#include <netinet/in.h>
#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
#include "asterisk/linkedlists.h"
Include dependency graph for rtp.h:
This graph shows which files directly or indirectly include this file:
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(), and ast_rtp_unset_m_type().
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 518 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 827 of file rtp.c.
References ast_rtcp::accumulated_transit, AST_CONTROL_VIDUPDATE, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), CRASH, 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_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().
00828 { 00829 socklen_t len; 00830 int position, i, packetwords; 00831 int res; 00832 struct sockaddr_in sin; 00833 unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET]; 00834 unsigned int *rtcpheader; 00835 int pt; 00836 struct timeval now; 00837 unsigned int length; 00838 int rc; 00839 double rttsec; 00840 uint64_t rtt = 0; 00841 unsigned int dlsr; 00842 unsigned int lsr; 00843 unsigned int msw; 00844 unsigned int lsw; 00845 unsigned int comp; 00846 struct ast_frame *f = &ast_null_frame; 00847 00848 if (!rtp || !rtp->rtcp) 00849 return &ast_null_frame; 00850 00851 len = sizeof(sin); 00852 00853 res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET, 00854 0, (struct sockaddr *)&sin, &len); 00855 rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET); 00856 00857 if (res < 0) { 00858 if (errno == EBADF) 00859 CRASH; 00860 if (errno != EAGAIN) { 00861 ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno)); 00862 return NULL; 00863 } 00864 return &ast_null_frame; 00865 } 00866 00867 packetwords = res / 4; 00868 00869 if (rtp->nat) { 00870 /* Send to whoever sent to us */ 00871 if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 00872 (rtp->rtcp->them.sin_port != sin.sin_port)) { 00873 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 00874 if (option_debug || rtpdebug) 00875 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)); 00876 } 00877 } 00878 00879 if (option_debug) 00880 ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res); 00881 00882 /* Process a compound packet */ 00883 position = 0; 00884 while (position < packetwords) { 00885 i = position; 00886 length = ntohl(rtcpheader[i]); 00887 pt = (length & 0xff0000) >> 16; 00888 rc = (length & 0x1f000000) >> 24; 00889 length &= 0xffff; 00890 00891 if ((i + length) > packetwords) { 00892 ast_log(LOG_WARNING, "RTCP Read too short\n"); 00893 return &ast_null_frame; 00894 } 00895 00896 if (rtcp_debug_test_addr(&sin)) { 00897 ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); 00898 ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown"); 00899 ast_verbose("Reception reports: %d\n", rc); 00900 ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]); 00901 } 00902 00903 i += 2; /* Advance past header and ssrc */ 00904 00905 switch (pt) { 00906 case RTCP_PT_SR: 00907 gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */ 00908 rtp->rtcp->spc = ntohl(rtcpheader[i+3]); 00909 rtp->rtcp->soc = ntohl(rtcpheader[i + 4]); 00910 rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/ 00911 00912 if (rtcp_debug_test_addr(&sin)) { 00913 ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096); 00914 ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2])); 00915 ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4])); 00916 } 00917 i += 5; 00918 if (rc < 1) 00919 break; 00920 /* Intentional fall through */ 00921 case RTCP_PT_RR: 00922 /* Don't handle multiple reception reports (rc > 1) yet */ 00923 /* Calculate RTT per RFC */ 00924 gettimeofday(&now, NULL); 00925 timeval2ntp(now, &msw, &lsw); 00926 if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */ 00927 comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16); 00928 lsr = ntohl(rtcpheader[i + 4]); 00929 dlsr = ntohl(rtcpheader[i + 5]); 00930 rtt = comp - lsr - dlsr; 00931 00932 /* Convert end to end delay to usec (keeping the calculation in 64bit space) 00933 sess->ee_delay = (eedelay * 1000) / 65536; */ 00934 if (rtt < 4294) { 00935 rtt = (rtt * 1000000) >> 16; 00936 } else { 00937 rtt = (rtt * 1000) >> 16; 00938 rtt *= 1000; 00939 } 00940 rtt = rtt / 1000.; 00941 rttsec = rtt / 1000.; 00942 00943 if (comp - dlsr >= lsr) { 00944 rtp->rtcp->accumulated_transit += rttsec; 00945 rtp->rtcp->rtt = rttsec; 00946 if (rtp->rtcp->maxrtt<rttsec) 00947 rtp->rtcp->maxrtt = rttsec; 00948 if (rtp->rtcp->minrtt>rttsec) 00949 rtp->rtcp->minrtt = rttsec; 00950 } else if (rtcp_debug_test_addr(&sin)) { 00951 ast_verbose("Internal RTCP NTP clock skew detected: " 00952 "lsr=%u, now=%u, dlsr=%u (%d:%03dms), " 00953 "diff=%d\n", 00954 lsr, comp, dlsr, dlsr / 65536, 00955 (dlsr % 65536) * 1000 / 65536, 00956 dlsr - (comp - lsr)); 00957 } 00958 } 00959 00960 rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]); 00961 rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff; 00962 if (rtcp_debug_test_addr(&sin)) { 00963 ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24)); 00964 ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost); 00965 ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff)); 00966 ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16); 00967 ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter); 00968 ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096); 00969 ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0); 00970 if (rtt) 00971 ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt); 00972 } 00973 break; 00974 case RTCP_PT_FUR: 00975 if (rtcp_debug_test_addr(&sin)) 00976 ast_verbose("Received an RTCP Fast Update Request\n"); 00977 rtp->f.frametype = AST_FRAME_CONTROL; 00978 rtp->f.subclass = AST_CONTROL_VIDUPDATE; 00979 rtp->f.datalen = 0; 00980 rtp->f.samples = 0; 00981 rtp->f.mallocd = 0; 00982 rtp->f.src = "RTP"; 00983 f = &rtp->f; 00984 break; 00985 case RTCP_PT_SDES: 00986 if (rtcp_debug_test_addr(&sin)) 00987 ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00988 break; 00989 case RTCP_PT_BYE: 00990 if (rtcp_debug_test_addr(&sin)) 00991 ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00992 break; 00993 default: 00994 if (option_debug) 00995 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)); 00996 break; 00997 } 00998 position += (length + 1); 00999 } 01000 01001 return f; 01002 }
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 2360 of file rtp.c.
References ast_rtcp_write(), ast_rtp::rtcp, and ast_rtcp::sendfur.
02361 { 02362 struct ast_rtp *rtp = data; 02363 int res; 02364 02365 rtp->rtcp->sendfur = 1; 02366 res = ast_rtcp_write(data); 02367 02368 return res; 02369 }
size_t ast_rtp_alloc_size | ( | void | ) |
Get the amount of space required to hold an RTP session.
Definition at line 398 of file rtp.c.
Referenced by process_sdp().
00399 { 00400 return sizeof(struct ast_rtp); 00401 }
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 3287 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_DTMF_COMPENSATE, 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, 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.
