE_INVALID;
899         }
900
901         endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
902                 ast_endpoint_get_resource(endpoint->persistent));
903
904         if (!endpoint_snapshot) {
905                 return AST_DEVICE_INVALID;
906         }
907
908         if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
909                 state = AST_DEVICE_UNAVAILABLE;
910         } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
911                 state = AST_DEVICE_NOT_INUSE;
912         }
913
914         if (!endpoint_snapshot->num_channels || !(cache = ast_channel_cache())) {
915                 return state;
916         }
917
918         ast_devstate_aggregate_init(&aggregate);
919
920         ao2_ref(cache, +1);
921
922         for (num = 0; num < endpoint_snapshot->num_channels; num++) {
923                 RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
924                 struct ast_channel_snapshot *snapshot;
925
926                 msg = stasis_cache_get(cache, ast_channel_snapshot_type(),
927                         endpoint_snapshot->channel_ids[num]);
928
929                 if (!msg) {
930                         continue;
931                 }
932
933                 snapshot = stasis_message_data(msg);
934
935                 if (snapshot->state == AST_STATE_DOWN) {
936                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_NOT_INUSE);
937                 } else if (snapshot->state == AST_STATE_RINGING) {
938                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_RINGING);
939                 } else if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
940                         (snapshot->state == AST_STATE_BUSY)) {
941                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_INUSE);
942                         inuse++;
943                 }
944         }
945
946         if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
947                 state = AST_DEVICE_BUSY;
948         } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
949                 state = ast_devstate_aggregate_result(&aggregate);
950         }
951
952         return state;
953 }
954
955 /*! \brief Function called to query options on a channel */
956 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
957 {
958         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
959         struct ast_sip_session *session = channel->session;
960         int res = -1;
961         enum ast_sip_session_t38state state = T38_STATE_UNAVAILABLE;
962
963         switch (option) {
964         case AST_OPTION_T38_STATE:
965                 if (session->endpoint->media.t38.enabled) {
966                         switch (session->t38state) {
967                         case T38_LOCAL_REINVITE:
968                         case T38_PEER_REINVITE:
969                                 state = T38_STATE_NEGOTIATING;
970                                 break;
971                         case T38_ENABLED:
972                                 state = T38_STATE_NEGOTIATED;
973                                 break;
974                         case T38_REJECTED:
975                                 state = T38_STATE_REJECTED;
976                                 break;
977                         default:
978                                 state = T38_STATE_UNKNOWN;
979                                 break;
980                         }
981                 }
982
983                 *((enum ast_t38_state *) data) = state;
984                 res = 0;
985
986                 break;
987         default:
988                 break;
989         }
990
991         return res;
992 }
993
994 struct indicate_data {
995         struct ast_sip_session *session;
996         int condition;
997         int response_code;
998         void *frame_data;
999         size_t datalen;
1000 };
1001
1002 static void indicate_data_destroy(void *obj)
1003 {
1004         struct indicate_data *ind_data = obj;
1005
1006         ast_free(ind_data->frame_data);
1007         ao2_ref(ind_data->session, -1);
1008 }
1009
1010 static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session,
1011                 int condition, int response_code, const void *frame_data, size_t datalen)
1012 {
1013         struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
1014
1015         if (!ind_data) {
1016                 return NULL;
1017         }
1018
1019         ind_data->frame_data = ast_malloc(datalen);
1020         if (!ind_data->frame_data) {
1021                 ao2_ref(ind_data, -1);
1022                 return NULL;
1023         }
1024
1025         memcpy(ind_data->frame_data, frame_data, datalen);
1026         ind_data->datalen = datalen;
1027         ind_data->condition = condition;
1028         ind_data->response_code = response_code;
1029         ao2_ref(session, +1);
1030         ind_data->session = session;
1031
1032         return ind_data;
1033 }
1034
1035 static int indicate(void *data)
1036 {
1037         pjsip_tx_data *packet = NULL;
1038         struct indicate_data *ind_data = data;
1039         struct ast_sip_session *session = ind_data->session;
1040         int response_code = ind_data->response_code;
1041
1042         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1043                 ast_sip_session_send_response(session, packet);
1044         }
1045
1046         ao2_ref(ind_data, -1);
1047
1048         return 0;
1049 }
1050
1051 /*! \brief Send SIP INFO with video update request */
1052 static int transmit_info_with_vidupdate(void *data)
1053 {
1054         const char * xml =
1055                 "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
1056                 " <media_control>\r\n"
1057                 "  <vc_primitive>\r\n"
1058                 "   <to_encoder>\r\n"
1059                 "    <picture_fast_update/>\r\n"
1060                 "   </to_encoder>\r\n"
1061                 "  </vc_primitive>\r\n"
1062                 " </media_control>\r\n";
