#include <netinet/in.h>
#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
#include "asterisk/linkedlists.h"
Go to the source code of this file.
Data Structures | |
struct | ast_rtp_protocol |
struct | ast_rtp_quality |
Defines | |
#define | AST_RTP_CISCO_DTMF (1 << 2) |
#define | AST_RTP_CN (1 << 1) |
#define | AST_RTP_DTMF (1 << 0) |
#define | AST_RTP_MAX AST_RTP_CISCO_DTMF |
#define | FLAG_3389_WARNING (1 << 0) |
#define | MAX_RTP_PT 256 |
Typedefs | |
typedef int(*) | ast_rtp_callback (struct ast_rtp *rtp, struct ast_frame *f, void *data) |
Enumerations | |
enum | ast_rtp_get_result { AST_RTP_GET_FAILED = 0, AST_RTP_TRY_PARTIAL, AST_RTP_TRY_NATIVE } |
enum | ast_rtp_options { AST_RTP_OPT_G726_NONSTANDARD = (1 << 0) } |
Functions | |
int | ast_rtcp_fd (struct ast_rtp *rtp) |
ast_frame * | ast_rtcp_read (struct ast_rtp *rtp) |
int | ast_rtcp_send_h261fur (void *data) |
Send an H.261 fast update request. Some devices need this rather than the XML message in SIP. | |
size_t | ast_rtp_alloc_size (void) |
Get the amount of space required to hold an RTP session. | |
int | ast_rtp_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms) |
Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk. | |
int | ast_rtp_codec_getformat (int pt) |
ast_codec_pref * | ast_rtp_codec_getpref (struct ast_rtp *rtp) |
int | ast_rtp_codec_setpref (struct ast_rtp *rtp, struct ast_codec_pref *prefs) |
void | ast_rtp_destroy (struct ast_rtp *rtp) |
int | ast_rtp_early_bridge (struct ast_channel *dest, struct ast_channel *src) |
If possible, create an early bridge directly between the devices without having to send a re-invite later. | |
int | ast_rtp_fd (struct ast_rtp *rtp) |
ast_rtp * | ast_rtp_get_bridged (struct ast_rtp *rtp) |
void | ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats) |
Return the union of all of the codecs that were set by rtp_set...() calls They're returned as two distinct sets: AST_FORMATs, and AST_RTPs. | |
int | ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them) |
char * | ast_rtp_get_quality (struct ast_rtp *rtp, struct ast_rtp_quality *qual) |
Return RTCP quality string. | |
int | ast_rtp_get_rtpholdtimeout (struct ast_rtp *rtp) |
Get rtp hold timeout. | |
int | ast_rtp_get_rtpkeepalive (struct ast_rtp *rtp) |
Get RTP keepalive interval. | |
int | ast_rtp_get_rtptimeout (struct ast_rtp *rtp) |
Get rtp timeout. | |
void | ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us) |
int | ast_rtp_getnat (struct ast_rtp *rtp) |
void | ast_rtp_init (void) |
Initialize the RTP system in Asterisk. | |
int | ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code) |
Looks up an RTP code out of our *static* outbound list. | |
char * | ast_rtp_lookup_mime_multiple (char *buf, size_t size, const int capability, const int isAstFormat, enum ast_rtp_options options) |
Build a string of MIME subtype names from a capability list. | |
const char * | ast_rtp_lookup_mime_subtype (int isAstFormat, int code, enum ast_rtp_options options) |
Mapping an Asterisk code into a MIME subtype (string):. | |
rtpPayloadType | ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt) |
Mapping between RTP payload format codes and Asterisk codes:. | |
int | ast_rtp_make_compatible (struct ast_channel *dest, struct ast_channel *src, int media) |
ast_rtp * | ast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode) |
Initializate a RTP session. | |
void | ast_rtp_new_init (struct ast_rtp *rtp) |
Initialize a new RTP structure. | |
void | ast_rtp_new_source (struct ast_rtp *rtp) |
ast_rtp * | ast_rtp_new_with_bindaddr (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in) |
Initializate a RTP session using an in_addr structure. | |
int | ast_rtp_proto_register (struct ast_rtp_protocol *proto) |
Register interface to channel driver. | |
void | ast_rtp_proto_unregister (struct ast_rtp_protocol *proto) |
Unregister interface to channel driver. | |
void | ast_rtp_pt_clear (struct ast_rtp *rtp) |
Setting RTP payload types from lines in a SDP description:. | |
void | ast_rtp_pt_copy (struct ast_rtp *dest, struct ast_rtp *src) |
Copy payload types between RTP structures. | |
void | ast_rtp_pt_default (struct ast_rtp *rtp) |
Set payload types to defaults. | |
ast_frame * | ast_rtp_read (struct ast_rtp *rtp) |
int | ast_rtp_reload (void) |
void | ast_rtp_reset (struct ast_rtp *rtp) |
int | ast_rtp_sendcng (struct ast_rtp *rtp, int level) |
generate comfort noice (CNG) | |
int | ast_rtp_senddigit_begin (struct ast_rtp *rtp, char digit) |
Send begin frames for DTMF. | |
int | ast_rtp_senddigit_end (struct ast_rtp *rtp, char digit) |
void | ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback) |
void | ast_rtp_set_data (struct ast_rtp *rtp, void *data) |
void | ast_rtp_set_m_type (struct ast_rtp *rtp, int pt) |
Activate payload type. | |
void | ast_rtp_set_peer (struct ast_rtp *rtp, struct sockaddr_in *them) |
void | ast_rtp_set_rtpholdtimeout (struct ast_rtp *rtp, int timeout) |
Set rtp hold timeout. | |
void | ast_rtp_set_rtpkeepalive (struct ast_rtp *rtp, int period) |
set RTP keepalive interval | |
int | ast_rtp_set_rtpmap_type (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype, enum ast_rtp_options options) |
Initiate payload type to a known MIME media type for a codec. | |
void | ast_rtp_set_rtptimeout (struct ast_rtp *rtp, int timeout) |
Set rtp timeout. | |
void | ast_rtp_set_rtptimers_onhold (struct ast_rtp *rtp) |
void | ast_rtp_setdtmf (struct ast_rtp *rtp, int dtmf) |
Indicate whether this RTP session is carrying DTMF or not. | |
void | ast_rtp_setdtmfcompensate (struct ast_rtp *rtp, int compensate) |
Compensate for devices that send RFC2833 packets all at once. | |
void | ast_rtp_setnat (struct ast_rtp *rtp, int nat) |
void | ast_rtp_setstun (struct ast_rtp *rtp, int stun_enable) |
Enable STUN capability. | |
int | ast_rtp_settos (struct ast_rtp *rtp, int tos) |
void | ast_rtp_stop (struct ast_rtp *rtp) |
void | ast_rtp_stun_request (struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username) |
void | ast_rtp_unset_m_type (struct ast_rtp *rtp, int pt) |
clear payload type | |
int | ast_rtp_write (struct ast_rtp *rtp, struct ast_frame *f) |
RTP is defined in RFC 3550.
Definition in file rtp.h.
#define AST_RTP_CISCO_DTMF (1 << 2) |
#define AST_RTP_CN (1 << 1) |
'Comfort Noise' (RFC3389)
Definition at line 45 of file rtp.h.
Referenced by ast_rtp_read(), and ast_rtp_sendcng().
#define AST_RTP_DTMF (1 << 0) |
DTMF (RFC2833)
Definition at line 43 of file rtp.h.
Referenced by add_noncodec_to_sdp(), ast_rtp_read(), ast_rtp_senddigit_begin(), bridge_p2p_rtp_write(), check_user_full(), create_addr(), create_addr_from_peer(), oh323_alloc(), oh323_request(), process_sdp(), sip_alloc(), and sip_dtmfmode().
#define AST_RTP_MAX AST_RTP_CISCO_DTMF |
Maximum RTP-specific code
Definition at line 49 of file rtp.h.
Referenced by add_sdp(), and ast_rtp_lookup_mime_multiple().
#define MAX_RTP_PT 256 |
Definition at line 51 of file rtp.h.
Referenced by ast_rtp_get_current_formats(), ast_rtp_lookup_code(), ast_rtp_lookup_pt(), ast_rtp_pt_clear(), ast_rtp_pt_copy(), ast_rtp_pt_default(), ast_rtp_set_m_type(), ast_rtp_set_rtpmap_type(), ast_rtp_unset_m_type(), and process_sdp().
typedef int(*) ast_rtp_callback(struct ast_rtp *rtp, struct ast_frame *f, void *data) |
enum ast_rtp_get_result |
Definition at line 57 of file rtp.h.
00057 { 00058 /*! Failed to find the RTP structure */ 00059 AST_RTP_GET_FAILED = 0, 00060 /*! RTP structure exists but true native bridge can not occur so try partial */ 00061 AST_RTP_TRY_PARTIAL, 00062 /*! RTP structure exists and native bridge can occur */ 00063 AST_RTP_TRY_NATIVE, 00064 };
enum ast_rtp_options |
int ast_rtcp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 517 of file rtp.c.
References ast_rtp::rtcp, and ast_rtcp::s.
Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), sip_new(), and start_rtp().
Definition at line 825 of file rtp.c.
References ast_rtcp::accumulated_transit, ast_assert, AST_CONTROL_VIDUPDATE, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), ast_frame::datalen, errno, ast_rtp::f, f, ast_frame::frametype, len(), LOG_DEBUG, LOG_WARNING, ast_frame::mallocd, ast_rtcp::maxrtt, ast_rtcp::minrtt, ast_rtp::nat, option_debug, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, rtcp_debug_test_addr(), RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, ast_rtcp::rtt, ast_rtcp::rxlsr, ast_rtp::s, ast_rtcp::s, ast_frame::samples, ast_rtcp::soc, ast_rtcp::spc, ast_frame::src, ast_frame::subclass, ast_rtcp::them, ast_rtcp::themrxlsr, and timeval2ntp().
Referenced by oh323_read(), sip_rtp_read(), and skinny_rtp_read().