03288 { 03289 struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ 03290 struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */ 03291 struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL; 03292 enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED; 03293 enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED; 03294 enum ast_bridge_result res = AST_BRIDGE_FAILED; 03295 int codec0 = 0, codec1 = 0; 03296 void *pvt0 = NULL, *pvt1 = NULL; 03297 03298 /* Lock channels */ 03299 ast_channel_lock(c0); 03300 while(ast_channel_trylock(c1)) { 03301 ast_channel_unlock(c0); 03302 usleep(1); 03303 ast_channel_lock(c0); 03304 } 03305 03306 /* Ensure neither channel got hungup during lock avoidance */ 03307 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 03308 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name); 03309 ast_channel_unlock(c0); 03310 ast_channel_unlock(c1); 03311 return AST_BRIDGE_FAILED; 03312 } 03313 03314 /* Find channel driver interfaces */ 03315 if (!(pr0 = get_proto(c0))) { 03316 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); 03317 ast_channel_unlock(c0); 03318 ast_channel_unlock(c1); 03319 return AST_BRIDGE_FAILED; 03320 } 03321 if (!(pr1 = get_proto(c1))) { 03322 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); 03323 ast_channel_unlock(c0); 03324 ast_channel_unlock(c1); 03325 return AST_BRIDGE_FAILED; 03326 } 03327 03328 /* Get channel specific interface structures */ 03329 pvt0 = c0->tech_pvt; 03330 pvt1 = c1->tech_pvt; 03331 03332 /* Get audio and video interface (if native bridge is possible) */ 03333 audio_p0_res = pr0->get_rtp_info(c0, &p0); 03334 video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED; 03335 audio_p1_res = pr1->get_rtp_info(c1, &p1); 03336 video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED; 03337 03338 /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */ 03339 if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE)) 03340 audio_p0_res = AST_RTP_GET_FAILED; 03341 if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE)) 03342 audio_p1_res = AST_RTP_GET_FAILED; 03343 03344 /* Check if a bridge is possible (partial/native) */ 03345 if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) { 03346 /* Somebody doesn't want to play... */ 03347 ast_channel_unlock(c0); 03348 ast_channel_unlock(c1); 03349 return AST_BRIDGE_FAILED_NOWARN; 03350 } 03351 03352 /* If we need to feed DTMF frames into the core then only do a partial native bridge */ 03353 if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) { 03354 ast_set_flag(p0, FLAG_P2P_NEED_DTMF); 03355 audio_p0_res = AST_RTP_TRY_PARTIAL; 03356 } 03357 03358 if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) { 03359 ast_set_flag(p1, FLAG_P2P_NEED_DTMF); 03360 audio_p1_res = AST_RTP_TRY_PARTIAL; 03361 } 03362 03363 /* If both sides are not using the same method of DTMF transmission 03364 * (ie: one is RFC2833, other is INFO... then we can not do direct media. 03365 * -------------------------------------------------- 03366 * | DTMF Mode | HAS_DTMF | Accepts Begin Frames | 03367 * |-----------|------------|-----------------------| 03368 * | Inband | False | True | 03369 * | RFC2833 | True | True | 03370 * | SIP INFO | False | False | 03371 * -------------------------------------------------- 03372 * However, if DTMF from both channels is being monitored by the core, then 03373 * we can still do packet-to-packet bridging, because passing through the 03374 * core will handle DTMF mode translation. 03375 */ 03376 if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) || 03377 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) { 03378 if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) { 03379 ast_channel_unlock(c0); 03380 ast_channel_unlock(c1); 03381 return AST_BRIDGE_FAILED_NOWARN; 03382 } 03383 audio_p0_res = AST_RTP_TRY_PARTIAL; 03384 audio_p1_res = AST_RTP_TRY_PARTIAL; 03385 } 03386 03387 /* If the core will need to compensate and the P2P bridge will need to feed up DTMF frames then we can not reliably do so yet, so do not P2P bridge */ 03388 if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF) && ast_test_flag(p0, FLAG_DTMF_COMPENSATE)) || 03389 (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF) && ast_test_flag(p1, FLAG_DTMF_COMPENSATE))) { 03390 ast_channel_unlock(c0); 03391 ast_channel_unlock(c1); 03392 return AST_BRIDGE_FAILED_NOWARN; 03393 } 03394 03395 /* Get codecs from both sides */ 03396 codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0; 03397 codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0; 03398 if (codec0 && codec1 && !(codec0 & codec1)) { 03399 /* Hey, we can't do native bridging if both parties speak different codecs */ 03400 if (option_debug) 03401 ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); 03402 ast_channel_unlock(c0); 03403 ast_channel_unlock(c1); 03404 return AST_BRIDGE_FAILED_NOWARN; 03405 } 03406 03407 /* If either side can only do a partial bridge, then don't try for a true native bridge */ 03408 if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) { 03409 struct ast_format_list fmt0, fmt1; 03410 03411 /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */ 03412 if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) { 03413 if (option_debug) 03414 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n"); 03415 ast_channel_unlock(c0); 03416 ast_channel_unlock(c1); 03417 return AST_BRIDGE_FAILED_NOWARN; 03418 } 03419 /* They must also be using the same packetization */ 03420 fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat); 03421 fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat); 03422 if (fmt0.cur_ms != fmt1.cur_ms) { 03423 if (option_debug) 03424 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n"); 03425 ast_channel_unlock(c0); 03426 ast_channel_unlock(c1); 03427 return AST_BRIDGE_FAILED_NOWARN; 03428 } 03429 03430 if (option_verbose > 2) 03431 ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name); 03432 res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1); 03433 } else { 03434 if (option_verbose > 2) 03435 ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name); 03436 res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1); 03437 } 03438 03439 return res; 03440 }
int ast_rtp_codec_getformat | ( | int | pt | ) |
Definition at line 2742 of file rtp.c.
References rtpPayloadType::code, and static_RTP_PT.
Referenced by process_sdp().
02743 { 02744 if (pt < 0 || pt > MAX_RTP_PT) 02745 return 0; /* bogus payload type */ 02746 02747 if (static_RTP_PT[pt].isAstFormat) 02748 return static_RTP_PT[pt].code; 02749 else 02750 return 0; 02751 }
struct ast_codec_pref* ast_rtp_codec_getpref | ( | struct ast_rtp * | rtp | ) |
Definition at line 2737 of file rtp.c.
References ast_rtp::pref.
Referenced by add_codec_to_sdp(), and process_sdp().
02738 { 02739 return &rtp->pref; 02740 }
int ast_rtp_codec_setpref | ( | struct ast_rtp * | rtp, | |
struct ast_codec_pref * | prefs | |||
) |
Definition at line 2724 of file rtp.c.
References ast_smoother_free(), ast_codec_pref::framing, ast_codec_pref::order, 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(), start_rtp(), and transmit_response_with_sdp().