1063
1064         const struct ast_sip_body body = {
1065                 .type = "application",
1066                 .subtype = "media_control+xml",
1067                 .body_text = xml
1068         };
1069
1070         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1071         struct pjsip_tx_data *tdata;
1072
1073         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1074                 ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
1075                 return -1;
1076         }
1077         if (ast_sip_add_body(tdata, &body)) {
1078                 ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
1079                 return -1;
1080         }
1081         ast_sip_session_send_request(session, tdata);
1082
1083         return 0;
1084 }
1085
1086 /*! \brief Update connected line information */
1087 static int update_connected_line_information(void *data)
1088 {
1089         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1090         struct ast_party_id connected_id;
1091
1092         if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
1093                 int response_code = 0;
1094
1095                 if (ast_channel_state(session->channel) == AST_STATE_RING) {
1096                         response_code = !session->endpoint->inband_progress ? 180 : 183;
1097                 } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
1098                         response_code = 183;
1099                 }
1100
1101                 if (response_code) {
1102                         struct pjsip_tx_data *packet = NULL;
1103
1104                         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1105                                 ast_sip_session_send_response(session, packet);
1106                         }
1107                 }
1108         } else {
1109                 enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
1110
1111                 if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
1112                         method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
1113                 }
1114
1115                 connected_id = ast_channel_connected_effective_id(session->channel);
1116                 if ((session->endpoint->id.send_pai || session->endpoint->id.send_rpid) &&
1117                     (session->endpoint->id.trust_outbound ||
1118                      ((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
1119                       (connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
1120                         ast_sip_session_refresh(session, NULL, NULL, NULL, method, 1);
1121                 }
1122         }
1123
1124         return 0;
1125 }
1126
1127 /*! \brief Function called by core to ask the channel to indicate some sort of condition */
1128 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
1129 {
1130         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1131         struct chan_pjsip_pvt *pvt = channel->pvt;
1132         struct ast_sip_session_media *media;
1133         int response_code = 0;
1134         int res = 0;
1135
1136         switch (condition) {
1137         case AST_CONTROL_RINGING:
1138                 if (ast_channel_state(ast) == AST_STATE_RING) {
1139                         if (channel->session->endpoint->inband_progress) {
1140                                 response_code = 183;
1141                                 res = -1;
1142                         } else {
1143                                 response_code = 180;
1144                         }
1145                 } else {
1146                         res = -1;
1147                 }
1148                 ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
1149                 break;
1150         case AST_CONTROL_BUSY:
1151                 if (ast_channel_state(ast) != AST_STATE_UP) {
1152                         response_code = 486;
1153                 } else {
1154                         res = -1;
1155                 }
1156                 break;
1157         case AST_CONTROL_CONGESTION:
1158                 if (ast_channel_state(ast) != AST_STATE_UP) {
1159                         response_code = 503;
1160                 } else {
1161                         res = -1;
1162                 }
1163                 break;
1164         case AST_CONTROL_INCOMPLETE:
1165                 if (ast_channel_state(ast) != AST_STATE_UP) {
1166                         response_code = 484;
1167                 } else {
1168                         res = -1;
1169                 }
1170                 break;
1171         case AST_CONTROL_PROCEEDING:
1172                 if (ast_channel_state(ast) != AST_STATE_UP) {
1173                         response_code = 100;
1174                 } else {
1175                         res = -1;
1176                 }
1177                 break;
1178         case AST_CONTROL_PROGRESS:
1179                 if (ast_channel_state(ast) != AST_STATE_UP) {
1180                         response_code = 183;
1181                 } else {
1182                         res = -1;
1183                 }
1184                 break;
1185         case AST_CONTROL_VIDUPDATE:
1186                 media = pvt->media[SIP_MEDIA_VIDEO];
1187                 if (media && media->rtp) {
1188                         /* FIXME: Only use this for VP8. Additional work would have to be done to
1189                          * fully support other video codecs */
1190                         struct ast_format_cap *fcap = ast_channel_nativeformats(ast);
1191                         struct ast_format vp8;
1192                         ast_format_set(&vp8, AST_FORMAT_VP8, 0);
1193                         if (ast_format_cap_iscompatible(fcap, &vp8)) {
1194                                 /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
1195                                  * RTP engine would provide a way to externally write/schedule RTCP
1196                                  * packets */