00826 { 00827 socklen_t len; 00828 int position, i, packetwords; 00829 int res; 00830 struct sockaddr_in sin; 00831 unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET]; 00832 unsigned int *rtcpheader; 00833 int pt; 00834 struct timeval now; 00835 unsigned int length; 00836 int rc; 00837 double rttsec; 00838 uint64_t rtt = 0; 00839 unsigned int dlsr; 00840 unsigned int lsr; 00841 unsigned int msw; 00842 unsigned int lsw; 00843 unsigned int comp; 00844 struct ast_frame *f = &ast_null_frame; 00845 00846 if (!rtp || !rtp->rtcp) 00847 return &ast_null_frame; 00848 00849 len = sizeof(sin); 00850 00851 res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET, 00852 0, (struct sockaddr *)&sin, &len); 00853 if (option_debug > 2) 00854 ast_log(LOG_DEBUG, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 00855 00856 rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET); 00857 00858 if (res < 0) { 00859 ast_assert(errno != EBADF); 00860 if (errno != EAGAIN) { 00861 ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno)); 00862 ast_log(LOG_WARNING, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 00863 return NULL; 00864 } 00865 return &ast_null_frame; 00866 } 00867 00868 packetwords = res / 4; 00869 00870 if (rtp->nat) { 00871 /* Send to whoever sent to us */ 00872 if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 00873 (rtp->rtcp->them.sin_port != sin.sin_port)) { 00874 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 00875 if (option_debug || rtpdebug) 00876 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)); 00877 } 00878 } 00879 00880 if (option_debug) 00881 ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res); 00882 00883 /* Process a compound packet */ 00884 position = 0; 00885 while (position < packetwords) { 00886 i = position; 00887 length = ntohl(rtcpheader[i]); 00888 pt = (length & 0xff0000) >> 16; 00889 rc = (length & 0x1f000000) >> 24; 00890 length &= 0xffff; 00891 00892 if ((i + length) > packetwords) { 00893 ast_log(LOG_WARNING, "RTCP Read too short\n"); 00894 return &ast_null_frame; 00895 } 00896 00897 if (rtcp_debug_test_addr(&sin)) { 00898 ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); 00899 ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown"); 00900 ast_verbose("Reception reports: %d\n", rc); 00901 ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]); 00902 } 00903 00904 i += 2; /* Advance past header and ssrc */ 00905 00906 switch (pt) { 00907 case RTCP_PT_SR: 00908 gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */ 00909 rtp->rtcp->spc = ntohl(rtcpheader[i+3]); 00910 rtp->rtcp->soc = ntohl(rtcpheader[i + 4]); 00911 rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/ 00912 00913 if (rtcp_debug_test_addr(&sin)) { 00914 ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096); 00915 ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2])); 00916 ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4])); 00917 } 00918 i += 5; 00919 if (rc < 1) 00920 break; 00921 /* Intentional fall through */ 00922 case RTCP_PT_RR: 00923 /* Don't handle multiple reception reports (rc > 1) yet */ 00924 /* Calculate RTT per RFC */ 00925 gettimeofday(&now, NULL); 00926 timeval2ntp(now, &msw, &lsw); 00927 if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */ 00928 comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16); 00929 lsr = ntohl(rtcpheader[i + 4]); 00930 dlsr = ntohl(rtcpheader[i + 5]); 00931 rtt = comp - lsr - dlsr; 00932 00933 /* Convert end to end delay to usec (keeping the calculation in 64bit space) 00934 sess->ee_delay = (eedelay * 1000) / 65536; */ 00935 if (rtt < 4294) { 00936 rtt = (rtt * 1000000) >> 16; 00937 } else { 00938 rtt = (rtt * 1000) >> 16; 00939 rtt *= 1000; 00940 } 00941 rtt = rtt / 1000.; 00942 rttsec = rtt / 1000.; 00943 00944 if (comp - dlsr >= lsr) { 00945 rtp->rtcp->accumulated_transit += rttsec; 00946 rtp->rtcp->rtt = rttsec; 00947 if (rtp->rtcp->maxrtt<rttsec) 00948 rtp->rtcp->maxrtt = rttsec; 00949 if (rtp->rtcp->minrtt>rttsec) 00950 rtp->rtcp->minrtt = rttsec; 00951 } else if (rtcp_debug_test_addr(&sin)) { 00952 ast_verbose("Internal RTCP NTP clock skew detected: " 00953 "lsr=%u, now=%u, dlsr=%u (%d:%03dms), " 00954 "diff=%d\n", 00955 lsr, comp, dlsr, dlsr / 65536, 00956 (dlsr % 65536) * 1000 / 65536, 00957 dlsr - (comp - lsr)); 00958 } 00959 } 00960 00961 rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]); 00962 rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff; 00963 if (rtcp_debug_test_addr(&sin)) { 00964 ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24)); 00965 ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost); 00966 ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff)); 00967 ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16); 00968 ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter); 00969 ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096); 00970 ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0); 00971 if (rtt) 00972 ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt); 00973 } 00974 break; 00975 case RTCP_PT_FUR: 00976 if (rtcp_debug_test_addr(&sin)) 00977 ast_verbose("Received an RTCP Fast Update Request\n"); 00978 rtp->f.frametype = AST_FRAME_CONTROL; 00979 rtp->f.subclass = AST_CONTROL_VIDUPDATE; 00980 rtp->f.datalen = 0; 00981 rtp->f.samples = 0; 00982 rtp->f.mallocd = 0; 00983 rtp->f.src = "RTP"; 00984 f = &rtp->f; 00985 break; 00986 case RTCP_PT_SDES: 00987 if (rtcp_debug_test_addr(&sin)) 00988 ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00989 break; 00990 case RTCP_PT_BYE: 00991 if (rtcp_debug_test_addr(&sin)) 00992 ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00993 break; 00994 default: 00995 if (option_debug) 00996 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)); 00997 break; 00998 } 00999 position += (length + 1); 01000 } 01001 01002 return f; 01003 }
int ast_rtcp_send_h261fur | ( | void * | data | ) |
Send an H.261 fast update request. Some devices need this rather than the XML message in SIP.
Definition at line 2369 of file rtp.c.
References ast_rtcp_write(), ast_rtp::rtcp, and ast_rtcp::sendfur.
02370 { 02371 struct ast_rtp *rtp = data; 02372 int res; 02373 02374 rtp->rtcp->sendfur = 1; 02375 res = ast_rtcp_write(data); 02376 02377 return res; 02378 }
size_t ast_rtp_alloc_size | ( | void | ) |
Get the amount of space required to hold an RTP session.
Definition at line 397 of file rtp.c.
Referenced by process_sdp().
00398 { 00399 return sizeof(struct ast_rtp); 00400 }
int ast_rtp_bridge | ( | struct ast_channel * | c0, | |
struct ast_channel * | c1, | |||
int | flags, | |||
struct ast_frame ** | fo, | |||
struct ast_channel ** | rc, | |||
int | timeoutms | |||
) |
Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk.
Definition at line 3347 of file rtp.c.
References AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_check_hangup(), ast_codec_pref_getsize(), ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_set_flag, ast_test_flag, ast_verbose(), bridge_native_loop(), bridge_p2p_loop(), ast_format_list::cur_ms, FLAG_HAS_DTMF, FLAG_P2P_NEED_DTMF, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, option_debug, option_verbose, ast_rtp::pref, ast_channel::rawreadformat, ast_channel::rawwriteformat, ast_channel_tech::send_digit_begin, ast_channel::tech, ast_channel::tech_pvt, and VERBOSE_PREFIX_3.
03348 { 03349 struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ 03350 struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */ 03351 struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL; 03352 enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED; 03353 enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED; 03354 enum ast_bridge_result res = AST_BRIDGE_FAILED; 03355 int codec0 = 0, codec1 = 0; 03356 void *pvt0 = NULL, *pvt1 = NULL; 03357 03358 /* Lock channels */ 03359 ast_channel_lock(c0); 03360 while(ast_channel_trylock(c1)) { 03361 ast_channel_unlock(c0); 03362 usleep(1); 03363 ast_channel_lock(c0); 03364 } 03365 03366 /* Ensure neither channel got hungup during lock avoidance */ 03367 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 03368 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name); 03369 ast_channel_unlock(c0); 03370 ast_channel_unlock(c1); 03371 return AST_BRIDGE_FAILED; 03372 } 03373 03374 /* Find channel driver interfaces */ 03375 if (!(pr0 = get_proto(c0))) { 03376 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); 03377 ast_channel_unlock(c0); 03378 ast_channel_unlock(c1); 03379 return AST_BRIDGE_FAILED; 03380 } 03381 if (!(pr1 = get_proto(c1))) { 03382 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); 03383 ast_channel_unlock(c0); 03384 ast_channel_unlock(c1); 03385 return AST_BRIDGE_FAILED; 03386 } 03387 03388 /* Get channel specific interface structures */ 03389 pvt0 = c0->tech_pvt; 03390 pvt1 = c1->tech_pvt; 03391 03392 /* Get audio and video interface (if native bridge is possible) */ 03393 audio_p0_res = pr0->get_rtp_info(c0, &p0); 03394 video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED; 03395 audio_p1_res = pr1->get_rtp_info(c1, &p1); 03396 video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED; 03397 03398 /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */ 03399 if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE)) 03400 audio_p0_res = AST_RTP_GET_FAILED; 03401 if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE)) 03402 audio_p1_res = AST_RTP_GET_FAILED; 03403 03404 /* Check if a bridge is possible (partial/native) */ 03405 if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) { 03406 /* Somebody doesn't want to play... */ 03407 ast_channel_unlock(c0); 03408 ast_channel_unlock(c1); 03409 return AST_BRIDGE_FAILED_NOWARN; 03410 } 03411 03412 /* If we need to feed DTMF frames into the core then only do a partial native bridge */ 03413 if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) { 03414 ast_set_flag(p0, FLAG_P2P_NEED_DTMF); 03415 audio_p0_res = AST_RTP_TRY_PARTIAL; 03416 } 03417 03418 if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) { 03419 ast_set_flag(p1, FLAG_P2P_NEED_DTMF); 03420 audio_p1_res = AST_RTP_TRY_PARTIAL; 03421 } 03422 03423 /* If both sides are not using the same method of DTMF transmission 03424 * (ie: one is RFC2833, other is INFO... then we can not do direct media. 03425 * -------------------------------------------------- 03426 * | DTMF Mode | HAS_DTMF | Accepts Begin Frames | 03427 * |-----------|------------|-----------------------| 03428 * | Inband | False | True | 03429 * | RFC2833 | True | True | 03430 * | SIP INFO | False | False | 03431 * -------------------------------------------------- 03432 * However, if DTMF from both channels is being monitored by the core, then 03433 * we can still do packet-to-packet bridging, because passing through the 03434 * core will handle DTMF mode translation. 03435 */ 03436 if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) || 03437 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) { 03438 if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) { 03439 ast_channel_unlock(c0); 03440 ast_channel_unlock(c1); 03441 return AST_BRIDGE_FAILED_NOWARN; 03442 } 03443 audio_p0_res = AST_RTP_TRY_PARTIAL; 03444 audio_p1_res = AST_RTP_TRY_PARTIAL; 03445 } 03446 03447 /* If we need to feed frames into the core don't do a P2P bridge */ 03448 if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) || 03449 (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) { 03450 ast_channel_unlock(c0); 03451 ast_channel_unlock(c1); 03452 return AST_BRIDGE_FAILED_NOWARN; 03453 } 03454 03455 /* Get codecs from both sides */ 03456 codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0; 03457 codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0; 03458 if (codec0 && codec1 && !(codec0 & codec1)) { 03459 /* Hey, we can't do native bridging if both parties speak different codecs */ 03460 if (option_debug) 03461 ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); 03462 ast_channel_unlock(c0); 03463 ast_channel_unlock(c1); 03464 return AST_BRIDGE_FAILED_NOWARN; 03465 } 03466 03467 /* If either side can only do a partial bridge, then don't try for a true native bridge */ 03468 if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) { 03469 struct ast_format_list fmt0, fmt1; 03470 03471 /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */ 03472 if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) { 03473 if (option_debug) 03474 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n"); 03475 ast_channel_unlock(c0); 03476 ast_channel_unlock(c1); 03477 return AST_BRIDGE_FAILED_NOWARN; 03478 } 03479 /* They must also be using the same packetization */ 03480 fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat); 03481 fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat); 03482 if (fmt0.cur_ms != fmt1.cur_ms) { 03483 if (option_debug) 03484 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n"); 03485 ast_channel_unlock(c0); 03486 ast_channel_unlock(c1); 03487 return AST_BRIDGE_FAILED_NOWARN; 03488 } 03489 03490 if (option_verbose > 2) 03491 ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name); 03492 res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1); 03493 } else { 03494 if (option_verbose > 2) 03495 ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name); 03496 res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1); 03497 } 03498 03499 return res; 03500 }
int ast_rtp_codec_getformat | ( | int | pt | ) |
Definition at line 2791 of file rtp.c.