02725 { 02726 int x; 02727 for (x = 0; x < 32; x++) { /* Ugly way */ 02728 rtp->pref.order[x] = prefs->order[x]; 02729 rtp->pref.framing[x] = prefs->framing[x]; 02730 } 02731 if (rtp->smoother) 02732 ast_smoother_free(rtp->smoother); 02733 rtp->smoother = NULL; 02734 return 0; 02735 }
void ast_rtp_destroy | ( | struct ast_rtp * | rtp | ) |
Definition at line 2143 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().
02144 { 02145 if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) { 02146 /*Print some info on the call here */ 02147 ast_verbose(" RTP-stats\n"); 02148 ast_verbose("* Our Receiver:\n"); 02149 ast_verbose(" SSRC: %u\n", rtp->themssrc); 02150 ast_verbose(" Received packets: %u\n", rtp->rxcount); 02151 ast_verbose(" Lost packets: %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior); 02152 ast_verbose(" Jitter: %.4f\n", rtp->rxjitter); 02153 ast_verbose(" Transit: %.4f\n", rtp->rxtransit); 02154 ast_verbose(" RR-count: %u\n", rtp->rtcp->rr_count); 02155 ast_verbose("* Our Sender:\n"); 02156 ast_verbose(" SSRC: %u\n", rtp->ssrc); 02157 ast_verbose(" Sent packets: %u\n", rtp->txcount); 02158 ast_verbose(" Lost packets: %u\n", rtp->rtcp->reported_lost); 02159 ast_verbose(" Jitter: %u\n", rtp->rtcp->reported_jitter / (unsigned int)65536.0); 02160 ast_verbose(" SR-count: %u\n", rtp->rtcp->sr_count); 02161 ast_verbose(" RTT: %f\n", rtp->rtcp->rtt); 02162 } 02163 02164 if (rtp->smoother) 02165 ast_smoother_free(rtp->smoother); 02166 if (rtp->ioid) 02167 ast_io_remove(rtp->io, rtp->ioid); 02168 if (rtp->s > -1) 02169 close(rtp->s); 02170 if (rtp->rtcp) { 02171 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02172 close(rtp->rtcp->s); 02173 free(rtp->rtcp); 02174 rtp->rtcp=NULL; 02175 } 02176 02177 ast_mutex_destroy(&rtp->bridge_lock); 02178 02179 free(rtp); 02180 }
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, 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) { 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 && srcpr->get_codec) 01551 srccodec = srcpr->get_codec(src); 01552 else 01553 srccodec = 0; 01554 if (audio_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 (!(srccodec & destcodec)) { 01560 ast_channel_unlock(dest); 01561 if (src) 01562 ast_channel_unlock(src); 01563 return 0; 01564 } 01565 /* Consider empty media as non-existant */ 01566 if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) 01567 srcp = NULL; 01568 /* If the client has NAT stuff turned on then just safe NAT is active */ 01569 if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01570 nat_active = 1; 01571 /* Bridge media early */ 01572 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active)) 01573 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01574 ast_channel_unlock(dest); 01575 if (src) 01576 ast_channel_unlock(src); 01577 if (option_debug) 01578 ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01579 return 1; 01580 }
int ast_rtp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 513 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().
00514 { 00515 return rtp->s; 00516 }
Definition at line 2054 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().
02055 { 02056 struct ast_rtp *bridged = NULL; 02057 02058 ast_mutex_lock(&rtp->bridge_lock); 02059 bridged = rtp->bridged; 02060 ast_mutex_unlock(&rtp->bridge_lock); 02061 02062 return bridged; 02063 }
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 1716 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 process_sdp().
01718 { 01719 int pt; 01720 01721 ast_mutex_lock(&rtp->bridge_lock); 01722 01723 *astFormats = *nonAstFormats = 0; 01724 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01725 if (rtp->current_RTP_PT[pt].isAstFormat) { 01726 *astFormats |= rtp->current_RTP_PT[pt].code; 01727 } else { 01728 *nonAstFormats |= rtp->current_RTP_PT[pt].code; 01729 } 01730 } 01731 01732 ast_mutex_unlock(&rtp->bridge_lock); 01733 01734 return; 01735 }
int ast_rtp_get_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2036 of file rtp.c.
References ast_rtp::them.
Referenced by add_sdp(), bridge_native_loop(), do_monitor(), gtalk_update_stun(), oh323_set_rtp_peer(), sip_set_rtp_peer(), and transmit_modify_with_sdp().
02037 { 02038 if ((them->sin_family != AF_INET) || 02039 (them->sin_port != rtp->them.sin_port) || 02040 (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) { 02041 them->sin_family = AF_INET; 02042 them->sin_port = rtp->them.sin_port; 02043 them->sin_addr = rtp->them.sin_addr; 02044 return 1; 02045 } 02046 return 0; 02047 }
char* ast_rtp_get_quality | ( | struct ast_rtp * | rtp, | |
struct ast_rtp_quality * | qual | |||
) |
Return RTCP quality string.
Definition at line 2099 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().
02100 { 02101 /* 02102 *ssrc our ssrc 02103 *themssrc their ssrc 02104 *lp lost packets 02105 *rxjitter our calculated jitter(rx) 02106 *rxcount no. received packets 02107 *txjitter reported jitter of the other end 02108 *txcount transmitted packets 02109 *rlp remote lost packets 02110 *rtt round trip time 02111 */ 02112 02113 if (qual && rtp) { 02114 qual->local_ssrc = rtp->ssrc; 02115 qual->local_jitter = rtp->rxjitter; 02116 qual->local_count = rtp->rxcount; 02117 qual->remote_ssrc = rtp->themssrc; 02118 qual->remote_count = rtp->txcount; 02119 if (rtp->rtcp) { 02120 qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior; 02121 qual->remote_lostpackets = rtp->rtcp->reported_lost; 02122 qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0; 02123 qual->rtt = rtp->rtcp->rtt; 02124 } 02125 } 02126 if (rtp->rtcp) { 02127 snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), 02128 "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", 02129 rtp->ssrc, 02130 rtp->themssrc, 02131 rtp->rtcp->expected_prior - rtp->rtcp->received_prior, 02132 rtp->rxjitter, 02133 rtp->rxcount, 02134 (double)rtp->rtcp->reported_jitter / 65536.0, 02135 rtp->txcount, 02136 rtp->rtcp->reported_lost, 02137 rtp->rtcp->rtt); 02138 return rtp->rtcp->quality; 02139 } else 02140 return "<Unknown> - RTP/RTCP has already been destroyed"; 02141 }
int ast_rtp_get_rtpholdtimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp hold timeout.
Definition at line 568 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by do_monitor().
00569 { 00570 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00571 return 0; 00572 return rtp->rtpholdtimeout; 00573 }
int ast_rtp_get_rtpkeepalive | ( | struct ast_rtp * | rtp | ) |
Get RTP keepalive interval.
Definition at line 576 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by do_monitor().
00577 { 00578 return rtp->rtpkeepalive; 00579 }
int ast_rtp_get_rtptimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp timeout.
Definition at line 560 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by do_monitor().
