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Merged revisions 111125 via svnmerge from
[asterisk-bristuff.git] / main / rtp.c
blob6cb7fb6bafca9616038ea20318099583dbc71b75
1 /*
2 * Asterisk -- An open source telephony toolkit.
3 *
4 * Copyright (C) 1999 - 2006, Digium, Inc.
5 *
6 * Mark Spencer <markster@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 /*!
20 * \file
21 *
22 * \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
23 *
24 * \author Mark Spencer <markster@digium.com>
25 *
26 * \note RTP is defined in RFC 3550.
27 */
28
29 #include "asterisk.h"
30
31 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
32
33 #include <stdio.h>
34 #include <stdlib.h>
35 #include <string.h>
36 #include <sys/time.h>
37 #include <signal.h>
38 #include <errno.h>
39 #include <unistd.h>
40 #include <netinet/in.h>
41 #include <sys/time.h>
42 #include <sys/socket.h>
43 #include <arpa/inet.h>
44 #include <fcntl.h>
45
46 #include "asterisk/rtp.h"
47 #include "asterisk/frame.h"
48 #include "asterisk/logger.h"
49 #include "asterisk/options.h"
50 #include "asterisk/channel.h"
51 #include "asterisk/acl.h"
52 #include "asterisk/channel.h"
53 #include "asterisk/config.h"
54 #include "asterisk/lock.h"
55 #include "asterisk/utils.h"
56 #include "asterisk/cli.h"
57 #include "asterisk/unaligned.h"
58 #include "asterisk/utils.h"
59
60 #define MAX_TIMESTAMP_SKEW 640
61
62 #define RTP_SEQ_MOD (1<<16) /*!< A sequence number can't be more than 16 bits */
63 #define RTCP_DEFAULT_INTERVALMS 5000 /*!< Default milli-seconds between RTCP reports we send */
64 #define RTCP_MIN_INTERVALMS 500 /*!< Min milli-seconds between RTCP reports we send */
65 #define RTCP_MAX_INTERVALMS 60000 /*!< Max milli-seconds between RTCP reports we send */
66
67 #define RTCP_PT_FUR 192
68 #define RTCP_PT_SR 200
69 #define RTCP_PT_RR 201
70 #define RTCP_PT_SDES 202
71 #define RTCP_PT_BYE 203
72 #define RTCP_PT_APP 204
73
74 #define RTP_MTU 1200
75
76 #define DEFAULT_DTMF_TIMEOUT 3000 /*!< samples */
77
78 static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
79
80 static int rtpstart; /*!< First port for RTP sessions (set in rtp.conf) */
81 static int rtpend; /*!< Last port for RTP sessions (set in rtp.conf) */
82 static int rtpdebug; /*!< Are we debugging? */
83 static int rtcpdebug; /*!< Are we debugging RTCP? */
84 static int rtcpstats; /*!< Are we debugging RTCP? */
85 static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
86 static int stundebug; /*!< Are we debugging stun? */
87 static struct sockaddr_in rtpdebugaddr; /*!< Debug packets to/from this host */
88 static struct sockaddr_in rtcpdebugaddr; /*!< Debug RTCP packets to/from this host */
89 #ifdef SO_NO_CHECK
90 static int nochecksums;
91 #endif
92
93 /* Uncomment this to enable more intense native bridging, but note: this is currently buggy */
94 /* #define P2P_INTENSE */
95
96 /*!
97 * \brief Structure representing a RTP session.
98 *
99 * RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]"
100 *
101 */
102 /*! \brief The value of each payload format mapping: */
103 struct rtpPayloadType {
104 int isAstFormat; /*!< whether the following code is an AST_FORMAT */
105 int code;
106 };
107
108
109 /*! \brief RTP session description */
110 struct ast_rtp {
111 int s;
112 struct ast_frame f;
113 unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
114 unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
115 unsigned int themssrc; /*!< Their SSRC */
116 unsigned int rxssrc;
117 unsigned int lastts;
118 unsigned int lastrxts;
119 unsigned int lastividtimestamp;
120 unsigned int lastovidtimestamp;
121 unsigned int lasteventseqn;
122 int lastrxseqno; /*!< Last received sequence number */
123 unsigned short seedrxseqno; /*!< What sequence number did they start with?*/
124 unsigned int seedrxts; /*!< What RTP timestamp did they start with? */
125 unsigned int rxcount; /*!< How many packets have we received? */
126 unsigned int rxoctetcount; /*!< How many octets have we received? should be rxcount *160*/
127 unsigned int txcount; /*!< How many packets have we sent? */
128 unsigned int txoctetcount; /*!< How many octets have we sent? (txcount*160)*/
129 unsigned int cycles; /*!< Shifted count of sequence number cycles */
130 double rxjitter; /*!< Interarrival jitter at the moment */
131 double rxtransit; /*!< Relative transit time for previous packet */
132 int lasttxformat;
133 int lastrxformat;
134
135 int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
136 int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
137 int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
138
139 /* DTMF Reception Variables */
140 char resp;
141 unsigned int lastevent;
142 int dtmfcount;
143 unsigned int dtmfsamples;
144 /* DTMF Transmission Variables */
145 unsigned int lastdigitts;
146 char sending_digit; /*!< boolean - are we sending digits */
147 char send_digit; /*!< digit we are sending */
148 int send_payload;
149 int send_duration;
150 int nat;
151 unsigned int flags;
152 struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
153 struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
154 struct timeval rxcore;
155 struct timeval txcore;
156 double drxcore; /*!< The double representation of the first received packet */
157 struct timeval lastrx; /*!< timeval when we last received a packet */
158 struct timeval dtmfmute;
159 struct ast_smoother *smoother;
160 int *ioid;
161 unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
162 unsigned short rxseqno;
163 struct sched_context *sched;
164 struct io_context *io;
165 void *data;
166 ast_rtp_callback callback;
167 ast_mutex_t bridge_lock;
168 struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
169 int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
170 int rtp_lookup_code_cache_code;
171 int rtp_lookup_code_cache_result;
172 struct ast_rtcp *rtcp;
173 struct ast_codec_pref pref;
174 struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
175 int set_marker_bit:1; /*!< Whether to set the marker bit or not */
176 };
177
178 /* Forward declarations */
179 static int ast_rtcp_write(const void *data);
180 static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw);
181 static int ast_rtcp_write_sr(const void *data);
182 static int ast_rtcp_write_rr(const void *data);
183 static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp);
184 static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp);
185 int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
186
187 #define FLAG_3389_WARNING (1 << 0)
188 #define FLAG_NAT_ACTIVE (3 << 1)
189 #define FLAG_NAT_INACTIVE (0 << 1)
190 #define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
191 #define FLAG_HAS_DTMF (1 << 3)
192 #define FLAG_P2P_SENT_MARK (1 << 4)
193 #define FLAG_P2P_NEED_DTMF (1 << 5)
194 #define FLAG_CALLBACK_MODE (1 << 6)
195 #define FLAG_DTMF_COMPENSATE (1 << 7)
196 #define FLAG_HAS_STUN (1 << 8)
197
198 /*!
199 * \brief Structure defining an RTCP session.
200 *
201 * The concept "RTCP session" is not defined in RFC 3550, but since
202 * this structure is analogous to ast_rtp, which tracks a RTP session,
203 * it is logical to think of this as a RTCP session.
204 *
205 * RTCP packet is defined on page 9 of RFC 3550.
206 *
207 */
208 struct ast_rtcp {
209 int s; /*!< Socket */
210 struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
211 struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
212 unsigned int soc; /*!< What they told us */
213 unsigned int spc; /*!< What they told us */
214 unsigned int themrxlsr; /*!< The middle 32 bits of the NTP timestamp in the last received SR*/
215 struct timeval rxlsr; /*!< Time when we got their last SR */
216 struct timeval txlsr; /*!< Time when we sent or last SR*/
217 unsigned int expected_prior; /*!< no. packets in previous interval */
218 unsigned int received_prior; /*!< no. packets received in previous interval */
219 int schedid; /*!< Schedid returned from ast_sched_add() to schedule RTCP-transmissions*/
220 unsigned int rr_count; /*!< number of RRs we've sent, not including report blocks in SR's */
221 unsigned int sr_count; /*!< number of SRs we've sent */
222 unsigned int lastsrtxcount; /*!< Transmit packet count when last SR sent */
223 double accumulated_transit; /*!< accumulated a-dlsr-lsr */
224 double rtt; /*!< Last reported rtt */
225 unsigned int reported_jitter; /*!< The contents of their last jitter entry in the RR */
226 unsigned int reported_lost; /*!< Reported lost packets in their RR */
227 char quality[AST_MAX_USER_FIELD];
228 double maxrxjitter;
229 double minrxjitter;
230 double maxrtt;
231 double minrtt;
232 int sendfur;
233 };
234
235
236 typedef struct { unsigned int id[4]; } __attribute__((packed)) stun_trans_id;
237
238 /* XXX Maybe stun belongs in another file if it ever has use outside of RTP */
239 struct stun_header {
240 unsigned short msgtype;
241 unsigned short msglen;
242 stun_trans_id id;
243 unsigned char ies[0];
244 } __attribute__((packed));
245
246 struct stun_attr {
247 unsigned short attr;
248 unsigned short len;
249 unsigned char value[0];
250 } __attribute__((packed));
251
252 struct stun_addr {
253 unsigned char unused;
254 unsigned char family;
255 unsigned short port;
256 unsigned int addr;
257 } __attribute__((packed));
258
259 #define STUN_IGNORE (0)
260 #define STUN_ACCEPT (1)
261
262 #define STUN_BINDREQ 0x0001
263 #define STUN_BINDRESP 0x0101
264 #define STUN_BINDERR 0x0111
265 #define STUN_SECREQ 0x0002
266 #define STUN_SECRESP 0x0102
267 #define STUN_SECERR 0x0112
268
269 #define STUN_MAPPED_ADDRESS 0x0001
270 #define STUN_RESPONSE_ADDRESS 0x0002
271 #define STUN_CHANGE_REQUEST 0x0003
272 #define STUN_SOURCE_ADDRESS 0x0004
273 #define STUN_CHANGED_ADDRESS 0x0005
274 #define STUN_USERNAME 0x0006
275 #define STUN_PASSWORD 0x0007
276 #define STUN_MESSAGE_INTEGRITY 0x0008
277 #define STUN_ERROR_CODE 0x0009
278 #define STUN_UNKNOWN_ATTRIBUTES 0x000a
279 #define STUN_REFLECTED_FROM 0x000b
280
281 static const char *stun_msg2str(int msg)
282 {
283 switch(msg) {
284 case STUN_BINDREQ:
285 return "Binding Request";
286 case STUN_BINDRESP:
287 return "Binding Response";
288 case STUN_BINDERR:
289 return "Binding Error Response";
290 case STUN_SECREQ:
291 return "Shared Secret Request";
292 case STUN_SECRESP:
293 return "Shared Secret Response";
294 case STUN_SECERR:
295 return "Shared Secret Error Response";
296 }
297 return "Non-RFC3489 Message";
298 }
299
300 static const char *stun_attr2str(int msg)
301 {
302 switch(msg) {
303 case STUN_MAPPED_ADDRESS:
304 return "Mapped Address";
305 case STUN_RESPONSE_ADDRESS:
306 return "Response Address";
307 case STUN_CHANGE_REQUEST:
308 return "Change Request";
309 case STUN_SOURCE_ADDRESS:
310 return "Source Address";
311 case STUN_CHANGED_ADDRESS:
312 return "Changed Address";
313 case STUN_USERNAME:
314 return "Username";
315 case STUN_PASSWORD:
316 return "Password";
317 case STUN_MESSAGE_INTEGRITY:
318 return "Message Integrity";
319 case STUN_ERROR_CODE:
320 return "Error Code";
321 case STUN_UNKNOWN_ATTRIBUTES:
322 return "Unknown Attributes";
323 case STUN_REFLECTED_FROM:
324 return "Reflected From";
325 }
326 return "Non-RFC3489 Attribute";
327 }
328
329 struct stun_state {
330 const char *username;
331 const char *password;
332 };
333
334 static int stun_process_attr(struct stun_state *state, struct stun_attr *attr)
335 {
336 if (stundebug)
337 ast_verbose("Found STUN Attribute %s (%04x), length %d\n",
338 stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
339 switch(ntohs(attr->attr)) {
340 case STUN_USERNAME:
341 state->username = (const char *) (attr->value);
342 break;
343 case STUN_PASSWORD:
344 state->password = (const char *) (attr->value);
345 break;
346 default:
347 if (stundebug)
348 ast_verbose("Ignoring STUN attribute %s (%04x), length %d\n",
349 stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
350 }
351 return 0;
352 }
353
354 static void append_attr_string(struct stun_attr **attr, int attrval, const char *s, int *len, int *left)
355 {
356 int size = sizeof(**attr) + strlen(s);
357 if (*left > size) {
358 (*attr)->attr = htons(attrval);
359 (*attr)->len = htons(strlen(s));
360 memcpy((*attr)->value, s, strlen(s));
361 (*attr) = (struct stun_attr *)((*attr)->value + strlen(s));
362 *len += size;
363 *left -= size;
364 }
365 }
366
367 static void append_attr_address(struct stun_attr **attr, int attrval, struct sockaddr_in *sin, int *len, int *left)
368 {
369 int size = sizeof(**attr) + 8;
370 struct stun_addr *addr;
371 if (*left > size) {
372 (*attr)->attr = htons(attrval);
373 (*attr)->len = htons(8);
374 addr = (struct stun_addr *)((*attr)->value);
375 addr->unused = 0;
376 addr->family = 0x01;
377 addr->port = sin->sin_port;
378 addr->addr = sin->sin_addr.s_addr;
379 (*attr) = (struct stun_attr *)((*attr)->value + 8);
380 *len += size;
381 *left -= size;
382 }
383 }
384
385 static int stun_send(int s, struct sockaddr_in *dst, struct stun_header *resp)
386 {
387 return sendto(s, resp, ntohs(resp->msglen) + sizeof(*resp), 0,
388 (struct sockaddr *)dst, sizeof(*dst));
389 }
390
391 static void stun_req_id(struct stun_header *req)
392 {
393 int x;
394 for (x=0;x<4;x++)
395 req->id.id[x] = ast_random();
396 }
397
398 size_t ast_rtp_alloc_size(void)
399 {
400 return sizeof(struct ast_rtp);
401 }
402
403 void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username)
404 {
405 struct stun_header *req;
406 unsigned char reqdata[1024];
407 int reqlen, reqleft;
408 struct stun_attr *attr;
409
410 req = (struct stun_header *)reqdata;
411 stun_req_id(req);
412 reqlen = 0;
413 reqleft = sizeof(reqdata) - sizeof(struct stun_header);
414 req->msgtype = 0;
415 req->msglen = 0;
416 attr = (struct stun_attr *)req->ies;
417 if (username)
418 append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
419 req->msglen = htons(reqlen);
420 req->msgtype = htons(STUN_BINDREQ);
421 stun_send(rtp->s, suggestion, req);
422 }
423
424 static int stun_handle_packet(int s, struct sockaddr_in *src, unsigned char *data, size_t len)
425 {
426 struct stun_header *resp, *hdr = (struct stun_header *)data;
427 struct stun_attr *attr;
428 struct stun_state st;
429 int ret = STUN_IGNORE;
430 unsigned char respdata[1024];
431 int resplen, respleft;
432
433 if (len < sizeof(struct stun_header)) {
434 if (option_debug)
435 ast_log(LOG_DEBUG, "Runt STUN packet (only %zd, wanting at least %zd)\n", len, sizeof(struct stun_header));
436 return -1;
437 }
438 if (stundebug)
439 ast_verbose("STUN Packet, msg %s (%04x), length: %d\n", stun_msg2str(ntohs(hdr->msgtype)), ntohs(hdr->msgtype), ntohs(hdr->msglen));
440 if (ntohs(hdr->msglen) > len - sizeof(struct stun_header)) {
441 if (option_debug)
442 ast_log(LOG_DEBUG, "Scrambled STUN packet length (got %d, expecting %zd)\n", ntohs(hdr->msglen), len - sizeof(struct stun_header));
443 } else
444 len = ntohs(hdr->msglen);
445 data += sizeof(struct stun_header);