References rtpPayloadType::code, and static_RTP_PT.
Referenced by process_sdp().
02792 { 02793 if (pt < 0 || pt > MAX_RTP_PT) 02794 return 0; /* bogus payload type */ 02795 02796 if (static_RTP_PT[pt].isAstFormat) 02797 return static_RTP_PT[pt].code; 02798 else 02799 return 0; 02800 }
struct ast_codec_pref* ast_rtp_codec_getpref | ( | struct ast_rtp * | rtp | ) |
Definition at line 2786 of file rtp.c.
References ast_rtp::pref.
Referenced by add_codec_to_sdp(), and process_sdp().
02787 { 02788 return &rtp->pref; 02789 }
int ast_rtp_codec_setpref | ( | struct ast_rtp * | rtp, | |
struct ast_codec_pref * | prefs | |||
) |
Definition at line 2739 of file rtp.c.
References ast_codec_pref_getsize(), ast_log(), ast_smoother_new(), ast_smoother_reconfigure(), ast_smoother_set_flags(), ast_format_list::cur_ms, ast_format_list::flags, ast_format_list::fr_len, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, option_debug, ast_rtp::pref, prefs, and ast_rtp::smoother.
Referenced by __oh323_rtp_create(), check_user_full(), create_addr_from_peer(), process_sdp(), register_verify(), set_peer_capabilities(), sip_alloc(), start_rtp(), and transmit_response_with_sdp().
02740 { 02741 struct ast_format_list current_format_old, current_format_new; 02742 02743 /* if no packets have been sent through this session yet, then 02744 * changing preferences does not require any extra work 02745 */ 02746 if (rtp->lasttxformat == 0) { 02747 rtp->pref = *prefs; 02748 return 0; 02749 } 02750 02751 current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat); 02752 02753 rtp->pref = *prefs; 02754 02755 current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat); 02756 02757 /* if the framing desired for the current format has changed, we may have to create 02758 * or adjust the smoother for this session 02759 */ 02760 if ((current_format_new.inc_ms != 0) && 02761 (current_format_new.cur_ms != current_format_old.cur_ms)) { 02762 int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms; 02763 02764 if (rtp->smoother) { 02765 ast_smoother_reconfigure(rtp->smoother, new_size); 02766 if (option_debug) { 02767 ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size); 02768 } 02769 } else { 02770 if (!(rtp->smoother = ast_smoother_new(new_size))) { 02771 ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size); 02772 return -1; 02773 } 02774 if (current_format_new.flags) { 02775 ast_smoother_set_flags(rtp->smoother, current_format_new.flags); 02776 } 02777 if (option_debug) { 02778 ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size); 02779 } 02780 } 02781 } 02782 02783 return 0; 02784 }
void ast_rtp_destroy | ( | struct ast_rtp * | rtp | ) |
Definition at line 2152 of file rtp.c.
References ast_io_remove(), ast_mutex_destroy(), AST_SCHED_DEL, ast_smoother_free(), ast_verbose(), ast_rtp::bridge_lock, ast_rtcp::expected_prior, free, ast_rtp::io, ast_rtp::ioid, ast_rtcp::received_prior, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtcp::rr_count, ast_rtp::rtcp, rtcp_debug_test_addr(), ast_rtcp::rtt, ast_rtp::rxcount, ast_rtp::rxjitter, ast_rtp::rxtransit, ast_rtcp::s, ast_rtp::s, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::smoother, ast_rtcp::sr_count, ast_rtp::ssrc, ast_rtp::them, ast_rtp::themssrc, and ast_rtp::txcount.
Referenced by __oh323_destroy(), __sip_destroy(), check_user_full(), cleanup_connection(), create_addr_from_peer(), destroy_endpoint(), gtalk_free_pvt(), mgcp_hangup(), oh323_alloc(), skinny_hangup(), start_rtp(), and unalloc_sub().
02153 { 02154 if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) { 02155 /*Print some info on the call here */ 02156 ast_verbose(" RTP-stats\n"); 02157 ast_verbose("* Our Receiver:\n"); 02158 ast_verbose(" SSRC: %u\n", rtp->themssrc); 02159 ast_verbose(" Received packets: %u\n", rtp->rxcount); 02160 ast_verbose(" Lost packets: %u\n", rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0); 02161 ast_verbose(" Jitter: %.4f\n", rtp->rxjitter); 02162 ast_verbose(" Transit: %.4f\n", rtp->rxtransit); 02163 ast_verbose(" RR-count: %u\n", rtp->rtcp ? rtp->rtcp->rr_count : 0); 02164 ast_verbose("* Our Sender:\n"); 02165 ast_verbose(" SSRC: %u\n", rtp->ssrc); 02166 ast_verbose(" Sent packets: %u\n", rtp->txcount); 02167 ast_verbose(" Lost packets: %u\n", rtp->rtcp ? rtp->rtcp->reported_lost : 0); 02168 ast_verbose(" Jitter: %u\n", rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int)65536.0) : 0); 02169 ast_verbose(" SR-count: %u\n", rtp->rtcp ? rtp->rtcp->sr_count : 0); 02170 ast_verbose(" RTT: %f\n", rtp->rtcp ? rtp->rtcp->rtt : 0); 02171 } 02172 02173 if (rtp->smoother) 02174 ast_smoother_free(rtp->smoother); 02175 if (rtp->ioid) 02176 ast_io_remove(rtp->io, rtp->ioid); 02177 if (rtp->s > -1) 02178 close(rtp->s); 02179 if (rtp->rtcp) { 02180 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02181 close(rtp->rtcp->s); 02182 free(rtp->rtcp); 02183 rtp->rtcp=NULL; 02184 } 02185 02186 ast_mutex_destroy(&rtp->bridge_lock); 02187 02188 free(rtp); 02189 }
int ast_rtp_early_bridge | ( | struct ast_channel * | dest, | |
struct ast_channel * | src | |||
) |
If possible, create an early bridge directly between the devices without having to send a re-invite later.
Definition at line 1494 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, ast_channel::name, option_debug, and ast_rtp_protocol::set_rtp_peer.
Referenced by wait_for_answer().
01495 { 01496 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01497 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01498 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01499 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01500 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01501 int srccodec, destcodec, nat_active = 0; 01502 01503 /* Lock channels */ 01504 ast_channel_lock(dest); 01505 if (src) { 01506 while(ast_channel_trylock(src)) { 01507 ast_channel_unlock(dest); 01508 usleep(1); 01509 ast_channel_lock(dest); 01510 } 01511 } 01512 01513 /* Find channel driver interfaces */ 01514 destpr = get_proto(dest); 01515 if (src) 01516 srcpr = get_proto(src); 01517 if (!destpr) { 01518 if (option_debug) 01519 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01520 ast_channel_unlock(dest); 01521 if (src) 01522 ast_channel_unlock(src); 01523 return 0; 01524 } 01525 if (!srcpr) { 01526 if (option_debug) 01527 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>"); 01528 ast_channel_unlock(dest); 01529 if (src) 01530 ast_channel_unlock(src); 01531 return 0; 01532 } 01533 01534 /* Get audio and video interface (if native bridge is possible) */ 01535 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01536 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01537 if (srcpr) { 01538 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01539 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01540 } 01541 01542 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01543 if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) { 01544 /* Somebody doesn't want to play... */ 01545 ast_channel_unlock(dest); 01546 if (src) 01547 ast_channel_unlock(src); 01548 return 0; 01549 } 01550 if (audio_src_res == AST_RTP_TRY_NATIVE && (video_src_res == AST_RTP_GET_FAILED || video_src_res == AST_RTP_TRY_NATIVE) && srcpr->get_codec) 01551 srccodec = srcpr->get_codec(src); 01552 else 01553 srccodec = 0; 01554 if (audio_dest_res == AST_RTP_TRY_NATIVE && (video_dest_res == AST_RTP_GET_FAILED || video_dest_res == AST_RTP_TRY_NATIVE) && destpr->get_codec) 01555 destcodec = destpr->get_codec(dest); 01556 else 01557 destcodec = 0; 01558 /* Ensure we have at least one matching codec */ 01559 if (srcp && !(srccodec & destcodec)) { 01560 ast_channel_unlock(dest); 01561 ast_channel_unlock(src); 01562 return 0; 01563 } 01564 /* Consider empty media as non-existant */ 01565 if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) 01566 srcp = NULL; 01567 /* If the client has NAT stuff turned on then just safe NAT is active */ 01568 if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01569 nat_active = 1; 01570 /* Bridge media early */ 01571 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active)) 01572 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01573 ast_channel_unlock(dest); 01574 if (src) 01575 ast_channel_unlock(src); 01576 if (option_debug) 01577 ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01578 return 1; 01579 }
int ast_rtp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 512 of file rtp.c.