00561 { 00562 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00563 return 0; 00564 return rtp->rtptimeout; 00565 }
void ast_rtp_get_us | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | us | |||
) |
Definition at line 2049 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 596 of file rtp.c.
References ast_test_flag, and FLAG_NAT_ACTIVE.
Referenced by sip_get_rtp_peer().
00597 { 00598 return ast_test_flag(rtp, FLAG_NAT_ACTIVE); 00599 }
void ast_rtp_init | ( | void | ) |
Initialize the RTP system in Asterisk.
Definition at line 3825 of file rtp.c.
References ast_cli_register_multiple(), ast_rtp_reload(), and cli_rtp.
Referenced by main().
03826 { 03827 ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry)); 03828 ast_rtp_reload(); 03829 }
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 1759 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().
01760 { 01761 int pt = 0; 01762 01763 ast_mutex_lock(&rtp->bridge_lock); 01764 01765 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && 01766 code == rtp->rtp_lookup_code_cache_code) { 01767 /* Use our cached mapping, to avoid the overhead of the loop below */ 01768 pt = rtp->rtp_lookup_code_cache_result; 01769 ast_mutex_unlock(&rtp->bridge_lock); 01770 return pt; 01771 } 01772 01773 /* Check the dynamic list first */ 01774 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01775 if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) { 01776 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01777 rtp->rtp_lookup_code_cache_code = code; 01778 rtp->rtp_lookup_code_cache_result = pt; 01779 ast_mutex_unlock(&rtp->bridge_lock); 01780 return pt; 01781 } 01782 } 01783 01784 /* Then the static list */ 01785 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01786 if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) { 01787 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01788 rtp->rtp_lookup_code_cache_code = code; 01789 rtp->rtp_lookup_code_cache_result = pt; 01790 ast_mutex_unlock(&rtp->bridge_lock); 01791 return pt; 01792 } 01793 } 01794 01795 ast_mutex_unlock(&rtp->bridge_lock); 01796 01797 return -1; 01798 }
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 1819 of file rtp.c.
References ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, format, len, and name.
Referenced by process_sdp().
01821 { 01822 int format; 01823 unsigned len; 01824 char *end = buf; 01825 char *start = buf; 01826 01827 if (!buf || !size) 01828 return NULL; 01829 01830 snprintf(end, size, "0x%x (", capability); 01831 01832 len = strlen(end); 01833 end += len; 01834 size -= len; 01835 start = end; 01836 01837 for (format = 1; format < AST_RTP_MAX; format <<= 1) { 01838 if (capability & format) { 01839 const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options); 01840 01841 snprintf(end, size, "%s|", name); 01842 len = strlen(end); 01843 end += len; 01844 size -= len; 01845 } 01846 } 01847 01848 if (start == end) 01849 snprintf(start, size, "nothing)"); 01850 else if (size > 1) 01851 *(end -1) = ')'; 01852 01853 return buf; 01854 }
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 1800 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().
01802 { 01803 unsigned int i; 01804 01805 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01806 if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) { 01807 if (isAstFormat && 01808 (code == AST_FORMAT_G726_AAL2) && 01809 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01810 return "G726-32"; 01811 else 01812 return mimeTypes[i].subtype; 01813 } 01814 } 01815 01816 return ""; 01817 }
struct rtpPayloadType ast_rtp_lookup_pt | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Mapping between RTP payload format codes and Asterisk codes:.
Definition at line 1737 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), MAX_RTP_PT, result, and static_RTP_PT.
Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and setup_rtp_connection().
01738 { 01739 struct rtpPayloadType result; 01740 01741 result.isAstFormat = result.code = 0; 01742 01743 if (pt < 0 || pt > MAX_RTP_PT) 01744 return result; /* bogus payload type */ 01745 01746 /* Start with negotiated codecs */ 01747 ast_mutex_lock(&rtp->bridge_lock); 01748 result = rtp->current_RTP_PT[pt]; 01749 ast_mutex_unlock(&rtp->bridge_lock); 01750 01751 /* If it doesn't exist, check our static RTP type list, just in case */ 01752 if (!result.code) 01753 result = static_RTP_PT[pt]; 01754 01755 return result; 01756 }
int ast_rtp_make_compatible | ( | struct ast_channel * | dest, | |
struct ast_channel * | src, | |||
int | media | |||
) |
Definition at line 1582 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, option_debug, and ast_rtp_protocol::set_rtp_peer.
Referenced by wait_for_answer().
01583 { 01584 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01585 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01586 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01587 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01588 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01589 int srccodec, destcodec; 01590 01591 /* Lock channels */ 01592 ast_channel_lock(dest); 01593 while(ast_channel_trylock(src)) { 01594 ast_channel_unlock(dest); 01595 usleep(1); 01596 ast_channel_lock(dest); 01597 } 01598 01599 /* Find channel driver interfaces */ 01600 if (!(destpr = get_proto(dest))) { 01601 if (option_debug) 01602 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01603 ast_channel_unlock(dest); 01604 ast_channel_unlock(src); 01605 return 0; 01606 } 01607 if (!(srcpr = get_proto(src))) { 01608 if (option_debug) 01609 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name); 01610 ast_channel_unlock(dest); 01611 ast_channel_unlock(src); 01612 return 0; 01613 } 01614 01615 /* Get audio and video interface (if native bridge is possible) */ 01616 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01617 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01618 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01619 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01620 01621 /* Ensure we have at least one matching codec */ 01622 if (srcpr->get_codec) 01623 srccodec = srcpr->get_codec(src); 01624 else 01625 srccodec = 0; 01626 if (destpr->get_codec) 01627 destcodec = destpr->get_codec(dest); 01628 else 01629 destcodec = 0; 01630 01631 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01632 if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) { 01633 /* Somebody doesn't want to play... */ 01634 ast_channel_unlock(dest); 01635 ast_channel_unlock(src); 01636 return 0; 01637 } 01638 ast_rtp_pt_copy(destp, srcp); 01639 if (vdestp && vsrcp) 01640 ast_rtp_pt_copy(vdestp, vsrcp); 01641 if (media) { 01642 /* Bridge early */ 01643 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01644 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name); 01645 } 01646 ast_channel_unlock(dest); 01647 ast_channel_unlock(src); 01648 if (option_debug) 01649 ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name); 01650 return 1; 01651 }
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 2001 of file rtp.c.
References ast_rtp_new_with_bindaddr(), io, and sched.
02002 { 02003 struct in_addr ia; 02004 02005 memset(&ia, 0, sizeof(ia)); 02006 return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia); 02007 }
void ast_rtp_new_init | ( | struct ast_rtp * | rtp | ) |
Initialize a new RTP structure.
Definition at line 1901 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().
01902 { 01903 ast_mutex_init(&rtp->bridge_lock); 01904 01905 rtp->them.sin_family = AF_INET; 01906 rtp->us.sin_family = AF_INET; 01907 rtp->ssrc = ast_random(); 01908 rtp->seqno = ast_random() & 0xffff; 01909 ast_set_flag(rtp, FLAG_HAS_DTMF); 01910 01911 return; 01912 }
void ast_rtp_new_source | ( | struct ast_rtp * | rtp | ) |
Definition at line 2018 of file rtp.c.