1197                                 struct ast_frame fr;
1198                                 fr.frametype = AST_FRAME_CONTROL;
1199                                 fr.subclass.integer = AST_CONTROL_VIDUPDATE;
1200                                 res = ast_rtp_instance_write(media->rtp, &fr);
1201                         } else {
1202                                 ao2_ref(channel->session, +1);
1203
1204                                 if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
1205                                         ao2_cleanup(channel->session);
1206                                 }
1207                         }
1208                 } else {
1209                         res = -1;
1210                 }
1211                 break;
1212         case AST_CONTROL_CONNECTED_LINE:
1213                 ao2_ref(channel->session, +1);
1214                 if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
1215                         ao2_cleanup(channel->session);
1216                 }
1217                 break;
1218         case AST_CONTROL_UPDATE_RTP_PEER:
1219                 break;
1220         case AST_CONTROL_PVT_CAUSE_CODE:
1221                 res = -1;
1222                 break;
1223         case AST_CONTROL_HOLD:
1224                 ast_moh_start(ast, data, NULL);
1225                 break;
1226         case AST_CONTROL_UNHOLD:
1227                 ast_moh_stop(ast);
1228                 break;
1229         case AST_CONTROL_SRCUPDATE:
1230                 break;
1231         case AST_CONTROL_SRCCHANGE:
1232                 break;
1233         case AST_CONTROL_REDIRECTING:
1234                 if (ast_channel_state(ast) != AST_STATE_UP) {
1235                         response_code = 181;
1236                 } else {
1237                         res = -1;
1238                 }
1239                 break;
1240         case AST_CONTROL_T38_PARAMETERS:
1241                 res = 0;
1242
1243                 if (channel->session->t38state == T38_PEER_REINVITE) {
1244                         const struct ast_control_t38_parameters *parameters = data;
1245
1246                         if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1247                                 res = AST_T38_REQUEST_PARMS;
1248                         }
1249                 }
1250
1251                 break;
1252         case -1:
1253                 res = -1;
1254                 break;
1255         default:
1256                 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1257                 res = -1;
1258                 break;
1259         }
1260
1261         if (response_code) {
1262                 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1263                 if (!ind_data || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1264                         ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1265                                         response_code, ast_sorcery_object_get_id(channel->session->endpoint));
1266                         ao2_cleanup(ind_data);
1267                         res = -1;
1268                 }
1269         }
1270
1271         return res;
1272 }
1273
1274 struct transfer_data {
1275         struct ast_sip_session *session;
1276         char *target;
1277 };
1278
1279 static void transfer_data_destroy(void *obj)
1280 {
1281         struct transfer_data *trnf_data = obj;
1282
1283         ast_free(trnf_data->target);
1284         ao2_cleanup(trnf_data->session);
1285 }
1286
1287 static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target)
1288 {
1289         struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
1290
1291         if (!trnf_data) {
1292                 return NULL;
1293         }
1294
1295         if (!(trnf_data->target = ast_strdup(target))) {
1296                 ao2_ref(trnf_data, -1);
1297                 return NULL;
1298         }
1299
1300         ao2_ref(session, +1);
1301         trnf_data->session = session;
1302
1303         return trnf_data;
1304 }
1305
1306 static void transfer_redirect(struct ast_sip_session *session, const char *target)
1307 {
1308         pjsip_tx_data *packet;
1309         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1310         pjsip_contact_hdr *contact;
1311         pj_str_t tmp;
1312
1313         if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS) {
1314                 message = AST_TRANSFER_FAILED;
1315                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1316
1317                 return;
1318         }
1319
1320         if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
1321                 contact = pjsip_contact_hdr_create(packet->pool);
1322         }
1323
1324         pj_strdup2_with_null(packet->pool, &tmp, target);
1325         if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
1326                 message = AST_TRANSFER_FAILED;
1327                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1328                 pjsip_tx_data_dec_ref(packet);
1329
1330                 return;
1331         }
1332         pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
1333
1334         ast_sip_session_send_response(session, packet);
1335         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1336 }
1337
1338 static void transfer_refer(struct ast_sip_session *session, const char *target)
1339 {
1340         pjsip_evsub *sub;
1341         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1342         pj_str_t tmp;
1343         pjsip_tx_data *packet;
1344
1345         if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {
1346                 message = AST_TRANSFER_FAILED;
1347                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1348
1349                 return;
1350         }
1351
1352         if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
1353                 message = AST_TRANSFER_FAILED;
1354                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1355                 pjsip_evsub_terminate(sub, PJ_FALSE);