References ast_rtp::s.
Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), mgcp_new(), sip_new(), skinny_new(), and start_rtp().
00513 { 00514 return rtp->s; 00515 }
Definition at line 2062 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, and ast_rtp::bridged.
Referenced by __sip_destroy(), and ast_rtp_read().
02063 { 02064 struct ast_rtp *bridged = NULL; 02065 02066 ast_mutex_lock(&rtp->bridge_lock); 02067 bridged = rtp->bridged; 02068 ast_mutex_unlock(&rtp->bridge_lock); 02069 02070 return bridged; 02071 }
void ast_rtp_get_current_formats | ( | struct ast_rtp * | rtp, | |
int * | astFormats, | |||
int * | nonAstFormats | |||
) |
Return the union of all of the codecs that were set by rtp_set...() calls They're returned as two distinct sets: AST_FORMATs, and AST_RTPs.
Definition at line 1715 of file rtp.c.
References ast_mutex_lock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, and MAX_RTP_PT.
Referenced by gtalk_is_answered(), gtalk_newcall(), and process_sdp().
01717 { 01718 int pt; 01719 01720 ast_mutex_lock(&rtp->bridge_lock); 01721 01722 *astFormats = *nonAstFormats = 0; 01723 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01724 if (rtp->current_RTP_PT[pt].isAstFormat) { 01725 *astFormats |= rtp->current_RTP_PT[pt].code; 01726 } else { 01727 *nonAstFormats |= rtp->current_RTP_PT[pt].code; 01728 } 01729 } 01730 01731 ast_mutex_unlock(&rtp->bridge_lock); 01732 01733 return; 01734 }
int ast_rtp_get_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2044 of file rtp.c.
References ast_rtp::them.
Referenced by add_sdp(), bridge_native_loop(), do_monitor(), gtalk_update_stun(), oh323_set_rtp_peer(), process_sdp(), sip_set_rtp_peer(), and transmit_modify_with_sdp().
02045 { 02046 if ((them->sin_family != AF_INET) || 02047 (them->sin_port != rtp->them.sin_port) || 02048 (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) { 02049 them->sin_family = AF_INET; 02050 them->sin_port = rtp->them.sin_port; 02051 them->sin_addr = rtp->them.sin_addr; 02052 return 1; 02053 } 02054 return 0; 02055 }
char* ast_rtp_get_quality | ( | struct ast_rtp * | rtp, | |
struct ast_rtp_quality * | qual | |||
) |
Return RTCP quality string.
Definition at line 2108 of file rtp.c.
References ast_rtcp::expected_prior, ast_rtp_quality::local_count, ast_rtp_quality::local_jitter, ast_rtp_quality::local_lostpackets, ast_rtp_quality::local_ssrc, ast_rtcp::quality, ast_rtcp::received_prior, ast_rtp_quality::remote_count, ast_rtp_quality::remote_jitter, ast_rtp_quality::remote_lostpackets, ast_rtp_quality::remote_ssrc, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, ast_rtcp::rtt, ast_rtp_quality::rtt, ast_rtp::rxcount, ast_rtp::rxjitter, ast_rtp::ssrc, ast_rtp::themssrc, and ast_rtp::txcount.
Referenced by acf_channel_read(), handle_request_bye(), and sip_hangup().
02109 { 02110 /* 02111 *ssrc our ssrc 02112 *themssrc their ssrc 02113 *lp lost packets 02114 *rxjitter our calculated jitter(rx) 02115 *rxcount no. received packets 02116 *txjitter reported jitter of the other end 02117 *txcount transmitted packets 02118 *rlp remote lost packets 02119 *rtt round trip time 02120 */ 02121 02122 if (qual && rtp) { 02123 qual->local_ssrc = rtp->ssrc; 02124 qual->local_jitter = rtp->rxjitter; 02125 qual->local_count = rtp->rxcount; 02126 qual->remote_ssrc = rtp->themssrc; 02127 qual->remote_count = rtp->txcount; 02128 if (rtp->rtcp) { 02129 qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior; 02130 qual->remote_lostpackets = rtp->rtcp->reported_lost; 02131 qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0; 02132 qual->rtt = rtp->rtcp->rtt; 02133 } 02134 } 02135 if (rtp->rtcp) { 02136 snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), 02137 "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", 02138 rtp->ssrc, 02139 rtp->themssrc, 02140 rtp->rtcp->expected_prior - rtp->rtcp->received_prior, 02141 rtp->rxjitter, 02142 rtp->rxcount, 02143 (double)rtp->rtcp->reported_jitter / 65536.0, 02144 rtp->txcount, 02145 rtp->rtcp->reported_lost, 02146 rtp->rtcp->rtt); 02147 return rtp->rtcp->quality; 02148 } else 02149 return "<Unknown> - RTP/RTCP has already been destroyed"; 02150 }
int ast_rtp_get_rtpholdtimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp hold timeout.
Definition at line 567 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by do_monitor().
00568 { 00569 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00570 return 0; 00571 return rtp->rtpholdtimeout; 00572 }
int ast_rtp_get_rtpkeepalive | ( | struct ast_rtp * | rtp | ) |
Get RTP keepalive interval.
Definition at line 575 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by do_monitor().
00576 { 00577 return rtp->rtpkeepalive; 00578 }
int ast_rtp_get_rtptimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp timeout.
Definition at line 559 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by do_monitor().
00560 { 00561 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00562 return 0; 00563 return rtp->rtptimeout; 00564 }
void ast_rtp_get_us | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | us | |||
) |
Definition at line 2057 of file rtp.c.
References ast_rtp::us.
Referenced by add_sdp(), external_rtp_create(), gtalk_create_candidates(), handle_open_receive_channel_ack_message(), and oh323_set_rtp_peer().
int ast_rtp_getnat | ( | struct ast_rtp * | rtp | ) |
Definition at line 595 of file rtp.c.
References ast_test_flag, and FLAG_NAT_ACTIVE.
Referenced by sip_get_rtp_peer().
00596 { 00597 return ast_test_flag(rtp, FLAG_NAT_ACTIVE); 00598 }
void ast_rtp_init | ( | void | ) |
Initialize the RTP system in Asterisk.
Definition at line 3885 of file rtp.c.
References ast_cli_register_multiple(), ast_rtp_reload(), and cli_rtp.
Referenced by main().
03886 { 03887 ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry)); 03888 ast_rtp_reload(); 03889 }
int ast_rtp_lookup_code | ( | struct ast_rtp * | rtp, | |
int | isAstFormat, | |||
int | code | |||
) |
Looks up an RTP code out of our *static* outbound list.
Definition at line 1758 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, 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 add_codec_to_answer(), add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), ast_rtp_sendcng(), ast_rtp_senddigit_begin(), ast_rtp_write(), and bridge_p2p_rtp_write().
01759 { 01760 int pt = 0; 01761 01762 ast_mutex_lock(&rtp->bridge_lock); 01763 01764 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && 01765 code == rtp->rtp_lookup_code_cache_code) { 01766 /* Use our cached mapping, to avoid the overhead of the loop below */ 01767 pt = rtp->rtp_lookup_code_cache_result; 01768 ast_mutex_unlock(&rtp->bridge_lock); 01769 return pt; 01770 } 01771 01772 /* Check the dynamic list first */ 01773 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01774 if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) { 01775 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01776 rtp->rtp_lookup_code_cache_code = code; 01777 rtp->rtp_lookup_code_cache_result = pt; 01778 ast_mutex_unlock(&rtp->bridge_lock); 01779 return pt; 01780 } 01781 } 01782 01783 /* Then the static list */ 01784 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01785 if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) { 01786 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01787 rtp->rtp_lookup_code_cache_code = code; 01788 rtp->rtp_lookup_code_cache_result = pt; 01789 ast_mutex_unlock(&rtp->bridge_lock); 01790 return pt; 01791 } 01792 } 01793 01794 ast_mutex_unlock(&rtp->bridge_lock); 01795 01796 return -1; 01797 }
char* ast_rtp_lookup_mime_multiple | ( | char * | buf, | |
size_t | size, | |||
const int | capability, | |||
const int | isAstFormat, | |||
enum ast_rtp_options | options | |||
) |
Build a string of MIME subtype names from a capability list.
Definition at line 1818 of file rtp.c.
References ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, format, len(), and name.
Referenced by process_sdp().
01820 { 01821 int format; 01822 unsigned len; 01823 char *end = buf; 01824 char *start = buf; 01825 01826 if (!buf || !size) 01827 return NULL; 01828 01829 snprintf(end, size, "0x%x (", capability); 01830 01831 len = strlen(end); 01832 end += len; 01833 size -= len; 01834 start = end; 01835 01836 for (format = 1; format < AST_RTP_MAX; format <<= 1) { 01837 if (capability & format) { 01838 const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options); 01839 01840 snprintf(end, size, "%s|", name); 01841 len = strlen(end); 01842 end += len; 01843 size -= len; 01844 } 01845 } 01846 01847 if (start == end) 01848 snprintf(start, size, "nothing)"); 01849 else if (size > 1) 01850 *(end -1) = ')'; 01851 01852 return buf; 01853 }
const char* ast_rtp_lookup_mime_subtype | ( | int | isAstFormat, | |
int | code, | |||
enum ast_rtp_options | options | |||
) |
Mapping an Asterisk code into a MIME subtype (string):.