References ast_random(), ast_rtp::set_marker_bit, and ast_rtp::ssrc.
Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), sip_write(), and skinny_indicate().
02019 { 02020 rtp->set_marker_bit = 1; 02021 rtp->ssrc = ast_random(); 02022 return; 02023 }
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 1914 of file rtp.c.
References ast_calloc, ast_io_add(), AST_IO_IN, ast_log(), ast_random(), ast_rtcp_new(), ast_rtp_new_init(), ast_rtp_pt_default(), ast_set_flag, errno, FLAG_CALLBACK_MODE, free, io, LOG_ERROR, rtp_socket(), rtpread(), and sched.
Referenced by __oh323_rtp_create(), ast_rtp_new(), gtalk_alloc(), sip_alloc(), and start_rtp().
01915 { 01916 struct ast_rtp *rtp; 01917 int x; 01918 int first; 01919 int startplace; 01920 01921 if (!(rtp = ast_calloc(1, sizeof(*rtp)))) 01922 return NULL; 01923 01924 ast_rtp_new_init(rtp); 01925 01926 rtp->s = rtp_socket(); 01927 if (rtp->s < 0) { 01928 free(rtp); 01929 ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno)); 01930 return NULL; 01931 } 01932 if (sched && rtcpenable) { 01933 rtp->sched = sched; 01934 rtp->rtcp = ast_rtcp_new(); 01935 } 01936 01937 /* Select a random port number in the range of possible RTP */ 01938 x = (ast_random() % (rtpend-rtpstart)) + rtpstart; 01939 x = x & ~1; 01940 /* Save it for future references. */ 01941 startplace = x; 01942 /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */ 01943 for (;;) { 01944 /* Must be an even port number by RTP spec */ 01945 rtp->us.sin_port = htons(x); 01946 rtp->us.sin_addr = addr; 01947 /* If there's rtcp, initialize it as well. */ 01948 if (rtp->rtcp) { 01949 rtp->rtcp->us.sin_port = htons(x + 1); 01950 rtp->rtcp->us.sin_addr = addr; 01951 } 01952 /* Try to bind it/them. */ 01953 if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) && 01954 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) 01955 break; 01956 if (!first) { 01957 /* Primary bind succeeded! Gotta recreate it */ 01958 close(rtp->s); 01959 rtp->s = rtp_socket(); 01960 } 01961 if (errno != EADDRINUSE) { 01962 /* We got an error that wasn't expected, abort! */ 01963 ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); 01964 close(rtp->s); 01965 if (rtp->rtcp) { 01966 close(rtp->rtcp->s); 01967 free(rtp->rtcp); 01968 } 01969 free(rtp); 01970 return NULL; 01971 } 01972 /* The port was used, increment it (by two). */ 01973 x += 2; 01974 /* Did we go over the limit ? */ 01975 if (x > rtpend) 01976 /* then, start from the begingig. */ 01977 x = (rtpstart + 1) & ~1; 01978 /* Check if we reached the place were we started. */ 01979 if (x == startplace) { 01980 /* If so, there's no ports available. */ 01981 ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); 01982 close(rtp->s); 01983 if (rtp->rtcp) { 01984 close(rtp->rtcp->s); 01985 free(rtp->rtcp); 01986 } 01987 free(rtp); 01988 return NULL; 01989 } 01990 } 01991 rtp->sched = sched; 01992 rtp->io = io; 01993 if (callbackmode) { 01994 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); 01995 ast_set_flag(rtp, FLAG_CALLBACK_MODE); 01996 } 01997 ast_rtp_pt_default(rtp); 01998 return rtp; 01999 }
int ast_rtp_proto_register | ( | struct ast_rtp_protocol * | proto | ) |
Register interface to channel driver.
Definition at line 2842 of file rtp.c.
References AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_log(), LOG_WARNING, protos, and ast_rtp_protocol::type.
Referenced by load_module().
02843 { 02844 struct ast_rtp_protocol *cur; 02845 02846 AST_LIST_LOCK(&protos); 02847 AST_LIST_TRAVERSE(&protos, cur, list) { 02848 if (!strcmp(cur->type, proto->type)) { 02849 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); 02850 AST_LIST_UNLOCK(&protos); 02851 return -1; 02852 } 02853 } 02854 AST_LIST_INSERT_HEAD(&protos, proto, list); 02855 AST_LIST_UNLOCK(&protos); 02856 02857 return 0; 02858 }
void ast_rtp_proto_unregister | ( | struct ast_rtp_protocol * | proto | ) |
Unregister interface to channel driver.
Definition at line 2834 of file rtp.c.
References AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, and protos.
Referenced by load_module(), and unload_module().
02835 { 02836 AST_LIST_LOCK(&protos); 02837 AST_LIST_REMOVE(&protos, proto, list); 02838 AST_LIST_UNLOCK(&protos); 02839 }
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 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.
Referenced by ast_rtp_new_with_bindaddr().
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 1106 of file rtp.c.
References ast_backtrace(), ast_codec_get_samples(), AST_FORMAT_MAX_AUDIO, ast_format_rate(), AST_FORMAT_SLINEAR, ast_frame_byteswap_be, 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, CRASH, ast_rtp::cycles, ast_frame::data, ast_frame::datalen, ast_frame::delivery, errno, event, 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_ERROR, 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::rtcp, rtp_debug_test_addr(), RTP_SEQ_MOD, ast_rtp::rxcount, ast_rtp::rxseqno, ast_rtp::rxssrc, ast_rtp::s, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, 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().