446 memset(&st, 0, sizeof(st));
447 while(len) {
448 if (len < sizeof(struct stun_attr)) {
449 if (option_debug)
450 ast_log(LOG_DEBUG, "Runt Attribute (got %zd, expecting %zd)\n", len, sizeof(struct stun_attr));
451 break;
452 }
453 attr = (struct stun_attr *)data;
454 if ((ntohs(attr->len) + sizeof(struct stun_attr)) > len) {
455 if (option_debug)
456 ast_log(LOG_DEBUG, "Inconsistent Attribute (length %d exceeds remaining msg len %d)\n", (int) (ntohs(attr->len) + sizeof(struct stun_attr)), (int) len);
457 break;
458 }
459 if (stun_process_attr(&st, attr)) {
460 if (option_debug)
461 ast_log(LOG_DEBUG, "Failed to handle attribute %s (%04x)\n", stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr));
462 break;
463 }
464 /* Clear attribute in case previous entry was a string */
465 attr->attr = 0;
466 data += ntohs(attr->len) + sizeof(struct stun_attr);
467 len -= ntohs(attr->len) + sizeof(struct stun_attr);
468 }
469 /* Null terminate any string */
470 *data = '\0';
471 resp = (struct stun_header *)respdata;
472 resplen = 0;
473 respleft = sizeof(respdata) - sizeof(struct stun_header);
474 resp->id = hdr->id;
475 resp->msgtype = 0;
476 resp->msglen = 0;
477 attr = (struct stun_attr *)resp->ies;
478 if (!len) {
479 switch(ntohs(hdr->msgtype)) {
480 case STUN_BINDREQ:
481 if (stundebug)
482 ast_verbose("STUN Bind Request, username: %s\n",
483 st.username ? st.username : "<none>");
484 if (st.username)
485 append_attr_string(&attr, STUN_USERNAME, st.username, &resplen, &respleft);
486 append_attr_address(&attr, STUN_MAPPED_ADDRESS, src, &resplen, &respleft);
487 resp->msglen = htons(resplen);
488 resp->msgtype = htons(STUN_BINDRESP);
489 stun_send(s, src, resp);
490 ret = STUN_ACCEPT;
491 break;
492 default:
493 if (stundebug)
494 ast_verbose("Dunno what to do with STUN message %04x (%s)\n", ntohs(hdr->msgtype), stun_msg2str(ntohs(hdr->msgtype)));
495 }
496 }
497 return ret;
498 }
499
500 /*! \brief List of current sessions */
501 static AST_LIST_HEAD_STATIC(protos, ast_rtp_protocol);
502
503 static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
504 {
505 unsigned int sec, usec, frac;
506 sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
507 usec = tv.tv_usec;
508 frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
509 *msw = sec;
510 *lsw = frac;
511 }
512
513 int ast_rtp_fd(struct ast_rtp *rtp)
514 {
515 return rtp->s;
516 }
517
518 int ast_rtcp_fd(struct ast_rtp *rtp)
519 {
520 if (rtp->rtcp)
521 return rtp->rtcp->s;
522 return -1;
523 }
524
525 unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
526 {
527 unsigned int interval;
528 /*! \todo XXX Do a more reasonable calculation on this one
529 * Look in RFC 3550 Section A.7 for an example*/
530 interval = rtcpinterval;
531 return interval;
532 }
533
534 /* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
535 void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp)
536 {
537 rtp->rtptimeout = (-1) * rtp->rtptimeout;
538 rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
539 }
540
541 /*! \brief Set rtp timeout */
542 void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout)
543 {
544 rtp->rtptimeout = timeout;
545 }
546
547 /*! \brief Set rtp hold timeout */
548 void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout)
549 {
550 rtp->rtpholdtimeout = timeout;
551 }
552
553 /*! \brief set RTP keepalive interval */
554 void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period)
555 {
556 rtp->rtpkeepalive = period;
557 }
558
559 /*! \brief Get rtp timeout */
560 int ast_rtp_get_rtptimeout(struct ast_rtp *rtp)
561 {
562 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
563 return 0;
564 return rtp->rtptimeout;
565 }
566
567 /*! \brief Get rtp hold timeout */
568 int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp)
569 {
570 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
571 return 0;
572 return rtp->rtpholdtimeout;
573 }
574
575 /*! \brief Get RTP keepalive interval */
576 int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp)
577 {
578 return rtp->rtpkeepalive;
579 }
580
581 void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
582 {
583 rtp->data = data;
584 }
585
586 void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback)
587 {
588 rtp->callback = callback;
589 }
590
591 void ast_rtp_setnat(struct ast_rtp *rtp, int nat)
592 {
593 rtp->nat = nat;
594 }
595
596 int ast_rtp_getnat(struct ast_rtp *rtp)
597 {
598 return ast_test_flag(rtp, FLAG_NAT_ACTIVE);
599 }
600
601 void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf)
602 {
603 ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
604 }
605
606 void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate)
607 {
608 ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
609 }
610
611 void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable)
612 {
613 ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
614 }
615
616 static struct ast_frame *send_dtmf(struct ast_rtp *rtp, enum ast_frame_type type)
617 {
618 if (((ast_test_flag(rtp, FLAG_DTMF_COMPENSATE) && type == AST_FRAME_DTMF_END) ||
619 (type == AST_FRAME_DTMF_BEGIN)) && ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) {
620 if (option_debug)
621 ast_log(LOG_DEBUG, "Ignore potential DTMF echo from '%s'\n", ast_inet_ntoa(rtp->them.sin_addr));
622 rtp->resp = 0;
623 rtp->dtmfsamples = 0;
624 return &ast_null_frame;
625 }
626 if (option_debug)
627 ast_log(LOG_DEBUG, "Sending dtmf: %d (%c), at %s\n", rtp->resp, rtp->resp, ast_inet_ntoa(rtp->them.sin_addr));
628 if (rtp->resp == 'X') {
629 rtp->f.frametype = AST_FRAME_CONTROL;
630 rtp->f.subclass = AST_CONTROL_FLASH;
631 } else {
632 rtp->f.frametype = type;
633 rtp->f.subclass = rtp->resp;
634 }
635 rtp->f.datalen = 0;
636 rtp->f.samples = 0;
637 rtp->f.mallocd = 0;
638 rtp->f.src = "RTP";
639 return &rtp->f;
640
641 }
642
643 static inline int rtp_debug_test_addr(struct sockaddr_in *addr)
644 {
645 if (rtpdebug == 0)
646 return 0;
647 if (rtpdebugaddr.sin_addr.s_addr) {
648 if (((ntohs(rtpdebugaddr.sin_port) != 0)
649 && (rtpdebugaddr.sin_port != addr->sin_port))
650 || (rtpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
651 return 0;
652 }
653 return 1;
654 }
655
656 static inline int rtcp_debug_test_addr(struct sockaddr_in *addr)
657 {
658 if (rtcpdebug == 0)
659 return 0;
660 if (rtcpdebugaddr.sin_addr.s_addr) {
661 if (((ntohs(rtcpdebugaddr.sin_port) != 0)
662 && (rtcpdebugaddr.sin_port != addr->sin_port))
663 || (rtcpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
664 return 0;
665 }
666 return 1;
667 }
668
669
670 static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *data, int len)
671 {
672 unsigned int event;
673 char resp = 0;
674 struct ast_frame *f = NULL;
675 event = ntohl(*((unsigned int *)(data)));
676 event &= 0x001F;
677 if (option_debug > 2 || rtpdebug)
678 ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
679 if (event < 10) {
680 resp = '0' + event;
681 } else if (event < 11) {
682 resp = '*';
683 } else if (event < 12) {
684 resp = '#';
685 } else if (event < 16) {
686 resp = 'A' + (event - 12);
687 } else if (event < 17) {
688 resp = 'X';
689 }
690 if (rtp->resp && (rtp->resp != resp)) {
691 f = send_dtmf(rtp, AST_FRAME_DTMF_END);
692 }
693 rtp->resp = resp;
694 rtp->dtmfcount = dtmftimeout;
695 return f;
696 }
697
698 /*!
699 * \brief Process RTP DTMF and events according to RFC 2833.
700 *
701 * RFC 2833 is "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals".
702 *
703 * \param rtp
704 * \param data
705 * \param len
706 * \param seqno
707 * \returns
708 */
709 static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp)
710 {
711 unsigned int event;
712 unsigned int event_end;
713 unsigned int samples;
714 char resp = 0;
715 struct ast_frame *f = NULL;
716
717 /* Figure out event, event end, and samples */
718 event = ntohl(*((unsigned int *)(data)));
719 event >>= 24;
720 event_end = ntohl(*((unsigned int *)(data)));
721 event_end <<= 8;
722 event_end >>= 24;
723 samples = ntohl(*((unsigned int *)(data)));
724 samples &= 0xFFFF;
725
726 /* Print out debug if turned on */
727 if (rtpdebug || option_debug > 2)
728 ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
729
730 /* Figure out what digit was pressed */
731 if (event < 10) {
732 resp = '0' + event;
733 } else if (event < 11) {
734 resp = '*';
735 } else if (event < 12) {
736 resp = '#';
737 } else if (event < 16) {
738 resp = 'A' + (event - 12);
739 } else if (event < 17) { /* Event 16: Hook flash */
740 resp = 'X';
741 } else {
742 /* Not a supported event */
743 ast_log(LOG_DEBUG, "Ignoring RTP 2833 Event: %08x. Not a DTMF Digit.\n", event);
744 return &ast_null_frame;
745 }
746
747 if (ast_test_flag(rtp, FLAG_DTMF_COMPENSATE)) {
748 if ((rtp->lastevent != timestamp) || (rtp->resp && rtp->resp != resp)) {
749 rtp->resp = resp;
750 f = send_dtmf(rtp, AST_FRAME_DTMF_END);
751 f->len = 0;
752 rtp->lastevent = timestamp;
753 }
754 } else {
755 if ((!(rtp->resp) && (!(event_end & 0x80))) || (rtp->resp && rtp->resp != resp)) {
756 rtp->resp = resp;
757 f = send_dtmf(rtp, AST_FRAME_DTMF_BEGIN);
758 } else if ((event_end & 0x80) && (rtp->lastevent != seqno) && rtp->resp) {
759 f = send_dtmf(rtp, AST_FRAME_DTMF_END);
760 f->len = ast_tvdiff_ms(ast_samp2tv(samples, 8000), ast_tv(0, 0)); /* XXX hard coded 8kHz */
761 rtp->resp = 0;
762 rtp->lastevent = seqno;
763 }
764 }
765
766 rtp->dtmfcount = dtmftimeout;
767 rtp->dtmfsamples = samples;
768
769 return f;
770 }
771
772 /*!
773 * \brief Process Comfort Noise RTP.
774 *
775 * This is incomplete at the moment.
776 *
777 */
778 static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *data, int len)
779 {
780 struct ast_frame *f = NULL;
781 /* Convert comfort noise into audio with various codecs. Unfortunately this doesn't
782 totally help us out becuase we don't have an engine to keep it going and we are not
783 guaranteed to have it every 20ms or anything */
784 if (rtpdebug)
785 ast_log(LOG_DEBUG, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len);
786
787 if (!(ast_test_flag(rtp, FLAG_3389_WARNING))) {
788 ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: %s\n",
789 ast_inet_ntoa(rtp->them.sin_addr));
790 ast_set_flag(rtp, FLAG_3389_WARNING);
791 }
792
793 /* Must have at least one byte */
794 if (!len)
795 return NULL;
796 if (len < 24) {
797 rtp->f.data = rtp->rawdata + AST_FRIENDLY_OFFSET;
798 rtp->f.datalen = len - 1;
799 rtp->f.offset = AST_FRIENDLY_OFFSET;
800 memcpy(rtp->f.data, data + 1, len - 1);
801 } else {
802 rtp->f.data = NULL;
803 rtp->f.offset = 0;
804 rtp->f.datalen = 0;
805 }
806 rtp->f.frametype = AST_FRAME_CNG;
807 rtp->f.subclass = data[0] & 0x7f;
808 rtp->f.datalen = len - 1;
809 rtp->f.samples = 0;
810 rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
811 f = &rtp->f;
812 return f;
813 }
814
815 static int rtpread(int *id, int fd, short events, void *cbdata)
816 {
817 struct ast_rtp *rtp = cbdata;
818 struct ast_frame *f;
819 f = ast_rtp_read(rtp);
820 if (f) {
821 if (rtp->callback)
822 rtp->callback(rtp, f, rtp->data);
823 }
824 return 1;
825 }
826
827 struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp)
828 {
829 socklen_t len;
830 int position, i, packetwords;
831 int res;
832 struct sockaddr_in sin;
833 unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
834 unsigned int *rtcpheader;
835 int pt;
836 struct timeval now;
837 unsigned int length;
838 int rc;
839 double rttsec;
840 uint64_t rtt = 0;
841 unsigned int dlsr;
842 unsigned int lsr;
843 unsigned int msw;
844 unsigned int lsw;
845 unsigned int comp;
846 struct ast_frame *f = &ast_null_frame;
847
848 if (!rtp || !rtp->rtcp)
849 return &ast_null_frame;
850
851 len = sizeof(sin);
852
853 res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
854 0, (struct sockaddr *)&sin, &len);
855 rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
856
857 if (res < 0) {
858 if (errno == EBADF)
859 CRASH;
860 if (errno != EAGAIN) {
861 ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno));
862 return NULL;
863 }
864 return &ast_null_frame;
865 }
866
867 packetwords = res / 4;
868
869 if (rtp->nat) {
870 /* Send to whoever sent to us */
871 if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
872 (rtp->rtcp->them.sin_port != sin.sin_port)) {
873 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
874 if (option_debug || rtpdebug)
875 ast_log(LOG_DEBUG, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
876 }
877 }
878
879 if (option_debug)
880 ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
881
882 /* Process a compound packet */
883 position = 0;
884 while (position < packetwords) {
885 i = position;
886 length = ntohl(rtcpheader[i]);
887 pt = (length & 0xff0000) >> 16;
888 rc = (length & 0x1f000000) >> 24;
889 length &= 0xffff;
890
891 if ((i + length) > packetwords) {
892 ast_log(LOG_WARNING, "RTCP Read too short\n");
893 return &ast_null_frame;
894 }
895
896 if (rtcp_debug_test_addr(&sin)) {
897 ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
898 ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
899 ast_verbose("Reception reports: %d\n", rc);
900 ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
901 }
902
903 i += 2; /* Advance past header and ssrc */
904
905 switch (pt) {
906 case RTCP_PT_SR:
907 gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
908 rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
909 rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
910 rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
911
912 if (rtcp_debug_test_addr(&sin)) {
913 ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
914 ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
915 ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
916 }
917 i += 5;
918 if (rc < 1)
919 break;
920 /* Intentional fall through */
921 case RTCP_PT_RR:
922 /* Don't handle multiple reception reports (rc > 1) yet */
923 /* Calculate RTT per RFC */
924 gettimeofday(&now, NULL);
925 timeval2ntp(now, &msw, &lsw);
926 if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */
927 comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
928 lsr = ntohl(rtcpheader[i + 4]);
929 dlsr = ntohl(rtcpheader[i + 5]);
930 rtt = comp - lsr - dlsr;
931
932 /* Convert end to end delay to usec (keeping the calculation in 64bit space)
933 sess->ee_delay = (eedelay * 1000) / 65536; */
934 if (rtt < 4294) {
935 rtt = (rtt * 1000000) >> 16;
936 } else {
937 rtt = (rtt * 1000) >> 16;
938 rtt *= 1000;
939 }
940 rtt = rtt / 1000.;
941 rttsec = rtt / 1000.;
942
943 if (comp - dlsr >= lsr) {
944 rtp->rtcp->accumulated_transit += rttsec;
945 rtp->rtcp->rtt = rttsec;
946 if (rtp->rtcp->maxrtt<rttsec)
947 rtp->rtcp->maxrtt = rttsec;
948 if (rtp->rtcp->minrtt>rttsec)
949 rtp->rtcp->minrtt = rttsec;
950 } else if (rtcp_debug_test_addr(&sin)) {
951 ast_verbose("Internal RTCP NTP clock skew detected: "
952 "lsr=%u, now=%u, dlsr=%u (%d:%03dms), "