Definition at line 1799 of file rtp.c.
References AST_FORMAT_G726_AAL2, AST_RTP_OPT_G726_NONSTANDARD, rtpPayloadType::code, mimeTypes, and payloadType.
Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), ast_rtp_lookup_mime_multiple(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
01801 { 01802 unsigned int i; 01803 01804 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01805 if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) { 01806 if (isAstFormat && 01807 (code == AST_FORMAT_G726_AAL2) && 01808 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01809 return "G726-32"; 01810 else 01811 return mimeTypes[i].subtype; 01812 } 01813 } 01814 01815 return ""; 01816 }
struct rtpPayloadType ast_rtp_lookup_pt | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Mapping between RTP payload format codes and Asterisk codes:.
Definition at line 1736 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), rtpPayloadType::isAstFormat, MAX_RTP_PT, and static_RTP_PT.
Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and setup_rtp_connection().
01737 { 01738 struct rtpPayloadType result; 01739 01740 result.isAstFormat = result.code = 0; 01741 01742 if (pt < 0 || pt > MAX_RTP_PT) 01743 return result; /* bogus payload type */ 01744 01745 /* Start with negotiated codecs */ 01746 ast_mutex_lock(&rtp->bridge_lock); 01747 result = rtp->current_RTP_PT[pt]; 01748 ast_mutex_unlock(&rtp->bridge_lock); 01749 01750 /* If it doesn't exist, check our static RTP type list, just in case */ 01751 if (!result.code) 01752 result = static_RTP_PT[pt]; 01753 01754 return result; 01755 }
int ast_rtp_make_compatible | ( | struct ast_channel * | dest, | |
struct ast_channel * | src, | |||
int | media | |||
) |
Definition at line 1581 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, ast_rtp_pt_copy(), AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, ast_channel::name, option_debug, and ast_rtp_protocol::set_rtp_peer.
Referenced by wait_for_answer().
01582 { 01583 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01584 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01585 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01586 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01587 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01588 int srccodec, destcodec; 01589 01590 /* Lock channels */ 01591 ast_channel_lock(dest); 01592 while(ast_channel_trylock(src)) { 01593 ast_channel_unlock(dest); 01594 usleep(1); 01595 ast_channel_lock(dest); 01596 } 01597 01598 /* Find channel driver interfaces */ 01599 if (!(destpr = get_proto(dest))) { 01600 if (option_debug) 01601 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01602 ast_channel_unlock(dest); 01603 ast_channel_unlock(src); 01604 return 0; 01605 } 01606 if (!(srcpr = get_proto(src))) { 01607 if (option_debug) 01608 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name); 01609 ast_channel_unlock(dest); 01610 ast_channel_unlock(src); 01611 return 0; 01612 } 01613 01614 /* Get audio and video interface (if native bridge is possible) */ 01615 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01616 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01617 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01618 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01619 01620 /* Ensure we have at least one matching codec */ 01621 if (srcpr->get_codec) 01622 srccodec = srcpr->get_codec(src); 01623 else 01624 srccodec = 0; 01625 if (destpr->get_codec) 01626 destcodec = destpr->get_codec(dest); 01627 else 01628 destcodec = 0; 01629 01630 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01631 if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) { 01632 /* Somebody doesn't want to play... */ 01633 ast_channel_unlock(dest); 01634 ast_channel_unlock(src); 01635 return 0; 01636 } 01637 ast_rtp_pt_copy(destp, srcp); 01638 if (vdestp && vsrcp) 01639 ast_rtp_pt_copy(vdestp, vsrcp); 01640 if (media) { 01641 /* Bridge early */ 01642 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01643 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name); 01644 } 01645 ast_channel_unlock(dest); 01646 ast_channel_unlock(src); 01647 if (option_debug) 01648 ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name); 01649 return 1; 01650 }
struct ast_rtp* ast_rtp_new | ( | struct sched_context * | sched, | |
struct io_context * | io, | |||
int | rtcpenable, | |||
int | callbackmode | |||
) |
Initializate a RTP session.
sched | ||
io | ||
rtcpenable | ||
callbackmode |
Definition at line 2008 of file rtp.c.
References ast_rtp_new_with_bindaddr(), io, and sched.
02009 { 02010 struct in_addr ia; 02011 02012 memset(&ia, 0, sizeof(ia)); 02013 return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia); 02014 }
void ast_rtp_new_init | ( | struct ast_rtp * | rtp | ) |
Initialize a new RTP structure.
Definition at line 1902 of file rtp.c.
References ast_mutex_init(), ast_random(), ast_set_flag, ast_rtp::bridge_lock, FLAG_HAS_DTMF, ast_rtp::seqno, ast_rtp::ssrc, ast_rtp::them, and ast_rtp::us.
Referenced by ast_rtp_new_with_bindaddr(), and process_sdp().
01903 { 01904 ast_mutex_init(&rtp->bridge_lock); 01905 01906 rtp->them.sin_family = AF_INET; 01907 rtp->us.sin_family = AF_INET; 01908 rtp->ssrc = ast_random(); 01909 rtp->seqno = ast_random() & 0xffff; 01910 ast_set_flag(rtp, FLAG_HAS_DTMF); 01911 01912 return; 01913 }
void ast_rtp_new_source | ( | struct ast_rtp * | rtp | ) |
Definition at line 2025 of file rtp.c.
References ast_rtp::set_marker_bit.
Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), sip_write(), and skinny_indicate().
struct ast_rtp* ast_rtp_new_with_bindaddr | ( | struct sched_context * | sched, | |
struct io_context * | io, | |||
int | rtcpenable, | |||
int | callbackmode, | |||
struct in_addr | in | |||
) |
Initializate a RTP session using an in_addr structure.
This fuction gets called by ast_rtp_new().
sched | ||
io | ||
rtcpenable | ||
callbackmode | ||
in |
Definition at line 1915 of file rtp.c.
References ast_calloc, ast_log(), ast_random(), ast_rtcp_new(), ast_rtp_new_init(), errno, first, free, LOG_DEBUG, LOG_ERROR, option_debug, rtp_socket(), and sched.
Referenced by __oh323_rtp_create(), ast_rtp_new(), gtalk_alloc(), sip_alloc(), and start_rtp().
01916 { 01917 struct ast_rtp *rtp; 01918 int x; 01919 int first; 01920 int startplace; 01921 01922 if (!(rtp = ast_calloc(1, sizeof(*rtp)))) 01923 return NULL; 01924 01925 ast_rtp_new_init(rtp); 01926 01927 rtp->s = rtp_socket(); 01928 if (option_debug > 2) 01929 ast_log(LOG_DEBUG, "socket RTP fd: %i\n", rtp->s); 01930 if (rtp->s < 0) { 01931 free(rtp); 01932 ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno)); 01933 return NULL; 01934 } 01935 if (sched && rtcpenable) { 01936 rtp->sched = sched; 01937 rtp->rtcp = ast_rtcp_new(); 01938 if (option_debug > 2) 01939 ast_log(LOG_DEBUG, "socket RTCP fd: %i\n", rtp->rtcp->s); 01940 } 01941 01942 /* Select a random port number in the range of possible RTP */ 01943 x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart; 01944 x = x & ~1; 01945 /* Save it for future references. */ 01946 startplace = x; 01947 /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */ 01948 for (;;) { 01949 /* Must be an even port number by RTP spec */ 01950 rtp->us.sin_port = htons(x); 01951 rtp->us.sin_addr = addr; 01952 /* If there's rtcp, initialize it as well. */ 01953 if (rtp->rtcp) { 01954 rtp->rtcp->us.sin_port = htons(x + 1); 01955 rtp->rtcp->us.sin_addr = addr; 01956 } 01957 /* Try to bind it/them. */ 01958 if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) && 01959 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) 01960 break; 01961 if (!first) { 01962 /* Primary bind succeeded! Gotta recreate it */ 01963 close(rtp->s); 01964 rtp->s = rtp_socket(); 01965 if (option_debug > 2) 01966 ast_log(LOG_DEBUG, "socket RTP2 fd: %i\n", rtp->s); 01967 } 01968 if (errno != EADDRINUSE) { 01969 /* We got an error that wasn't expected, abort! */ 01970 ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); 01971 close(rtp->s); 01972 if (rtp->rtcp) { 01973 close(rtp->rtcp->s); 01974 free(rtp->rtcp); 01975 } 01976 free(rtp); 01977 return NULL; 01978 } 01979 /* The port was used, increment it (by two). */ 01980 x += 2; 01981 /* Did we go over the limit ? */ 01982 if (x > rtpend) 01983 /* then, start from the begingig. */ 01984 x = (rtpstart + 1) & ~1; 01985 /* Check if we reached the place were we started. */ 01986 if (x == startplace) { 01987 /* If so, there's no ports available. */ 01988 ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); 01989 close(rtp->s); 01990 if (rtp->rtcp) { 01991 close(rtp->rtcp->s); 01992 free(rtp->rtcp); 01993 } 01994 free(rtp); 01995 return NULL; 01996 } 01997 } 01998 rtp->sched = sched; 01999 rtp->io = io; 02000 if (callbackmode) { 02001 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); 02002 ast_set_flag(rtp, FLAG_CALLBACK_MODE); 02003 } 02004 ast_rtp_pt_default(rtp); 02005 return rtp; 02006 }
int ast_rtp_proto_register | ( | struct ast_rtp_protocol * | proto | ) |
Register interface to channel driver.
Definition at line 2902 of file rtp.c.
References AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_log(), ast_rtp_protocol::list, LOG_WARNING, and ast_rtp_protocol::type.
Referenced by load_module().
02903 { 02904 struct ast_rtp_protocol *cur; 02905 02906 AST_LIST_LOCK(&protos); 02907 AST_LIST_TRAVERSE(&protos, cur, list) { 02908 if (!strcmp(cur->type, proto->type)) { 02909 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); 02910 AST_LIST_UNLOCK(&protos); 02911 return -1; 02912 } 02913 } 02914 AST_LIST_INSERT_HEAD(&protos, proto, list); 02915 AST_LIST_UNLOCK(&protos); 02916 02917 return 0; 02918 }
void ast_rtp_proto_unregister | ( | struct ast_rtp_protocol * | proto | ) |
Unregister interface to channel driver.