01107 { 01108 int res; 01109 struct sockaddr_in sin; 01110 socklen_t len; 01111 unsigned int seqno; 01112 int version; 01113 int payloadtype; 01114 int hdrlen = 12; 01115 int padding; 01116 int mark; 01117 int ext; 01118 int cc; 01119 unsigned int ssrc; 01120 unsigned int timestamp; 01121 unsigned int *rtpheader; 01122 struct rtpPayloadType rtpPT; 01123 struct ast_rtp *bridged = NULL; 01124 01125 if( !rtp ) { 01126 ast_log(LOG_ERROR, "ast_rtp_read(): called with rtp == NULL\n"); 01127 ast_backtrace(); 01128 return &ast_null_frame; 01129 } 01130 01131 /* If time is up, kill it */ 01132 if (rtp->sending_digit) 01133 ast_rtp_senddigit_continuation(rtp); 01134 01135 len = sizeof(sin); 01136 01137 /* Cache where the header will go */ 01138 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 01139 0, (struct sockaddr *)&sin, &len); 01140 01141 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); 01142 if (res < 0) { 01143 if (errno == EBADF) 01144 CRASH; 01145 if (errno != EAGAIN) { 01146 ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno)); 01147 return NULL; 01148 } 01149 return &ast_null_frame; 01150 } 01151 01152 if (res < hdrlen) { 01153 ast_log(LOG_WARNING, "RTP Read too short\n"); 01154 return &ast_null_frame; 01155 } 01156 01157 /* Get fields */ 01158 seqno = ntohl(rtpheader[0]); 01159 01160 /* Check RTP version */ 01161 version = (seqno & 0xC0000000) >> 30; 01162 if (!version) { 01163 if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) && 01164 (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) { 01165 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 01166 } 01167 return &ast_null_frame; 01168 } 01169 01170 #if 0 /* Allow to receive RTP stream with closed transmission path */ 01171 /* If we don't have the other side's address, then ignore this */ 01172 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 01173 return &ast_null_frame; 01174 #endif 01175 01176 /* Send to whoever send to us if NAT is turned on */ 01177 if (rtp->nat) { 01178 if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 01179 (rtp->them.sin_port != sin.sin_port)) { 01180 rtp->them = sin; 01181 if (rtp->rtcp) { 01182 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 01183 rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1); 01184 } 01185 rtp->rxseqno = 0; 01186 ast_set_flag(rtp, FLAG_NAT_ACTIVE); 01187 if (option_debug || rtpdebug) 01188 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)); 01189 } 01190 } 01191 01192 /* If we are bridged to another RTP stream, send direct */ 01193 if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen)) 01194 return &ast_null_frame; 01195 01196 if (version != 2) 01197 return &ast_null_frame; 01198 01199 payloadtype = (seqno & 0x7f0000) >> 16; 01200 padding = seqno & (1 << 29); 01201 mark = seqno & (1 << 23); 01202 ext = seqno & (1 << 28); 01203 cc = (seqno & 0xF000000) >> 24; 01204 seqno &= 0xffff; 01205 timestamp = ntohl(rtpheader[1]); 01206 ssrc = ntohl(rtpheader[2]); 01207 01208 if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) { 01209 if (option_debug || rtpdebug) 01210 ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n"); 01211 mark = 1; 01212 } 01213 01214 rtp->rxssrc = ssrc; 01215 01216 if (padding) { 01217 /* Remove padding bytes */ 01218 res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; 01219 } 01220 01221 if (cc) { 01222 /* CSRC fields present */ 01223 hdrlen += cc*4; 01224 } 01225 01226 if (ext) { 01227 /* RTP Extension present */ 01228 hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2; 01229 hdrlen += 4; 01230 if (option_debug) { 01231 int profile; 01232 profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16; 01233 if (profile == 0x505a) 01234 ast_log(LOG_DEBUG, "Found Zfone extension in RTP stream - zrtp - not supported.\n"); 01235 else 01236 ast_log(LOG_DEBUG, "Found unknown RTP Extensions %x\n", profile); 01237 } 01238 } 01239 01240 if (res < hdrlen) { 01241 ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); 01242 return &ast_null_frame; 01243 } 01244 01245 rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */ 01246 01247 if (rtp->rxcount==1) { 01248 /* This is the first RTP packet successfully received from source */ 01249 rtp->seedrxseqno = seqno; 01250 } 01251 01252 /* Do not schedule RR if RTCP isn't run */ 01253 if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) { 01254 /* Schedule transmission of Receiver Report */ 01255 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); 01256 } 01257 if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */ 01258 rtp->cycles += RTP_SEQ_MOD; 01259 01260 rtp->lastrxseqno = seqno; 01261 01262 if (rtp->themssrc==0) 01263 rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */ 01264 01265 if (rtp_debug_test_addr(&sin)) 01266 ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 01267 ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); 01268 01269 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); 01270 if (!rtpPT.isAstFormat) { 01271 struct ast_frame *f = NULL; 01272 01273 /* This is special in-band data that's not one of our codecs */ 01274 if (rtpPT.code == AST_RTP_DTMF) { 01275 /* It's special -- rfc2833 process it */ 01276 if (rtp_debug_test_addr(&sin)) { 01277 unsigned char *data; 01278 unsigned int event; 01279 unsigned int event_end; 01280 unsigned int duration; 01281 data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen; 01282 event = ntohl(*((unsigned int *)(data))); 01283 event >>= 24; 01284 event_end = ntohl(*((unsigned int *)(data))); 01285 event_end <<= 8; 01286 event_end >>= 24; 01287 duration = ntohl(*((unsigned int *)(data))); 01288 duration &= 0xFFFF; 01289 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); 01290 } 01291 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp); 01292 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { 01293 /* It's really special -- process it the Cisco way */ 01294 if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) { 01295 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01296 rtp->lastevent = seqno; 01297 } 01298 } else if (rtpPT.code == AST_RTP_CN) { 01299 /* Comfort Noise */ 01300 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01301 } else { 01302 ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr)); 01303 } 01304 return f ? f : &ast_null_frame; 01305 } 01306 rtp->lastrxformat = rtp->f.subclass = rtpPT.code; 01307 rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO; 01308 01309 if (!rtp->lastrxts) 01310 rtp->lastrxts = timestamp; 01311 01312 rtp->rxseqno = seqno; 01313 01314 /* Record received timestamp as last received now */ 01315 rtp->lastrxts = timestamp; 01316 01317 rtp->f.mallocd = 0; 01318 rtp->f.datalen = res - hdrlen; 01319 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; 01320 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; 01321 rtp->f.seqno = seqno; 01322 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { 01323 rtp->f.samples = ast_codec_get_samples(&rtp->f); 01324 if (rtp->f.subclass == AST_FORMAT_SLINEAR) 01325 ast_frame_byteswap_be(&rtp->f); 01326 calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark); 01327 /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */ 01328 ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO); 01329 rtp->f.ts = timestamp / 8; 01330 rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 ); 01331 } else { 01332 /* Video -- samples is # of samples vs. 90000 */ 01333 if (!rtp->lastividtimestamp) 01334 rtp->lastividtimestamp = timestamp; 01335 rtp->f.samples = timestamp - rtp->lastividtimestamp; 01336 rtp->lastividtimestamp = timestamp; 01337 rtp->f.delivery.tv_sec = 0; 01338 rtp->f.delivery.tv_usec = 0; 01339 if (mark) 01340 rtp->f.subclass |= 0x1; 01341 01342 } 01343 rtp->f.src = "RTP"; 01344 return &rtp->f; 01345 }
int ast_rtp_reload | ( | void | ) |
Definition at line 3760 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().