1356
1357                 return;
1358         }
1359
1360         pjsip_xfer_send_request(sub, packet);
1361         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1362 }
1363
1364 static int transfer(void *data)
1365 {
1366         struct transfer_data *trnf_data = data;
1367
1368         if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
1369                 transfer_redirect(trnf_data->session, trnf_data->target);
1370         } else {
1371                 transfer_refer(trnf_data->session, trnf_data->target);
1372         }
1373
1374         ao2_ref(trnf_data, -1);
1375         return 0;
1376 }
1377
1378 /*! \brief Function called by core for Asterisk initiated transfer */
1379 static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
1380 {
1381         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1382         struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
1383
1384         if (!trnf_data) {
1385                 return -1;
1386         }
1387
1388         if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
1389                 ast_log(LOG_WARNING, "Error requesting transfer\n");
1390                 ao2_cleanup(trnf_data);
1391                 return -1;
1392         }
1393
1394         return 0;
1395 }
1396
1397 /*! \brief Function called by core to start a DTMF digit */
1398 static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
1399 {
1400         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1401         struct chan_pjsip_pvt *pvt = channel->pvt;
1402         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1403         int res = 0;
1404
1405         switch (channel->session->endpoint->dtmf) {
1406         case AST_SIP_DTMF_RFC_4733:
1407                 if (!media || !media->rtp) {
1408                         return -1;
1409                 }
1410
1411                 ast_rtp_instance_dtmf_begin(media->rtp, digit);
1412         case AST_SIP_DTMF_NONE:
1413                 break;
1414         case AST_SIP_DTMF_INBAND:
1415                 res = -1;
1416                 break;
1417         default:
1418                 break;
1419         }
1420
1421         return res;
1422 }
1423
1424 struct info_dtmf_data {
1425         struct ast_sip_session *session;
1426         char digit;
1427         unsigned int duration;
1428 };
1429
1430 static void info_dtmf_data_destroy(void *obj)
1431 {
1432         struct info_dtmf_data *dtmf_data = obj;
1433         ao2_ref(dtmf_data->session, -1);
1434 }
1435
1436 static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
1437 {
1438         struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
1439         if (!dtmf_data) {
1440                 return NULL;
1441         }
1442         ao2_ref(session, +1);
1443         dtmf_data->session = session;
1444         dtmf_data->digit = digit;
1445         dtmf_data->duration = duration;
1446         return dtmf_data;
1447 }
1448
1449 static int transmit_info_dtmf(void *data)
1450 {
1451         RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
1452
1453         struct ast_sip_session *session = dtmf_data->session;
1454         struct pjsip_tx_data *tdata;
1455
1456         RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
1457
1458         struct ast_sip_body body = {
1459                 .type = "application",
1460                 .subtype = "dtmf-relay",
1461         };
1462
1463         if (!(body_text = ast_str_create(32))) {
1464                 ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
1465                 return -1;
1466         }
1467         ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
1468
1469         body.body_text = ast_str_buffer(body_text);
1470
1471         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1472                 ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
1473                 return -1;
1474         }
1475         if (ast_sip_add_body(tdata, &body)) {
1476                 ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
1477                 pjsip_tx_data_dec_ref(tdata);
1478                 return -1;
1479         }
1480         ast_sip_session_send_request(session, tdata);
1481
1482         return 0;
1483 }
1484
1485 /*! \brief Function called by core to stop a DTMF digit */
1486 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
1487 {
1488         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1489         struct chan_pjsip_pvt *pvt = channel->pvt;
1490         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1491         int res = 0;
1492
1493         switch (channel->session->endpoint->dtmf) {
1494         case AST_SIP_DTMF_INFO:
1495         {
1496                 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
1497
1498                 if (!dtmf_data) {
1499                         return -1;
1500                 }
1501
1502                 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
1503                         ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
1504                         ao2_cleanup(dtmf_data);
1505                         return -1;
1506                 }
1507                 break;
1508         }
1509         case AST_SIP_DTMF_RFC_4733:
1510                 if (!media || !media->rtp) {
1511                         return -1;
1512                 }
1513
1514                 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
1515         case AST_SIP_DTMF_NONE:
1516                 break;
1517         case AST_SIP_DTMF_INBAND:
1518                 res = -1;
1519                 break;
1520         }
1521
1522         return res;
1523 }
1524
1525 static int call(void *data)
1526 {
1527         struct ast_sip_session *session = data;
1528         pjsip_tx_data *tdata;
1529
1530         int res = ast_sip_session_create_invite(session, &tdata);
1531
1532         if (res) {
1533                 ast_queue_hangup(session->channel);
1534         } else {
1535                 ast_sip_session_send_request(session, tdata);