Definition at line 2894 of file rtp.c.
References AST_LIST_LOCK, AST_LIST_REMOVE, and AST_LIST_UNLOCK.
Referenced by load_module(), and unload_module().
02895 { 02896 AST_LIST_LOCK(&protos); 02897 AST_LIST_REMOVE(&protos, proto, list); 02898 AST_LIST_UNLOCK(&protos); 02899 }
void ast_rtp_pt_clear | ( | struct ast_rtp * | rtp | ) |
Setting RTP payload types from lines in a SDP description:.
Definition at line 1418 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, 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 gtalk_alloc(), and process_sdp().
01419 { 01420 int i; 01421 01422 if (!rtp) 01423 return; 01424 01425 ast_mutex_lock(&rtp->bridge_lock); 01426 01427 for (i = 0; i < MAX_RTP_PT; ++i) { 01428 rtp->current_RTP_PT[i].isAstFormat = 0; 01429 rtp->current_RTP_PT[i].code = 0; 01430 } 01431 01432 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01433 rtp->rtp_lookup_code_cache_code = 0; 01434 rtp->rtp_lookup_code_cache_result = 0; 01435 01436 ast_mutex_unlock(&rtp->bridge_lock); 01437 }
Copy payload types between RTP structures.
Definition at line 1458 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, 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_make_compatible(), and process_sdp().
01459 { 01460 unsigned int i; 01461 01462 ast_mutex_lock(&dest->bridge_lock); 01463 ast_mutex_lock(&src->bridge_lock); 01464 01465 for (i=0; i < MAX_RTP_PT; ++i) { 01466 dest->current_RTP_PT[i].isAstFormat = 01467 src->current_RTP_PT[i].isAstFormat; 01468 dest->current_RTP_PT[i].code = 01469 src->current_RTP_PT[i].code; 01470 } 01471 dest->rtp_lookup_code_cache_isAstFormat = 0; 01472 dest->rtp_lookup_code_cache_code = 0; 01473 dest->rtp_lookup_code_cache_result = 0; 01474 01475 ast_mutex_unlock(&src->bridge_lock); 01476 ast_mutex_unlock(&dest->bridge_lock); 01477 }
void ast_rtp_pt_default | ( | struct ast_rtp * | rtp | ) |
Set payload types to defaults.
Definition at line 1439 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, 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, ast_rtp::rtp_lookup_code_cache_result, and static_RTP_PT.
01440 { 01441 int i; 01442 01443 ast_mutex_lock(&rtp->bridge_lock); 01444 01445 /* Initialize to default payload types */ 01446 for (i = 0; i < MAX_RTP_PT; ++i) { 01447 rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat; 01448 rtp->current_RTP_PT[i].code = static_RTP_PT[i].code; 01449 } 01450 01451 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01452 rtp->rtp_lookup_code_cache_code = 0; 01453 rtp->rtp_lookup_code_cache_result = 0; 01454 01455 ast_mutex_unlock(&rtp->bridge_lock); 01456 }
Definition at line 1108 of file rtp.c.
References ast_assert, ast_codec_get_samples(), AST_FORMAT_MAX_AUDIO, ast_format_rate(), AST_FORMAT_SLINEAR, ast_frame_byteswap_be, AST_FRAME_DTMF_END, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_get_bridged(), ast_rtp_lookup_pt(), ast_rtp_senddigit_continuation(), ast_sched_add(), ast_set_flag, ast_verbose(), bridge_p2p_rtp_write(), ast_rtp::bridged, calc_rxstamp(), rtpPayloadType::code, ast_rtp::cycles, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp::dtmfcount, errno, ext, ast_rtp::f, f, FLAG_NAT_ACTIVE, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lastevent, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len(), LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_rtp::rawdata, ast_rtp::resp, ast_rtp::rtcp, rtp_debug_test_addr(), RTP_SEQ_MOD, ast_rtp::rxcount, ast_rtp::rxseqno, ast_rtp::rxssrc, ast_rtcp::s, ast_rtp::s, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, send_dtmf(), ast_rtp::sending_digit, ast_frame::seqno, ast_frame::src, STUN_ACCEPT, stun_handle_packet(), ast_frame::subclass, ast_rtcp::them, ast_rtp::them, ast_rtp::themssrc, and ast_frame::ts.
Referenced by gtalk_rtp_read(), mgcp_rtp_read(), oh323_rtp_read(), rtpread(), sip_rtp_read(), and skinny_rtp_read().
01109 { 01110 int res; 01111 struct sockaddr_in sin; 01112 socklen_t len; 01113 unsigned int seqno; 01114 int version; 01115 int payloadtype; 01116 int hdrlen = 12; 01117 int padding; 01118 int mark; 01119 int ext; 01120 int cc; 01121 unsigned int ssrc; 01122 unsigned int timestamp; 01123 unsigned int *rtpheader; 01124 struct rtpPayloadType rtpPT; 01125 struct ast_rtp *bridged = NULL; 01126 01127 /* If time is up, kill it */ 01128 if (rtp->sending_digit) 01129 ast_rtp_senddigit_continuation(rtp); 01130 01131 len = sizeof(sin); 01132 01133 /* Cache where the header will go */ 01134 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 01135 0, (struct sockaddr *)&sin, &len); 01136 if (option_debug > 3) 01137 ast_log(LOG_DEBUG, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 01138 01139 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); 01140 if (res < 0) { 01141 ast_assert(errno != EBADF); 01142 if (errno != EAGAIN) { 01143 ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno)); 01144 ast_log(LOG_WARNING, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 01145 return NULL; 01146 } 01147 return &ast_null_frame; 01148 } 01149 01150 if (res < hdrlen) { 01151 ast_log(LOG_WARNING, "RTP Read too short\n"); 01152 return &ast_null_frame; 01153 } 01154 01155 /* Get fields */ 01156 seqno = ntohl(rtpheader[0]); 01157 01158 /* Check RTP version */ 01159 version = (seqno & 0xC0000000) >> 30; 01160 if (!version) { 01161 if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) && 01162 (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) { 01163 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 01164 } 01165 return &ast_null_frame; 01166 } 01167 01168 #if 0 /* Allow to receive RTP stream with closed transmission path */ 01169 /* If we don't have the other side's address, then ignore this */ 01170 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 01171 return &ast_null_frame; 01172 #endif 01173 01174 /* Send to whoever send to us if NAT is turned on */ 01175 if (rtp->nat) { 01176 if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 01177 (rtp->them.sin_port != sin.sin_port)) { 01178 rtp->them = sin; 01179 if (rtp->rtcp) { 01180 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 01181 rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1); 01182 } 01183 rtp->rxseqno = 0; 01184 ast_set_flag(rtp, FLAG_NAT_ACTIVE); 01185 if (option_debug || rtpdebug) 01186 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)); 01187 } 01188 } 01189 01190 /* If we are bridged to another RTP stream, send direct */ 01191 if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen)) 01192 return &ast_null_frame; 01193 01194 if (version != 2) 01195 return &ast_null_frame; 01196 01197 payloadtype = (seqno & 0x7f0000) >> 16; 01198 padding = seqno & (1 << 29); 01199 mark = seqno & (1 << 23); 01200 ext = seqno & (1 << 28); 01201 cc = (seqno & 0xF000000) >> 24; 01202 seqno &= 0xffff; 01203 timestamp = ntohl(rtpheader[1]); 01204 ssrc = ntohl(rtpheader[2]); 01205 01206 if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) { 01207 if (option_debug || rtpdebug) 01208 ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n"); 01209 mark = 1; 01210 } 01211 01212 rtp->rxssrc = ssrc; 01213 01214 if (padding) { 01215 /* Remove padding bytes */ 01216 res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; 01217 } 01218 01219 if (cc) { 01220 /* CSRC fields present */ 01221 hdrlen += cc*4; 01222 } 01223 01224 if (ext) { 01225 /* RTP Extension present */ 01226 hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2; 01227 hdrlen += 4; 01228 } 01229 01230 if (res < hdrlen) { 01231 ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); 01232 return &ast_null_frame; 01233 } 01234 01235 rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */ 01236 01237 if (rtp->rxcount==1) { 01238 /* This is the first RTP packet successfully received from source */ 01239 rtp->seedrxseqno = seqno; 01240 } 01241 01242 /* Do not schedule RR if RTCP isn't run */ 01243 if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) { 01244 /* Schedule transmission of Receiver Report */ 01245 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); 01246 } 01247 if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */ 01248 rtp->cycles += RTP_SEQ_MOD; 01249 01250 rtp->lastrxseqno = seqno; 01251 01252 if (rtp->themssrc==0) 01253 rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */ 01254 01255 if (rtp_debug_test_addr(&sin)) 01256 ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 01257 ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); 01258 01259 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); 01260 if (!rtpPT.isAstFormat) { 01261 struct ast_frame *f = NULL; 01262 01263 /* This is special in-band data that's not one of our codecs */ 01264 if (rtpPT.code == AST_RTP_DTMF) { 01265 /* It's special -- rfc2833 process it */ 01266 if (rtp_debug_test_addr(&sin)) { 01267 unsigned char *data; 01268 unsigned int event; 01269 unsigned int event_end; 01270 unsigned int duration; 01271 data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen; 01272 event = ntohl(*((unsigned int *)(data))); 01273 event >>= 24; 01274 event_end = ntohl(*((unsigned int *)(data))); 01275 event_end <<= 8; 01276 event_end >>= 24; 01277 duration = ntohl(*((unsigned int *)(data))); 01278 duration &= 0xFFFF; 01279 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); 01280 } 01281 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp); 01282 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { 01283 /* It's really special -- process it the Cisco way */ 01284 if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) { 01285 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01286 rtp->lastevent = seqno; 01287 } 01288 } else if (rtpPT.code == AST_RTP_CN) { 01289 /* Comfort Noise */ 01290 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01291 } else { 01292 ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr)); 01293 } 01294 return f ? f : &ast_null_frame; 01295 } 01296 rtp->lastrxformat = rtp->f.subclass = rtpPT.code; 01297 rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO; 01298 01299 rtp->rxseqno = seqno; 01300 01301 if (rtp->dtmfcount) { 01302 rtp->dtmfcount -= (timestamp - rtp->lastrxts); 01303 01304 if (rtp->dtmfcount < 0) { 01305 rtp->dtmfcount = 0; 01306 } 01307 01308 if (rtp->resp && !rtp->dtmfcount) { 01309 struct ast_frame *f; 01310 f = send_dtmf(rtp, AST_FRAME_DTMF_END); 01311 rtp->resp = 0; 01312 return f; 01313 } 01314 } 01315 01316 /* Record received timestamp as last received now */ 01317 rtp->lastrxts = timestamp; 01318 01319 rtp->f.mallocd = 0; 01320 rtp->f.datalen = res - hdrlen; 01321 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; 01322 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; 01323 rtp->f.seqno = seqno; 01324 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { 01325 rtp->f.samples = ast_codec_get_samples(&rtp->f); 01326 if (rtp->f.subclass == AST_FORMAT_SLINEAR) 01327 ast_frame_byteswap_be(&rtp->f); 01328 calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark); 01329 /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */ 01330 ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO); 01331 rtp->f.ts = timestamp / 8; 01332 rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000); 01333 } else { 01334 /* Video -- samples is # of samples vs. 90000 */ 01335 if (!rtp->lastividtimestamp) 01336 rtp->lastividtimestamp = timestamp; 01337 rtp->f.samples = timestamp - rtp->lastividtimestamp; 01338 rtp->lastividtimestamp = timestamp; 01339 rtp->f.delivery.tv_sec = 0; 01340 rtp->f.delivery.tv_usec = 0; 01341 if (mark) 01342 rtp->f.subclass |= 0x1; 01343 } 01344 rtp->f.src = "RTP"; 01345 return &rtp->f; 01346 }
int ast_rtp_reload | ( | void | ) |
Definition at line 3820 of file rtp.c.
