#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(), 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 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_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_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 ast_assert(errno != EBADF); 00859 if (errno != EAGAIN) { 00860 ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno)); 00861 return NULL; 00862 } 00863 return &ast_null_frame; 00864 } 00865 00866 packetwords = res / 4; 00867 00868 if (rtp->nat) { 00869 /* Send to whoever sent to us */ 00870 if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 00871 (rtp->rtcp->them.sin_port != sin.sin_port)) { 00872 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 00873 if (option_debug || rtpdebug) 00874 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)); 00875 } 00876 } 00877 00878 if (option_debug) 00879 ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res); 00880 00881 /* Process a compound packet */ 00882 position = 0; 00883 while (position < packetwords) { 00884 i = position; 00885 length = ntohl(rtcpheader[i]); 00886 pt = (length & 0xff0000) >> 16; 00887 rc = (length & 0x1f000000) >> 24; 00888 length &= 0xffff; 00889 00890 if ((i + length) > packetwords) { 00891 ast_log(LOG_WARNING, "RTCP Read too short\n"); 00892 return &ast_null_frame; 00893 } 00894 00895 if (rtcp_debug_test_addr(&sin)) { 00896 ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); 00897 ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown"); 00898 ast_verbose("Reception reports: %d\n", rc); 00899 ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]); 00900 } 00901 00902 i += 2; /* Advance past header and ssrc */ 00903 00904 switch (pt) { 00905 case RTCP_PT_SR: 00906 gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */ 00907 rtp->rtcp->spc = ntohl(rtcpheader[i+3]); 00908 rtp->rtcp->soc = ntohl(rtcpheader[i + 4]); 00909 rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/ 00910 00911 if (rtcp_debug_test_addr(&sin)) { 00912 ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096); 00913 ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2])); 00914 ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4])); 00915 } 00916 i += 5; 00917 if (rc < 1) 00918 break; 00919 /* Intentional fall through */ 00920 case RTCP_PT_RR: 00921 /* Don't handle multiple reception reports (rc > 1) yet */ 00922 /* Calculate RTT per RFC */ 00923 gettimeofday(&now, NULL); 00924 timeval2ntp(now, &msw, &lsw); 00925 if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */ 00926 comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16); 00927 lsr = ntohl(rtcpheader[i + 4]); 00928 dlsr = ntohl(rtcpheader[i + 5]); 00929 rtt = comp - lsr - dlsr; 00930 00931 /* Convert end to end delay to usec (keeping the calculation in 64bit space) 00932 sess->ee_delay = (eedelay * 1000) / 65536; */ 00933 if (rtt < 4294) { 00934 rtt = (rtt * 1000000) >> 16; 00935 } else { 00936 rtt = (rtt * 1000) >> 16; 00937 rtt *= 1000; 00938 } 00939 rtt = rtt / 1000.; 00940 rttsec = rtt / 1000.; 00941 00942 if (comp - dlsr >= lsr) { 00943 rtp->rtcp->accumulated_transit += rttsec; 00944 rtp->rtcp->rtt = rttsec; 00945 if (rtp->rtcp->maxrtt<rttsec) 00946 rtp->rtcp->maxrtt = rttsec; 00947 if (rtp->rtcp->minrtt>rttsec) 00948 rtp->rtcp->minrtt = rttsec; 00949 } else if (rtcp_debug_test_addr(&sin)) { 00950 ast_verbose("Internal RTCP NTP clock skew detected: " 00951 "lsr=%u, now=%u, dlsr=%u (%d:%03dms), " 00952 "diff=%d\n", 00953 lsr, comp, dlsr, dlsr / 65536, 00954 (dlsr % 65536) * 1000 / 65536, 00955 dlsr - (comp - lsr)); 00956 } 00957 } 00958 00959 rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]); 00960 rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff; 00961 if (rtcp_debug_test_addr(&sin)) { 00962 ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24)); 00963 ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost); 00964 ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff)); 00965 ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16); 00966 ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter); 00967 ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096); 00968 ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0); 00969 if (rtt) 00970 ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt); 00971 } 00972 break; 00973 case RTCP_PT_FUR: 00974 if (rtcp_debug_test_addr(&sin)) 00975 ast_verbose("Received an RTCP Fast Update Request\n"); 00976 rtp->f.frametype = AST_FRAME_CONTROL; 00977 rtp->f.subclass = AST_CONTROL_VIDUPDATE; 00978 rtp->f.datalen = 0; 00979 rtp->f.samples = 0; 00980 rtp->f.mallocd = 0; 00981 rtp->f.src = "RTP"; 00982 f = &rtp->f; 00983 break; 00984 case RTCP_PT_SDES: 00985 if (rtcp_debug_test_addr(&sin)) 00986 ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00987 break; 00988 case RTCP_PT_BYE: 00989 if (rtcp_debug_test_addr(&sin)) 00990 ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00991 break; 00992 default: 00993 if (option_debug) 00994 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)); 00995 break; 00996 } 00997 position += (length + 1); 00998 } 00999 01000 return f; 01001 }
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 2341 of file rtp.c.
References ast_rtcp_write(), ast_rtp::rtcp, and ast_rtcp::sendfur.
02342 { 02343 struct ast_rtp *rtp = data; 02344 int res; 02345 02346 rtp->rtcp->sendfur = 1; 02347 res = ast_rtcp_write(data); 02348 02349 return res; 02350 }
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 3279 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, 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.