953 "diff=%d\n",
954 lsr, comp, dlsr, dlsr / 65536,
955 (dlsr % 65536) * 1000 / 65536,
956 dlsr - (comp - lsr));
957 }
958 }
959
960 rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
961 rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
962 if (rtcp_debug_test_addr(&sin)) {
963 ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
964 ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost);
965 ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
966 ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
967 ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
968 ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
969 ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
970 if (rtt)
971 ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt);
972 }
973 break;
974 case RTCP_PT_FUR:
975 if (rtcp_debug_test_addr(&sin))
976 ast_verbose("Received an RTCP Fast Update Request\n");
977 rtp->f.frametype = AST_FRAME_CONTROL;
978 rtp->f.subclass = AST_CONTROL_VIDUPDATE;
979 rtp->f.datalen = 0;
980 rtp->f.samples = 0;
981 rtp->f.mallocd = 0;
982 rtp->f.src = "RTP";
983 f = &rtp->f;
984 break;
985 case RTCP_PT_SDES:
986 if (rtcp_debug_test_addr(&sin))
987 ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
988 break;
989 case RTCP_PT_BYE:
990 if (rtcp_debug_test_addr(&sin))
991 ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
992 break;
993 default:
994 if (option_debug)
995 ast_log(LOG_DEBUG, "Unknown RTCP packet (pt=%d) received from %s:%d\n", pt, ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
996 break;
997 }
998 position += (length + 1);
999 }
1000
1001 return f;
1002 }
1003
1004 static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
1005 {
1006 struct timeval now;
1007 double transit;
1008 double current_time;
1009 double d;
1010 double dtv;
1011 double prog;
1012
1013 if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
1014 gettimeofday(&rtp->rxcore, NULL);
1015 rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
1016 /* map timestamp to a real time */
1017 rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
1018 rtp->rxcore.tv_sec -= timestamp / 8000;
1019 rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
1020 /* Round to 0.1ms for nice, pretty timestamps */
1021 rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
1022 if (rtp->rxcore.tv_usec < 0) {
1023 /* Adjust appropriately if necessary */
1024 rtp->rxcore.tv_usec += 1000000;
1025 rtp->rxcore.tv_sec -= 1;
1026 }
1027 }
1028
1029 gettimeofday(&now,NULL);
1030 /* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
1031 tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
1032 tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
1033 if (tv->tv_usec >= 1000000) {
1034 tv->tv_usec -= 1000000;
1035 tv->tv_sec += 1;
1036 }
1037 prog = (double)((timestamp-rtp->seedrxts)/8000.);
1038 dtv = (double)rtp->drxcore + (double)(prog);
1039 current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
1040 transit = current_time - dtv;
1041 d = transit - rtp->rxtransit;
1042 rtp->rxtransit = transit;
1043 if (d<0)
1044 d=-d;
1045 rtp->rxjitter += (1./16.) * (d - rtp->rxjitter);
1046 if (rtp->rtcp && rtp->rxjitter > rtp->rtcp->maxrxjitter)
1047 rtp->rtcp->maxrxjitter = rtp->rxjitter;
1048 if (rtp->rtcp && rtp->rxjitter < rtp->rtcp->minrxjitter)
1049 rtp->rtcp->minrxjitter = rtp->rxjitter;
1050 }
1051
1052 /*! \brief Perform a Packet2Packet RTP write */
1053 static int bridge_p2p_rtp_write(struct ast_rtp *rtp, struct ast_rtp *bridged, unsigned int *rtpheader, int len, int hdrlen)
1054 {
1055 int res = 0, payload = 0, bridged_payload = 0, mark;
1056 struct rtpPayloadType rtpPT;
1057 int reconstruct = ntohl(rtpheader[0]);
1058
1059 /* Get fields from packet */
1060 payload = (reconstruct & 0x7f0000) >> 16;
1061 mark = (((reconstruct & 0x800000) >> 23) != 0);
1062
1063 /* Check what the payload value should be */
1064 rtpPT = ast_rtp_lookup_pt(rtp, payload);
1065
1066 /* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
1067 if (!bridged->current_RTP_PT[payload].code)
1068 return -1;
1069
1070 /* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
1071 if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
1072 return -1;
1073
1074 /* Otherwise adjust bridged payload to match */
1075 bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
1076
1077 /* If the mark bit has not been sent yet... do it now */
1078 if (!ast_test_flag(rtp, FLAG_P2P_SENT_MARK)) {
1079 mark = 1;
1080 ast_set_flag(rtp, FLAG_P2P_SENT_MARK);
1081 }
1082
1083 /* Reconstruct part of the packet */
1084 reconstruct &= 0xFF80FFFF;
1085 reconstruct |= (bridged_payload << 16);
1086 reconstruct |= (mark << 23);
1087 rtpheader[0] = htonl(reconstruct);
1088
1089 /* Send the packet back out */
1090 res = sendto(bridged->s, (void *)rtpheader, len, 0, (struct sockaddr *)&bridged->them, sizeof(bridged->them));
1091 if (res < 0) {
1092 if (!bridged->nat || (bridged->nat && (ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
1093 ast_log(LOG_DEBUG, "RTP Transmission error of packet to %s:%d: %s\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), strerror(errno));
1094 } else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) {
1095 if (option_debug || rtpdebug)
1096 ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port));
1097 ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
1098 }
1099 return 0;
1100 } else if (rtp_debug_test_addr(&bridged->them))
1101 ast_verbose("Sent RTP P2P packet to %s:%u (type %-2.2d, len %-6.6u)\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), bridged_payload, len - hdrlen);
1102
1103 return 0;
1104 }
1105
1106 struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
1107 {
1108 int res;
1109 struct sockaddr_in sin;
1110 socklen_t len;
1111 unsigned int seqno;
1112 int version;
1113 int payloadtype;
1114 int hdrlen = 12;
1115 int padding;
1116 int mark;
1117 int ext;
1118 int cc;
1119 unsigned int ssrc;
1120 unsigned int timestamp;
1121 unsigned int *rtpheader;
1122 struct rtpPayloadType rtpPT;
1123 struct ast_rtp *bridged = NULL;
1124
1125 /* If time is up, kill it */
1126 if (rtp->sending_digit)
1127 ast_rtp_senddigit_continuation(rtp);
1128
1129 len = sizeof(sin);
1130
1131 /* Cache where the header will go */
1132 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
1133 0, (struct sockaddr *)&sin, &len);
1134
1135 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
1136 if (res < 0) {
1137 if (errno == EBADF)
1138 CRASH;
1139 if (errno != EAGAIN) {
1140 ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno));
1141 return NULL;
1142 }
1143 return &ast_null_frame;
1144 }
1145
1146 if (res < hdrlen) {
1147 ast_log(LOG_WARNING, "RTP Read too short\n");
1148 return &ast_null_frame;
1149 }
1150
1151 /* Get fields */
1152 seqno = ntohl(rtpheader[0]);
1153
1154 /* Check RTP version */
1155 version = (seqno & 0xC0000000) >> 30;
1156 if (!version) {
1157 if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) &&
1158 (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
1159 memcpy(&rtp->them, &sin, sizeof(rtp->them));
1160 }
1161 return &ast_null_frame;
1162 }
1163
1164 #if 0 /* Allow to receive RTP stream with closed transmission path */
1165 /* If we don't have the other side's address, then ignore this */
1166 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
1167 return &ast_null_frame;
1168 #endif
1169
1170 /* Send to whoever send to us if NAT is turned on */
1171 if (rtp->nat) {
1172 if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
1173 (rtp->them.sin_port != sin.sin_port)) {
1174 rtp->them = sin;
1175 if (rtp->rtcp) {
1176 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
1177 rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
1178 }
1179 rtp->rxseqno = 0;
1180 ast_set_flag(rtp, FLAG_NAT_ACTIVE);
1181 if (option_debug || rtpdebug)
1182 ast_log(LOG_DEBUG, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
1183 }
1184 }
1185
1186 /* If we are bridged to another RTP stream, send direct */
1187 if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
1188 return &ast_null_frame;
1189
1190 if (version != 2)
1191 return &ast_null_frame;
1192
1193 payloadtype = (seqno & 0x7f0000) >> 16;
1194 padding = seqno & (1 << 29);
1195 mark = seqno & (1 << 23);
1196 ext = seqno & (1 << 28);
1197 cc = (seqno & 0xF000000) >> 24;
1198 seqno &= 0xffff;
1199 timestamp = ntohl(rtpheader[1]);
1200 ssrc = ntohl(rtpheader[2]);
1201
1202 if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
1203 if (option_debug || rtpdebug)
1204 ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
1205 mark = 1;
1206 }
1207
1208 rtp->rxssrc = ssrc;
1209
1210 if (padding) {
1211 /* Remove padding bytes */
1212 res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
1213 }
1214
1215 if (cc) {
1216 /* CSRC fields present */
1217 hdrlen += cc*4;
1218 }
1219
1220 if (ext) {
1221 /* RTP Extension present */
1222 hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
1223 hdrlen += 4;
1224 }
1225
1226 if (res < hdrlen) {
1227 ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
1228 return &ast_null_frame;
1229 }
1230
1231 rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
1232
1233 if (rtp->rxcount==1) {
1234 /* This is the first RTP packet successfully received from source */
1235 rtp->seedrxseqno = seqno;
1236 }
1237
1238 /* Do not schedule RR if RTCP isn't run */
1239 if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
1240 /* Schedule transmission of Receiver Report */
1241 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
1242 }
1243 if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
1244 rtp->cycles += RTP_SEQ_MOD;
1245
1246 rtp->lastrxseqno = seqno;
1247
1248 if (rtp->themssrc==0)
1249 rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
1250
1251 if (rtp_debug_test_addr(&sin))
1252 ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
1253 ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
1254
1255 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
1256 if (!rtpPT.isAstFormat) {
1257 struct ast_frame *f = NULL;
1258
1259 /* This is special in-band data that's not one of our codecs */
1260 if (rtpPT.code == AST_RTP_DTMF) {
1261 /* It's special -- rfc2833 process it */
1262 if (rtp_debug_test_addr(&sin)) {
1263 unsigned char *data;
1264 unsigned int event;
1265 unsigned int event_end;
1266 unsigned int duration;
1267 data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
1268 event = ntohl(*((unsigned int *)(data)));
1269 event >>= 24;
1270 event_end = ntohl(*((unsigned int *)(data)));
1271 event_end <<= 8;
1272 event_end >>= 24;
1273 duration = ntohl(*((unsigned int *)(data)));
1274 duration &= 0xFFFF;
1275 ast_verbose("Got RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
1276 }
1277 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
1278 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
1279 /* It's really special -- process it the Cisco way */
1280 if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
1281 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
1282 rtp->lastevent = seqno;
1283 }
1284 } else if (rtpPT.code == AST_RTP_CN) {
1285 /* Comfort Noise */
1286 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
1287 } else {
1288 ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
1289 }
1290 return f ? f : &ast_null_frame;
1291 }
1292 rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
1293 rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
1294
1295 if (!rtp->lastrxts)
1296 rtp->lastrxts = timestamp;
1297
1298 rtp->rxseqno = seqno;
1299
1300 /* Record received timestamp as last received now */
1301 rtp->lastrxts = timestamp;
1302
1303 rtp->f.mallocd = 0;
1304 rtp->f.datalen = res - hdrlen;
1305 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
1306 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
1307 rtp->f.seqno = seqno;
1308 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
1309 rtp->f.samples = ast_codec_get_samples(&rtp->f);
1310 if (rtp->f.subclass == AST_FORMAT_SLINEAR)
1311 ast_frame_byteswap_be(&rtp->f);
1312 calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
1313 /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
1314 ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
1315 rtp->f.ts = timestamp / 8;
1316 rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
1317 } else {
1318 /* Video -- samples is # of samples vs. 90000 */
1319 if (!rtp->lastividtimestamp)
1320 rtp->lastividtimestamp = timestamp;
1321 rtp->f.samples = timestamp - rtp->lastividtimestamp;
1322 rtp->lastividtimestamp = timestamp;
1323 rtp->f.delivery.tv_sec = 0;
1324 rtp->f.delivery.tv_usec = 0;
1325 if (mark)
1326 rtp->f.subclass |= 0x1;
1327
1328 }
1329 rtp->f.src = "RTP";
1330 return &rtp->f;
1331 }
1332
1333 /* The following array defines the MIME Media type (and subtype) for each
1334 of our codecs, or RTP-specific data type. */
1335 static struct {
1336 struct rtpPayloadType payloadType;
1337 char* type;
1338 char* subtype;
1339 } mimeTypes[] = {
1340 {{1, AST_FORMAT_G723_1}, "audio", "G723"},
1341 {{1, AST_FORMAT_GSM}, "audio", "GSM"},
1342 {{1, AST_FORMAT_ULAW}, "audio", "PCMU"},
1343 {{1, AST_FORMAT_ULAW}, "audio", "G711U"},
1344 {{1, AST_FORMAT_ALAW}, "audio", "PCMA"},
1345 {{1, AST_FORMAT_ALAW}, "audio", "G711A"},
1346 {{1, AST_FORMAT_G726}, "audio", "G726-32"},
1347 {{1, AST_FORMAT_ADPCM}, "audio", "DVI4"},
1348 {{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
1349 {{1, AST_FORMAT_LPC10}, "audio", "LPC"},
1350 {{1, AST_FORMAT_G729A}, "audio", "G729"},
1351 {{1, AST_FORMAT_G729A}, "audio", "G729A"},
1352 {{1, AST_FORMAT_SPEEX}, "audio", "speex"},
1353 {{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
1354 {{1, AST_FORMAT_G722}, "audio", "G722"},
1355 {{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32"},
1356 {{0, AST_RTP_DTMF}, "audio", "telephone-event"},
1357 {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
1358 {{0, AST_RTP_CN}, "audio", "CN"},
1359 {{1, AST_FORMAT_JPEG}, "video", "JPEG"},
1360 {{1, AST_FORMAT_PNG}, "video", "PNG"},
1361 {{1, AST_FORMAT_H261}, "video", "H261"},
1362 {{1, AST_FORMAT_H263}, "video", "H263"},
1363 {{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998"},
1364 {{1, AST_FORMAT_H264}, "video", "H264"},
1365 };
1366
1367 /* Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
1368 also, our own choices for dynamic payload types. This is our master
1369 table for transmission */
1370 static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
1371 [0] = {1, AST_FORMAT_ULAW},
1372 #ifdef USE_DEPRECATED_G726
1373 [2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
1374 #endif
1375 [3] = {1, AST_FORMAT_GSM},
1376 [4] = {1, AST_FORMAT_G723_1},
1377 [5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
1378 [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
1379 [7] = {1, AST_FORMAT_LPC10},
1380 [8] = {1, AST_FORMAT_ALAW},
1381 [9] = {1, AST_FORMAT_G722},