References ast_config_destroy(), ast_config_load(), ast_false(), ast_log(), ast_variable_retrieve(), ast_verbose(), DEFAULT_DTMF_TIMEOUT, LOG_WARNING, option_verbose, RTCP_MAX_INTERVALMS, RTCP_MIN_INTERVALMS, s, and VERBOSE_PREFIX_2.
Referenced by ast_rtp_init().
03821 { 03822 struct ast_config *cfg; 03823 const char *s; 03824 03825 rtpstart = 5000; 03826 rtpend = 31000; 03827 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03828 cfg = ast_config_load("rtp.conf"); 03829 if (cfg) { 03830 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { 03831 rtpstart = atoi(s); 03832 if (rtpstart < 1024) 03833 rtpstart = 1024; 03834 if (rtpstart > 65535) 03835 rtpstart = 65535; 03836 } 03837 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { 03838 rtpend = atoi(s); 03839 if (rtpend < 1024) 03840 rtpend = 1024; 03841 if (rtpend > 65535) 03842 rtpend = 65535; 03843 } 03844 if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) { 03845 rtcpinterval = atoi(s); 03846 if (rtcpinterval == 0) 03847 rtcpinterval = 0; /* Just so we're clear... it's zero */ 03848 if (rtcpinterval < RTCP_MIN_INTERVALMS) 03849 rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */ 03850 if (rtcpinterval > RTCP_MAX_INTERVALMS) 03851 rtcpinterval = RTCP_MAX_INTERVALMS; 03852 } 03853 if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) { 03854 #ifdef SO_NO_CHECK 03855 if (ast_false(s)) 03856 nochecksums = 1; 03857 else 03858 nochecksums = 0; 03859 #else 03860 if (ast_false(s)) 03861 ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n"); 03862 #endif 03863 } 03864 if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { 03865 dtmftimeout = atoi(s); 03866 if ((dtmftimeout < 0) || (dtmftimeout > 20000)) { 03867 ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", 03868 dtmftimeout, DEFAULT_DTMF_TIMEOUT); 03869 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03870 }; 03871 } 03872 ast_config_destroy(cfg); 03873 } 03874 if (rtpstart >= rtpend) { 03875 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n"); 03876 rtpstart = 5000; 03877 rtpend = 31000; 03878 } 03879 if (option_verbose > 1) 03880 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); 03881 return 0; 03882 }
void ast_rtp_reset | ( | struct ast_rtp * | rtp | ) |
Definition at line 2089 of file rtp.c.
References ast_rtp::dtmfcount, ast_rtp::dtmfmute, ast_rtp::lastdigitts, ast_rtp::lastevent, ast_rtp::lasteventseqn, ast_rtp::lastividtimestamp, ast_rtp::lastovidtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxts, ast_rtp::lastts, ast_rtp::lasttxformat, ast_rtp::rxcore, ast_rtp::rxseqno, ast_rtp::seqno, and ast_rtp::txcore.
02090 { 02091 memset(&rtp->rxcore, 0, sizeof(rtp->rxcore)); 02092 memset(&rtp->txcore, 0, sizeof(rtp->txcore)); 02093 memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute)); 02094 rtp->lastts = 0; 02095 rtp->lastdigitts = 0; 02096 rtp->lastrxts = 0; 02097 rtp->lastividtimestamp = 0; 02098 rtp->lastovidtimestamp = 0; 02099 rtp->lasteventseqn = 0; 02100 rtp->lastevent = 0; 02101 rtp->lasttxformat = 0; 02102 rtp->lastrxformat = 0; 02103 rtp->dtmfcount = 0; 02104 rtp->seqno = 0; 02105 rtp->rxseqno = 0; 02106 }
int ast_rtp_sendcng | ( | struct ast_rtp * | rtp, | |
int | level | |||
) |
generate comfort noice (CNG)
Definition at line 2603 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_CN, ast_rtp_lookup_code(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), ast_rtp::data, ast_rtp::dtmfmute, errno, ast_rtp::lastts, LOG_ERROR, rtp_debug_test_addr(), ast_rtp::s, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.
Referenced by do_monitor().
02604 { 02605 unsigned int *rtpheader; 02606 int hdrlen = 12; 02607 int res; 02608 int payload; 02609 char data[256]; 02610 level = 127 - (level & 0x7f); 02611 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN); 02612 02613 /* If we have no peer, return immediately */ 02614 if (!rtp->them.sin_addr.s_addr) 02615 return 0; 02616 02617 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02618 02619 /* Get a pointer to the header */ 02620 rtpheader = (unsigned int *)data; 02621 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++)); 02622 rtpheader[1] = htonl(rtp->lastts); 02623 rtpheader[2] = htonl(rtp->ssrc); 02624 data[12] = level; 02625 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { 02626 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); 02627 if (res <0) 02628 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)); 02629 if (rtp_debug_test_addr(&rtp->them)) 02630 ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n" 02631 , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); 02632 02633 } 02634 return 0; 02635 }
int ast_rtp_senddigit_begin | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
Send begin frames for DTMF.
Definition at line 2211 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), ast_rtp::dtmfmute, errno, ast_rtp::lastdigitts, ast_rtp::lastts, LOG_ERROR, LOG_WARNING, rtp_debug_test_addr(), ast_rtp::s, ast_rtp::send_digit, ast_rtp::send_duration, ast_rtp::send_payload, ast_rtp::sending_digit, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.
Referenced by mgcp_senddigit_begin(), oh323_digit_begin(), and sip_senddigit_begin().
02212 { 02213 unsigned int *rtpheader; 02214 int hdrlen = 12, res = 0, i = 0, payload = 0; 02215 char data[256]; 02216 02217 if ((digit <= '9') && (digit >= '0')) 02218 digit -= '0'; 02219 else if (digit == '*') 02220 digit = 10; 02221 else if (digit == '#') 02222 digit = 11; 02223 else if ((digit >= 'A') && (digit <= 'D')) 02224 digit = digit - 'A' + 12; 02225 else if ((digit >= 'a') && (digit <= 'd')) 02226 digit = digit - 'a' + 12; 02227 else { 02228 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); 02229 return 0; 02230 } 02231 02232 /* If we have no peer, return immediately */ 02233 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 02234 return 0; 02235 02236 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); 02237 02238 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02239 rtp->send_duration = 160; 02240 rtp->lastdigitts = rtp->lastts + rtp->send_duration; 02241 02242 /* Get a pointer to the header */ 02243 rtpheader = (unsigned int *)data; 02244 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno)); 02245 rtpheader[1] = htonl(rtp->lastdigitts); 02246 rtpheader[2] = htonl(rtp->ssrc); 02247 02248 for (i = 0; i < 2; i++) { 02249 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration)); 02250 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); 02251 if (res < 0) 02252 ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n", 02253 ast_inet_ntoa(rtp->them.sin_addr), 02254 ntohs(rtp->them.sin_port), strerror(errno)); 02255 if (rtp_debug_test_addr(&rtp->them)) 02256 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 02257 ast_inet_ntoa(rtp->them.sin_addr), 02258 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 02259 /* Increment sequence number */ 02260 rtp->seqno++; 02261 /* Increment duration */ 02262 rtp->send_duration += 160; 02263 /* Clear marker bit and set seqno */ 02264 rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); 02265 } 02266 02267 /* Since we received a begin, we can safely store the digit and disable any compensation */ 02268 rtp->sending_digit = 1; 02269 rtp->send_digit = digit; 02270 rtp->send_payload = payload; 02271 02272 return 0; 02273 }
int ast_rtp_senddigit_end | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
void ast_rtp_set_callback | ( | struct ast_rtp * | rtp, | |
ast_rtp_callback | callback | |||
) |
Definition at line 585 of file rtp.c.
References ast_rtp::callback.
Referenced by start_rtp().
00586 { 00587 rtp->callback = callback; 00588 }
void ast_rtp_set_data | ( | struct ast_rtp * | rtp, | |
void * | data | |||
) |
Definition at line 580 of file rtp.c.
References ast_rtp::data.
Referenced by start_rtp().
00581 { 00582 rtp->data = data; 00583 }
void ast_rtp_set_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Activate payload type.
Definition at line 1656 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, ast_rtp::current_RTP_PT, MAX_RTP_PT, and static_RTP_PT.
Referenced by gtalk_is_answered(), gtalk_newcall(), and process_sdp().
01657 { 01658 if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 01659 return; /* bogus payload type */ 01660 01661 ast_mutex_lock(&rtp->bridge_lock); 01662 rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; 01663 ast_mutex_unlock(&rtp->bridge_lock); 01664 }
void ast_rtp_set_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2033 of file rtp.c.