03280 { 03281 struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ 03282 struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */ 03283 struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL; 03284 enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED; 03285 enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED; 03286 enum ast_bridge_result res = AST_BRIDGE_FAILED; 03287 int codec0 = 0, codec1 = 0; 03288 void *pvt0 = NULL, *pvt1 = NULL; 03289 03290 /* Lock channels */ 03291 ast_channel_lock(c0); 03292 while(ast_channel_trylock(c1)) { 03293 ast_channel_unlock(c0); 03294 usleep(1); 03295 ast_channel_lock(c0); 03296 } 03297 03298 /* Ensure neither channel got hungup during lock avoidance */ 03299 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 03300 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name); 03301 ast_channel_unlock(c0); 03302 ast_channel_unlock(c1); 03303 return AST_BRIDGE_FAILED; 03304 } 03305 03306 /* Find channel driver interfaces */ 03307 if (!(pr0 = get_proto(c0))) { 03308 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); 03309 ast_channel_unlock(c0); 03310 ast_channel_unlock(c1); 03311 return AST_BRIDGE_FAILED; 03312 } 03313 if (!(pr1 = get_proto(c1))) { 03314 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); 03315 ast_channel_unlock(c0); 03316 ast_channel_unlock(c1); 03317 return AST_BRIDGE_FAILED; 03318 } 03319 03320 /* Get channel specific interface structures */ 03321 pvt0 = c0->tech_pvt; 03322 pvt1 = c1->tech_pvt; 03323 03324 /* Get audio and video interface (if native bridge is possible) */ 03325 audio_p0_res = pr0->get_rtp_info(c0, &p0); 03326 video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED; 03327 audio_p1_res = pr1->get_rtp_info(c1, &p1); 03328 video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED; 03329 03330 /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */ 03331 if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE)) 03332 audio_p0_res = AST_RTP_GET_FAILED; 03333 if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE)) 03334 audio_p1_res = AST_RTP_GET_FAILED; 03335 03336 /* Check if a bridge is possible (partial/native) */ 03337 if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) { 03338 /* Somebody doesn't want to play... */ 03339 ast_channel_unlock(c0); 03340 ast_channel_unlock(c1); 03341 return AST_BRIDGE_FAILED_NOWARN; 03342 } 03343 03344 /* If we need to feed DTMF frames into the core then only do a partial native bridge */ 03345 if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) { 03346 ast_set_flag(p0, FLAG_P2P_NEED_DTMF); 03347 audio_p0_res = AST_RTP_TRY_PARTIAL; 03348 } 03349 03350 if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) { 03351 ast_set_flag(p1, FLAG_P2P_NEED_DTMF); 03352 audio_p1_res = AST_RTP_TRY_PARTIAL; 03353 } 03354 03355 /* If both sides are not using the same method of DTMF transmission 03356 * (ie: one is RFC2833, other is INFO... then we can not do direct media. 03357 * -------------------------------------------------- 03358 * | DTMF Mode | HAS_DTMF | Accepts Begin Frames | 03359 * |-----------|------------|-----------------------| 03360 * | Inband | False | True | 03361 * | RFC2833 | True | True | 03362 * | SIP INFO | False | False | 03363 * -------------------------------------------------- 03364 * However, if DTMF from both channels is being monitored by the core, then 03365 * we can still do packet-to-packet bridging, because passing through the 03366 * core will handle DTMF mode translation. 03367 */ 03368 if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) || 03369 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) { 03370 if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) { 03371 ast_channel_unlock(c0); 03372 ast_channel_unlock(c1); 03373 return AST_BRIDGE_FAILED_NOWARN; 03374 } 03375 audio_p0_res = AST_RTP_TRY_PARTIAL; 03376 audio_p1_res = AST_RTP_TRY_PARTIAL; 03377 } 03378 03379 /* If we need to feed frames into the core don't do a P2P bridge */ 03380 if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) || 03381 (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) { 03382 ast_channel_unlock(c0); 03383 ast_channel_unlock(c1); 03384 return AST_BRIDGE_FAILED_NOWARN; 03385 } 03386 03387 /* Get codecs from both sides */ 03388 codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0; 03389 codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0; 03390 if (codec0 && codec1 && !(codec0 & codec1)) { 03391 /* Hey, we can't do native bridging if both parties speak different codecs */ 03392 if (option_debug) 03393 ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); 03394 ast_channel_unlock(c0); 03395 ast_channel_unlock(c1); 03396 return AST_BRIDGE_FAILED_NOWARN; 03397 } 03398 03399 /* If either side can only do a partial bridge, then don't try for a true native bridge */ 03400 if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) { 03401 struct ast_format_list fmt0, fmt1; 03402 03403 /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */ 03404 if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) { 03405 if (option_debug) 03406 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n"); 03407 ast_channel_unlock(c0); 03408 ast_channel_unlock(c1); 03409 return AST_BRIDGE_FAILED_NOWARN; 03410 } 03411 /* They must also be using the same packetization */ 03412 fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat); 03413 fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat); 03414 if (fmt0.cur_ms != fmt1.cur_ms) { 03415 if (option_debug) 03416 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n"); 03417 ast_channel_unlock(c0); 03418 ast_channel_unlock(c1); 03419 return AST_BRIDGE_FAILED_NOWARN; 03420 } 03421 03422 if (option_verbose > 2) 03423 ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name); 03424 res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1); 03425 } else { 03426 if (option_verbose > 2) 03427 ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name); 03428 res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1); 03429 } 03430 03431 return res; 03432 }
int ast_rtp_codec_getformat | ( | int | pt | ) |
Definition at line 2723 of file rtp.c.
References rtpPayloadType::code, and static_RTP_PT.
Referenced by process_sdp().
02724 { 02725 if (pt < 0 || pt > MAX_RTP_PT) 02726 return 0; /* bogus payload type */ 02727 02728 if (static_RTP_PT[pt].isAstFormat) 02729 return static_RTP_PT[pt].code; 02730 else 02731 return 0; 02732 }
struct ast_codec_pref* ast_rtp_codec_getpref | ( | struct ast_rtp * | rtp | ) |
Definition at line 2718 of file rtp.c.
References ast_rtp::pref.
Referenced by add_codec_to_sdp(), and process_sdp().