1382 [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
1383 [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
1384 [13] = {0, AST_RTP_CN},
1385 [16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
1386 [17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
1387 [18] = {1, AST_FORMAT_G729A},
1388 [19] = {0, AST_RTP_CN}, /* Also used for CN */
1389 [26] = {1, AST_FORMAT_JPEG},
1390 [31] = {1, AST_FORMAT_H261},
1391 [34] = {1, AST_FORMAT_H263},
1392 [103] = {1, AST_FORMAT_H263_PLUS},
1393 [97] = {1, AST_FORMAT_ILBC},
1394 [99] = {1, AST_FORMAT_H264},
1395 [101] = {0, AST_RTP_DTMF},
1396 [110] = {1, AST_FORMAT_SPEEX},
1397 [111] = {1, AST_FORMAT_G726},
1398 [112] = {1, AST_FORMAT_G726_AAL2},
1399 [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
1400 };
1401
1402 void ast_rtp_pt_clear(struct ast_rtp* rtp)
1403 {
1404 int i;
1405
1406 if (!rtp)
1407 return;
1408
1409 ast_mutex_lock(&rtp->bridge_lock);
1410
1411 for (i = 0; i < MAX_RTP_PT; ++i) {
1412 rtp->current_RTP_PT[i].isAstFormat = 0;
1413 rtp->current_RTP_PT[i].code = 0;
1414 }
1415
1416 rtp->rtp_lookup_code_cache_isAstFormat = 0;
1417 rtp->rtp_lookup_code_cache_code = 0;
1418 rtp->rtp_lookup_code_cache_result = 0;
1419
1420 ast_mutex_unlock(&rtp->bridge_lock);
1421 }
1422
1423 void ast_rtp_pt_default(struct ast_rtp* rtp)
1424 {
1425 int i;
1426
1427 ast_mutex_lock(&rtp->bridge_lock);
1428
1429 /* Initialize to default payload types */
1430 for (i = 0; i < MAX_RTP_PT; ++i) {
1431 rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
1432 rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
1433 }
1434
1435 rtp->rtp_lookup_code_cache_isAstFormat = 0;
1436 rtp->rtp_lookup_code_cache_code = 0;
1437 rtp->rtp_lookup_code_cache_result = 0;
1438
1439 ast_mutex_unlock(&rtp->bridge_lock);
1440 }
1441
1442 void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
1443 {
1444 unsigned int i;
1445
1446 ast_mutex_lock(&dest->bridge_lock);
1447 ast_mutex_lock(&src->bridge_lock);
1448
1449 for (i=0; i < MAX_RTP_PT; ++i) {
1450 dest->current_RTP_PT[i].isAstFormat =
1451 src->current_RTP_PT[i].isAstFormat;
1452 dest->current_RTP_PT[i].code =
1453 src->current_RTP_PT[i].code;
1454 }
1455 dest->rtp_lookup_code_cache_isAstFormat = 0;
1456 dest->rtp_lookup_code_cache_code = 0;
1457 dest->rtp_lookup_code_cache_result = 0;
1458
1459 ast_mutex_unlock(&src->bridge_lock);
1460 ast_mutex_unlock(&dest->bridge_lock);
1461 }
1462
1463 /*! \brief Get channel driver interface structure */
1464 static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
1465 {
1466 struct ast_rtp_protocol *cur = NULL;
1467
1468 AST_LIST_LOCK(&protos);
1469 AST_LIST_TRAVERSE(&protos, cur, list) {
1470 if (cur->type == chan->tech->type)
1471 break;
1472 }
1473 AST_LIST_UNLOCK(&protos);
1474
1475 return cur;
1476 }
1477
1478 int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
1479 {
1480 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
1481 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
1482 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
1483 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
1484 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
1485 int srccodec, destcodec, nat_active = 0;
1486
1487 /* Lock channels */
1488 ast_channel_lock(dest);
1489 if (src) {
1490 while(ast_channel_trylock(src)) {
1491 ast_channel_unlock(dest);
1492 usleep(1);
1493 ast_channel_lock(dest);
1494 }
1495 }
1496
1497 /* Find channel driver interfaces */
1498 destpr = get_proto(dest);
1499 if (src)
1500 srcpr = get_proto(src);
1501 if (!destpr) {
1502 if (option_debug)
1503 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
1504 ast_channel_unlock(dest);
1505 if (src)
1506 ast_channel_unlock(src);
1507 return 0;
1508 }
1509 if (!srcpr) {
1510 if (option_debug)
1511 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
1512 ast_channel_unlock(dest);
1513 if (src)
1514 ast_channel_unlock(src);
1515 return 0;
1516 }
1517
1518 /* Get audio and video interface (if native bridge is possible) */
1519 audio_dest_res = destpr->get_rtp_info(dest, &destp);
1520 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
1521 if (srcpr) {
1522 audio_src_res = srcpr->get_rtp_info(src, &srcp);
1523 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
1524 }
1525
1526 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
1527 if (audio_dest_res != AST_RTP_TRY_NATIVE) {
1528 /* Somebody doesn't want to play... */
1529 ast_channel_unlock(dest);
1530 if (src)
1531 ast_channel_unlock(src);
1532 return 0;
1533 }
1534 if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
1535 srccodec = srcpr->get_codec(src);
1536 else
1537 srccodec = 0;
1538 if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
1539 destcodec = destpr->get_codec(dest);
1540 else
1541 destcodec = 0;
1542 /* Ensure we have at least one matching codec */
1543 if (!(srccodec & destcodec)) {
1544 ast_channel_unlock(dest);
1545 if (src)
1546 ast_channel_unlock(src);
1547 return 0;
1548 }
1549 /* Consider empty media as non-existant */
1550 if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
1551 srcp = NULL;
1552 /* If the client has NAT stuff turned on then just safe NAT is active */
1553 if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
1554 nat_active = 1;
1555 /* Bridge media early */
1556 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active))
1557 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>");
1558 ast_channel_unlock(dest);
1559 if (src)
1560 ast_channel_unlock(src);
1561 if (option_debug)
1562 ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
1563 return 1;
1564 }
1565
1566 int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media)
1567 {
1568 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
1569 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
1570 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
1571 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
1572 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
1573 int srccodec, destcodec;
1574
1575 /* Lock channels */
1576 ast_channel_lock(dest);
1577 while(ast_channel_trylock(src)) {
1578 ast_channel_unlock(dest);
1579 usleep(1);
1580 ast_channel_lock(dest);
1581 }
1582
1583 /* Find channel driver interfaces */
1584 if (!(destpr = get_proto(dest))) {
1585 if (option_debug)
1586 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
1587 ast_channel_unlock(dest);
1588 ast_channel_unlock(src);
1589 return 0;
1590 }
1591 if (!(srcpr = get_proto(src))) {
1592 if (option_debug)
1593 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
1594 ast_channel_unlock(dest);
1595 ast_channel_unlock(src);
1596 return 0;
1597 }
1598
1599 /* Get audio and video interface (if native bridge is possible) */
1600 audio_dest_res = destpr->get_rtp_info(dest, &destp);
1601 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
1602 audio_src_res = srcpr->get_rtp_info(src, &srcp);
1603 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
1604
1605 /* Ensure we have at least one matching codec */
1606 if (srcpr->get_codec)
1607 srccodec = srcpr->get_codec(src);
1608 else
1609 srccodec = 0;
1610 if (destpr->get_codec)
1611 destcodec = destpr->get_codec(dest);
1612 else
1613 destcodec = 0;
1614
1615 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
1616 if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
1617 /* Somebody doesn't want to play... */
1618 ast_channel_unlock(dest);
1619 ast_channel_unlock(src);
1620 return 0;
1621 }
1622 ast_rtp_pt_copy(destp, srcp);
1623 if (vdestp && vsrcp)
1624 ast_rtp_pt_copy(vdestp, vsrcp);
1625 if (media) {
1626 /* Bridge early */
1627 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
1628 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
1629 }
1630 ast_channel_unlock(dest);
1631 ast_channel_unlock(src);
1632 if (option_debug)
1633 ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
1634 return 1;
1635 }
1636
1637 /*! \brief Make a note of a RTP payload type that was seen in a SDP "m=" line.
1638 * By default, use the well-known value for this type (although it may
1639 * still be set to a different value by a subsequent "a=rtpmap:" line)
1640 */
1641 void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
1642 {
1643 if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0)
1644 return; /* bogus payload type */
1645
1646 ast_mutex_lock(&rtp->bridge_lock);
1647 rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
1648 ast_mutex_unlock(&rtp->bridge_lock);
1649 }
1650
1651 /*! \brief remove setting from payload type list if the rtpmap header indicates
1652 an unknown media type */
1653 void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt)
1654 {
1655 if (pt < 0 || pt > MAX_RTP_PT)
1656 return; /* bogus payload type */
1657
1658 ast_mutex_lock(&rtp->bridge_lock);
1659 rtp->current_RTP_PT[pt].isAstFormat = 0;
1660 rtp->current_RTP_PT[pt].code = 0;
1661 ast_mutex_unlock(&rtp->bridge_lock);
1662 }
1663
1664 /*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
1665 * an SDP "a=rtpmap:" line.
1666 * \return 0 if the MIME type was found and set, -1 if it wasn't found
1667 */
1668 int ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
1669 char *mimeType, char *mimeSubtype,
1670 enum ast_rtp_options options)
1671 {
1672 unsigned int i;
1673 int found = 0;
1674
1675 if (pt < 0 || pt > MAX_RTP_PT)
1676 return -1; /* bogus payload type */
1677
1678 ast_mutex_lock(&rtp->bridge_lock);
1679
1680 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
1681 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
1682 strcasecmp(mimeType, mimeTypes[i].type) == 0) {
1683 found = 1;
1684 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
1685 if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
1686 mimeTypes[i].payloadType.isAstFormat &&
1687 (options & AST_RTP_OPT_G726_NONSTANDARD))
1688 rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
1689 break;
1690 }
1691 }
1692
1693 ast_mutex_unlock(&rtp->bridge_lock);
1694
1695 return (found ? 0 : -1);
1696 }
1697
1698 /*! \brief Return the union of all of the codecs that were set by rtp_set...() calls
1699 * They're returned as two distinct sets: AST_FORMATs, and AST_RTPs */
1700 void ast_rtp_get_current_formats(struct ast_rtp* rtp,
1701 int* astFormats, int* nonAstFormats)
1702 {
1703 int pt;
1704
1705 ast_mutex_lock(&rtp->bridge_lock);
1706
1707 *astFormats = *nonAstFormats = 0;
1708 for (pt = 0; pt < MAX_RTP_PT; ++pt) {
1709 if (rtp->current_RTP_PT[pt].isAstFormat) {
1710 *astFormats |= rtp->current_RTP_PT[pt].code;
1711 } else {
1712 *nonAstFormats |= rtp->current_RTP_PT[pt].code;
1713 }
1714 }
1715
1716 ast_mutex_unlock(&rtp->bridge_lock);
1717
1718 return;
1719 }
1720
1721 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
1722 {
1723 struct rtpPayloadType result;
1724
1725 result.isAstFormat = result.code = 0;
1726
1727 if (pt < 0 || pt > MAX_RTP_PT)
1728 return result; /* bogus payload type */
1729
1730 /* Start with negotiated codecs */
1731 ast_mutex_lock(&rtp->bridge_lock);
1732 result = rtp->current_RTP_PT[pt];
1733 ast_mutex_unlock(&rtp->bridge_lock);
1734
1735 /* If it doesn't exist, check our static RTP type list, just in case */
1736 if (!result.code)
1737 result = static_RTP_PT[pt];
1738
1739 return result;
1740 }
1741
1742 /*! \brief Looks up an RTP code out of our *static* outbound list */
1743 int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int code)
1744 {
1745 int pt = 0;
1746
1747 ast_mutex_lock(&rtp->bridge_lock);
1748
1749 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
1750 code == rtp->rtp_lookup_code_cache_code) {
1751 /* Use our cached mapping, to avoid the overhead of the loop below */
1752 pt = rtp->rtp_lookup_code_cache_result;
1753 ast_mutex_unlock(&rtp->bridge_lock);
1754 return pt;
1755 }
1756
1757 /* Check the dynamic list first */
1758 for (pt = 0; pt < MAX_RTP_PT; ++pt) {
1759 if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
1760 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
1761 rtp->rtp_lookup_code_cache_code = code;
1762 rtp->rtp_lookup_code_cache_result = pt;
1763 ast_mutex_unlock(&rtp->bridge_lock);
1764 return pt;
1765 }
1766 }
1767
1768 /* Then the static list */
1769 for (pt = 0; pt < MAX_RTP_PT; ++pt) {
1770 if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
1771 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
1772 rtp->rtp_lookup_code_cache_code = code;
1773 rtp->rtp_lookup_code_cache_result = pt;
1774 ast_mutex_unlock(&rtp->bridge_lock);
1775 return pt;
1776 }
1777 }
1778
1779 ast_mutex_unlock(&rtp->bridge_lock);
1780
1781 return -1;
1782 }
1783
1784 const char *ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code,
1785 enum ast_rtp_options options)
1786 {
1787 unsigned int i;
1788
1789 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
1790 if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
1791 if (isAstFormat &&
1792 (code == AST_FORMAT_G726_AAL2) &&
1793 (options & AST_RTP_OPT_G726_NONSTANDARD))
1794 return "G726-32";
1795 else
1796 return mimeTypes[i].subtype;
1797 }
1798 }
1799
1800 return "";
1801 }
1802
1803 char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
1804 const int isAstFormat, enum ast_rtp_options options)
1805 {
1806 int format;
1807 unsigned len;
1808 char *end = buf;
1809 char *start = buf;
1810
1811 if (!buf || !size)
1812 return NULL;
1813
1814 snprintf(end, size, "0x%x (", capability);
1815
1816 len = strlen(end);
1817 end += len;
1818 size -= len;
1819 start = end;
1820
1821 for (format = 1; format < AST_RTP_MAX; format <<= 1) {
1822 if (capability & format) {
1823 const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
1824
1825 snprintf(end, size, "%s|", name);
1826 len = strlen(end);
1827 end += len;
1828 size -= len;
1829 }
1830 }
1831
1832 if (start == end)
1833 snprintf(start, size, "nothing)");
1834 else if (size > 1)
1835 *(end -1) = ')';
1836
1837 return buf;
1838 }
1839
1840 static int rtp_socket(void)
1841 {
1842 int s;
1843 long flags;
1844 s = socket(AF_INET, SOCK_DGRAM, 0);
1845 if (s > -1) {
1846 flags = fcntl(s, F_GETFL);
1847 fcntl(s, F_SETFL, flags | O_NONBLOCK);
1848 #ifdef SO_NO_CHECK
1849 if (nochecksums)
1850 setsockopt(s, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
1851 #endif
1852 }
1853 return s;
1854 }
1855
1856 /*!
1857 * \brief Initialize a new RTCP session.
1858 *
1859 * \returns The newly initialized RTCP session.
1860 */
1861 static struct ast_rtcp *ast_rtcp_new(void)
1862 {
1863 struct ast_rtcp *rtcp;
1864
1865 if (!(rtcp = ast_calloc(1, sizeof(*rtcp))))
1866 return NULL;
1867 rtcp->s = rtp_socket();
1868 rtcp->us.sin_family = AF_INET;
1869 rtcp->them.sin_family = AF_INET;
1870 rtcp->schedid = -1;
1871
1872 if (rtcp->s < 0) {
1873 free(rtcp);
1874 ast_log(LOG_WARNING, "Unable to allocate RTCP socket: %s\n", strerror(errno));
1875 return NULL;
1876 }
1877
1878 return rtcp;
1879 }
1880
1881 /*!
1882 * \brief Initialize a new RTP structure.