References ast_rtp::rtcp, ast_rtp::rxseqno, ast_rtcp::them, and ast_rtp::them.
Referenced by handle_open_receive_channel_ack_message(), process_sdp(), and setup_rtp_connection().
02034 { 02035 rtp->them.sin_port = them->sin_port; 02036 rtp->them.sin_addr = them->sin_addr; 02037 if (rtp->rtcp) { 02038 rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1); 02039 rtp->rtcp->them.sin_addr = them->sin_addr; 02040 } 02041 rtp->rxseqno = 0; 02042 }
void ast_rtp_set_rtpholdtimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp hold timeout.
Definition at line 547 of file rtp.c.
References ast_rtp::rtpholdtimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00548 { 00549 rtp->rtpholdtimeout = timeout; 00550 }
void ast_rtp_set_rtpkeepalive | ( | struct ast_rtp * | rtp, | |
int | period | |||
) |
set RTP keepalive interval
Definition at line 553 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by create_addr_from_peer(), and sip_alloc().
00554 { 00555 rtp->rtpkeepalive = period; 00556 }
int ast_rtp_set_rtpmap_type | ( | struct ast_rtp * | rtp, | |
int | pt, | |||
char * | mimeType, | |||
char * | mimeSubtype, | |||
enum ast_rtp_options | options | |||
) |
Initiate payload type to a known MIME media type for a codec.
Definition at line 1683 of file rtp.c.
References AST_FORMAT_G726, AST_FORMAT_G726_AAL2, ast_mutex_lock(), ast_mutex_unlock(), AST_RTP_OPT_G726_NONSTANDARD, ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, MAX_RTP_PT, mimeTypes, payloadType, subtype, and type.
Referenced by __oh323_rtp_create(), gtalk_is_answered(), gtalk_newcall(), process_sdp(), and set_dtmf_payload().
01686 { 01687 unsigned int i; 01688 int found = 0; 01689 01690 if (pt < 0 || pt > MAX_RTP_PT) 01691 return -1; /* bogus payload type */ 01692 01693 ast_mutex_lock(&rtp->bridge_lock); 01694 01695 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01696 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && 01697 strcasecmp(mimeType, mimeTypes[i].type) == 0) { 01698 found = 1; 01699 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; 01700 if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) && 01701 mimeTypes[i].payloadType.isAstFormat && 01702 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01703 rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2; 01704 break; 01705 } 01706 } 01707 01708 ast_mutex_unlock(&rtp->bridge_lock); 01709 01710 return (found ? 0 : -1); 01711 }
void ast_rtp_set_rtptimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp timeout.
Definition at line 541 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00542 { 00543 rtp->rtptimeout = timeout; 00544 }
void ast_rtp_set_rtptimers_onhold | ( | struct ast_rtp * | rtp | ) |
Definition at line 534 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by handle_response_invite().
00535 { 00536 rtp->rtptimeout = (-1) * rtp->rtptimeout; 00537 rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout; 00538 }
void ast_rtp_setdtmf | ( | struct ast_rtp * | rtp, | |
int | dtmf | |||
) |
Indicate whether this RTP session is carrying DTMF or not.
Definition at line 600 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_DTMF.
Referenced by create_addr_from_peer(), handle_request_invite(), process_sdp(), sip_alloc(), and sip_dtmfmode().
00601 { 00602 ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF); 00603 }
void ast_rtp_setdtmfcompensate | ( | struct ast_rtp * | rtp, | |
int | compensate | |||
) |
Compensate for devices that send RFC2833 packets all at once.
Definition at line 605 of file rtp.c.
References ast_set2_flag, and FLAG_DTMF_COMPENSATE.
Referenced by create_addr_from_peer(), handle_request_invite(), process_sdp(), and sip_alloc().
00606 { 00607 ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE); 00608 }
void ast_rtp_setnat | ( | struct ast_rtp * | rtp, | |
int | nat | |||
) |
Definition at line 590 of file rtp.c.
References ast_rtp::nat.
Referenced by __oh323_rtp_create(), do_setnat(), oh323_rtp_read(), and start_rtp().
void ast_rtp_setstun | ( | struct ast_rtp * | rtp, | |
int | stun_enable | |||
) |
Enable STUN capability.
Definition at line 610 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_STUN.
Referenced by gtalk_new().
00611 { 00612 ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN); 00613 }
int ast_rtp_settos | ( | struct ast_rtp * | rtp, | |
int | tos | |||
) |
Definition at line 2016 of file rtp.c.
References ast_log(), LOG_WARNING, and ast_rtp::s.
Referenced by __oh323_rtp_create(), and sip_alloc().
02017 { 02018 int res; 02019 02020 if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 02021 ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); 02022 return res; 02023 }
void ast_rtp_stop | ( | struct ast_rtp * | rtp | ) |
Definition at line 2073 of file rtp.c.
References ast_clear_flag, AST_SCHED_DEL, FLAG_P2P_SENT_MARK, ast_rtp::rtcp, ast_rtp::sched, ast_rtcp::schedid, ast_rtcp::them, and ast_rtp::them.
Referenced by process_sdp(), setup_rtp_connection(), and stop_media_flows().
02074 { 02075 if (rtp->rtcp) { 02076 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02077 } 02078 02079 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 02080 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 02081 if (rtp->rtcp) { 02082 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr)); 02083 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port)); 02084 } 02085 02086 ast_clear_flag(rtp, FLAG_P2P_SENT_MARK); 02087 }
void ast_rtp_stun_request | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | suggestion, | |||
const char * | username | |||
) |
Definition at line 402 of file rtp.c.
References append_attr_string(), stun_attr::attr, ast_rtp::s, STUN_BINDREQ, stun_req_id(), stun_send(), and STUN_USERNAME.
Referenced by gtalk_update_stun().
00403 { 00404 struct stun_header *req; 00405 unsigned char reqdata[1024]; 00406 int reqlen, reqleft; 00407 struct stun_attr *attr; 00408 00409 req = (struct stun_header *)reqdata; 00410 stun_req_id(req); 00411 reqlen = 0; 00412 reqleft = sizeof(reqdata) - sizeof(struct stun_header); 00413 req->msgtype = 0; 00414 req->msglen = 0; 00415 attr = (struct stun_attr *)req->ies; 00416 if (username) 00417 append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft); 00418 req->msglen = htons(reqlen); 00419 req->msgtype = htons(STUN_BINDREQ); 00420 stun_send(rtp->s, suggestion, req); 00421 }
void ast_rtp_unset_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
clear payload type
Definition at line 1668 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, and MAX_RTP_PT.
Referenced by process_sdp().
01669 { 01670 if (pt < 0 || pt > MAX_RTP_PT) 01671 return; /* bogus payload type */ 01672 01673 ast_mutex_lock(&rtp->bridge_lock); 01674 rtp->current_RTP_PT[pt].isAstFormat = 0; 01675 rtp->current_RTP_PT[pt].code = 0; 01676 ast_mutex_unlock(&rtp->bridge_lock); 01677 }
Definition at line 2802 of file rtp.c.
References ast_codec_pref_getsize(), AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_SPEEX, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frdup(), ast_frfree, ast_getformatname(), ast_log(), ast_rtp_lookup_code(), ast_rtp_raw_write(), ast_smoother_feed, ast_smoother_feed_be, AST_SMOOTHER_FLAG_BE, ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_smoother_set_flags(), ast_smoother_test_flag(), ast_format_list::cur_ms, ast_frame::datalen, f, ast_format_list::flags, ast_format_list::fr_len, ast_frame::frametype, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, option_debug, ast_rtp::pref, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.
Referenced by gtalk_write(), mgcp_write(), oh323_write(), sip_write(), and skinny_write().
02803 { 02804 struct ast_frame *f; 02805 int codec; 02806 int hdrlen = 12; 02807 int subclass; 02808 02809 02810 /* If we have no peer, return immediately */ 02811 if (!rtp->them.sin_addr.s_addr) 02812 return 0; 02813 02814 /* If there is no data length, return immediately */ 02815 if (!_f->datalen) 02816 return 0; 02817 02818 /* Make sure we have enough space for RTP header */ 02819 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { 02820 ast_log(LOG_WARNING, "RTP can only send voice and video\n"); 02821 return -1; 02822 } 02823 02824 subclass = _f->subclass; 02825 if (_f->frametype == AST_FRAME_VIDEO) 02826 subclass &= ~0x1; 02827 02828 codec = ast_rtp_lookup_code(rtp, 1, subclass); 02829 if (codec < 0) { 02830 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); 02831 return -1; 02832 } 02833 02834 if (rtp->lasttxformat != subclass) { 02835 /* New format, reset the smoother */ 02836 if (option_debug) 02837 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); 02838 rtp->lasttxformat = subclass; 02839 if (rtp->smoother) 02840 ast_smoother_free(rtp->smoother); 02841 rtp->smoother = NULL; 02842 } 02843 02844 if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) { 02845 struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass); 02846 if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */ 02847 if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) { 02848 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)); 02849 return -1; 02850 } 02851 if (fmt.flags) 02852 ast_smoother_set_flags(rtp->smoother, fmt.flags); 02853 if (option_debug) 02854 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)); 02855 } 02856 } 02857 if (rtp->smoother) { 02858 if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) { 02859 ast_smoother_feed_be(rtp->smoother, _f); 02860 } else { 02861 ast_smoother_feed(rtp->smoother, _f); 02862 } 02863 02864 while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) { 02865 if (f->subclass == AST_FORMAT_G722) { 02866 /* G.722 is silllllllllllllly */ 02867 f->samples /= 2; 02868 } 02869 02870 ast_rtp_raw_write(rtp, f, codec); 02871 } 02872 } else { 02873 /* Don't buffer outgoing frames; send them one-per-packet: */ 02874 if (_f->offset < hdrlen) { 02875 f = ast_frdup(_f); 02876 } else { 02877 f = _f; 02878 } 02879 if (f->data) { 02880 if (f->subclass == AST_FORMAT_G722) { 02881 /* G.722 is silllllllllllllly */ 02882 f->samples /= 2; 02883 } 02884 ast_rtp_raw_write(rtp, f, codec); 02885 } 02886 if (f != _f) 02887 ast_frfree(f); 02888 } 02889 02890 return 0; 02891 }