02719 { 02720 return &rtp->pref; 02721 }
int ast_rtp_codec_setpref | ( | struct ast_rtp * | rtp, | |
struct ast_codec_pref * | prefs | |||
) |
Definition at line 2705 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(), sip_alloc(), start_rtp(), and transmit_response_with_sdp().
02706 { 02707 int x; 02708 for (x = 0; x < 32; x++) { /* Ugly way */ 02709 rtp->pref.order[x] = prefs->order[x]; 02710 rtp->pref.framing[x] = prefs->framing[x]; 02711 } 02712 if (rtp->smoother) 02713 ast_smoother_free(rtp->smoother); 02714 rtp->smoother = NULL; 02715 return 0; 02716 }
void ast_rtp_destroy | ( | struct ast_rtp * | rtp | ) |
Definition at line 2124 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().
02125 { 02126 if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) { 02127 /*Print some info on the call here */ 02128 ast_verbose(" RTP-stats\n"); 02129 ast_verbose("* Our Receiver:\n"); 02130 ast_verbose(" SSRC: %u\n", rtp->themssrc); 02131 ast_verbose(" Received packets: %u\n", rtp->rxcount); 02132 ast_verbose(" Lost packets: %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior); 02133 ast_verbose(" Jitter: %.4f\n", rtp->rxjitter); 02134 ast_verbose(" Transit: %.4f\n", rtp->rxtransit); 02135 ast_verbose(" RR-count: %u\n", rtp->rtcp->rr_count); 02136 ast_verbose("* Our Sender:\n"); 02137 ast_verbose(" SSRC: %u\n", rtp->ssrc); 02138 ast_verbose(" Sent packets: %u\n", rtp->txcount); 02139 ast_verbose(" Lost packets: %u\n", rtp->rtcp->reported_lost); 02140 ast_verbose(" Jitter: %u\n", rtp->rtcp->reported_jitter / (unsigned int)65536.0); 02141 ast_verbose(" SR-count: %u\n", rtp->rtcp->sr_count); 02142 ast_verbose(" RTT: %f\n", rtp->rtcp->rtt); 02143 } 02144 02145 if (rtp->smoother) 02146 ast_smoother_free(rtp->smoother); 02147 if (rtp->ioid) 02148 ast_io_remove(rtp->io, rtp->ioid); 02149 if (rtp->s > -1) 02150 close(rtp->s); 02151 if (rtp->rtcp) { 02152 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02153 close(rtp->rtcp->s); 02154 free(rtp->rtcp); 02155 rtp->rtcp=NULL; 02156 } 02157 02158 ast_mutex_destroy(&rtp->bridge_lock); 02159 02160 free(rtp); 02161 }
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 1476 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().
01477 { 01478 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01479 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01480 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01481 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01482 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01483 int srccodec, destcodec, nat_active = 0; 01484 01485 /* Lock channels */ 01486 ast_channel_lock(dest); 01487 if (src) { 01488 while(ast_channel_trylock(src)) { 01489 ast_channel_unlock(dest); 01490 usleep(1); 01491 ast_channel_lock(dest); 01492 } 01493 } 01494 01495 /* Find channel driver interfaces */ 01496 destpr = get_proto(dest); 01497 if (src) 01498 srcpr = get_proto(src); 01499 if (!destpr) { 01500 if (option_debug) 01501 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01502 ast_channel_unlock(dest); 01503 if (src) 01504 ast_channel_unlock(src); 01505 return 0; 01506 } 01507 if (!srcpr) { 01508 if (option_debug) 01509 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>"); 01510 ast_channel_unlock(dest); 01511 if (src) 01512 ast_channel_unlock(src); 01513 return 0; 01514 } 01515 01516 /* Get audio and video interface (if native bridge is possible) */ 01517 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01518 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01519 if (srcpr) { 01520 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01521 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01522 } 01523 01524 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01525 if (audio_dest_res != AST_RTP_TRY_NATIVE) { 01526 /* Somebody doesn't want to play... */ 01527 ast_channel_unlock(dest); 01528 if (src) 01529 ast_channel_unlock(src); 01530 return 0; 01531 } 01532 if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec) 01533 srccodec = srcpr->get_codec(src); 01534 else 01535 srccodec = 0; 01536 if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec) 01537 destcodec = destpr->get_codec(dest); 01538 else 01539 destcodec = 0; 01540 /* Ensure we have at least one matching codec */ 01541 if (!(srccodec & destcodec)) { 01542 ast_channel_unlock(dest); 01543 if (src) 01544 ast_channel_unlock(src); 01545 return 0; 01546 } 01547 /* Consider empty media as non-existant */ 01548 if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) 01549 srcp = NULL; 01550 /* If the client has NAT stuff turned on then just safe NAT is active */ 01551 if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01552 nat_active = 1; 01553 /* Bridge media early */ 01554 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active)) 01555 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01556 ast_channel_unlock(dest); 01557 if (src) 01558 ast_channel_unlock(src); 01559 if (option_debug) 01560 ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01561 return 1; 01562 }
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 2035 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().
02036 { 02037 struct ast_rtp *bridged = NULL; 02038 02039 ast_mutex_lock(&rtp->bridge_lock); 02040 bridged = rtp->bridged; 02041 ast_mutex_unlock(&rtp->bridge_lock); 02042 02043 return bridged; 02044 }
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 1698 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().
01700 { 01701 int pt; 01702 01703 ast_mutex_lock(&rtp->bridge_lock); 01704 01705 *astFormats = *nonAstFormats = 0; 01706 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01707 if (rtp->current_RTP_PT[pt].isAstFormat) { 01708 *astFormats |= rtp->current_RTP_PT[pt].code; 01709 } else { 01710 *nonAstFormats |= rtp->current_RTP_PT[pt].code; 01711 } 01712 } 01713 01714 ast_mutex_unlock(&rtp->bridge_lock); 01715 01716 return; 01717 }
int ast_rtp_get_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2017 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().