1883 *
1884 */
1885 void ast_rtp_new_init(struct ast_rtp *rtp)
1886 {
1887 ast_mutex_init(&rtp->bridge_lock);
1888
1889 rtp->them.sin_family = AF_INET;
1890 rtp->us.sin_family = AF_INET;
1891 rtp->ssrc = ast_random();
1892 rtp->seqno = ast_random() & 0xffff;
1893 ast_set_flag(rtp, FLAG_HAS_DTMF);
1894
1895 return;
1896 }
1897
1898 struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr addr)
1899 {
1900 struct ast_rtp *rtp;
1901 int x;
1902 int first;
1903 int startplace;
1904
1905 if (!(rtp = ast_calloc(1, sizeof(*rtp))))
1906 return NULL;
1907
1908 ast_rtp_new_init(rtp);
1909
1910 rtp->s = rtp_socket();
1911 if (rtp->s < 0) {
1912 free(rtp);
1913 ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
1914 return NULL;
1915 }
1916 if (sched && rtcpenable) {
1917 rtp->sched = sched;
1918 rtp->rtcp = ast_rtcp_new();
1919 }
1920
1921 /* Select a random port number in the range of possible RTP */
1922 x = (ast_random() % (rtpend-rtpstart)) + rtpstart;
1923 x = x & ~1;
1924 /* Save it for future references. */
1925 startplace = x;
1926 /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
1927 for (;;) {
1928 /* Must be an even port number by RTP spec */
1929 rtp->us.sin_port = htons(x);
1930 rtp->us.sin_addr = addr;
1931 /* If there's rtcp, initialize it as well. */
1932 if (rtp->rtcp) {
1933 rtp->rtcp->us.sin_port = htons(x + 1);
1934 rtp->rtcp->us.sin_addr = addr;
1935 }
1936 /* Try to bind it/them. */
1937 if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
1938 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
1939 break;
1940 if (!first) {
1941 /* Primary bind succeeded! Gotta recreate it */
1942 close(rtp->s);
1943 rtp->s = rtp_socket();
1944 }
1945 if (errno != EADDRINUSE) {
1946 /* We got an error that wasn't expected, abort! */
1947 ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
1948 close(rtp->s);
1949 if (rtp->rtcp) {
1950 close(rtp->rtcp->s);
1951 free(rtp->rtcp);
1952 }
1953 free(rtp);
1954 return NULL;
1955 }
1956 /* The port was used, increment it (by two). */
1957 x += 2;
1958 /* Did we go over the limit ? */
1959 if (x > rtpend)
1960 /* then, start from the begingig. */
1961 x = (rtpstart + 1) & ~1;
1962 /* Check if we reached the place were we started. */
1963 if (x == startplace) {
1964 /* If so, there's no ports available. */
1965 ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
1966 close(rtp->s);
1967 if (rtp->rtcp) {
1968 close(rtp->rtcp->s);
1969 free(rtp->rtcp);
1970 }
1971 free(rtp);
1972 return NULL;
1973 }
1974 }
1975 rtp->sched = sched;
1976 rtp->io = io;
1977 if (callbackmode) {
1978 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
1979 ast_set_flag(rtp, FLAG_CALLBACK_MODE);
1980 }
1981 ast_rtp_pt_default(rtp);
1982 return rtp;
1983 }
1984
1985 struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
1986 {
1987 struct in_addr ia;
1988
1989 memset(&ia, 0, sizeof(ia));
1990 return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
1991 }
1992
1993 int ast_rtp_settos(struct ast_rtp *rtp, int tos)
1994 {
1995 int res;
1996
1997 if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos))))
1998 ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
1999 return res;
2000 }
2001
2002 void ast_rtp_new_source(struct ast_rtp *rtp)
2003 {
2004 rtp->set_marker_bit = 1;
2005 rtp->ssrc = ast_random();
2006 return;
2007 }
2008
2009 void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
2010 {
2011 rtp->them.sin_port = them->sin_port;
2012 rtp->them.sin_addr = them->sin_addr;
2013 if (rtp->rtcp) {
2014 rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
2015 rtp->rtcp->them.sin_addr = them->sin_addr;
2016 }
2017 rtp->rxseqno = 0;
2018 }
2019
2020 int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
2021 {
2022 if ((them->sin_family != AF_INET) ||
2023 (them->sin_port != rtp->them.sin_port) ||
2024 (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
2025 them->sin_family = AF_INET;
2026 them->sin_port = rtp->them.sin_port;
2027 them->sin_addr = rtp->them.sin_addr;
2028 return 1;
2029 }
2030 return 0;
2031 }
2032
2033 void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
2034 {
2035 *us = rtp->us;
2036 }
2037
2038 struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp)
2039 {
2040 struct ast_rtp *bridged = NULL;
2041
2042 ast_mutex_lock(&rtp->bridge_lock);
2043 bridged = rtp->bridged;
2044 ast_mutex_unlock(&rtp->bridge_lock);
2045
2046 return bridged;
2047 }
2048
2049 void ast_rtp_stop(struct ast_rtp *rtp)
2050 {
2051 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
2052
2053 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
2054 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
2055 if (rtp->rtcp) {
2056 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
2057 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
2058 }
2059
2060 ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
2061 }
2062
2063 void ast_rtp_reset(struct ast_rtp *rtp)
2064 {
2065 memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
2066 memset(&rtp->txcore, 0, sizeof(rtp->txcore));
2067 memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
2068 rtp->lastts = 0;
2069 rtp->lastdigitts = 0;
2070 rtp->lastrxts = 0;
2071 rtp->lastividtimestamp = 0;
2072 rtp->lastovidtimestamp = 0;
2073 rtp->lasteventseqn = 0;
2074 rtp->lastevent = 0;
2075 rtp->lasttxformat = 0;
2076 rtp->lastrxformat = 0;
2077 rtp->dtmfcount = 0;
2078 rtp->dtmfsamples = 0;
2079 rtp->seqno = 0;
2080 rtp->rxseqno = 0;
2081 }
2082
2083 char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual)
2084 {
2085 /*
2086 *ssrc our ssrc
2087 *themssrc their ssrc
2088 *lp lost packets
2089 *rxjitter our calculated jitter(rx)
2090 *rxcount no. received packets
2091 *txjitter reported jitter of the other end
2092 *txcount transmitted packets
2093 *rlp remote lost packets
2094 *rtt round trip time
2095 */
2096
2097 if (qual && rtp) {
2098 qual->local_ssrc = rtp->ssrc;
2099 qual->local_jitter = rtp->rxjitter;
2100 qual->local_count = rtp->rxcount;
2101 qual->remote_ssrc = rtp->themssrc;
2102 qual->remote_count = rtp->txcount;
2103 if (rtp->rtcp) {
2104 qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
2105 qual->remote_lostpackets = rtp->rtcp->reported_lost;
2106 qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
2107 qual->rtt = rtp->rtcp->rtt;
2108 }
2109 }
2110 if (rtp->rtcp) {
2111 snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality),
2112 "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
2113 rtp->ssrc,
2114 rtp->themssrc,
2115 rtp->rtcp->expected_prior - rtp->rtcp->received_prior,
2116 rtp->rxjitter,
2117 rtp->rxcount,
2118 (double)rtp->rtcp->reported_jitter / 65536.0,
2119 rtp->txcount,
2120 rtp->rtcp->reported_lost,
2121 rtp->rtcp->rtt);
2122 return rtp->rtcp->quality;
2123 } else
2124 return "<Unknown> - RTP/RTCP has already been destroyed";
2125 }
2126
2127 void ast_rtp_destroy(struct ast_rtp *rtp)
2128 {
2129 if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
2130 /*Print some info on the call here */
2131 ast_verbose(" RTP-stats\n");
2132 ast_verbose("* Our Receiver:\n");
2133 ast_verbose(" SSRC: %u\n", rtp->themssrc);
2134 ast_verbose(" Received packets: %u\n", rtp->rxcount);
2135 ast_verbose(" Lost packets: %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
2136 ast_verbose(" Jitter: %.4f\n", rtp->rxjitter);
2137 ast_verbose(" Transit: %.4f\n", rtp->rxtransit);
2138 ast_verbose(" RR-count: %u\n", rtp->rtcp->rr_count);
2139 ast_verbose("* Our Sender:\n");
2140 ast_verbose(" SSRC: %u\n", rtp->ssrc);
2141 ast_verbose(" Sent packets: %u\n", rtp->txcount);
2142 ast_verbose(" Lost packets: %u\n", rtp->rtcp->reported_lost);
2143 ast_verbose(" Jitter: %u\n", rtp->rtcp->reported_jitter / (unsigned int)65536.0);
2144 ast_verbose(" SR-count: %u\n", rtp->rtcp->sr_count);
2145 ast_verbose(" RTT: %f\n", rtp->rtcp->rtt);
2146 }
2147
2148 if (rtp->smoother)
2149 ast_smoother_free(rtp->smoother);
2150 if (rtp->ioid)
2151 ast_io_remove(rtp->io, rtp->ioid);
2152 if (rtp->s > -1)
2153 close(rtp->s);
2154 if (rtp->rtcp) {
2155 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
2156 close(rtp->rtcp->s);
2157 free(rtp->rtcp);
2158 rtp->rtcp=NULL;
2159 }
2160
2161 ast_mutex_destroy(&rtp->bridge_lock);
2162
2163 free(rtp);
2164 }
2165
2166 static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
2167 {
2168 struct timeval t;
2169 long ms;
2170 if (ast_tvzero(rtp->txcore)) {
2171 rtp->txcore = ast_tvnow();
2172 /* Round to 20ms for nice, pretty timestamps */
2173 rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
2174 }
2175 /* Use previous txcore if available */
2176 t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
2177 ms = ast_tvdiff_ms(t, rtp->txcore);
2178 if (ms < 0)
2179 ms = 0;
2180 /* Use what we just got for next time */
2181 rtp->txcore = t;
2182 return (unsigned int) ms;
2183 }
2184
2185 /*! \brief Send begin frames for DTMF */
2186 int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit)
2187 {
2188 unsigned int *rtpheader;
2189 int hdrlen = 12, res = 0, i = 0, payload = 0;
2190 char data[256];
2191
2192 if ((digit <= '9') && (digit >= '0'))
2193 digit -= '0';
2194 else if (digit == '*')
2195 digit = 10;
2196 else if (digit == '#')
2197 digit = 11;
2198 else if ((digit >= 'A') && (digit <= 'D'))
2199 digit = digit - 'A' + 12;
2200 else if ((digit >= 'a') && (digit <= 'd'))
2201 digit = digit - 'a' + 12;
2202 else {
2203 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
2204 return 0;
2205 }
2206
2207 /* If we have no peer, return immediately */
2208 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
2209 return 0;
2210
2211 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
2212
2213 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
2214 rtp->send_duration = 160;
2215
2216 /* Get a pointer to the header */
2217 rtpheader = (unsigned int *)data;
2218 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
2219 rtpheader[1] = htonl(rtp->lastdigitts);
2220 rtpheader[2] = htonl(rtp->ssrc);
2221
2222 for (i = 0; i < 2; i++) {
2223 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
2224 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
2225 if (res < 0)
2226 ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
2227 ast_inet_ntoa(rtp->them.sin_addr),
2228 ntohs(rtp->them.sin_port), strerror(errno));
2229 if (rtp_debug_test_addr(&rtp->them))
2230 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
2231 ast_inet_ntoa(rtp->them.sin_addr),
2232 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
2233 /* Increment sequence number */
2234 rtp->seqno++;
2235 /* Increment duration */
2236 rtp->send_duration += 160;
2237 /* Clear marker bit and set seqno */
2238 rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
2239 }
2240
2241 /* Since we received a begin, we can safely store the digit and disable any compensation */
2242 rtp->sending_digit = 1;
2243 rtp->send_digit = digit;
2244 rtp->send_payload = payload;
2245
2246 return 0;
2247 }
2248
2249 /*! \brief Send continuation frame for DTMF */
2250 static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp)
2251 {
2252 unsigned int *rtpheader;
2253 int hdrlen = 12, res = 0;
2254 char data[256];
2255
2256 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
2257 return 0;
2258
2259 /* Setup packet to send */
2260 rtpheader = (unsigned int *)data;
2261 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
2262 rtpheader[1] = htonl(rtp->lastdigitts);
2263 rtpheader[2] = htonl(rtp->ssrc);
2264 rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
2265 rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
2266
2267 /* Transmit */
2268 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
2269 if (res < 0)
2270 ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
2271 ast_inet_ntoa(rtp->them.sin_addr),
2272 ntohs(rtp->them.sin_port), strerror(errno));
2273 if (rtp_debug_test_addr(&rtp->them))
2274 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
2275 ast_inet_ntoa(rtp->them.sin_addr),
2276 ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
2277
2278 /* Increment sequence number */
2279 rtp->seqno++;
2280 /* Increment duration */
2281 rtp->send_duration += 160;
2282
2283 return 0;
2284 }
2285
2286 /*! \brief Send end packets for DTMF */
2287 int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit)
2288 {
2289 unsigned int *rtpheader;
2290 int hdrlen = 12, res = 0, i = 0;
2291 char data[256];
2292
2293 /* If no address, then bail out */
2294 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
2295 return 0;
2296
2297 if ((digit <= '9') && (digit >= '0'))
2298 digit -= '0';
2299 else if (digit == '*')
2300 digit = 10;
2301 else if (digit == '#')
2302 digit = 11;
2303 else if ((digit >= 'A') && (digit <= 'D'))
2304 digit = digit - 'A' + 12;
2305 else if ((digit >= 'a') && (digit <= 'd'))
2306 digit = digit - 'a' + 12;
2307 else {
2308 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
2309 return 0;
2310 }
2311
2312 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
2313
2314 rtpheader = (unsigned int *)data;
2315 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
2316 rtpheader[1] = htonl(rtp->lastdigitts);
2317 rtpheader[2] = htonl(rtp->ssrc);
2318 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
2319 /* Set end bit */
2320 rtpheader[3] |= htonl((1 << 23));
2321 rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
2322 /* Send 3 termination packets */
2323 for (i = 0; i < 3; i++) {
2324 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
2325 if (res < 0)
2326 ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
2327 ast_inet_ntoa(rtp->them.sin_addr),
2328 ntohs(rtp->them.sin_port), strerror(errno));
2329 if (rtp_debug_test_addr(&rtp->them))
2330 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
2331 ast_inet_ntoa(rtp->them.sin_addr),
2332 ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
2333 }
2334 rtp->sending_digit = 0;
2335 rtp->send_digit = 0;
2336 /* Increment lastdigitts */
2337 rtp->lastdigitts += 960;
2338 rtp->seqno++;
2339
2340 return res;
2341 }
2342
2343 /*! \brief Public function: Send an H.261 fast update request, some devices need this rather than SIP XML */
2344 int ast_rtcp_send_h261fur(void *data)
2345 {
2346 struct ast_rtp *rtp = data;
2347 int res;
2348
2349 rtp->rtcp->sendfur = 1;
2350 res = ast_rtcp_write(data);
2351
2352 return res;
2353 }
2354
2355 /*! \brief Send RTCP sender's report */
2356 static int ast_rtcp_write_sr(const void *data)
2357 {
2358 struct ast_rtp *rtp = (struct ast_rtp *)data;
2359 int res;
2360 int len = 0;
2361 struct timeval now;
2362 unsigned int now_lsw;
2363 unsigned int now_msw;
2364 unsigned int *rtcpheader;
2365 unsigned int lost;
2366 unsigned int extended;
2367 unsigned int expected;
2368 unsigned int expected_interval;
2369 unsigned int received_interval;
2370 int lost_interval;
2371 int fraction;
2372 struct timeval dlsr;
2373 char bdata[512];
2374
2375 /* Commented condition is always not NULL if rtp->rtcp is not NULL */
2376 if (!rtp || !rtp->rtcp/* || (&rtp->rtcp->them.sin_addr == 0)*/)
2377 return 0;
2378
2379 if (!rtp->rtcp->them.sin_addr.s_addr) { /* This'll stop rtcp for this rtp session */
2380 ast_verbose("RTCP SR transmission error, rtcp halted\n");
2381 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
2382 return 0;
2383 }
2384
2385 gettimeofday(&now, NULL);
2386 timeval2ntp(now, &now_msw, &now_lsw); /* fill thses ones in from utils.c*/
2387 rtcpheader = (unsigned int *)bdata;
2388 rtcpheader[1] = htonl(rtp->ssrc); /* Our SSRC */
2389 rtcpheader[2] = htonl(now_msw); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970*/
2390 rtcpheader[3] = htonl(now_lsw); /* now, LSW */
2391 rtcpheader[4] = htonl(rtp->lastts); /* FIXME shouldn't be that, it should be now */
2392 rtcpheader[5] = htonl(rtp->txcount); /* No. packets sent */
2393 rtcpheader[6] = htonl(rtp->txoctetcount); /* No. bytes sent */
2394 len += 28;
2395
2396 extended = rtp->cycles + rtp->lastrxseqno;
2397 expected = extended - rtp->seedrxseqno + 1;
2398 if (rtp->rxcount > expected)
2399 expected += rtp->rxcount - expected;
2400 lost = expected - rtp->rxcount;
2401 expected_interval = expected - rtp->rtcp->expected_prior;
2402 rtp->rtcp->expected_prior = expected;
2403 received_interval = rtp->rxcount - rtp->rtcp->received_prior;
2404 rtp->rtcp->received_prior = rtp->rxcount;
2405 lost_interval = expected_interval - received_interval;
2406 if (expected_interval == 0 || lost_interval <= 0)
2407 fraction = 0;
2408 else
2409 fraction = (lost_interval << 8) / expected_interval;
2410 timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
2411 rtcpheader[7] = htonl(rtp->themssrc);
2412 rtcpheader[8] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
2413 rtcpheader[9] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