1536         }
1537         ao2_ref(session, -1);
1538         return res;
1539 }
1540
1541 /*! \brief Function called by core to actually start calling a remote party */
1542 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
1543 {
1544         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1545
1546         ao2_ref(channel->session, +1);
1547         if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
1548                 ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
1549                 ao2_cleanup(channel->session);
1550                 return -1;
1551         }
1552
1553         return 0;
1554 }
1555
1556 /*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
1557 static int hangup_cause2sip(int cause)
1558 {
1559         switch (cause) {
1560         case AST_CAUSE_UNALLOCATED:             /* 1 */
1561         case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
1562         case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
1563                 return 404;
1564         case AST_CAUSE_CONGESTION:              /* 34 */
1565         case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
1566                 return 503;
1567         case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
1568                 return 408;
1569         case AST_CAUSE_NO_ANSWER:               /* 19 */
1570         case AST_CAUSE_UNREGISTERED:        /* 20 */
1571                 return 480;
1572         case AST_CAUSE_CALL_REJECTED:           /* 21 */
1573                 return 403;
1574         case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
1575                 return 410;
1576         case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
1577                 return 480;
1578         case AST_CAUSE_INVALID_NUMBER_FORMAT:
1579                 return 484;
1580         case AST_CAUSE_USER_BUSY:
1581                 return 486;
1582         case AST_CAUSE_FAILURE:
1583                 return 500;
1584         case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
1585                 return 501;
1586         case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
1587                 return 503;
1588         case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
1589                 return 502;
1590         case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
1591                 return 488;
1592         case AST_CAUSE_INTERWORKING:    /* Unspecified Interworking issues */
1593                 return 500;
1594         case AST_CAUSE_NOTDEFINED:
1595         default:
1596                 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
1597                 return 0;
1598         }
1599
1600         /* Never reached */
1601         return 0;
1602 }
1603
1604 struct hangup_data {
1605         int cause;
1606         struct ast_channel *chan;
1607 };
1608
1609 static void hangup_data_destroy(void *obj)
1610 {
1611         struct hangup_data *h_data = obj;
1612
1613         h_data->chan = ast_channel_unref(h_data->chan);
1614 }
1615
1616 static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
1617 {
1618         struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
1619
1620         if (!h_data) {
1621                 return NULL;
1622         }
1623
1624         h_data->cause = cause;
1625         h_data->chan = ast_channel_ref(chan);
1626
1627         return h_data;
1628 }
1629
1630 /*! \brief Clear a channel from a session along with its PVT */
1631 static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct chan_pjsip_pvt *pvt)
1632 {
1633         session->channel = NULL;
1634         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
1635                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, "");
1636         }
1637         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
1638                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, "");
1639         }
1640         ast_channel_tech_pvt_set(ast, NULL);
1641 }
1642
1643 static int hangup(void *data)
1644 {
1645         pj_status_t status;
1646         pjsip_tx_data *packet = NULL;
1647         struct hangup_data *h_data = data;
1648         struct ast_channel *ast = h_data->chan;
1649         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1650         struct chan_pjsip_pvt *pvt = channel->pvt;
1651         struct ast_sip_session *session = channel->session;
1652         int cause = h_data->cause;
1653
1654         if (!session->defer_terminate &&
1655                 ((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS) && packet) {
1656                 if (packet->msg->type == PJSIP_RESPONSE_MSG) {
1657                         ast_sip_session_send_response(session, packet);
1658                 } else {
1659                         ast_sip_session_send_request(session, packet);
1660                 }
1661         }
1662
1663         clear_session_and_channel(session, ast, pvt);
1664         ao2_cleanup(channel);
1665         ao2_cleanup(h_data);
1666
1667         return 0;
1668 }
1669
1670 /*! \brief Function called by core to hang up a PJSIP session */
1671 static int chan_pjsip_hangup(struct ast_channel *ast)
1672 {
1673         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1674         struct chan_pjsip_pvt *pvt = channel->pvt;
1675         int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
1676         struct hangup_data *h_data = hangup_data_alloc(cause, ast);
1677
1678         if (!h_data) {
1679                 goto failure;
1680         }
1681
1682         if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
1683                 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
1684                 goto failure;
1685         }
1686
1687         return 0;
1688
1689 failure:
1690         /* Go ahead and do our cleanup of the session and channel even if we're not going
1691          * to be able to send our SIP request/response
1692          */
1693         clear_session_and_channel(channel->session, ast, pvt);