03761 { 03762 struct ast_config *cfg; 03763 const char *s; 03764 03765 rtpstart = 5000; 03766 rtpend = 31000; 03767 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03768 cfg = ast_config_load("rtp.conf"); 03769 if (cfg) { 03770 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { 03771 rtpstart = atoi(s); 03772 if (rtpstart < 1024) 03773 rtpstart = 1024; 03774 if (rtpstart > 65535) 03775 rtpstart = 65535; 03776 } 03777 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { 03778 rtpend = atoi(s); 03779 if (rtpend < 1024) 03780 rtpend = 1024; 03781 if (rtpend > 65535) 03782 rtpend = 65535; 03783 } 03784 if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) { 03785 rtcpinterval = atoi(s); 03786 if (rtcpinterval == 0) 03787 rtcpinterval = 0; /* Just so we're clear... it's zero */ 03788 if (rtcpinterval < RTCP_MIN_INTERVALMS) 03789 rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */ 03790 if (rtcpinterval > RTCP_MAX_INTERVALMS) 03791 rtcpinterval = RTCP_MAX_INTERVALMS; 03792 } 03793 if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) { 03794 #ifdef SO_NO_CHECK 03795 if (ast_false(s)) 03796 nochecksums = 1; 03797 else 03798 nochecksums = 0; 03799 #else 03800 if (ast_false(s)) 03801 ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n"); 03802 #endif 03803 } 03804 if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { 03805 dtmftimeout = atoi(s); 03806 if ((dtmftimeout < 0) || (dtmftimeout > 20000)) { 03807 ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", 03808 dtmftimeout, DEFAULT_DTMF_TIMEOUT); 03809 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03810 }; 03811 } 03812 ast_config_destroy(cfg); 03813 } 03814 if (rtpstart >= rtpend) { 03815 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n"); 03816 rtpstart = 5000; 03817 rtpend = 31000; 03818 } 03819 if (option_verbose > 1) 03820 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); 03821 return 0; 03822 }
void ast_rtp_reset | ( | struct ast_rtp * | rtp | ) |
Definition at line 2079 of file rtp.c.
References ast_rtp::dtmfcount, ast_rtp::dtmfmute, ast_rtp::dtmfsamples, 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.
02080 { 02081 memset(&rtp->rxcore, 0, sizeof(rtp->rxcore)); 02082 memset(&rtp->txcore, 0, sizeof(rtp->txcore)); 02083 memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute)); 02084 rtp->lastts = 0; 02085 rtp->lastdigitts = 0; 02086 rtp->lastrxts = 0; 02087 rtp->lastividtimestamp = 0; 02088 rtp->lastovidtimestamp = 0; 02089 rtp->lasteventseqn = 0; 02090 rtp->lastevent = 0; 02091 rtp->lasttxformat = 0; 02092 rtp->lastrxformat = 0; 02093 rtp->dtmfcount = 0; 02094 rtp->dtmfsamples = 0; 02095 rtp->seqno = 0; 02096 rtp->rxseqno = 0; 02097 }
int ast_rtp_sendcng | ( | struct ast_rtp * | rtp, | |
int | level | |||
) |
generate comfort noice (CNG)
Definition at line 2594 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_CN, ast_rtp_lookup_code(), ast_tvadd(), 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().
02595 { 02596 unsigned int *rtpheader; 02597 int hdrlen = 12; 02598 int res; 02599 int payload; 02600 char data[256]; 02601 level = 127 - (level & 0x7f); 02602 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN); 02603 02604 /* If we have no peer, return immediately */ 02605 if (!rtp->them.sin_addr.s_addr) 02606 return 0; 02607 02608 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02609 02610 /* Get a pointer to the header */ 02611 rtpheader = (unsigned int *)data; 02612 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++)); 02613 rtpheader[1] = htonl(rtp->lastts); 02614 rtpheader[2] = htonl(rtp->ssrc); 02615 data[12] = level; 02616 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { 02617 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); 02618 if (res <0) 02619 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)); 02620 if (rtp_debug_test_addr(&rtp->them)) 02621 ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n" 02622 , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); 02623 02624 } 02625 return 0; 02626 }
int ast_rtp_senddigit_begin | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
Send begin frames for DTMF.
Definition at line 2202 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_tvadd(), ast_verbose(), ast_rtp::dtmfmute, errno, ast_rtp::lastdigitts, 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().
02203 { 02204 unsigned int *rtpheader; 02205 int hdrlen = 12, res = 0, i = 0, payload = 0; 02206 char data[256]; 02207 02208 if ((digit <= '9') && (digit >= '0')) 02209 digit -= '0'; 02210 else if (digit == '*') 02211 digit = 10; 02212 else if (digit == '#') 02213 digit = 11; 02214 else if ((digit >= 'A') && (digit <= 'D')) 02215 digit = digit - 'A' + 12; 02216 else if ((digit >= 'a') && (digit <= 'd')) 02217 digit = digit - 'a' + 12; 02218 else { 02219 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); 02220 return 0; 02221 } 02222 02223 /* If we have no peer, return immediately */ 02224 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 02225 return 0; 02226 02227 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); 02228 02229 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02230 rtp->send_duration = 160; 02231 02232 /* Get a pointer to the header */ 02233 rtpheader = (unsigned int *)data; 02234 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno)); 02235 rtpheader[1] = htonl(rtp->lastdigitts); 02236 rtpheader[2] = htonl(rtp->ssrc); 02237 02238 for (i = 0; i < 2; i++) { 02239 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration)); 02240 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); 02241 if (res < 0) 02242 ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n", 02243 ast_inet_ntoa(rtp->them.sin_addr), 02244 ntohs(rtp->them.sin_port), strerror(errno)); 02245 if (rtp_debug_test_addr(&rtp->them)) 02246 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 02247 ast_inet_ntoa(rtp->them.sin_addr), 02248 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 02249 /* Increment sequence number */ 02250 rtp->seqno++; 02251 /* Increment duration */ 02252 rtp->send_duration += 160; 02253 /* Clear marker bit and set seqno */ 02254 rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); 02255 } 02256 02257 /* Since we received a begin, we can safely store the digit and disable any compensation */ 02258 rtp->sending_digit = 1; 02259 rtp->send_digit = digit; 02260 rtp->send_payload = payload; 02261 02262 return 0; 02263 }
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 586 of file rtp.c.
References ast_rtp::callback.
Referenced by start_rtp().
00587 { 00588 rtp->callback = callback; 00589 }
void ast_rtp_set_data | ( | struct ast_rtp * | rtp, | |
void * | data | |||
) |
Definition at line 581 of file rtp.c.
References ast_rtp::data.
Referenced by start_rtp().
00582 { 00583 rtp->data = data; 00584 }
void ast_rtp_set_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Activate payload type.
Definition at line 1657 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_newcall(), and process_sdp().
01658 { 01659 if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 01660 return; /* bogus payload type */ 01661 01662 ast_mutex_lock(&rtp->bridge_lock); 01663 rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; 01664 ast_mutex_unlock(&rtp->bridge_lock); 01665 }
void ast_rtp_set_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2025 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().
02026 { 02027 rtp->them.sin_port = them->sin_port; 02028 rtp->them.sin_addr = them->sin_addr; 02029 if (rtp->rtcp) { 02030 rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1); 02031 rtp->rtcp->them.sin_addr = them->sin_addr; 02032 } 02033 rtp->rxseqno = 0; 02034 }
void ast_rtp_set_rtpholdtimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp hold timeout.