02018 { 02019 if ((them->sin_family != AF_INET) || 02020 (them->sin_port != rtp->them.sin_port) || 02021 (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) { 02022 them->sin_family = AF_INET; 02023 them->sin_port = rtp->them.sin_port; 02024 them->sin_addr = rtp->them.sin_addr; 02025 return 1; 02026 } 02027 return 0; 02028 }
char* ast_rtp_get_quality | ( | struct ast_rtp * | rtp, | |
struct ast_rtp_quality * | qual | |||
) |
Return RTCP quality string.
Definition at line 2080 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().
02081 { 02082 /* 02083 *ssrc our ssrc 02084 *themssrc their ssrc 02085 *lp lost packets 02086 *rxjitter our calculated jitter(rx) 02087 *rxcount no. received packets 02088 *txjitter reported jitter of the other end 02089 *txcount transmitted packets 02090 *rlp remote lost packets 02091 *rtt round trip time 02092 */ 02093 02094 if (qual && rtp) { 02095 qual->local_ssrc = rtp->ssrc; 02096 qual->local_jitter = rtp->rxjitter; 02097 qual->local_count = rtp->rxcount; 02098 qual->remote_ssrc = rtp->themssrc; 02099 qual->remote_count = rtp->txcount; 02100 if (rtp->rtcp) { 02101 qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior; 02102 qual->remote_lostpackets = rtp->rtcp->reported_lost; 02103 qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0; 02104 qual->rtt = rtp->rtcp->rtt; 02105 } 02106 } 02107 if (rtp->rtcp) { 02108 snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), 02109 "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", 02110 rtp->ssrc, 02111 rtp->themssrc, 02112 rtp->rtcp->expected_prior - rtp->rtcp->received_prior, 02113 rtp->rxjitter, 02114 rtp->rxcount, 02115 (double)rtp->rtcp->reported_jitter / 65536.0, 02116 rtp->txcount, 02117 rtp->rtcp->reported_lost, 02118 rtp->rtcp->rtt); 02119 return rtp->rtcp->quality; 02120 } else 02121 return "<Unknown> - RTP/RTCP has already been destroyed"; 02122 }
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 2030 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 3817 of file rtp.c.
References ast_cli_register_multiple(), ast_rtp_reload(), and cli_rtp.
Referenced by main().
03818 { 03819 ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry)); 03820 ast_rtp_reload(); 03821 }
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 1741 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().
01742 { 01743 int pt = 0; 01744 01745 ast_mutex_lock(&rtp->bridge_lock); 01746 01747 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && 01748 code == rtp->rtp_lookup_code_cache_code) { 01749 /* Use our cached mapping, to avoid the overhead of the loop below */ 01750 pt = rtp->rtp_lookup_code_cache_result; 01751 ast_mutex_unlock(&rtp->bridge_lock); 01752 return pt; 01753 } 01754 01755 /* Check the dynamic list first */ 01756 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01757 if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) { 01758 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01759 rtp->rtp_lookup_code_cache_code = code; 01760 rtp->rtp_lookup_code_cache_result = pt; 01761 ast_mutex_unlock(&rtp->bridge_lock); 01762 return pt; 01763 } 01764 } 01765 01766 /* Then the static list */ 01767 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01768 if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) { 01769 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01770 rtp->rtp_lookup_code_cache_code = code; 01771 rtp->rtp_lookup_code_cache_result = pt; 01772 ast_mutex_unlock(&rtp->bridge_lock); 01773 return pt; 01774 } 01775 } 01776 01777 ast_mutex_unlock(&rtp->bridge_lock); 01778 01779 return -1; 01780 }
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 1801 of file rtp.c.
References ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, format, len, and name.
Referenced by process_sdp().
01803 { 01804 int format; 01805 unsigned len; 01806 char *end = buf; 01807 char *start = buf; 01808 01809 if (!buf || !size) 01810 return NULL; 01811 01812 snprintf(end, size, "0x%x (", capability); 01813 01814 len = strlen(end); 01815 end += len; 01816 size -= len; 01817 start = end; 01818 01819 for (format = 1; format < AST_RTP_MAX; format <<= 1) { 01820 if (capability & format) { 01821 const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options); 01822 01823 snprintf(end, size, "%s|", name); 01824 len = strlen(end); 01825 end += len; 01826 size -= len; 01827 } 01828 } 01829 01830 if (start == end) 01831 snprintf(start, size, "nothing)"); 01832 else if (size > 1) 01833 *(end -1) = ')'; 01834 01835 return buf; 01836 }
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 1782 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().
01784 { 01785 unsigned int i; 01786 01787 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01788 if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) { 01789 if (isAstFormat && 01790 (code == AST_FORMAT_G726_AAL2) && 01791 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01792 return "G726-32"; 01793 else 01794 return mimeTypes[i].subtype; 01795 } 01796 } 01797 01798 return ""; 01799 }
struct rtpPayloadType ast_rtp_lookup_pt | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Mapping between RTP payload format codes and Asterisk codes:.
Definition at line 1719 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().
01720 { 01721 struct rtpPayloadType result; 01722 01723 result.isAstFormat = result.code = 0; 01724 01725 if (pt < 0 || pt > MAX_RTP_PT) 01726 return result; /* bogus payload type */ 01727 01728 /* Start with negotiated codecs */ 01729 ast_mutex_lock(&rtp->bridge_lock); 01730 result = rtp->current_RTP_PT[pt]; 01731 ast_mutex_unlock(&rtp->bridge_lock); 01732 01733 /* If it doesn't exist, check our static RTP type list, just in case */ 01734 if (!result.code) 01735 result = static_RTP_PT[pt]; 01736 01737 return result; 01738 }
int ast_rtp_make_compatible | ( | struct ast_channel * | dest, | |
struct ast_channel * | src, | |||
int | media | |||
) |
Definition at line 1564 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().
