#include <asterisk/frame.h>
#include <asterisk/io.h>
#include <asterisk/sched.h>
#include <asterisk/channel.h>
#include <netinet/in.h>
Go to the source code of this file.
Data Structures | |
struct | ast_rtp_protocol |
Defines | |
#define | AST_RTP_DTMF (1 << 0) |
#define | AST_RTP_CN (1 << 1) |
#define | AST_RTP_CISCO_DTMF (1 << 2) |
#define | AST_RTP_MAX AST_RTP_CISCO_DTMF |
Typedefs | |
typedef int(* | ast_rtp_callback )(struct ast_rtp *rtp, struct ast_frame *f, void *data) |
Functions | |
ast_rtp * | ast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode) |
void | ast_rtp_set_peer (struct ast_rtp *rtp, struct sockaddr_in *them) |
void | ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them) |
void | ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us) |
void | ast_rtp_destroy (struct ast_rtp *rtp) |
void | ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback) |
void | ast_rtp_set_data (struct ast_rtp *rtp, void *data) |
int | ast_rtp_write (struct ast_rtp *rtp, struct ast_frame *f) |
ast_frame * | ast_rtp_read (struct ast_rtp *rtp) |
ast_frame * | ast_rtcp_read (struct ast_rtp *rtp) |
int | ast_rtp_fd (struct ast_rtp *rtp) |
int | ast_rtcp_fd (struct ast_rtp *rtp) |
int | ast_rtp_senddigit (struct ast_rtp *rtp, char digit) |
int | ast_rtp_settos (struct ast_rtp *rtp, int tos) |
void | ast_rtp_pt_clear (struct ast_rtp *rtp) |
void | ast_rtp_pt_default (struct ast_rtp *rtp) |
void | ast_rtp_set_m_type (struct ast_rtp *rtp, int pt) |
void | ast_rtp_set_rtpmap_type (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype) |
rtpPayloadType | ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt) |
int | ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code) |
void | ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats) |
char * | ast_rtp_lookup_mime_subtype (int isAstFormat, int code) |
void | ast_rtp_setnat (struct ast_rtp *rtp, int nat) |
int | ast_rtp_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc) |
int | ast_rtp_proto_register (struct ast_rtp_protocol *proto) |
void | ast_rtp_proto_unregister (struct ast_rtp_protocol *proto) |
void | ast_rtp_stop (struct ast_rtp *rtp) |
void | ast_rtp_init (void) |
void | ast_rtp_reload (void) |
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DTMF (Cisco Proprietary) Definition at line 34 of file rtp.h. Referenced by ast_rtp_read(). |
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'Comfort Noise' (RFC3389) Definition at line 32 of file rtp.h. Referenced by ast_rtp_read(). |
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DTMF (RFC2833) Definition at line 30 of file rtp.h. Referenced by ast_rtp_read(). |
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Maximum RTP-specific code |
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Definition at line 48 of file rtp.h. Referenced by ast_rtp_set_callback(). |
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Definition at line 105 of file rtp.c. References ast_rtp::rtcp, and ast_rtcp::s.
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Definition at line 309 of file rtp.c. References AST_FRAME_NULL, ast_log(), CRASH, LOG_DEBUG, LOG_WARNING, ast_rtp::nat, option_debug, ast_rtp::rtcp, ast_rtcp::s, ast_rtp::them, and ast_rtcp::them.
00310 { 00311 static struct ast_frame null_frame = { AST_FRAME_NULL, }; 00312 int len; 00313 int hdrlen = 8; 00314 int res; 00315 struct sockaddr_in sin; 00316 unsigned int rtcpdata[1024]; 00317 00318 if (!rtp->rtcp) 00319 return &null_frame; 00320 00321 len = sizeof(sin); 00322 00323 res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata), 00324 0, (struct sockaddr *)&sin, &len); 00325 00326 if (res < 0) { 00327 ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno)); 00328 if (errno == EBADF) 00329 CRASH; 00330 return &null_frame; 00331 } 00332 00333 if (res < hdrlen) { 00334 ast_log(LOG_WARNING, "RTP Read too short\n"); 00335 return &null_frame; 00336 } 00337 00338 if (rtp->nat) { 00339 /* Send to whoever sent to us */ 00340 if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 00341 (rtp->rtcp->them.sin_port != sin.sin_port)) { 00342 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 00343 ast_log(LOG_DEBUG, "RTP NAT: Using address %s:%d\n", inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00344 } 00345 } 00346 if (option_debug) 00347 ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res); 00348 return &null_frame; 00349 } |
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Definition at line 1133 of file rtp.c. References ast_autoservice_start(), ast_autoservice_stop(), AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_FRAME_DTMF, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frfree(), ast_log(), ast_mutex_lock, ast_mutex_unlock, ast_read(), ast_rtp_get_peer(), ast_waitfor_n(), ast_write(), ast_frame::frametype, inaddrcmp(), ast_channel::lock, LOG_DEBUG, LOG_WARNING, ast_channel::name, ast_channel_pvt::pvt, and ast_channel::pvt.