2414 rtcpheader[10] = htonl((unsigned int)(rtp->rxjitter * 65536.));
2415 rtcpheader[11] = htonl(rtp->rtcp->themrxlsr);
2416 rtcpheader[12] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
2417 len += 24;
2418
2419 rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SR << 16) | ((len/4)-1));
2420
2421 if (rtp->rtcp->sendfur) {
2422 rtcpheader[13] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1);
2423 rtcpheader[14] = htonl(rtp->ssrc); /* Our SSRC */
2424 len += 8;
2425 rtp->rtcp->sendfur = 0;
2426 }
2427
2428 /* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
2429 /* it can change mid call, and SDES can't) */
2430 rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
2431 rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
2432 rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
2433 len += 12;
2434
2435 res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
2436 if (res < 0) {
2437 ast_log(LOG_ERROR, "RTCP SR transmission error to %s:%d, rtcp halted %s\n",ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port), strerror(errno));
2438 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
2439 return 0;
2440 }
2441
2442 /* FIXME Don't need to get a new one */
2443 gettimeofday(&rtp->rtcp->txlsr, NULL);
2444 rtp->rtcp->sr_count++;
2445
2446 rtp->rtcp->lastsrtxcount = rtp->txcount;
2447
2448 if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
2449 ast_verbose("* Sent RTCP SR to %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
2450 ast_verbose(" Our SSRC: %u\n", rtp->ssrc);
2451 ast_verbose(" Sent(NTP): %u.%010u\n", (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096);
2452 ast_verbose(" Sent(RTP): %u\n", rtp->lastts);
2453 ast_verbose(" Sent packets: %u\n", rtp->txcount);
2454 ast_verbose(" Sent octets: %u\n", rtp->txoctetcount);
2455 ast_verbose(" Report block:\n");
2456 ast_verbose(" Fraction lost: %u\n", fraction);
2457 ast_verbose(" Cumulative loss: %u\n", lost);
2458 ast_verbose(" IA jitter: %.4f\n", rtp->rxjitter);
2459 ast_verbose(" Their last SR: %u\n", rtp->rtcp->themrxlsr);
2460 ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(ntohl(rtcpheader[12])/65536.0));
2461 }
2462 return res;
2463 }
2464
2465 /*! \brief Send RTCP recepient's report */
2466 static int ast_rtcp_write_rr(const void *data)
2467 {
2468 struct ast_rtp *rtp = (struct ast_rtp *)data;
2469 int res;
2470 int len = 32;
2471 unsigned int lost;
2472 unsigned int extended;
2473 unsigned int expected;
2474 unsigned int expected_interval;
2475 unsigned int received_interval;
2476 int lost_interval;
2477 struct timeval now;
2478 unsigned int *rtcpheader;
2479 char bdata[1024];
2480 struct timeval dlsr;
2481 int fraction;
2482
2483 if (!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
2484 return 0;
2485
2486 if (!rtp->rtcp->them.sin_addr.s_addr) {
2487 ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted\n");
2488 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
2489 return 0;
2490 }
2491
2492 extended = rtp->cycles + rtp->lastrxseqno;
2493 expected = extended - rtp->seedrxseqno + 1;
2494 lost = expected - rtp->rxcount;
2495 expected_interval = expected - rtp->rtcp->expected_prior;
2496 rtp->rtcp->expected_prior = expected;
2497 received_interval = rtp->rxcount - rtp->rtcp->received_prior;
2498 rtp->rtcp->received_prior = rtp->rxcount;
2499 lost_interval = expected_interval - received_interval;
2500 if (expected_interval == 0 || lost_interval <= 0)
2501 fraction = 0;
2502 else
2503 fraction = (lost_interval << 8) / expected_interval;
2504 gettimeofday(&now, NULL);
2505 timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
2506 rtcpheader = (unsigned int *)bdata;
2507 rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_RR << 16) | ((len/4)-1));
2508 rtcpheader[1] = htonl(rtp->ssrc);
2509 rtcpheader[2] = htonl(rtp->themssrc);
2510 rtcpheader[3] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
2511 rtcpheader[4] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
2512 rtcpheader[5] = htonl((unsigned int)(rtp->rxjitter * 65536.));
2513 rtcpheader[6] = htonl(rtp->rtcp->themrxlsr);
2514 rtcpheader[7] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
2515
2516 if (rtp->rtcp->sendfur) {
2517 rtcpheader[8] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1); /* Header from page 36 in RFC 3550 */
2518 rtcpheader[9] = htonl(rtp->ssrc); /* Our SSRC */
2519 len += 8;
2520 rtp->rtcp->sendfur = 0;
2521 }
2522
2523 /*! \note Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos
2524 it can change mid call, and SDES can't) */
2525 rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
2526 rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
2527 rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
2528 len += 12;
2529
2530 res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
2531
2532 if (res < 0) {
2533 ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted: %s\n",strerror(errno));
2534 /* Remove the scheduler */
2535 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
2536 return 0;
2537 }
2538
2539 rtp->rtcp->rr_count++;
2540
2541 if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
2542 ast_verbose("\n* Sending RTCP RR to %s:%d\n"
2543 " Our SSRC: %u\nTheir SSRC: %u\niFraction lost: %d\nCumulative loss: %u\n"
2544 " IA jitter: %.4f\n"
2545 " Their last SR: %u\n"
2546 " DLSR: %4.4f (sec)\n\n",
2547 ast_inet_ntoa(rtp->rtcp->them.sin_addr),
2548 ntohs(rtp->rtcp->them.sin_port),
2549 rtp->ssrc, rtp->themssrc, fraction, lost,
2550 rtp->rxjitter,
2551 rtp->rtcp->themrxlsr,
2552 (double)(ntohl(rtcpheader[7])/65536.0));
2553 }
2554
2555 return res;
2556 }
2557
2558 /*! \brief Write and RTCP packet to the far end
2559 * \note Decide if we are going to send an SR (with Reception Block) or RR
2560 * RR is sent if we have not sent any rtp packets in the previous interval */
2561 static int ast_rtcp_write(const void *data)
2562 {
2563 struct ast_rtp *rtp = (struct ast_rtp *)data;
2564 int res;
2565
2566 if (!rtp || !rtp->rtcp)
2567 return 0;
2568
2569 if (rtp->txcount > rtp->rtcp->lastsrtxcount)
2570 res = ast_rtcp_write_sr(data);
2571 else
2572 res = ast_rtcp_write_rr(data);
2573
2574 return res;
2575 }
2576
2577 /*! \brief generate comfort noice (CNG) */
2578 int ast_rtp_sendcng(struct ast_rtp *rtp, int level)
2579 {
2580 unsigned int *rtpheader;
2581 int hdrlen = 12;
2582 int res;
2583 int payload;
2584 char data[256];
2585 level = 127 - (level & 0x7f);
2586 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
2587
2588 /* If we have no peer, return immediately */
2589 if (!rtp->them.sin_addr.s_addr)
2590 return 0;
2591
2592 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
2593
2594 /* Get a pointer to the header */
2595 rtpheader = (unsigned int *)data;
2596 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
2597 rtpheader[1] = htonl(rtp->lastts);
2598 rtpheader[2] = htonl(rtp->ssrc);
2599 data[12] = level;
2600 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
2601 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
2602 if (res <0)
2603 ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s:%d: %s\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
2604 if (rtp_debug_test_addr(&rtp->them))
2605 ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
2606 , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);
2607
2608 }
2609 return 0;
2610 }
2611
2612 static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
2613 {
2614 unsigned char *rtpheader;
2615 int hdrlen = 12;
2616 int res;
2617 unsigned int ms;
2618 int pred;
2619 int mark = 0;
2620
2621 ms = calc_txstamp(rtp, &f->delivery);
2622 /* Default prediction */
2623 if (f->frametype == AST_FRAME_VOICE) {
2624 pred = rtp->lastts + f->samples;
2625
2626 /* Re-calculate last TS */
2627 rtp->lastts = rtp->lastts + ms * 8;
2628 if (ast_tvzero(f->delivery)) {
2629 /* If this isn't an absolute delivery time, Check if it is close to our prediction,
2630 and if so, go with our prediction */
2631 if (abs(rtp->lastts - pred) < MAX_TIMESTAMP_SKEW)
2632 rtp->lastts = pred;
2633 else {
2634 if (option_debug > 2)
2635 ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
2636 mark = 1;
2637 }
2638 }
2639 } else if (f->frametype == AST_FRAME_VIDEO) {
2640 mark = f->subclass & 0x1;
2641 pred = rtp->lastovidtimestamp + f->samples;
2642 /* Re-calculate last TS */
2643 rtp->lastts = rtp->lastts + ms * 90;
2644 /* If it's close to our prediction, go for it */
2645 if (ast_tvzero(f->delivery)) {
2646 if (abs(rtp->lastts - pred) < 7200) {
2647 rtp->lastts = pred;
2648 rtp->lastovidtimestamp += f->samples;
2649 } else {
2650 if (option_debug > 2)
2651 ast_log(LOG_DEBUG, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, f->samples);
2652 rtp->lastovidtimestamp = rtp->lastts;
2653 }
2654 }
2655 }
2656
2657 /* If we have been explicitly told to set the marker bit do so */
2658 if (rtp->set_marker_bit) {
2659 mark = 1;
2660 rtp->set_marker_bit = 0;
2661 }
2662
2663 /* If the timestamp for non-digit packets has moved beyond the timestamp
2664 for digits, update the digit timestamp.
2665 */
2666 if (rtp->lastts > rtp->lastdigitts)
2667 rtp->lastdigitts = rtp->lastts;
2668
2669 if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO))
2670 rtp->lastts = f->ts * 8;
2671
2672 /* Get a pointer to the header */
2673 rtpheader = (unsigned char *)(f->data - hdrlen);
2674
2675 put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (rtp->seqno) | (mark << 23)));
2676 put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
2677 put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc));
2678
2679 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
2680 res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
2681 if (res <0) {
2682 if (!rtp->nat || (rtp->nat && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
2683 ast_log(LOG_DEBUG, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
2684 } else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
2685 /* Only give this error message once if we are not RTP debugging */
2686 if (option_debug || rtpdebug)
2687 ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
2688 ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
2689 }
2690 } else {
2691 rtp->txcount++;
2692 rtp->txoctetcount +=(res - hdrlen);
2693
2694 if (rtp->rtcp && rtp->rtcp->schedid < 1)
2695 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
2696 }
2697
2698 if (rtp_debug_test_addr(&rtp->them))
2699 ast_verbose("Sent RTP packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
2700 ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), codec, rtp->seqno, rtp->lastts,res - hdrlen);
2701 }
2702
2703 rtp->seqno++;
2704
2705 return 0;
2706 }
2707
2708 int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs)
2709 {
2710 int x;
2711 for (x = 0; x < 32; x++) { /* Ugly way */
2712 rtp->pref.order[x] = prefs->order[x];
2713 rtp->pref.framing[x] = prefs->framing[x];
2714 }
2715 if (rtp->smoother)
2716 ast_smoother_free(rtp->smoother);
2717 rtp->smoother = NULL;
2718 return 0;
2719 }
2720
2721 struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp)
2722 {
2723 return &rtp->pref;
2724 }
2725
2726 int ast_rtp_codec_getformat(int pt)
2727 {
2728 if (pt < 0 || pt > MAX_RTP_PT)
2729 return 0; /* bogus payload type */
2730
2731 if (static_RTP_PT[pt].isAstFormat)
2732 return static_RTP_PT[pt].code;
2733 else
2734 return 0;
2735 }
2736
2737 int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
2738 {
2739 struct ast_frame *f;
2740 int codec;
2741 int hdrlen = 12;
2742 int subclass;
2743
2744
2745 /* If we have no peer, return immediately */
2746 if (!rtp->them.sin_addr.s_addr)
2747 return 0;
2748
2749 /* If there is no data length, return immediately */
2750 if (!_f->datalen)
2751 return 0;
2752
2753 /* Make sure we have enough space for RTP header */
2754 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
2755 ast_log(LOG_WARNING, "RTP can only send voice and video\n");
2756 return -1;
2757 }
2758
2759 subclass = _f->subclass;
2760 if (_f->frametype == AST_FRAME_VIDEO)
2761 subclass &= ~0x1;
2762
2763 codec = ast_rtp_lookup_code(rtp, 1, subclass);
2764 if (codec < 0) {
2765 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
2766 return -1;
2767 }
2768
2769 if (rtp->lasttxformat != subclass) {
2770 /* New format, reset the smoother */
2771 if (option_debug)
2772 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
2773 rtp->lasttxformat = subclass;
2774 if (rtp->smoother)
2775 ast_smoother_free(rtp->smoother);
2776 rtp->smoother = NULL;
2777 }
2778
2779 if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) {
2780 struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
2781 if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
2782 if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
2783 ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
2784 return -1;
2785 }
2786 if (fmt.flags)
2787 ast_smoother_set_flags(rtp->smoother, fmt.flags);
2788 if (option_debug)
2789 ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
2790 }
2791 }
2792 if (rtp->smoother) {
2793 if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
2794 ast_smoother_feed_be(rtp->smoother, _f);
2795 } else {
2796 ast_smoother_feed(rtp->smoother, _f);
2797 }
2798
2799 while((f = ast_smoother_read(rtp->smoother)) && (f->data))
2800 ast_rtp_raw_write(rtp, f, codec);
2801 } else {
2802 /* Don't buffer outgoing frames; send them one-per-packet: */
2803 if (_f->offset < hdrlen) {
2804 f = ast_frdup(_f);
2805 } else {
2806 f = _f;
2807 }
2808 if (f->data)
2809 ast_rtp_raw_write(rtp, f, codec);
2810 if (f != _f)
2811 ast_frfree(f);
2812 }
2813
2814 return 0;
2815 }
2816
2817 /*! \brief Unregister interface to channel driver */
2818 void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
2819 {
2820 AST_LIST_LOCK(&protos);
2821 AST_LIST_REMOVE(&protos, proto, list);
2822 AST_LIST_UNLOCK(&protos);
2823 }
2824
2825 /*! \brief Register interface to channel driver */
2826 int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
2827 {
2828 struct ast_rtp_protocol *cur;
2829
2830 AST_LIST_LOCK(&protos);
2831 AST_LIST_TRAVERSE(&protos, cur, list) {
2832 if (!strcmp(cur->type, proto->type)) {
2833 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
2834 AST_LIST_UNLOCK(&protos);
2835 return -1;
2836 }
2837 }
2838 AST_LIST_INSERT_HEAD(&protos, proto, list);
2839 AST_LIST_UNLOCK(&protos);
2840
2841 return 0;
2842 }
2843
2844 /*! \brief Bridge loop for true native bridge (reinvite) */
2845 static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, struct ast_rtp_protocol *pr0, struct ast_rtp_protocol *pr1, int codec0, int codec1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
2846 {
2847 struct ast_frame *fr = NULL;
2848 struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
2849 int oldcodec0 = codec0, oldcodec1 = codec1;
2850 struct sockaddr_in ac1 = {0,}, vac1 = {0,}, ac0 = {0,}, vac0 = {0,};
2851 struct sockaddr_in t1 = {0,}, vt1 = {0,}, t0 = {0,}, vt0 = {0,};
2852
2853 /* Set it up so audio goes directly between the two endpoints */
2854
2855 /* Test the first channel */
2856 if (!(pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))) {
2857 ast_rtp_get_peer(p1, &ac1);
2858 if (vp1)
2859 ast_rtp_get_peer(vp1, &vac1);
2860 } else
2861 ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
2862
2863 /* Test the second channel */
2864 if (!(pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))) {
2865 ast_rtp_get_peer(p0, &ac0);
2866 if (vp0)
2867 ast_rtp_get_peer(vp0, &vac0);
2868 } else
2869 ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
2870
2871 /* Now we can unlock and move into our loop */
2872 ast_channel_unlock(c0);
2873 ast_channel_unlock(c1);
2874
2875 /* Throw our channels into the structure and enter the loop */
2876 cs[0] = c0;
2877 cs[1] = c1;
2878 cs[2] = NULL;
2879 for (;;) {
2880 /* Check if anything changed */
2881 if ((c0->tech_pvt != pvt0) ||
2882 (c1->tech_pvt != pvt1) ||
2883 (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
2884 (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
2885 ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
2886 if (c0->tech_pvt == pvt0)
2887 if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
2888 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
2889 if (c1->tech_pvt == pvt1)
2890 if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
2891 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
2892 return AST_BRIDGE_RETRY;
2893 }
2894
2895 /* Check if they have changed their address */
2896 ast_rtp_get_peer(p1, &t1);
2897 if (vp1)
2898 ast_rtp_get_peer(vp1, &vt1);
2899 if (pr1->get_codec)
2900 codec1 = pr1->get_codec(c1);
2901 ast_rtp_get_peer(p0, &t0);
2902 if (vp0)
2903 ast_rtp_get_peer(vp0, &vt0);
2904 if (pr0->get_codec)
2905 codec0 = pr0->get_codec(c0);
2906 if ((inaddrcmp(&t1, &ac1)) ||