01565 { 01566 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01567 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01568 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01569 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01570 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01571 int srccodec, destcodec; 01572 01573 /* Lock channels */ 01574 ast_channel_lock(dest); 01575 while(ast_channel_trylock(src)) { 01576 ast_channel_unlock(dest); 01577 usleep(1); 01578 ast_channel_lock(dest); 01579 } 01580 01581 /* Find channel driver interfaces */ 01582 if (!(destpr = get_proto(dest))) { 01583 if (option_debug) 01584 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01585 ast_channel_unlock(dest); 01586 ast_channel_unlock(src); 01587 return 0; 01588 } 01589 if (!(srcpr = get_proto(src))) { 01590 if (option_debug) 01591 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name); 01592 ast_channel_unlock(dest); 01593 ast_channel_unlock(src); 01594 return 0; 01595 } 01596 01597 /* Get audio and video interface (if native bridge is possible) */ 01598 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01599 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01600 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01601 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01602 01603 /* Ensure we have at least one matching codec */ 01604 if (srcpr->get_codec) 01605 srccodec = srcpr->get_codec(src); 01606 else 01607 srccodec = 0; 01608 if (destpr->get_codec) 01609 destcodec = destpr->get_codec(dest); 01610 else 01611 destcodec = 0; 01612 01613 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01614 if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) { 01615 /* Somebody doesn't want to play... */ 01616 ast_channel_unlock(dest); 01617 ast_channel_unlock(src); 01618 return 0; 01619 } 01620 ast_rtp_pt_copy(destp, srcp); 01621 if (vdestp && vsrcp) 01622 ast_rtp_pt_copy(vdestp, vsrcp); 01623 if (media) { 01624 /* Bridge early */ 01625 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01626 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name); 01627 } 01628 ast_channel_unlock(dest); 01629 ast_channel_unlock(src); 01630 if (option_debug) 01631 ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name); 01632 return 1; 01633 }
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 1983 of file rtp.c.
References ast_rtp_new_with_bindaddr(), io, and sched.
01984 { 01985 struct in_addr ia; 01986 01987 memset(&ia, 0, sizeof(ia)); 01988 return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia); 01989 }
void ast_rtp_new_init | ( | struct ast_rtp * | rtp | ) |
Initialize a new RTP structure.
Definition at line 1883 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().
01884 { 01885 ast_mutex_init(&rtp->bridge_lock); 01886 01887 rtp->them.sin_family = AF_INET; 01888 rtp->us.sin_family = AF_INET; 01889 rtp->ssrc = ast_random(); 01890 rtp->seqno = ast_random() & 0xffff; 01891 ast_set_flag(rtp, FLAG_HAS_DTMF); 01892 01893 return; 01894 }
void ast_rtp_new_source | ( | struct ast_rtp * | rtp | ) |
Definition at line 2000 of file rtp.c.
References ast_rtp::set_marker_bit.
Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), sip_write(), and skinny_indicate().
02001 { 02002 rtp->set_marker_bit = 1; 02003 return; 02004 }
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 1896 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().
01897 { 01898 struct ast_rtp *rtp; 01899 int x; 01900 int first; 01901 int startplace; 01902 01903 if (!(rtp = ast_calloc(1, sizeof(*rtp)))) 01904 return NULL; 01905 01906 ast_rtp_new_init(rtp); 01907 01908 rtp->s = rtp_socket(); 01909 if (rtp->s < 0) { 01910 free(rtp); 01911 ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno)); 01912 return NULL; 01913 } 01914 if (sched && rtcpenable) { 01915 rtp->sched = sched; 01916 rtp->rtcp = ast_rtcp_new(); 01917 } 01918 01919 /* Select a random port number in the range of possible RTP */ 01920 x = (ast_random() % (rtpend-rtpstart)) + rtpstart; 01921 x = x & ~1; 01922 /* Save it for future references. */ 01923 startplace = x; 01924 /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */ 01925 for (;;) { 01926 /* Must be an even port number by RTP spec */ 01927 rtp->us.sin_port = htons(x); 01928 rtp->us.sin_addr = addr; 01929 /* If there's rtcp, initialize it as well. */ 01930 if (rtp->rtcp) { 01931 rtp->rtcp->us.sin_port = htons(x + 1); 01932 rtp->rtcp->us.sin_addr = addr; 01933 } 01934 /* Try to bind it/them. */ 01935 if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) && 01936 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) 01937 break; 01938 if (!first) { 01939 /* Primary bind succeeded! Gotta recreate it */ 01940 close(rtp->s); 01941 rtp->s = rtp_socket(); 01942 } 01943 if (errno != EADDRINUSE) { 01944 /* We got an error that wasn't expected, abort! */ 01945 ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); 01946 close(rtp->s); 01947 if (rtp->rtcp) { 01948 close(rtp->rtcp->s); 01949 free(rtp->rtcp); 01950 } 01951 free(rtp); 01952 return NULL; 01953 } 01954 /* The port was used, increment it (by two). */ 01955 x += 2; 01956 /* Did we go over the limit ? */ 01957 if (x > rtpend) 01958 /* then, start from the begingig. */ 01959 x = (rtpstart + 1) & ~1; 01960 /* Check if we reached the place were we started. */ 01961 if (x == startplace) { 01962 /* If so, there's no ports available. */ 01963 ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); 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 } 01973 rtp->sched = sched; 01974 rtp->io = io; 01975 if (callbackmode) { 01976 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); 01977 ast_set_flag(rtp, FLAG_CALLBACK_MODE); 01978 } 01979 ast_rtp_pt_default(rtp); 01980 return rtp; 01981 }
int ast_rtp_proto_register | ( | struct ast_rtp_protocol * | proto | ) |
Register interface to channel driver.
Definition at line 2834 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().