01134 { 01135 struct ast_frame *f; 01136 struct ast_channel *who, *cs[3]; 01137 struct ast_rtp *p0, *p1; 01138 struct ast_rtp *vp0, *vp1; 01139 struct ast_rtp_protocol *pr0, *pr1; 01140 struct sockaddr_in ac0, ac1; 01141 struct sockaddr_in vac0, vac1; 01142 struct sockaddr_in t0, t1; 01143 struct sockaddr_in vt0, vt1; 01144 01145 void *pvt0, *pvt1; 01146 int to; 01147 01148 memset(&vt0, 0, sizeof(vt0)); 01149 memset(&vt1, 0, sizeof(vt1)); 01150 memset(&vac0, 0, sizeof(vac0)); 01151 memset(&vac1, 0, sizeof(vac1)); 01152 01153 /* XXX Wait a half a second for things to settle up 01154 this really should be fixed XXX */ 01155 ast_autoservice_start(c0); 01156 ast_autoservice_start(c1); 01157 usleep(500000); 01158 ast_autoservice_stop(c0); 01159 ast_autoservice_stop(c1); 01160 01161 /* if need DTMF, cant native bridge */ 01162 if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1)) 01163 return -2; 01164 ast_mutex_lock(&c0->lock); 01165 ast_mutex_lock(&c1->lock); 01166 pr0 = get_proto(c0); 01167 pr1 = get_proto(c1); 01168 if (!pr0) { 01169 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); 01170 ast_mutex_unlock(&c0->lock); 01171 ast_mutex_unlock(&c1->lock); 01172 return -1; 01173 } 01174 if (!pr1) { 01175 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); 01176 ast_mutex_unlock(&c0->lock); 01177 ast_mutex_unlock(&c1->lock); 01178 return -1; 01179 } 01180 pvt0 = c0->pvt->pvt; 01181 pvt1 = c1->pvt->pvt; 01182 p0 = pr0->get_rtp_info(c0); 01183 if (pr0->get_vrtp_info) 01184 vp0 = pr0->get_vrtp_info(c0); 01185 else 01186 vp0 = NULL; 01187 p1 = pr1->get_rtp_info(c1); 01188 if (pr1->get_vrtp_info) 01189 vp1 = pr1->get_vrtp_info(c1); 01190 else 01191 vp1 = NULL; 01192 if (!p0 || !p1) { 01193 /* Somebody doesn't want to play... */ 01194 ast_mutex_unlock(&c0->lock); 01195 ast_mutex_unlock(&c1->lock); 01196 return -2; 01197 } 01198 if (pr0->set_rtp_peer(c0, p1, vp1)) 01199 ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name); 01200 else { 01201 /* Store RTP peer */ 01202 ast_rtp_get_peer(p1, &ac1); 01203 if (vp1) 01204 ast_rtp_get_peer(p1, &vac1); 01205 } 01206 if (pr1->set_rtp_peer(c1, p0, vp0)) 01207 ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name); 01208 else { 01209 /* Store RTP peer */ 01210 ast_rtp_get_peer(p0, &ac0); 01211 if (vp0) 01212 ast_rtp_get_peer(p0, &vac0); 01213 } 01214 ast_mutex_unlock(&c0->lock); 01215 ast_mutex_unlock(&c1->lock); 01216 cs[0] = c0; 01217 cs[1] = c1; 01218 cs[2] = NULL; 01219 for (;;) { 01220 if ((c0->pvt->pvt != pvt0) || 01221 (c1->pvt->pvt != pvt1) || 01222 (c0->masq || c0->masqr || c1->masq || c1->masqr)) { 01223 ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n"); 01224 if (c0->pvt->pvt == pvt0) { 01225 if (pr0->set_rtp_peer(c0, NULL, NULL)) 01226 ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name); 01227 } 01228 if (c1->pvt->pvt == pvt1) { 01229 if (pr1->set_rtp_peer(c1, NULL, NULL)) 01230 ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name); 01231 } 01232 /* Tell it to try again later */ 01233 return -3; 01234 } 01235 to = -1; 01236 ast_rtp_get_peer(p1, &t1); 01237 ast_rtp_get_peer(p0, &t0); 01238 if (vp1) 01239 ast_rtp_get_peer(vp1, &vt1); 01240 if (vp0) 01241 ast_rtp_get_peer(vp0, &vt0); 01242 if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1))) { 01243 ast_log(LOG_DEBUG, "Oooh, '%s' changed end address\n", c1->name); 01244 if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL)) 