2907 (vp1 && inaddrcmp(&vt1, &vac1)) ||
2908 (codec1 != oldcodec1)) {
2909 if (option_debug > 1) {
2910 ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
2911 c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
2912 ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
2913 c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
2914 ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
2915 c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
2916 ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
2917 c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
2918 }
2919 if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
2920 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
2921 memcpy(&ac1, &t1, sizeof(ac1));
2922 memcpy(&vac1, &vt1, sizeof(vac1));
2923 oldcodec1 = codec1;
2924 }
2925 if ((inaddrcmp(&t0, &ac0)) ||
2926 (vp0 && inaddrcmp(&vt0, &vac0))) {
2927 if (option_debug > 1) {
2928 ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
2929 c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
2930 ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
2931 c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
2932 }
2933 if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
2934 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
2935 memcpy(&ac0, &t0, sizeof(ac0));
2936 memcpy(&vac0, &vt0, sizeof(vac0));
2937 oldcodec0 = codec0;
2938 }
2939
2940 /* Wait for frame to come in on the channels */
2941 if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
2942 if (!timeoutms) {
2943 if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
2944 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
2945 if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
2946 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
2947 return AST_BRIDGE_RETRY;
2948 }
2949 if (option_debug)
2950 ast_log(LOG_DEBUG, "Ooh, empty read...\n");
2951 if (ast_check_hangup(c0) || ast_check_hangup(c1))
2952 break;
2953 continue;
2954 }
2955 fr = ast_read(who);
2956 other = (who == c0) ? c1 : c0;
2957 if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
2958 (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
2959 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
2960 /* Break out of bridge */
2961 *fo = fr;
2962 *rc = who;
2963 if (option_debug)
2964 ast_log(LOG_DEBUG, "Oooh, got a %s\n", fr ? "digit" : "hangup");
2965 if (c0->tech_pvt == pvt0)
2966 if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
2967 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
2968 if (c1->tech_pvt == pvt1)
2969 if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
2970 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
2971 return AST_BRIDGE_COMPLETE;
2972 } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
2973 if ((fr->subclass == AST_CONTROL_HOLD) ||
2974 (fr->subclass == AST_CONTROL_UNHOLD) ||
2975 (fr->subclass == AST_CONTROL_VIDUPDATE) ||
2976 (fr->subclass == AST_CONTROL_SRCUPDATE)) {
2977 if (fr->subclass == AST_CONTROL_HOLD) {
2978 /* If we someone went on hold we want the other side to reinvite back to us */
2979 if (who == c0)
2980 pr1->set_rtp_peer(c1, NULL, NULL, 0, 0);
2981 else
2982 pr0->set_rtp_peer(c0, NULL, NULL, 0, 0);
2983 } else if (fr->subclass == AST_CONTROL_UNHOLD) {
2984 /* If they went off hold they should go back to being direct */
2985 if (who == c0)
2986 pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE));
2987 else
2988 pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE));
2989 }
2990 /* Update local address information */
2991 ast_rtp_get_peer(p0, &t0);
2992 memcpy(&ac0, &t0, sizeof(ac0));
2993 ast_rtp_get_peer(p1, &t1);
2994 memcpy(&ac1, &t1, sizeof(ac1));
2995 /* Update codec information */
2996 if (pr0->get_codec && c0->tech_pvt)
2997 oldcodec0 = codec0 = pr0->get_codec(c0);
2998 if (pr1->get_codec && c1->tech_pvt)
2999 oldcodec1 = codec1 = pr1->get_codec(c1);
3000 ast_indicate_data(other, fr->subclass, fr->data, fr->datalen);
3001 ast_frfree(fr);
3002 } else {
3003 *fo = fr;
3004 *rc = who;
3005 ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
3006 return AST_BRIDGE_COMPLETE;
3007 }
3008 } else {
3009 if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
3010 (fr->frametype == AST_FRAME_DTMF_END) ||
3011 (fr->frametype == AST_FRAME_VOICE) ||
3012 (fr->frametype == AST_FRAME_VIDEO) ||
3013 (fr->frametype == AST_FRAME_IMAGE) ||
3014 (fr->frametype == AST_FRAME_HTML) ||
3015 (fr->frametype == AST_FRAME_MODEM) ||
3016 (fr->frametype == AST_FRAME_TEXT)) {
3017 ast_write(other, fr);
3018 }
3019 ast_frfree(fr);
3020 }
3021 /* Swap priority */
3022 cs[2] = cs[0];
3023 cs[0] = cs[1];
3024 cs[1] = cs[2];
3025 }
3026
3027 if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
3028 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
3029 if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
3030 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
3031
3032 return AST_BRIDGE_FAILED;
3033 }
3034
3035 /*! \brief P2P RTP Callback */
3036 #ifdef P2P_INTENSE
3037 static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata)
3038 {
3039 int res = 0, hdrlen = 12;
3040 struct sockaddr_in sin;
3041 socklen_t len;
3042 unsigned int *header;
3043 struct ast_rtp *rtp = cbdata, *bridged = NULL;
3044
3045 if (!rtp)
3046 return 1;
3047
3048 len = sizeof(sin);
3049 if ((res = recvfrom(fd, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 0, (struct sockaddr *)&sin, &len)) < 0)
3050 return 1;
3051
3052 header = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
3053
3054 /* If NAT support is turned on, then see if we need to change their address */
3055 if ((rtp->nat) &&
3056 ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
3057 (rtp->them.sin_port != sin.sin_port))) {
3058 rtp->them = sin;
3059 rtp->rxseqno = 0;
3060 ast_set_flag(rtp, FLAG_NAT_ACTIVE);
3061 if (option_debug || rtpdebug)
3062 ast_log(LOG_DEBUG, "P2P RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
3063 }
3064
3065 /* Write directly out to other RTP stream if bridged */
3066 if ((bridged = ast_rtp_get_bridged(rtp)))
3067 bridge_p2p_rtp_write(rtp, bridged, header, res, hdrlen);
3068
3069 return 1;
3070 }
3071
3072 /*! \brief Helper function to switch a channel and RTP stream into callback mode */
3073 static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int *fds, int **iod)
3074 {
3075 /* If we need DTMF, are looking for STUN, or we have no IO structure then we can't do direct callback */
3076 if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || ast_test_flag(rtp, FLAG_HAS_STUN) || !rtp->io)
3077 return 0;
3078
3079 /* If the RTP structure is already in callback mode, remove it temporarily */
3080 if (rtp->ioid) {
3081 ast_io_remove(rtp->io, rtp->ioid);
3082 rtp->ioid = NULL;
3083 }
3084
3085 /* Steal the file descriptors from the channel and stash them away */
3086 fds[0] = chan->fds[0];
3087 chan->fds[0] = -1;
3088
3089 /* Now, fire up callback mode */
3090 iod[0] = ast_io_add(rtp->io, fds[0], p2p_rtp_callback, AST_IO_IN, rtp);
3091
3092 return 1;
3093 }
3094 #else
3095 static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int *fds, int **iod)
3096 {
3097 return 0;
3098 }
3099 #endif
3100
3101 /*! \brief Helper function to switch a channel and RTP stream out of callback mode */
3102 static int p2p_callback_disable(struct ast_channel *chan, struct ast_rtp *rtp, int *fds, int **iod)
3103 {
3104 ast_channel_lock(chan);
3105
3106 /* Remove the callback from the IO context */
3107 ast_io_remove(rtp->io, iod[0]);
3108
3109 /* Restore file descriptors */
3110 chan->fds[0] = fds[0];
3111 ast_channel_unlock(chan);
3112
3113 /* Restore callback mode if previously used */
3114 if (ast_test_flag(rtp, FLAG_CALLBACK_MODE))
3115 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
3116
3117 return 0;
3118 }
3119
3120 /*! \brief Helper function that sets what an RTP structure is bridged to */
3121 static void p2p_set_bridge(struct ast_rtp *rtp0, struct ast_rtp *rtp1)
3122 {
3123 ast_mutex_lock(&rtp0->bridge_lock);
3124 rtp0->bridged = rtp1;
3125 ast_mutex_unlock(&rtp0->bridge_lock);
3126
3127 return;
3128 }
3129
3130 /*! \brief Bridge loop for partial native bridge (packet2packet) */
3131 static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
3132 {
3133 struct ast_frame *fr = NULL;
3134 struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
3135 int p0_fds[2] = {-1, -1}, p1_fds[2] = {-1, -1};
3136 int *p0_iod[2] = {NULL, NULL}, *p1_iod[2] = {NULL, NULL};
3137 int p0_callback = 0, p1_callback = 0;
3138 enum ast_bridge_result res = AST_BRIDGE_FAILED;
3139
3140 /* Okay, setup each RTP structure to do P2P forwarding */
3141 ast_clear_flag(p0, FLAG_P2P_SENT_MARK);
3142 p2p_set_bridge(p0, p1);
3143 ast_clear_flag(p1, FLAG_P2P_SENT_MARK);
3144 p2p_set_bridge(p1, p0);
3145
3146 /* Activate callback modes if possible */
3147 p0_callback = p2p_callback_enable(c0, p0, &p0_fds[0], &p0_iod[0]);
3148 p1_callback = p2p_callback_enable(c1, p1, &p1_fds[0], &p1_iod[0]);
3149
3150 /* Now let go of the channel locks and be on our way */
3151 ast_channel_unlock(c0);
3152 ast_channel_unlock(c1);
3153
3154 /* Go into a loop forwarding frames until we don't need to anymore */
3155 cs[0] = c0;
3156 cs[1] = c1;
3157 cs[2] = NULL;
3158 for (;;) {
3159 /* If the underlying formats have changed force this bridge to break */
3160 if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) {
3161 ast_log(LOG_DEBUG, "Oooh, formats changed, backing out\n");
3162 res = AST_BRIDGE_FAILED_NOWARN;
3163 break;
3164 }
3165 /* Check if anything changed */
3166 if ((c0->tech_pvt != pvt0) ||
3167 (c1->tech_pvt != pvt1) ||
3168 (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
3169 (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
3170 ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
3171 if ((c0->masq || c0->masqr) && (fr = ast_read(c0)))
3172 ast_frfree(fr);
3173 if ((c1->masq || c1->masqr) && (fr = ast_read(c1)))
3174 ast_frfree(fr);
3175 res = AST_BRIDGE_RETRY;
3176 break;
3177 }
3178 /* Wait on a channel to feed us a frame */
3179 if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
3180 if (!timeoutms) {
3181 res = AST_BRIDGE_RETRY;
3182 break;
3183 }
3184 if (option_debug)
3185 ast_log(LOG_NOTICE, "Ooh, empty read...\n");
3186 if (ast_check_hangup(c0) || ast_check_hangup(c1))
3187 break;
3188 continue;
3189 }
3190 /* Read in frame from channel */
3191 fr = ast_read(who);
3192 other = (who == c0) ? c1 : c0;
3193 /* Dependong on the frame we may need to break out of our bridge */
3194 if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
3195 ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
3196 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
3197 /* Record received frame and who */
3198 *fo = fr;
3199 *rc = who;
3200 if (option_debug)
3201 ast_log(LOG_DEBUG, "Oooh, got a %s\n", fr ? "digit" : "hangup");
3202 res = AST_BRIDGE_COMPLETE;
3203 break;
3204 } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
3205 if ((fr->subclass == AST_CONTROL_HOLD) ||
3206 (fr->subclass == AST_CONTROL_UNHOLD) ||
3207 (fr->subclass == AST_CONTROL_VIDUPDATE) ||
3208 (fr->subclass == AST_CONTROL_SRCUPDATE)) {
3209 /* If we are going on hold, then break callback mode and P2P bridging */
3210 if (fr->subclass == AST_CONTROL_HOLD) {
3211 if (p0_callback)
3212 p0_callback = p2p_callback_disable(c0, p0, &p0_fds[0], &p0_iod[0]);
3213 if (p1_callback)
3214 p1_callback = p2p_callback_disable(c1, p1, &p1_fds[0], &p1_iod[0]);
3215 p2p_set_bridge(p0, NULL);
3216 p2p_set_bridge(p1, NULL);
3217 } else if (fr->subclass == AST_CONTROL_UNHOLD) {
3218 /* If we are off hold, then go back to callback mode and P2P bridging */
3219 ast_clear_flag(p0, FLAG_P2P_SENT_MARK);
3220 p2p_set_bridge(p0, p1);
3221 ast_clear_flag(p1, FLAG_P2P_SENT_MARK);
3222 p2p_set_bridge(p1, p0);
3223 p0_callback = p2p_callback_enable(c0, p0, &p0_fds[0], &p0_iod[0]);
3224 p1_callback = p2p_callback_enable(c1, p1, &p1_fds[0], &p1_iod[0]);
3225 }
3226 ast_indicate_data(other, fr->subclass, fr->data, fr->datalen);
3227 ast_frfree(fr);
3228 } else {
3229 *fo = fr;
3230 *rc = who;
3231 ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
3232 res = AST_BRIDGE_COMPLETE;
3233 break;
3234 }
3235 } else {
3236 if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
3237 (fr->frametype == AST_FRAME_DTMF_END) ||
3238 (fr->frametype == AST_FRAME_VOICE) ||
3239 (fr->frametype == AST_FRAME_VIDEO) ||
3240 (fr->frametype == AST_FRAME_IMAGE) ||
3241 (fr->frametype == AST_FRAME_HTML) ||
3242 (fr->frametype == AST_FRAME_MODEM) ||
3243 (fr->frametype == AST_FRAME_TEXT)) {
3244 ast_write(other, fr);
3245 }
3246
3247 ast_frfree(fr);
3248 }
3249 /* Swap priority */
3250 cs[2] = cs[0];
3251 cs[0] = cs[1];
3252 cs[1] = cs[2];
3253 }
3254
3255 /* If we are totally avoiding the core, then restore our link to it */
3256 if (p0_callback)
3257 p0_callback = p2p_callback_disable(c0, p0, &p0_fds[0], &p0_iod[0]);
3258 if (p1_callback)
3259 p1_callback = p2p_callback_disable(c1, p1, &p1_fds[0], &p1_iod[0]);
3260
3261 /* Break out of the direct bridge */
3262 p2p_set_bridge(p0, NULL);
3263 p2p_set_bridge(p1, NULL);
3264
3265 return res;
3266 }
3267
3268 /*! \brief Bridge calls. If possible and allowed, initiate
3269 re-invite so the peers exchange media directly outside
3270 of Asterisk. */
3271 enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
3272 {
3273 struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */
3274 struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */
3275 struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
3276 enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
3277 enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
3278 enum ast_bridge_result res = AST_BRIDGE_FAILED;
3279 int codec0 = 0, codec1 = 0;
3280 void *pvt0 = NULL, *pvt1 = NULL;
3281
3282 /* Lock channels */
3283 ast_channel_lock(c0);
3284 while(ast_channel_trylock(c1)) {
3285 ast_channel_unlock(c0);
3286 usleep(1);
3287 ast_channel_lock(c0);
3288 }
3289
3290 /* Ensure neither channel got hungup during lock avoidance */
3291 if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
3292 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
3293 ast_channel_unlock(c0);
3294 ast_channel_unlock(c1);
3295 return AST_BRIDGE_FAILED;
3296 }
3297
3298 /* Find channel driver interfaces */
3299 if (!(pr0 = get_proto(c0))) {
3300 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
3301 ast_channel_unlock(c0);
3302 ast_channel_unlock(c1);
3303 return AST_BRIDGE_FAILED;
3304 }
3305 if (!(pr1 = get_proto(c1))) {
3306 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
3307 ast_channel_unlock(c0);
3308 ast_channel_unlock(c1);
3309 return AST_BRIDGE_FAILED;
3310 }
3311
3312 /* Get channel specific interface structures */
3313 pvt0 = c0->tech_pvt;
3314 pvt1 = c1->tech_pvt;
3315
3316 /* Get audio and video interface (if native bridge is possible) */
3317 audio_p0_res = pr0->get_rtp_info(c0, &p0);
3318 video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
3319 audio_p1_res = pr1->get_rtp_info(c1, &p1);
3320 video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
3321
3322 /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
3323 if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
3324 audio_p0_res = AST_RTP_GET_FAILED;
3325 if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
3326 audio_p1_res = AST_RTP_GET_FAILED;
3327
3328 /* Check if a bridge is possible (partial/native) */
3329 if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
3330 /* Somebody doesn't want to play... */
3331 ast_channel_unlock(c0);
3332 ast_channel_unlock(c1);
3333 return AST_BRIDGE_FAILED_NOWARN;
3334 }
3335
3336 /* If we need to feed DTMF frames into the core then only do a partial native bridge */
3337 if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
3338 ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
3339 audio_p0_res = AST_RTP_TRY_PARTIAL;
3340 }
3341
3342 if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
3343 ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
3344 audio_p1_res = AST_RTP_TRY_PARTIAL;
3345 }
3346
3347 /* If both sides are not using the same method of DTMF transmission
3348 * (ie: one is RFC2833, other is INFO... then we can not do direct media.
3349 * --------------------------------------------------
3350 * | DTMF Mode | HAS_DTMF | Accepts Begin Frames |
3351 * |-----------|------------|-----------------------|
3352 * | Inband | False | True |
3353 * | RFC2833 | True | True |
3354 * | SIP INFO | False | False |
3355 * --------------------------------------------------
3356 * However, if DTMF from both channels is being monitored by the core, then
3357 * we can still do packet-to-packet bridging, because passing through the
3358 * core will handle DTMF mode translation.