02835 { 02836 struct ast_rtp_protocol *cur; 02837 02838 AST_LIST_LOCK(&protos); 02839 AST_LIST_TRAVERSE(&protos, cur, list) { 02840 if (!strcmp(cur->type, proto->type)) { 02841 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); 02842 AST_LIST_UNLOCK(&protos); 02843 return -1; 02844 } 02845 } 02846 AST_LIST_INSERT_HEAD(&protos, proto, list); 02847 AST_LIST_UNLOCK(&protos); 02848 02849 return 0; 02850 }
void ast_rtp_proto_unregister | ( | struct ast_rtp_protocol * | proto | ) |
Unregister interface to channel driver.
Definition at line 2826 of file rtp.c.
References AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, and protos.
Referenced by load_module(), and unload_module().
02827 { 02828 AST_LIST_LOCK(&protos); 02829 AST_LIST_REMOVE(&protos, proto, list); 02830 AST_LIST_UNLOCK(&protos); 02831 }
void ast_rtp_pt_clear | ( | struct ast_rtp * | rtp | ) |
Setting RTP payload types from lines in a SDP description:.
Definition at line 1400 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().
01401 { 01402 int i; 01403 01404 if (!rtp) 01405 return; 01406 01407 ast_mutex_lock(&rtp->bridge_lock); 01408 01409 for (i = 0; i < MAX_RTP_PT; ++i) { 01410 rtp->current_RTP_PT[i].isAstFormat = 0; 01411 rtp->current_RTP_PT[i].code = 0; 01412 } 01413 01414 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01415 rtp->rtp_lookup_code_cache_code = 0; 01416 rtp->rtp_lookup_code_cache_result = 0; 01417 01418 ast_mutex_unlock(&rtp->bridge_lock); 01419 }
Copy payload types between RTP structures.
Definition at line 1440 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().
01441 { 01442 unsigned int i; 01443 01444 ast_mutex_lock(&dest->bridge_lock); 01445 ast_mutex_lock(&src->bridge_lock); 01446 01447 for (i=0; i < MAX_RTP_PT; ++i) { 01448 dest->current_RTP_PT[i].isAstFormat = 01449 src->current_RTP_PT[i].isAstFormat; 01450 dest->current_RTP_PT[i].code = 01451 src->current_RTP_PT[i].code; 01452 } 01453 dest->rtp_lookup_code_cache_isAstFormat = 0; 01454 dest->rtp_lookup_code_cache_code = 0; 01455 dest->rtp_lookup_code_cache_result = 0; 01456 01457 ast_mutex_unlock(&src->bridge_lock); 01458 ast_mutex_unlock(&dest->bridge_lock); 01459 }
void ast_rtp_pt_default | ( | struct ast_rtp * | rtp | ) |
Set payload types to defaults.
Definition at line 1421 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().
01422 { 01423 int i; 01424 01425 ast_mutex_lock(&rtp->bridge_lock); 01426 01427 /* Initialize to default payload types */ 01428 for (i = 0; i < MAX_RTP_PT; ++i) { 01429 rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat; 01430 rtp->current_RTP_PT[i].code = static_RTP_PT[i].code; 01431 } 01432 01433 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01434 rtp->rtp_lookup_code_cache_code = 0; 01435 rtp->rtp_lookup_code_cache_result = 0; 01436 01437 ast_mutex_unlock(&rtp->bridge_lock); 01438 }
Definition at line 1105 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_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, 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_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().
01106 { 01107 int res; 01108 struct sockaddr_in sin; 01109 socklen_t len; 01110 unsigned int seqno; 01111 int version; 01112 int payloadtype; 01113 int hdrlen = 12; 01114 int padding; 01115 int mark; 01116 int ext; 01117 int cc; 01118 unsigned int ssrc; 01119 unsigned int timestamp; 01120 unsigned int *rtpheader; 01121 struct rtpPayloadType rtpPT; 01122 struct ast_rtp *bridged = NULL; 01123 01124 /* If time is up, kill it */ 01125 if (rtp->sending_digit) 01126 ast_rtp_senddigit_continuation(rtp); 01127 01128 len = sizeof(sin); 01129 01130 /* Cache where the header will go */ 01131 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 01132 0, (struct sockaddr *)&sin, &len); 01133 01134 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); 01135 if (res < 0) { 01136 ast_assert(errno != EBADF); 01137 if (errno != EAGAIN) { 01138 ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno)); 01139 return NULL; 01140 } 01141 return &ast_null_frame; 01142 } 01143 01144 if (res < hdrlen) { 01145 ast_log(LOG_WARNING, "RTP Read too short\n"); 01146 return &ast_null_frame; 01147 } 01148 01149 /* Get fields */ 01150 seqno = ntohl(rtpheader[0]); 01151 01152 /* Check RTP version */ 01153 version = (seqno & 0xC0000000) >> 30; 01154 if (!version) { 01155 if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) && 01156 (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) { 01157 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 01158 } 01159 return &ast_null_frame; 01160 } 01161 01162 #if 0 /* Allow to receive RTP stream with closed transmission path */ 01163 /* If we don't have the other side's address, then ignore this */ 01164 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 01165 return &ast_null_frame; 01166 #endif 01167 01168 /* Send to whoever send to us if NAT is turned on */ 01169 if (rtp->nat) { 01170 if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 01171 (rtp->them.sin_port != sin.sin_port)) { 01172 rtp->them = sin; 01173 if (rtp->rtcp) { 01174 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 01175 rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1); 01176 } 01177 rtp->rxseqno = 0; 01178 ast_set_flag(rtp, FLAG_NAT_ACTIVE); 01179 if (option_debug || rtpdebug) 01180 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)); 01181 } 01182 } 01183 01184 /* If we are bridged to another RTP stream, send direct */ 01185 if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen)) 01186 return &ast_null_frame; 01187 01188 if (version != 2) 01189 return &ast_null_frame; 01190 01191 payloadtype = (seqno & 0x7f0000) >> 16; 01192 padding = seqno & (1 << 29); 01193 mark = seqno & (1 << 23); 01194 ext = seqno & (1 << 28); 01195 cc = (seqno & 0xF000000) >> 24; 01196 seqno &= 0xffff; 01197 timestamp = ntohl(rtpheader[1]); 01198 ssrc = ntohl(rtpheader[2]); 01199 01200 if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) { 01201 if (option_debug || rtpdebug) 01202 ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n"); 01203 mark = 1; 01204 } 01205 01206 rtp->rxssrc = ssrc; 01207 01208 if (padding) { 01209 /* Remove padding bytes */ 01210 res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; 01211 } 01212 01213 if (cc) { 