Definition at line 548 of file rtp.c.
References ast_rtp::rtpholdtimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00549 { 00550 rtp->rtpholdtimeout = timeout; 00551 }
void ast_rtp_set_rtpkeepalive | ( | struct ast_rtp * | rtp, | |
int | period | |||
) |
set RTP keepalive interval
Definition at line 554 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by create_addr_from_peer(), and sip_alloc().
00555 { 00556 rtp->rtpkeepalive = period; 00557 }
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 1684 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_newcall(), process_sdp(), and set_dtmf_payload().
01687 { 01688 unsigned int i; 01689 int found = 0; 01690 01691 if (pt < 0 || pt > MAX_RTP_PT) 01692 return -1; /* bogus payload type */ 01693 01694 ast_mutex_lock(&rtp->bridge_lock); 01695 01696 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01697 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && 01698 strcasecmp(mimeType, mimeTypes[i].type) == 0) { 01699 found = 1; 01700 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; 01701 if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) && 01702 mimeTypes[i].payloadType.isAstFormat && 01703 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01704 rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2; 01705 break; 01706 } 01707 } 01708 01709 ast_mutex_unlock(&rtp->bridge_lock); 01710 01711 return (found ? 0 : -1); 01712 }
void ast_rtp_set_rtptimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp timeout.
Definition at line 542 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00543 { 00544 rtp->rtptimeout = timeout; 00545 }
void ast_rtp_set_rtptimers_onhold | ( | struct ast_rtp * | rtp | ) |
Definition at line 535 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by handle_response_invite().
00536 { 00537 rtp->rtptimeout = (-1) * rtp->rtptimeout; 00538 rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout; 00539 }
void ast_rtp_setdtmf | ( | struct ast_rtp * | rtp, | |
int | dtmf | |||
) |
Indicate whether this RTP session is carrying DTMF or not.
Definition at line 601 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().
00602 { 00603 ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF); 00604 }
void ast_rtp_setdtmfcompensate | ( | struct ast_rtp * | rtp, | |
int | compensate | |||
) |
Compensate for devices that send RFC2833 packets all at once.
Definition at line 606 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().
00607 { 00608 ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE); 00609 }
void ast_rtp_setnat | ( | struct ast_rtp * | rtp, | |
int | nat | |||
) |
Definition at line 591 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 611 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_STUN.
Referenced by gtalk_new().
00612 { 00613 ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN); 00614 }
int ast_rtp_settos | ( | struct ast_rtp * | rtp, | |
int | tos | |||
) |
Definition at line 2009 of file rtp.c.
References ast_log(), LOG_WARNING, and ast_rtp::s.
Referenced by __oh323_rtp_create(), and sip_alloc().
02010 { 02011 int res; 02012 02013 if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 02014 ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); 02015 return res; 02016 }
void ast_rtp_stop | ( | struct ast_rtp * | rtp | ) |
Definition at line 2065 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().
02066 { 02067 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02068 02069 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 02070 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 02071 if (rtp->rtcp) { 02072 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr)); 02073 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port)); 02074 } 02075 02076 ast_clear_flag(rtp, FLAG_P2P_SENT_MARK); 02077 }
void ast_rtp_stun_request | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | suggestion, | |||
const char * | username | |||
) |
Definition at line 403 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().
00404 { 00405 struct stun_header *req; 00406 unsigned char reqdata[1024]; 00407 int reqlen, reqleft; 00408 struct stun_attr *attr; 00409 00410 req = (struct stun_header *)reqdata; 00411 stun_req_id(req); 00412 reqlen = 0; 00413 reqleft = sizeof(reqdata) - sizeof(struct stun_header); 00414 req->msgtype = 0; 00415 req->msglen = 0; 00416 attr = (struct stun_attr *)req->ies; 00417 if (username) 00418 append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft); 00419 req->msglen = htons(reqlen); 00420 req->msgtype = htons(STUN_BINDREQ); 00421 stun_send(rtp->s, suggestion, req); 00422 }
void ast_rtp_unset_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
clear payload type
Definition at line 1669 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().
01670 { 01671 if (pt < 0 || pt > MAX_RTP_PT) 01672 return; /* bogus payload type */ 01673 01674 ast_mutex_lock(&rtp->bridge_lock); 01675 rtp->current_RTP_PT[pt].isAstFormat = 0; 01676 rtp->current_RTP_PT[pt].code = 0; 01677 ast_mutex_unlock(&rtp->bridge_lock); 01678 }
Definition at line 2753 of file rtp.c.
References ast_codec_pref_getsize(), 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_frame::datalen, f, fmt, ast_frame::frametype, 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().
02754 { 02755 struct ast_frame *f; 02756 int codec; 02757 int hdrlen = 12; 02758 int subclass; 02759 02760 02761 /* If we have no peer, return immediately */ 02762 if (!rtp->them.sin_addr.s_addr) 02763 return 0; 02764 02765 /* If there is no data length, return immediately */ 02766 if (!_f->datalen) 02767 return 0; 02768 02769 /* Make sure we have enough space for RTP header */ 02770 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { 02771 ast_log(LOG_WARNING, "RTP can only send voice and video\n"); 02772 return -1; 02773 } 02774 02775 subclass = _f->subclass; 02776 if (_f->frametype == AST_FRAME_VIDEO) 02777 subclass &= ~0x1; 02778 02779 codec = ast_rtp_lookup_code(rtp, 1, subclass); 02780 if (codec < 0) { 02781 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); 02782 return -1; 02783 } 02784 02785 if (rtp->lasttxformat != subclass) { 02786 /* New format, reset the smoother */ 02787 if (option_debug) 02788 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); 02789 rtp->lasttxformat = subclass; 02790 if (rtp->smoother) 02791 ast_smoother_free(rtp->smoother); 02792 rtp->smoother = NULL; 02793 } 02794 02795 if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) { 02796 struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass); 02797 if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */ 02798 if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) { 02799 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)); 02800 return -1; 02801 } 02802 if (fmt.flags) 02803 ast_smoother_set_flags(rtp->smoother, fmt.flags); 02804 if (option_debug) 02805 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)); 02806 } 02807 } 02808 if (rtp->smoother) { 02809 if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) { 02810 ast_smoother_feed_be(rtp->smoother, _f); 02811 } else { 02812 ast_smoother_feed(rtp->smoother, _f); 02813 } 02814 02815 while((f = ast_smoother_read(rtp->smoother)) && (f->data)) 02816 ast_rtp_raw_write(rtp, f, codec); 02817 } else { 02818 /* Don't buffer outgoing frames; send them one-per-packet: */ 02819 if (_f->offset < hdrlen) { 02820 f = ast_frdup(_f); 02821 } else { 02822 f = _f; 02823 } 02824 if (f->data) 02825 ast_rtp_raw_write(rtp, f, codec); 02826 if (f != _f) 02827 ast_frfree(f); 02828 } 02829 02830 return 0; 02831 }