01245 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name); 01246 memcpy(&ac1, &t1, sizeof(ac1)); 01247 memcpy(&vac1, &vt1, sizeof(vac1)); 01248 } 01249 if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) { 01250 ast_log(LOG_DEBUG, "Oooh, '%s' changed end address\n", c0->name); 01251 if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL)) 01252 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name); 01253 memcpy(&ac0, &t0, sizeof(ac0)); 01254 memcpy(&vac0, &vt0, sizeof(vac0)); 01255 } 01256 who = ast_waitfor_n(cs, 2, &to); 01257 if (!who) { 01258 ast_log(LOG_DEBUG, "Ooh, empty read...\n"); 01259 continue; 01260 } 01261 f = ast_read(who); 01262 if (!f || ((f->frametype == AST_FRAME_DTMF) && 01263 (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 01264 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) { 01265 *fo = f; 01266 *rc = who; 01267 ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup"); 01268 if ((c0->pvt->pvt == pvt0) && (!c0->_softhangup)) { 01269 if (pr0->set_rtp_peer(c0, NULL, NULL)) 01270 ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name); 01271 } 01272 if ((c1->pvt->pvt == pvt1) && (!c1->_softhangup)) { 01273 if (pr1->set_rtp_peer(c1, NULL, NULL)) 01274 ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name); 01275 } 01276 /* That's all we needed */ 01277 return 0; 01278 } else { 01279 if ((f->frametype == AST_FRAME_DTMF) || 01280 (f->frametype == AST_FRAME_VOICE) || 01281 (f->frametype == AST_FRAME_VIDEO)) { 01282 /* Forward voice or DTMF frames if they happen upon us */ 01283 if (who == c0) { 01284 ast_write(c1, f); 01285 } else if (who == c1) { 01286 ast_write(c0, f); 01287 } 01288 } 01289 ast_frfree(f); 01290 } 01291 /* Swap priority not that it's a big deal at this point */ 01292 cs[2] = cs[0]; 01293 cs[0] = cs[1]; 01294 cs[1] = cs[2]; 01295 01296 } 01297 return -1; 01298 } |
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Definition at line 803 of file rtp.c. References ast_io_remove(), ast_smoother_free(), free, ast_rtp::io, ast_rtp::ioid, ast_rtp::rtcp, ast_rtcp::s, ast_rtp::s, and ast_rtp::smoother.
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Definition at line 100 of file rtp.c. References ast_rtp::s.
00101 { 00102 return rtp->s; 00103 } |
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Definition at line 612 of file rtp.c. References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, and MAX_RTP_PT.
00613 { 00614 int pt; 00615 00616 *astFormats = *nonAstFormats = 0; 00617 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 00618 if (rtp->current_RTP_PT[pt].isAstFormat) { 00619 *astFormats |= rtp->current_RTP_PT[pt].code; 00620 } else { 00621 *nonAstFormats |= rtp->current_RTP_PT[pt].code; 00622 } 00623 } 00624 } |
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Definition at line 781 of file rtp.c. References ast_rtp::them. Referenced by ast_rtp_bridge().
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Definition at line 788 of file rtp.c. References ast_rtp::us.
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Definition at line 1333 of file rtp.c. References ast_rtp_reload(). Referenced by main().
01334 { 01335 ast_rtp_reload(); 01336 } |
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Definition at line 636 of file rtp.c. References rtpPayloadType::code, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result. Referenced by ast_rtp_write().