01214 /* CSRC fields present */ 01215 hdrlen += cc*4; 01216 } 01217 01218 if (ext) { 01219 /* RTP Extension present */ 01220 hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2; 01221 hdrlen += 4; 01222 } 01223 01224 if (res < hdrlen) { 01225 ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); 01226 return &ast_null_frame; 01227 } 01228 01229 rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */ 01230 01231 if (rtp->rxcount==1) { 01232 /* This is the first RTP packet successfully received from source */ 01233 rtp->seedrxseqno = seqno; 01234 } 01235 01236 /* Do not schedule RR if RTCP isn't run */ 01237 if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) { 01238 /* Schedule transmission of Receiver Report */ 01239 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); 01240 } 01241 if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */ 01242 rtp->cycles += RTP_SEQ_MOD; 01243 01244 rtp->lastrxseqno = seqno; 01245 01246 if (rtp->themssrc==0) 01247 rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */ 01248 01249 if (rtp_debug_test_addr(&sin)) 01250 ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 01251 ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); 01252 01253 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); 01254 if (!rtpPT.isAstFormat) { 01255 struct ast_frame *f = NULL; 01256 01257 /* This is special in-band data that's not one of our codecs */ 01258 if (rtpPT.code == AST_RTP_DTMF) { 01259 /* It's special -- rfc2833 process it */ 01260 if (rtp_debug_test_addr(&sin)) { 01261 unsigned char *data; 01262 unsigned int event; 01263 unsigned int event_end; 01264 unsigned int duration; 01265 data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen; 01266 event = ntohl(*((unsigned int *)(data))); 01267 event >>= 24; 01268 event_end = ntohl(*((unsigned int *)(data))); 01269 event_end <<= 8; 01270 event_end >>= 24; 01271 duration = ntohl(*((unsigned int *)(data))); 01272 duration &= 0xFFFF; 01273 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); 01274 } 01275 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp); 01276 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { 01277 /* It's really special -- process it the Cisco way */ 01278 if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) { 01279 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01280 rtp->lastevent = seqno; 01281 } 01282 } else if (rtpPT.code == AST_RTP_CN) { 01283 /* Comfort Noise */ 01284 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01285 } else { 01286 ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr)); 01287 } 01288 return f ? f : &ast_null_frame; 01289 } 01290 rtp->lastrxformat = rtp->f.subclass = rtpPT.code; 01291 rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO; 01292 01293 if (!rtp->lastrxts) 01294 rtp->lastrxts = timestamp; 01295 01296 rtp->rxseqno = seqno; 01297 01298 /* Record received timestamp as last received now */ 01299 rtp->lastrxts = timestamp; 01300 01301 rtp->f.mallocd = 0; 01302 rtp->f.datalen = res - hdrlen; 01303 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; 01304 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; 01305 rtp->f.seqno = seqno; 01306 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { 01307 rtp->f.samples = ast_codec_get_samples(&rtp->f); 01308 if (rtp->f.subclass == AST_FORMAT_SLINEAR) 01309 ast_frame_byteswap_be(&rtp->f); 01310 calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark); 01311 /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */ 01312 ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO); 01313 rtp->f.ts = timestamp / 8; 01314 rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000); 01315 } else { 01316 /* Video -- samples is # of samples vs. 90000 */ 01317 if (!rtp->lastividtimestamp) 01318 rtp->lastividtimestamp = timestamp; 01319 rtp->f.samples = timestamp - rtp->lastividtimestamp; 01320 rtp->lastividtimestamp = timestamp; 01321 rtp->f.delivery.tv_sec = 0; 01322 rtp->f.delivery.tv_usec = 0; 01323 if (mark) 01324 rtp->f.subclass |= 0x1; 01325 01326 } 01327 rtp->f.src = "RTP"; 01328 return &rtp->f; 01329 }
int ast_rtp_reload | ( | void | ) |
Definition at line 3752 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().
03753 { 03754 struct ast_config *cfg; 03755 const char *s; 03756 03757 rtpstart = 5000; 03758 rtpend = 31000; 03759 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03760 cfg = ast_config_load("rtp.conf"); 03761 if (cfg) { 03762 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { 03763 rtpstart = atoi(s); 03764 if (rtpstart < 1024) 03765 rtpstart = 1024; 03766 if (rtpstart > 65535) 03767 rtpstart = 65535; 03768 } 03769 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { 03770 rtpend = atoi(s); 03771 if (rtpend < 1024) 03772 rtpend = 1024; 03773 if (rtpend > 65535) 03774 rtpend = 65535; 03775 } 03776 if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) { 03777 rtcpinterval = atoi(s); 03778 if (rtcpinterval == 0) 03779 rtcpinterval = 0; /* Just so we're clear... it's zero */ 03780 if (rtcpinterval < RTCP_MIN_INTERVALMS) 03781 rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */ 03782 if (rtcpinterval > RTCP_MAX_INTERVALMS) 03783 rtcpinterval = RTCP_MAX_INTERVALMS; 03784 } 03785 if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) { 03786 #ifdef SO_NO_CHECK 03787 if (ast_false(s)) 03788 nochecksums = 1; 03789 else 03790 nochecksums = 0; 03791 #else 03792 if (ast_false(s)) 03793 ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n"); 03794 #endif 03795 } 03796 if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { 03797 dtmftimeout = atoi(s); 03798 if ((dtmftimeout < 0) || (dtmftimeout > 20000)) { 03799 ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", 03800 dtmftimeout, DEFAULT_DTMF_TIMEOUT); 03801 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03802 }; 03803 } 03804 ast_config_destroy(cfg); 03805 } 03806 if (rtpstart >= rtpend) { 03807 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n"); 03808 rtpstart = 5000; 03809 rtpend = 31000; 03810 } 03811 if (option_verbose > 1) 03812 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); 03813 return 0; 03814 }
void ast_rtp_reset | ( | struct ast_rtp * | rtp | ) |
Definition at line 2060 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.