1694         ao2_cleanup(channel);
1695         ao2_cleanup(h_data);
1696
1697         return -1;
1698 }
1699
1700 struct request_data {
1701         struct ast_sip_session *session;
1702         struct ast_format_cap *caps;
1703         const char *dest;
1704         int cause;
1705 };
1706
1707 static int request(void *obj)
1708 {
1709         struct request_data *req_data = obj;
1710         struct ast_sip_session *session = NULL;
1711         char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
1712         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
1713
1714         AST_DECLARE_APP_ARGS(args,
1715                 AST_APP_ARG(endpoint);
1716                 AST_APP_ARG(aor);
1717         );
1718
1719         if (ast_strlen_zero(tmp)) {
1720                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
1721                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1722                 return -1;
1723         }
1724
1725         AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
1726
1727         /* If a request user has been specified extract it from the endpoint name portion */
1728         if ((endpoint_name = strchr(args.endpoint, '@'))) {
1729                 request_user = args.endpoint;
1730                 *endpoint_name++ = '\0';
1731         } else {
1732                 endpoint_name = args.endpoint;
1733         }
1734
1735         if (ast_strlen_zero(endpoint_name)) {
1736                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
1737                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1738         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
1739                 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
1740                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1741                 return -1;
1742         }
1743
1744         if (!(session = ast_sip_session_create_outgoing(endpoint, args.aor, request_user, req_data->caps))) {
1745                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1746                 return -1;
1747         }
1748
1749         req_data->session = session;
1750
1751         return 0;
1752 }
1753
1754 /*! \brief Function called by core to create a new outgoing PJSIP session */
1755 static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
1756 {
1757         struct request_data req_data;
1758         RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
1759
1760         req_data.caps = cap;
1761         req_data.dest = data;
1762
1763         if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
1764                 *cause = req_data.cause;
1765                 return NULL;
1766         }
1767
1768         session = req_data.session;
1769
1770         if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
1771                 /* Session needs to be terminated prematurely */
1772                 return NULL;
1773         }
1774
1775         return session->channel;
1776 }
1777
1778 struct sendtext_data {
1779         struct ast_sip_session *session;
1780         char text[0];
1781 };
1782
1783 static void sendtext_data_destroy(void *obj)
1784 {
1785         struct sendtext_data *data = obj;
1786         ao2_ref(data->session, -1);
1787 }
1788
1789 static struct sendtext_data* sendtext_data_create(struct ast_sip_session *session, const char *text)
1790 {
1791         int size = strlen(text) + 1;
1792         struct sendtext_data *data = ao2_alloc(sizeof(*data)+size, sendtext_data_destroy);
1793
1794         if (!data) {
1795                 return NULL;
1796         }
1797
1798         data->session = session;
1799         ao2_ref(data->session, +1);
1800         ast_copy_string(data->text, text, size);
1801         return data;
1802 }
1803
1804 static int sendtext(void *obj)
1805 {
1806         RAII_VAR(struct sendtext_data *, data, obj, ao2_cleanup);
1807         pjsip_tx_data *tdata;
1808
1809         const struct ast_sip_body body = {
1810                 .type = "text",
1811                 .subtype = "plain",
1812                 .body_text = data->text
1813         };
1814
1815         /* NOT ast_strlen_zero, because a zero-length message is specifically
1816          * allowed by RFC 3428 (See section 10, Examples) */
1817         if (!data->text) {
1818                 return 0;
1819         }
1820
1821         ast_debug(3, "Sending in dialog SIP message\n");
1822
1823         ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, &tdata);
1824         ast_sip_add_body(tdata, &body);
1825         ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint);
1826
1827         return 0;
1828 }
1829
1830 /*! \brief Function called by core to send text on PJSIP session */
1831 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
1832 {
1833         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1834         struct sendtext_data *data = sendtext_data_create(channel->session, text);
1835
1836         if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
1837                 ao2_ref(data, -1);
1838                 return -1;
1839         }
1840         return 0;
1841 }
1842
1843 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
1844 static int hangup_sip2cause(int cause)
1845 {
1846         /* Possible values taken from causes.h */
1847
1848         switch(cause) {
1849         case 401:       /* Unauthorized */
1850                 return AST_CAUSE_CALL_REJECTED;
1851         case 403:       /* Not found */
1852                 return AST_CAUSE_CALL_REJECTED;
1853         case 404:       /* Not found */
1854                 return AST_CAUSE_UNALLOCATED;
1855         case 405:       /* Method not allowed */
1856                 return AST_CAUSE_INTERWORKING;
1857         case 407:       /* Proxy authentication required */
1858                 return AST_CAUSE_CALL_REJECTED;
1859         case 408:       /* No reaction */
1860                 return AST_CAUSE_NO_USER_RESPONSE;
1861         case 409:       /* Conflict */