00636 { 00637 int pt; 00638 00639 /* Looks up an RTP code out of our *static* outbound list */ 00640 00641 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && 00642 code == rtp->rtp_lookup_code_cache_code) { 00643 // Use our cached mapping, to avoid the overhead of the loop below 00644 return rtp->rtp_lookup_code_cache_result; 00645 } 00646 00647 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 00648 if (static_RTP_PT[pt].code == code && 00649 static_RTP_PT[pt].isAstFormat == isAstFormat) { 00650 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 00651 rtp->rtp_lookup_code_cache_code = code; 00652 rtp->rtp_lookup_code_cache_result = pt; 00653 return pt; 00654 } 00655 } 00656 return -1; 00657 } |
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Definition at line 659 of file rtp.c.
00659 { 00660 int i; 00661 00662 for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) { 00663 if (mimeTypes[i].payloadType.code == code && 00664 mimeTypes[i].payloadType.isAstFormat == isAstFormat) { 00665 return mimeTypes[i].subtype; 00666 } 00667 } 00668 return ""; 00669 } |
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Definition at line 626 of file rtp.c. References ast_rtp_lookup_pt(), rtpPayloadType::code, rtpPayloadType::isAstFormat, and MAX_RTP_PT. Referenced by ast_rtp_lookup_pt(), and ast_rtp_read().
00626 { 00627 if (pt < 0 || pt > MAX_RTP_PT) { 00628 struct rtpPayloadType result; 00629 result.isAstFormat = result.code = 0; 00630 return result; // bogus payload type 00631 } 00632 /* Gotta use our static one, since that's what we sent against */ 00633 return static_RTP_PT[pt]; 00634 } |
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Definition at line 691 of file rtp.c. References ast_io_add(), AST_IO_IN, ast_log(), ast_rtp_pt_default(), free, LOG_WARNING, and malloc.
00692 { 00693 struct ast_rtp *rtp; 00694 int x; 00695 int flags; 00696 int startplace; 00697 rtp = malloc(sizeof(struct ast_rtp)); 00698 if (!rtp) 00699 return NULL; 00700 memset(rtp, 0, sizeof(struct ast_rtp)); 00701 rtp->them.sin_family = AF_INET; 00702 rtp->us.sin_family = AF_INET; 00703 rtp->s = socket(AF_INET, SOCK_DGRAM, 0); 00704 rtp->ssrc = rand(); 00705 rtp->seqno = rand() & 0xffff; 00706 if (rtp->s < 0) { 00707 free(rtp); 00708 ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno)); 00709 return NULL; 00710 } 00711 if (sched && rtcpenable) { 00712 rtp->sched = sched; 00713 rtp->rtcp = ast_rtcp_new(); 00714 } 00715 flags = fcntl(rtp->s, F_GETFL); 00716 fcntl(rtp->s, F_SETFL, flags | O_NONBLOCK); 00717 /* Find us a place */ 00718 x = (rand() % (rtpend-rtpstart)) + rtpstart; 00719 x = x & ~1; 00720 startplace = x; 00721 for (;;) { 00722 /* Must be an even port number by RTP spec */ 00723 rtp->us.sin_port = htons(x); 00724 if (rtp->rtcp) 00725 rtp->rtcp->us.sin_port = htons(x + 1); 00726 if (!bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us)) && 00727 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) 00728 break; 00729 if (errno != EADDRINUSE) { 00730 ast_log(LOG_WARNING, "Unexpected bind error: %s\n", strerror(errno)); 00731 close(rtp->s); 00732 if (rtp->rtcp) { 00733 close(rtp->rtcp->s); 00734 free(rtp->rtcp); 00735 } 00736 free(rtp); 00737 return NULL; 00738 } 00739 x += 2; 00740 if (x > rtpend) 00741 x = (rtpstart + 1) & ~1; 00742 if (x == startplace) { 00743 ast_log(LOG_WARNING, "No RTP ports remaining\n"); 00744 close(rtp->s); 00745 if (rtp->rtcp) { 00746 close(rtp->rtcp->s); 00747 free(rtp->rtcp); 00748 } 00749 free(rtp); 00750 return NULL; 00751 } 00752 } 00753 if (io && sched && callbackmode) { 00754 /* Operate this one in a callback mode */ 00755 rtp->sched = sched; 00756 rtp->io = io; 00757 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); 00758 } 00759 ast_rtp_pt_default(rtp); 00760 return rtp; 00761 } |
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Definition at line 1104 of file rtp.c. References ast_log(), LOG_WARNING, ast_rtp_protocol::next, and ast_rtp_protocol::type.