3359 */
3360 if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
3361 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
3362 if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
3363 ast_channel_unlock(c0);
3364 ast_channel_unlock(c1);
3365 return AST_BRIDGE_FAILED_NOWARN;
3366 }
3367 audio_p0_res = AST_RTP_TRY_PARTIAL;
3368 audio_p1_res = AST_RTP_TRY_PARTIAL;
3369 }
3370
3371 /* If the core will need to compensate and the P2P bridge will need to feed up DTMF frames then we can not reliably do so yet, so do not P2P bridge */
3372 if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF) && ast_test_flag(p0, FLAG_DTMF_COMPENSATE)) ||
3373 (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF) && ast_test_flag(p1, FLAG_DTMF_COMPENSATE))) {
3374 ast_channel_unlock(c0);
3375 ast_channel_unlock(c1);
3376 return AST_BRIDGE_FAILED_NOWARN;
3377 }
3378
3379 /* Get codecs from both sides */
3380 codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
3381 codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
3382 if (codec0 && codec1 && !(codec0 & codec1)) {
3383 /* Hey, we can't do native bridging if both parties speak different codecs */
3384 if (option_debug)
3385 ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
3386 ast_channel_unlock(c0);
3387 ast_channel_unlock(c1);
3388 return AST_BRIDGE_FAILED_NOWARN;
3389 }
3390
3391 /* If either side can only do a partial bridge, then don't try for a true native bridge */
3392 if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
3393 struct ast_format_list fmt0, fmt1;
3394
3395 /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
3396 if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
3397 if (option_debug)
3398 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n");
3399 ast_channel_unlock(c0);
3400 ast_channel_unlock(c1);
3401 return AST_BRIDGE_FAILED_NOWARN;
3402 }
3403 /* They must also be using the same packetization */
3404 fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
3405 fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
3406 if (fmt0.cur_ms != fmt1.cur_ms) {
3407 if (option_debug)
3408 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n");
3409 ast_channel_unlock(c0);
3410 ast_channel_unlock(c1);
3411 return AST_BRIDGE_FAILED_NOWARN;
3412 }
3413
3414 if (option_verbose > 2)
3415 ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
3416 res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
3417 } else {
3418 if (option_verbose > 2)
3419 ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
3420 res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
3421 }
3422
3423 return res;
3424 }
3425
3426 static int rtp_do_debug_ip(int fd, int argc, char *argv[])
3427 {
3428 struct hostent *hp;
3429 struct ast_hostent ahp;
3430 int port = 0;
3431 char *p, *arg;
3432
3433 if (argc != 4)
3434 return RESULT_SHOWUSAGE;
3435 arg = argv[3];
3436 p = strstr(arg, ":");
3437 if (p) {
3438 *p = '\0';
3439 p++;
3440 port = atoi(p);
3441 }
3442 hp = ast_gethostbyname(arg, &ahp);
3443 if (hp == NULL)
3444 return RESULT_SHOWUSAGE;
3445 rtpdebugaddr.sin_family = AF_INET;
3446 memcpy(&rtpdebugaddr.sin_addr, hp->h_addr, sizeof(rtpdebugaddr.sin_addr));
3447 rtpdebugaddr.sin_port = htons(port);
3448 if (port == 0)
3449 ast_cli(fd, "RTP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtpdebugaddr.sin_addr));
3450 else
3451 ast_cli(fd, "RTP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtpdebugaddr.sin_addr), port);
3452 rtpdebug = 1;
3453 return RESULT_SUCCESS;
3454 }
3455
3456 static int rtcp_do_debug_ip_deprecated(int fd, int argc, char *argv[])
3457 {
3458 struct hostent *hp;
3459 struct ast_hostent ahp;
3460 int port = 0;
3461 char *p, *arg;
3462 if (argc != 5)
3463 return RESULT_SHOWUSAGE;
3464
3465 arg = argv[4];
3466 p = strstr(arg, ":");
3467 if (p) {
3468 *p = '\0';
3469 p++;
3470 port = atoi(p);
3471 }
3472 hp = ast_gethostbyname(arg, &ahp);
3473 if (hp == NULL)
3474 return RESULT_SHOWUSAGE;
3475 rtcpdebugaddr.sin_family = AF_INET;
3476 memcpy(&rtcpdebugaddr.sin_addr, hp->h_addr, sizeof(rtcpdebugaddr.sin_addr));
3477 rtcpdebugaddr.sin_port = htons(port);
3478 if (port == 0)
3479 ast_cli(fd, "RTCP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr));
3480 else
3481 ast_cli(fd, "RTCP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr), port);
3482 rtcpdebug = 1;
3483 return RESULT_SUCCESS;
3484 }
3485
3486 static int rtcp_do_debug_ip(int fd, int argc, char *argv[])
3487 {
3488 struct hostent *hp;
3489 struct ast_hostent ahp;
3490 int port = 0;
3491 char *p, *arg;
3492 if (argc != 4)
3493 return RESULT_SHOWUSAGE;
3494
3495 arg = argv[3];
3496 p = strstr(arg, ":");
3497 if (p) {
3498 *p = '\0';
3499 p++;
3500 port = atoi(p);
3501 }
3502 hp = ast_gethostbyname(arg, &ahp);
3503 if (hp == NULL)
3504 return RESULT_SHOWUSAGE;
3505 rtcpdebugaddr.sin_family = AF_INET;
3506 memcpy(&rtcpdebugaddr.sin_addr, hp->h_addr, sizeof(rtcpdebugaddr.sin_addr));
3507 rtcpdebugaddr.sin_port = htons(port);
3508 if (port == 0)
3509 ast_cli(fd, "RTCP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr));
3510 else
3511 ast_cli(fd, "RTCP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr), port);
3512 rtcpdebug = 1;
3513 return RESULT_SUCCESS;
3514 }
3515
3516 static int rtp_do_debug(int fd, int argc, char *argv[])
3517 {
3518 if (argc != 2) {
3519 if (argc != 4)
3520 return RESULT_SHOWUSAGE;
3521 return rtp_do_debug_ip(fd, argc, argv);
3522 }
3523 rtpdebug = 1;
3524 memset(&rtpdebugaddr,0,sizeof(rtpdebugaddr));
3525 ast_cli(fd, "RTP Debugging Enabled\n");
3526 return RESULT_SUCCESS;
3527 }
3528
3529 static int rtcp_do_debug_deprecated(int fd, int argc, char *argv[]) {
3530 if (argc != 3) {
3531 if (argc != 5)
3532 return RESULT_SHOWUSAGE;
3533 return rtcp_do_debug_ip_deprecated(fd, argc, argv);
3534 }
3535 rtcpdebug = 1;
3536 memset(&rtcpdebugaddr,0,sizeof(rtcpdebugaddr));
3537 ast_cli(fd, "RTCP Debugging Enabled\n");
3538 return RESULT_SUCCESS;
3539 }
3540
3541 static int rtcp_do_debug(int fd, int argc, char *argv[]) {
3542 if (argc != 2) {
3543 if (argc != 4)
3544 return RESULT_SHOWUSAGE;
3545 return rtcp_do_debug_ip(fd, argc, argv);
3546 }
3547 rtcpdebug = 1;
3548 memset(&rtcpdebugaddr,0,sizeof(rtcpdebugaddr));
3549 ast_cli(fd, "RTCP Debugging Enabled\n");
3550 return RESULT_SUCCESS;
3551 }
3552
3553 static int rtcp_do_stats_deprecated(int fd, int argc, char *argv[]) {
3554 if (argc != 3) {
3555 return RESULT_SHOWUSAGE;
3556 }
3557 rtcpstats = 1;
3558 ast_cli(fd, "RTCP Stats Enabled\n");
3559 return RESULT_SUCCESS;
3560 }
3561
3562 static int rtcp_do_stats(int fd, int argc, char *argv[]) {
3563 if (argc != 2) {
3564 return RESULT_SHOWUSAGE;
3565 }
3566 rtcpstats = 1;
3567 ast_cli(fd, "RTCP Stats Enabled\n");
3568 return RESULT_SUCCESS;
3569 }
3570
3571 static int rtp_no_debug(int fd, int argc, char *argv[])
3572 {
3573 if (argc != 3)
3574 return RESULT_SHOWUSAGE;
3575 rtpdebug = 0;
3576 ast_cli(fd,"RTP Debugging Disabled\n");
3577 return RESULT_SUCCESS;
3578 }
3579
3580 static int rtcp_no_debug_deprecated(int fd, int argc, char *argv[])
3581 {
3582 if (argc != 4)
3583 return RESULT_SHOWUSAGE;
3584 rtcpdebug = 0;
3585 ast_cli(fd,"RTCP Debugging Disabled\n");
3586 return RESULT_SUCCESS;
3587 }
3588
3589 static int rtcp_no_debug(int fd, int argc, char *argv[])
3590 {
3591 if (argc != 3)
3592 return RESULT_SHOWUSAGE;
3593 rtcpdebug = 0;
3594 ast_cli(fd,"RTCP Debugging Disabled\n");
3595 return RESULT_SUCCESS;
3596 }
3597
3598 static int rtcp_no_stats_deprecated(int fd, int argc, char *argv[])
3599 {
3600 if (argc != 4)
3601 return RESULT_SHOWUSAGE;
3602 rtcpstats = 0;
3603 ast_cli(fd,"RTCP Stats Disabled\n");
3604 return RESULT_SUCCESS;
3605 }
3606
3607 static int rtcp_no_stats(int fd, int argc, char *argv[])
3608 {
3609 if (argc != 3)
3610 return RESULT_SHOWUSAGE;
3611 rtcpstats = 0;
3612 ast_cli(fd,"RTCP Stats Disabled\n");
3613 return RESULT_SUCCESS;
3614 }
3615
3616 static int stun_do_debug(int fd, int argc, char *argv[])
3617 {
3618 if (argc != 2) {
3619 return RESULT_SHOWUSAGE;
3620 }
3621 stundebug = 1;
3622 ast_cli(fd, "STUN Debugging Enabled\n");
3623 return RESULT_SUCCESS;
3624 }
3625
3626 static int stun_no_debug(int fd, int argc, char *argv[])
3627 {
3628 if (argc != 3)
3629 return RESULT_SHOWUSAGE;
3630 stundebug = 0;
3631 ast_cli(fd, "STUN Debugging Disabled\n");
3632 return RESULT_SUCCESS;
3633 }
3634
3635 static char debug_usage[] =
3636 "Usage: rtp debug [ip host[:port]]\n"
3637 " Enable dumping of all RTP packets to and from host.\n";
3638
3639 static char no_debug_usage[] =
3640 "Usage: rtp debug off\n"
3641 " Disable all RTP debugging\n";
3642
3643 static char stun_debug_usage[] =
3644 "Usage: stun debug\n"
3645 " Enable STUN (Simple Traversal of UDP through NATs) debugging\n";
3646
3647 static char stun_no_debug_usage[] =
3648 "Usage: stun debug off\n"
3649 " Disable STUN debugging\n";
3650
3651 static char rtcp_debug_usage[] =
3652 "Usage: rtcp debug [ip host[:port]]\n"
3653 " Enable dumping of all RTCP packets to and from host.\n";
3654
3655 static char rtcp_no_debug_usage[] =
3656 "Usage: rtcp debug off\n"
3657 " Disable all RTCP debugging\n";
3658
3659 static char rtcp_stats_usage[] =
3660 "Usage: rtcp stats\n"
3661 " Enable dumping of RTCP stats.\n";
3662
3663 static char rtcp_no_stats_usage[] =
3664 "Usage: rtcp stats off\n"
3665 " Disable all RTCP stats\n";
3666
3667 static struct ast_cli_entry cli_rtp_no_debug_deprecated = {
3668 { "rtp", "no", "debug", NULL },
3669 rtp_no_debug, NULL,
3670 NULL };
3671
3672 static struct ast_cli_entry cli_rtp_rtcp_debug_ip_deprecated = {
3673 { "rtp", "rtcp", "debug", "ip", NULL },
3674 rtcp_do_debug_deprecated, NULL,
3675 NULL };
3676
3677 static struct ast_cli_entry cli_rtp_rtcp_debug_deprecated = {
3678 { "rtp", "rtcp", "debug", NULL },
3679 rtcp_do_debug_deprecated, NULL,
3680 NULL };
3681
3682 static struct ast_cli_entry cli_rtp_rtcp_no_debug_deprecated = {
3683 { "rtp", "rtcp", "no", "debug", NULL },
3684 rtcp_no_debug_deprecated, NULL,
3685 NULL };
3686
3687 static struct ast_cli_entry cli_rtp_rtcp_stats_deprecated = {
3688 { "rtp", "rtcp", "stats", NULL },
3689 rtcp_do_stats_deprecated, NULL,
3690 NULL };
3691
3692 static struct ast_cli_entry cli_rtp_rtcp_no_stats_deprecated = {
3693 { "rtp", "rtcp", "no", "stats", NULL },
3694 rtcp_no_stats_deprecated, NULL,
3695 NULL };
3696
3697 static struct ast_cli_entry cli_stun_no_debug_deprecated = {
3698 { "stun", "no", "debug", NULL },
3699 stun_no_debug, NULL,
3700 NULL };
3701
3702 static struct ast_cli_entry cli_rtp[] = {
3703 { { "rtp", "debug", "ip", NULL },
3704 rtp_do_debug, "Enable RTP debugging on IP",
3705 debug_usage },
3706
3707 { { "rtp", "debug", NULL },
3708 rtp_do_debug, "Enable RTP debugging",
3709 debug_usage },
3710
3711 { { "rtp", "debug", "off", NULL },
3712 rtp_no_debug, "Disable RTP debugging",
3713 no_debug_usage, NULL, &cli_rtp_no_debug_deprecated },
3714
3715 { { "rtcp", "debug", "ip", NULL },
3716 rtcp_do_debug, "Enable RTCP debugging on IP",
3717 rtcp_debug_usage, NULL, &cli_rtp_rtcp_debug_ip_deprecated },
3718
3719 { { "rtcp", "debug", NULL },
3720 rtcp_do_debug, "Enable RTCP debugging",
3721 rtcp_debug_usage, NULL, &cli_rtp_rtcp_debug_deprecated },
3722
3723 { { "rtcp", "debug", "off", NULL },
3724 rtcp_no_debug, "Disable RTCP debugging",
3725 rtcp_no_debug_usage, NULL, &cli_rtp_rtcp_no_debug_deprecated },
3726
3727 { { "rtcp", "stats", NULL },
3728 rtcp_do_stats, "Enable RTCP stats",
3729 rtcp_stats_usage, NULL, &cli_rtp_rtcp_stats_deprecated },
3730
3731 { { "rtcp", "stats", "off", NULL },
3732 rtcp_no_stats, "Disable RTCP stats",
3733 rtcp_no_stats_usage, NULL, &cli_rtp_rtcp_no_stats_deprecated },
3734
3735 { { "stun", "debug", NULL },
3736 stun_do_debug, "Enable STUN debugging",
3737 stun_debug_usage },
3738
3739 { { "stun", "debug", "off", NULL },
3740 stun_no_debug, "Disable STUN debugging",
3741 stun_no_debug_usage, NULL, &cli_stun_no_debug_deprecated },
3742 };
3743
3744 int ast_rtp_reload(void)
3745 {
3746 struct ast_config *cfg;
3747 const char *s;
3748
3749 rtpstart = 5000;
3750 rtpend = 31000;
3751 dtmftimeout = DEFAULT_DTMF_TIMEOUT;
3752 cfg = ast_config_load("rtp.conf");
3753 if (cfg) {
3754 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
3755 rtpstart = atoi(s);
3756 if (rtpstart < 1024)
3757 rtpstart = 1024;
3758 if (rtpstart > 65535)
3759 rtpstart = 65535;
3760 }
3761 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
3762 rtpend = atoi(s);
3763 if (rtpend < 1024)
3764 rtpend = 1024;
3765 if (rtpend > 65535)
3766 rtpend = 65535;
3767 }
3768 if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
3769 rtcpinterval = atoi(s);
3770 if (rtcpinterval == 0)
3771 rtcpinterval = 0; /* Just so we're clear... it's zero */
3772 if (rtcpinterval < RTCP_MIN_INTERVALMS)
3773 rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
3774 if (rtcpinterval > RTCP_MAX_INTERVALMS)
3775 rtcpinterval = RTCP_MAX_INTERVALMS;
3776 }
3777 if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
3778 #ifdef SO_NO_CHECK
3779 if (ast_false(s))
3780 nochecksums = 1;
3781 else
3782 nochecksums = 0;
3783 #else
3784 if (ast_false(s))
3785 ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
3786 #endif
3787 }
3788 if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
3789 dtmftimeout = atoi(s);
3790 if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
3791 ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
3792 dtmftimeout, DEFAULT_DTMF_TIMEOUT);
3793 dtmftimeout = DEFAULT_DTMF_TIMEOUT;
3794 };
3795 }
3796 ast_config_destroy(cfg);
3797 }
3798 if (rtpstart >= rtpend) {
3799 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
3800 rtpstart = 5000;
3801 rtpend = 31000;
3802 }
3803 if (option_verbose > 1)
3804 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
3805 return 0;
3806 }
3807
3808 /*! \brief Initialize the RTP system in Asterisk */
3809 void ast_rtp_init(void)
3810 {
3811 ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
3812 ast_rtp_reload();
3813 }
3814
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