1862                 return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
1863         case 410:       /* Gone */
1864                 return AST_CAUSE_NUMBER_CHANGED;
1865         case 411:       /* Length required */
1866                 return AST_CAUSE_INTERWORKING;
1867         case 413:       /* Request entity too large */
1868                 return AST_CAUSE_INTERWORKING;
1869         case 414:       /* Request URI too large */
1870                 return AST_CAUSE_INTERWORKING;
1871         case 415:       /* Unsupported media type */
1872                 return AST_CAUSE_INTERWORKING;
1873         case 420:       /* Bad extension */
1874                 return AST_CAUSE_NO_ROUTE_DESTINATION;
1875         case 480:       /* No answer */
1876                 return AST_CAUSE_NO_ANSWER;
1877         case 481:       /* No answer */
1878                 return AST_CAUSE_INTERWORKING;
1879         case 482:       /* Loop detected */
1880                 return AST_CAUSE_INTERWORKING;
1881         case 483:       /* Too many hops */
1882                 return AST_CAUSE_NO_ANSWER;
1883         case 484:       /* Address incomplete */
1884                 return AST_CAUSE_INVALID_NUMBER_FORMAT;
1885         case 485:       /* Ambiguous */
1886                 return AST_CAUSE_UNALLOCATED;
1887         case 486:       /* Busy everywhere */
1888                 return AST_CAUSE_BUSY;
1889         case 487:       /* Request terminated */
1890                 return AST_CAUSE_INTERWORKING;
1891         case 488:       /* No codecs approved */
1892                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1893         case 491:       /* Request pending */
1894                 return AST_CAUSE_INTERWORKING;
1895         case 493:       /* Undecipherable */
1896                 return AST_CAUSE_INTERWORKING;
1897         case 500:       /* Server internal failure */
1898                 return AST_CAUSE_FAILURE;
1899         case 501:       /* Call rejected */
1900                 return AST_CAUSE_FACILITY_REJECTED;
1901         case 502:
1902                 return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
1903         case 503:       /* Service unavailable */
1904                 return AST_CAUSE_CONGESTION;
1905         case 504:       /* Gateway timeout */
1906                 return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
1907         case 505:       /* SIP version not supported */
1908                 return AST_CAUSE_INTERWORKING;
1909         case 600:       /* Busy everywhere */
1910                 return AST_CAUSE_USER_BUSY;
1911         case 603:       /* Decline */
1912                 return AST_CAUSE_CALL_REJECTED;
1913         case 604:       /* Does not exist anywhere */
1914                 return AST_CAUSE_UNALLOCATED;
1915         case 606:       /* Not acceptable */
1916                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1917         default:
1918                 if (cause < 500 && cause >= 400) {
1919                         /* 4xx class error that is unknown - someting wrong with our request */
1920                         return AST_CAUSE_INTERWORKING;
1921                 } else if (cause < 600 && cause >= 500) {
1922                         /* 5xx class error - problem in the remote end */
1923                         return AST_CAUSE_CONGESTION;
1924                 } else if (cause < 700 && cause >= 600) {
1925                         /* 6xx - global errors in the 4xx class */
1926                         return AST_CAUSE_INTERWORKING;
1927                 }
1928                 return AST_CAUSE_NORMAL;
1929         }
1930         /* Never reached */
1931         return 0;
1932 }
1933
1934 static void chan_pjsip_session_begin(struct ast_sip_session *session)
1935 {
1936         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
1937
1938         if (session->endpoint->media.direct_media.glare_mitigation ==
1939                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
1940                 return;
1941         }
1942
1943         datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
1944                         "direct_media_glare_mitigation");
1945
1946         if (!datastore) {
1947                 return;
1948         }
1949
1950         ast_sip_session_add_datastore(session, datastore);
1951 }
1952
1953 /*! \brief Function called when the session ends */
1954 static void chan_pjsip_session_end(struct ast_sip_session *session)
1955 {
1956         if (!session->channel) {
1957                 return;
1958         }
1959
1960         if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
1961                 int cause = hangup_sip2cause(session->inv_session->cause);
1962
1963                 ast_queue_hangup_with_cause(session->channel, cause);
1964         } else {
1965                 ast_queue_hangup(session->channel);
1966         }
1967 }
1968
1969 /*! \brief Function called when a request is received on the session */
1970 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1971 {
1972         pjsip_tx_data *packet = NULL;
1973
1974         if (session->channel) {
1975                 return 0;
1976         }
1977
1978         if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
1979                 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
1980                         ast_sip_session_send_response(session, packet);
1981                 }
1982
1983                 ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n");
1984                 return -1;
1985         }
1986         /* channel gets created on incoming request, but we wait to call start
1987            so other supplements have a chance to run */
1988         return 0;
1989 }
1990
1991 static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
1992 {
1993         int res;
1994
1995         res = ast_pbx_start(session->channel);
1996
1997         switch (res) {
1998         case AST_PBX_FAILED:
1999                 ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