01105 { 01106 struct ast_rtp_protocol *cur; 01107 cur = protos; 01108 while(cur) { 01109 if (cur->type == proto->type) { 01110 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); 01111 return -1; 01112 } 01113 cur = cur->next; 01114 } 01115 proto->next = protos; 01116 protos = proto; 01117 return 0; 01118 } |
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Definition at line 1086 of file rtp.c. References ast_rtp_protocol::next.
01087 { 01088 struct ast_rtp_protocol *cur, *prev; 01089 cur = protos; 01090 prev = NULL; 01091 while(cur) { 01092 if (cur == proto) { 01093 if (prev) 01094 prev->next = proto->next; 01095 else 01096 protos = proto->next; 01097 return; 01098 } 01099 prev = cur; 01100 cur = cur->next; 01101 } 01102 } |
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Definition at line 554 of file rtp.c. References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.
00555 { 00556 int i; 00557 00558 for (i = 0; i < MAX_RTP_PT; ++i) { 00559 rtp->current_RTP_PT[i].isAstFormat = 0; 00560 rtp->current_RTP_PT[i].code = 0; 00561 } 00562 00563 rtp->rtp_lookup_code_cache_isAstFormat = 0; 00564 rtp->rtp_lookup_code_cache_code = 0; 00565 rtp->rtp_lookup_code_cache_result = 0; 00566 } |
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Definition at line 568 of file rtp.c. References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result. Referenced by ast_rtp_new().
00569 { 00570 int i; 00571 /* Initialize to default payload types */ 00572 for (i = 0; i < MAX_RTP_PT; ++i) { 00573 rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat; 00574 rtp->current_RTP_PT[i].code = static_RTP_PT[i].code; 00575 } 00576 00577 rtp->rtp_lookup_code_cache_isAstFormat = 0; 00578 rtp->rtp_lookup_code_cache_code = 0; 00579 rtp->rtp_lookup_code_cache_result = 0; 00580 } |
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Definition at line 351 of file rtp.c. References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G723_1, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_MAX_AUDIO, AST_FORMAT_SLINEAR, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, AST_FRAME_NULL, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_getformatname(), ast_log(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_lookup_pt(), rtpPayloadType::code, CRASH, ast_frame::data, ast_frame::datalen, ast_rtp::dtmfcount, ast_rtp::f, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxts, LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, ast_rtp::rawdata, ast_rtp::resp, ast_rtp::s, ast_frame::samples, ast_frame::src, ast_frame::subclass, and ast_rtp::them.
00352 { 00353 int res; 00354 struct sockaddr_in sin; 00355 int len; 00356 unsigned int seqno; 00357 int payloadtype; 00358 int hdrlen = 12; 00359 int mark; 00360 unsigned int timestamp; 00361 unsigned int *rtpheader; 00362 static struct ast_frame *f, null_frame = { AST_FRAME_NULL, }; 00363 struct rtpPayloadType rtpPT; 00364 00365 len = sizeof(sin); 00366 00367 /* Cache where the header will go */ 00368 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 00369 0, (struct sockaddr *)&sin, &len); 00370 00371 00372 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); 00373 if (res < 0) { 00374 ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno)); 00375 if (errno == EBADF) 00376 CRASH; 00377 return &null_frame; 00378 } 00379 if (res < hdrlen) { 00380 ast_log(LOG_WARNING, "RTP Read too short\n"); 00381 return &null_frame; 00382 } 00383 if (rtp->nat) { 00384 /* Send to whoever sent to us */ 00385 if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 00386 (rtp->them.sin_port != sin.sin_port)) { 00387 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 00388 ast_log(LOG_DEBUG, "RTP NAT: Using address %s:%d\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port)); 00389 } 00390 } 00391 /* Get fields */ 00392 seqno = ntohl(rtpheader[0]); 00393 payloadtype = (seqno & 0x7f0000) >> 16; 00394 mark = seqno & (1 << 23); 00395 seqno &= 0xffff; 00396 timestamp = ntohl(rtpheader[1]); 00397 00398 #if 0 00399 printf("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len = %d)\n", inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); 00400 #endif 00401 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); 00402 if (!rtpPT.isAstFormat) { 00403 // This is special in-band data that's not one of our codecs 00404 if (rtpPT.code == AST_RTP_DTMF) { 00405 /* It's special -- rfc2833 process it */ 00406 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 00407 if (f) return f; else return &null_frame; 00408 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { 00409 /* It's really special -- process it the Cisco way */ 00410 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 00411 if (f) return f; else return &null_frame; 00412 } else if (rtpPT.code == AST_RTP_CN) { 00413 /* Comfort Noise */ 00414 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 00415 if (f) return f; else return &null_frame; 00416 } else { 00417 ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype); 00418 return &null_frame; 00419 } 00420 } 00421 rtp->f.subclass = rtpPT.code; 00422 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) 00423 rtp->f.frametype = AST_FRAME_VOICE; 00424 else 00425 rtp->f.frametype = AST_FRAME_VIDEO; 00426 rtp->lastrxformat = rtp->f.subclass; 00427 00428 if (!rtp->lastrxts) 00429 rtp->lastrxts = timestamp; 00430 00431 if (rtp->dtmfcount) { 00432 #if 0 00433 printf("dtmfcount was %d\n", rtp->dtmfcount); 00434 #endif 00435 rtp->dtmfcount -= (timestamp - rtp->lastrxts); 00436 if (rtp->dtmfcount < 0) 00437 rtp->dtmfcount = 0; 00438 #if 0 00439 if (dtmftimeout != rtp->dtmfcount) 00440 printf("dtmfcount is %d\n", rtp->dtmfcount); 00441 #endif 00442 } 00443 rtp->lastrxts = timestamp; 00444 00445 /* Send any pending DTMF */ 00446 if (rtp->resp && !rtp->dtmfcount) { 00447 ast_log(LOG_DEBUG, "Sending pending DTMF\n"); 00448 return send_dtmf(rtp); 00449 } 00450 rtp->f.mallocd = 0; 00451 rtp->f.datalen = res - hdrlen; 00452 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; 00453 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; 00454 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { 00455 switch(rtp->f.subclass) { 00456 case AST_FORMAT_ULAW: 00457 case AST_FORMAT_ALAW: 00458 rtp->f.samples = rtp->f.datalen; 00459 break; 00460 case AST_FORMAT_SLINEAR: 00461 rtp->f.samples = rtp->f.datalen / 2; 00462 break; 00463 case AST_FORMAT_GSM: 00464 rtp->f.samples = 160 * (rtp->f.datalen / 33); 00465 break; 00466 case AST_FORMAT_ILBC: 00467 rtp->f.samples = 240 * (rtp->f.datalen / 50); 00468 break; 00469 case AST_FORMAT_ADPCM: 00470 rtp->f.samples = rtp->f.datalen * 2; 00471 break; 00472 case AST_FORMAT_G729A: 00473 rtp->f.samples = rtp->f.datalen * 8; 00474 break; 00475 case AST_FORMAT_G723_1: 00476 rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen); 00477 break; 00478 case AST_FORMAT_SPEEX: 00479 rtp->f.samples = 160; 00480 // assumes that the RTP packet contained one Speex frame 00481 break; 00482 default: 00483 ast_log(LOG_NOTICE, "Unable to calculate samples for format %s\n", ast_getformatname(rtp->f.subclass)); 00484 break; 00485 } 00486 } else { 00487 /* Video -- samples is # of samples vs. 90000 */ 00488 if (!rtp->lastividtimestamp) 00489 rtp->lastividtimestamp = timestamp; 00490 rtp->f.samples = timestamp - rtp->lastividtimestamp; 00491 rtp->lastividtimestamp = timestamp; 00492 if (mark) 00493 rtp->f.subclass |= 0x1; 00494 00495 } 00496 rtp->f.src = "RTP"; 00497 return &rtp->f; 00498 } |
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Definition at line 1300 of file rtp.c. References ast_destroy(), ast_load(), ast_log(), ast_variable_retrieve(), ast_verbose(), LOG_WARNING, option_verbose, and VERBOSE_PREFIX_2. Referenced by ast_module_reload(), and ast_rtp_init().
01301 { 01302 struct ast_config *cfg; 01303 char *s; 01304 rtpstart = 5000; 01305 rtpend = 31000; 01306 cfg = ast_load("rtp.conf"); 01307 if (cfg) { 01308 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { 01309 rtpstart = atoi(s); 01310 if (rtpstart < 1024) 01311 rtpstart = 1024; 01312 if (rtpstart > 65535) 01313 rtpstart = 65535; 01314 } 01315 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { 01316 rtpend = atoi(s); 01317 if (rtpend < 1024) 01318 rtpend = 1024; 01319 if (rtpend > 65535) 01320 rtpend = 65535; 01321 } 01322 ast_destroy(cfg); 01323 } 01324 if (rtpstart >= rtpend) { 01325 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end\n"); 01326 rtpstart = 5000; 01327 rtpend = 31000; 01328 } 01329 if (option_verbose > 1) 01330 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); 01331 } |
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Definition at line 834 of file rtp.c. References ast_log(), ast_rtp::lastts, LOG_NOTICE, LOG_WARNING, ast_rtp::s, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.
00835 { 00836 unsigned int *rtpheader; 00837 int hdrlen = 12; 00838 int res; 00839 int ms; 00840 int pred; 00841 int x; 00842 char data[256]; 00843 00844 if ((digit <= '9') && (digit >= '0')) 00845 digit -= '0'; 00846 else if (digit == '*') 00847 digit = 10; 00848 else if (digit == '#') 00849 digit = 11; 00850 else if ((digit >= 'A') && (digit <= 'D')) 00851 digit = digit - 'A' + 12; 00852 else if ((digit >= 'a') && (digit <= 'd')) 00853 digit = digit - 'a' + 12; 00854 else { 00855 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); 00856 return -1; 00857 } 00858 00859 00860 /* If we have no peer, return immediately */ 00861 if (!rtp->them.sin_addr.s_addr) 00862 return 0; 00863 00864 ms = calc_txstamp(rtp); 00865 /* Default prediction */ 00866 pred = rtp->lastts + ms * 8; 00867 00868 /* Get a pointer to the header */ 00869 rtpheader = (unsigned int *)data; 00870 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (101 << 16) | (rtp->seqno++)); 00871 rtpheader[1] = htonl(rtp->lastts); 00872 rtpheader[2] = htonl(rtp->ssrc); 00873 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0)); 00874 for (x=0;x<4;x++) { 00875 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { 00876 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 4, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); 00877 if (res <0) 00878 ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno)); 00879 #if 0 00880 printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port)); 00881 #endif 00882 } 00883 if (x ==0) { 00884 /* Clear marker bit and increment seqno */ 00885 rtpheader[0] = htonl((2 << 30) | (101 << 16) | (rtp->seqno++)); 00886 /* Make duration 240 */ 00887 rtpheader[3] |= htonl((240)); 00888 /* Set the End bit for the last 3 */ 00889 rtpheader[3] |= htonl((1 << 23)); 00890 } 00891 } 00892 return 0; 00893 } |
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Definition at line 153 of file rtp.c. References ast_rtp_callback, and ast_rtp::callback.
00154 { 00155 rtp->callback = callback; 00156 } |
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Definition at line 148 of file rtp.c. References ast_rtp::data.
00149 { 00150 rtp->data = data; 00151 } |
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Definition at line 585 of file rtp.c. References rtpPayloadType::code, ast_rtp::current_RTP_PT, and MAX_RTP_PT.
00585 { 00586 if (pt < 0 || pt > MAX_RTP_PT) return; // bogus payload type 00587 00588 if (static_RTP_PT[pt].code != 0) { 00589 rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; 00590 } 00591 } |
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Definition at line 771 of file rtp.c. References ast_rtp::rtcp, ast_rtcp::them, and ast_rtp::them.
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Definition at line 595 of file rtp.c. References ast_rtp::current_RTP_PT, MAX_RTP_PT, subtype, and type.
00596 { 00597 int i; 00598 00599 if (pt < 0 || pt > MAX_RTP_PT) return; // bogus payload type 00600 00601 for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) { 00602 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && 00603 strcasecmp(mimeType, mimeTypes[i].type) == 0) { 00604 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; 00605 return; 00606 } 00607 } 00608 } |
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Definition at line 158 of file rtp.c. References ast_rtp::nat.
00159 { 00160 rtp->nat = nat; 00161 } |
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Definition at line 763 of file rtp.c. References ast_log(), LOG_WARNING, and ast_rtp::s.
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Definition at line 793 of file rtp.c. References ast_rtp::rtcp, ast_rtcp::them, and ast_rtp::them.