2000                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
2001                 ast_hangup(session->channel);
2002                 break;
2003         case AST_PBX_CALL_LIMIT:
2004                 ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
2005                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
2006                 ast_hangup(session->channel);
2007                 break;
2008         case AST_PBX_SUCCESS:
2009         default:
2010                 break;
2011         }
2012
2013         ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
2014
2015         return (res == AST_PBX_SUCCESS) ? 0 : -1;
2016 }
2017
2018 static struct ast_sip_session_supplement pbx_start_supplement = {
2019         .method = "INVITE",
2020         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_LAST,
2021         .incoming_request = pbx_start_incoming_request,
2022 };
2023
2024 /*! \brief Function called when a response is received on the session */
2025 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
2026 {
2027         struct pjsip_status_line status = rdata->msg_info.msg->line.status;
2028
2029         if (!session->channel) {
2030                 return;
2031         }
2032
2033         switch (status.code) {
2034         case 180:
2035                 ast_queue_control(session->channel, AST_CONTROL_RINGING);
2036                 ast_channel_lock(session->channel);
2037                 if (ast_channel_state(session->channel) != AST_STATE_UP) {
2038                         ast_setstate(session->channel, AST_STATE_RINGING);
2039                 }
2040                 ast_channel_unlock(session->channel);
2041                 break;
2042         case 183:
2043                 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
2044                 break;
2045         case 200:
2046                 ast_queue_control(session->channel, AST_CONTROL_ANSWER);
2047                 break;
2048         default:
2049                 break;
2050         }
2051 }
2052
2053 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
2054 {
2055         if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
2056                 if (session->endpoint->media.direct_media.enabled && session->channel) {
2057                         ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
2058                 }
2059         }
2060         return 0;
2061 }
2062
2063 /*!
2064  * \brief Load the module
2065  *
2066  * Module loading including tests for configuration or dependencies.
2067  * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
2068  * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
2069  * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
2070  * configuration file or other non-critical problem return
2071  * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
2072  */
2073 static int load_module(void)
2074 {
2075         if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc(0))) {
2076                 return AST_MODULE_LOAD_DECLINE;
2077         }
2078
2079         ast_format_cap_add_all_by_type(chan_pjsip_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
2080
2081         ast_rtp_glue_register(&chan_pjsip_rtp_glue);
2082
2083         if (ast_channel_register(&chan_pjsip_tech)) {
2084                 ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
2085                 goto end;
2086         }
2087
2088         if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
2089                 ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
2090                 goto end;
2091         }
2092
2093         if (ast_custom_function_register(&media_offer_function)) {
2094                 ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
2095         }
2096
2097         if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
2098                 ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
2099                 goto end;
2100         }
2101
2102         if (ast_sip_session_register_supplement(&pbx_start_supplement)) {
2103                 ast_log(LOG_ERROR, "Unable to register PJSIP pbx start supplement\n");
2104                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2105                 goto end;
2106         }
2107
2108         if (ast_sip_session_register_supplement(&chan_pjsip_ack_supplement)) {
2109                 ast_log(LOG_ERROR, "Unable to register PJSIP ACK supplement\n");
2110                 ast_sip_session_unregister_supplement(&pbx_start_supplement);
2111                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2112                 goto end;
2113         }
2114
2115         return 0;
2116
2117 end:
2118         ast_custom_function_unregister(&media_offer_function);
2119         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2120         ast_channel_unregister(&chan_pjsip_tech);
2121         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2122
2123         return AST_MODULE_LOAD_FAILURE;
2124 }
2125
2126 /*! \brief Reload module */
2127 static int reload(void)
2128 {
2129         return -1;
2130 }
2131
2132 /*! \brief Unload the PJSIP channel from Asterisk */
2133 static int unload_module(void)
2134 {
2135         ast_custom_function_unregister(&media_offer_function);
2136
2137         ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2138         ast_sip_session_unregister_supplement(&pbx_start_supplement);
2139         ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
2140
2141         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2142         ast_channel_unregister(&chan_pjsip_tech);
2143         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2144
2145         return 0;
2146 }
2147
2148 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
2149                 .load = load_module,
2150                 .unload = unload_module,
2151                 .reload = reload,
2152                 .load_pri = AST_MODPRI_CHANNEL_DRIVER,
2153                );