00794 { 00795 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 00796 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 00797 if (rtp->rtcp) { 00798 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 00799 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 00800 } 00801 } |
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Definition at line 973 of file rtp.c. References AST_FORMAT_ALAW, AST_FORMAT_G723_1, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_H261, AST_FORMAT_H263, AST_FORMAT_ILBC, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frdup(), ast_getformatname(), ast_log(), ast_rtp_lookup_code(), ast_smoother_feed(), ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_frame::datalen, ast_frame::frametype, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.
00974 { 00975 struct ast_frame *f; 00976 int codec; 00977 int hdrlen = 12; 00978 int subclass; 00979 00980 00981 /* If we have no peer, return immediately */ 00982 if (!rtp->them.sin_addr.s_addr) 00983 return 0; 00984 00985 /* If there is no data length, return immediately */ 00986 if (!_f->datalen) 00987 return 0; 00988 00989 /* Make sure we have enough space for RTP header */ 00990 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { 00991 ast_log(LOG_WARNING, "RTP can only send voice\n"); 00992 return -1; 00993 } 00994 00995 subclass = _f->subclass; 00996 if (_f->frametype == AST_FRAME_VIDEO) 00997 subclass &= ~0x1; 00998 00999 codec = ast_rtp_lookup_code(rtp, 1, subclass); 01000 if (codec < 0) { 01001 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); 01002 return -1; 01003 } 01004 01005 if (rtp->lasttxformat != subclass) { 01006 /* New format, reset the smoother */ 01007 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); 01008 rtp->lasttxformat = subclass; 01009 if (rtp->smoother) 01010 ast_smoother_free(rtp->smoother); 01011 rtp->smoother = NULL; 01012 } 01013 01014 01015 switch(subclass) { 01016 case AST_FORMAT_ULAW: 01017 case AST_FORMAT_ALAW: 01018 if (!rtp->smoother) { 01019 rtp->smoother = ast_smoother_new(160); 01020 } 01021 if (!rtp->smoother) { 01022 ast_log(LOG_WARNING, "Unable to create smoother :(\n"); 01023 return -1; 01024 } 01025 ast_smoother_feed(rtp->smoother, _f); 01026 01027 while((f = ast_smoother_read(rtp->smoother))) 01028 ast_rtp_raw_write(rtp, f, codec); 01029 break; 01030 case AST_FORMAT_G729A: 01031 if (!rtp->smoother) { 01032 rtp->smoother = ast_smoother_new(20); 01033 } 01034 if (!rtp->smoother) { 01035 ast_log(LOG_WARNING, "Unable to create g729 smoother :(\n"); 01036 return -1; 01037 } 01038 ast_smoother_feed(rtp->smoother, _f); 01039 01040 while((f = ast_smoother_read(rtp->smoother))) 01041 ast_rtp_raw_write(rtp, f, codec); 01042 break; 01043 case AST_FORMAT_GSM: 01044 if (!rtp->smoother) { 01045 rtp->smoother = ast_smoother_new(33); 01046 } 01047 if (!rtp->smoother) { 01048 ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n"); 01049 return -1; 01050 } 01051 ast_smoother_feed(rtp->smoother, _f); 01052 while((f = ast_smoother_read(rtp->smoother))) 01053 ast_rtp_raw_write(rtp, f, codec); 01054 break; 01055 case AST_FORMAT_ILBC: 01056 if (!rtp->smoother) { 01057 rtp->smoother = ast_smoother_new(50); 01058 } 01059 if (!rtp->smoother) { 01060 ast_log(LOG_WARNING, "Unable to create ILBC smoother :(\n"); 01061 return -1; 01062 } 01063 ast_smoother_feed(rtp->smoother, _f); 01064 while((f = ast_smoother_read(rtp->smoother))) 01065 ast_rtp_raw_write(rtp, f, codec); 01066 break; 01067 default: 01068 ast_log(LOG_WARNING, "Not sure about sending format %s packets\n", ast_getformatname(subclass)); 01069 // fall through to... 01070 case AST_FORMAT_H261: 01071 case AST_FORMAT_H263: 01072 case AST_FORMAT_G723_1: 01073 case AST_FORMAT_SPEEX: 01074 // Don't buffer outgoing frames; send them one-per-packet: 01075 if (_f->offset < hdrlen) { 01076 f = ast_frdup(_f); 01077 } else { 01078 f = _f; 01079 } 01080 ast_rtp_raw_write(rtp, f, codec); 01081 } 01082 01083 return 0; 01084 } |