02061 { 02062 memset(&rtp->rxcore, 0, sizeof(rtp->rxcore)); 02063 memset(&rtp->txcore, 0, sizeof(rtp->txcore)); 02064 memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute)); 02065 rtp->lastts = 0; 02066 rtp->lastdigitts = 0; 02067 rtp->lastrxts = 0; 02068 rtp->lastividtimestamp = 0; 02069 rtp->lastovidtimestamp = 0; 02070 rtp->lasteventseqn = 0; 02071 rtp->lastevent = 0; 02072 rtp->lasttxformat = 0; 02073 rtp->lastrxformat = 0; 02074 rtp->dtmfcount = 0; 02075 rtp->dtmfsamples = 0; 02076 rtp->seqno = 0; 02077 rtp->rxseqno = 0; 02078 }
int ast_rtp_sendcng | ( | struct ast_rtp * | rtp, | |
int | level | |||
) |
generate comfort noice (CNG)
Definition at line 2575 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().
02576 { 02577 unsigned int *rtpheader; 02578 int hdrlen = 12; 02579 int res; 02580 int payload; 02581 char data[256]; 02582 level = 127 - (level & 0x7f); 02583 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN); 02584 02585 /* If we have no peer, return immediately */ 02586 if (!rtp->them.sin_addr.s_addr) 02587 return 0; 02588 02589 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02590 02591 /* Get a pointer to the header */ 02592 rtpheader = (unsigned int *)data; 02593 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++)); 02594 rtpheader[1] = htonl(rtp->lastts); 02595 rtpheader[2] = htonl(rtp->ssrc); 02596 data[12] = level; 02597 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { 02598 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); 02599 if (res <0) 02600 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)); 02601 if (rtp_debug_test_addr(&rtp->them)) 02602 ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n" 02603 , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); 02604 02605 } 02606 return 0; 02607 }
int ast_rtp_senddigit_begin | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
Send begin frames for DTMF.
Definition at line 2183 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().
02184 { 02185 unsigned int *rtpheader; 02186 int hdrlen = 12, res = 0, i = 0, payload = 0; 02187 char data[256]; 02188 02189 if ((digit <= '9') && (digit >= '0')) 02190 digit -= '0'; 02191 else if (digit == '*') 02192 digit = 10; 02193 else if (digit == '#') 02194 digit = 11; 02195 else if ((digit >= 'A') && (digit <= 'D')) 02196 digit = digit - 'A' + 12; 02197 else if ((digit >= 'a') && (digit <= 'd')) 02198 digit = digit - 'a' + 12; 02199 else { 02200 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); 02201 return 0; 02202 } 02203 02204 /* If we have no peer, return immediately */ 02205 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 02206 return 0; 02207 02208 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); 02209 02210 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02211 rtp->send_duration = 160; 02212 02213 /* Get a pointer to the header */ 02214 rtpheader = (unsigned int *)data; 02215 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno)); 02216 rtpheader[1] = htonl(rtp->lastdigitts); 02217 rtpheader[2] = htonl(rtp->ssrc); 02218 02219 for (i = 0; i < 2; i++) { 02220 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration)); 02221 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); 02222 if (res < 0) 02223 ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n", 02224 ast_inet_ntoa(rtp->them.sin_addr), 02225 ntohs(rtp->them.sin_port), strerror(errno)); 02226 if (rtp_debug_test_addr(&rtp->them)) 02227 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 02228 ast_inet_ntoa(rtp->them.sin_addr), 02229 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 02230 /* Increment sequence number */ 02231 rtp->seqno++; 02232 /* Increment duration */ 02233 rtp->send_duration += 160; 02234 /* Clear marker bit and set seqno */ 02235 rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); 02236 } 02237 02238 /* Since we received a begin, we can safely store the digit and disable any compensation */ 02239 rtp->sending_digit = 1; 02240 rtp->send_digit = digit; 02241 rtp->send_payload = payload; 02242 02243 return 0; 02244 }
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 1639 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().
01640 { 01641 if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 01642 return; /* bogus payload type */ 01643 01644 ast_mutex_lock(&rtp->bridge_lock); 01645 rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; 01646 ast_mutex_unlock(&rtp->bridge_lock); 01647 }
void ast_rtp_set_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2006 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().
02007 { 02008 rtp->them.sin_port = them->sin_port; 02009 rtp->them.sin_addr = them->sin_addr; 02010 if (rtp->rtcp) { 02011 rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1); 02012 rtp->rtcp->them.sin_addr = them->sin_addr; 02013 } 02014 rtp->rxseqno = 0; 02015 }
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 1666 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().
01669 { 01670 unsigned int i; 01671 int found = 0; 01672 01673 if (pt < 0 || pt > MAX_RTP_PT) 01674 return -1; /* bogus payload type */ 01675 01676 ast_mutex_lock(&rtp->bridge_lock); 01677 01678 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01679 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && 01680 strcasecmp(mimeType, mimeTypes[i].type) == 0) { 01681 found = 1; 01682 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; 01683 if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) && 01684 mimeTypes[i].payloadType.isAstFormat && 01685 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01686 rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2; 01687 break; 01688 } 01689 } 01690 01691 ast_mutex_unlock(&rtp->bridge_lock); 01692 01693 return (found ? 0 : -1); 01694 }
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 1991 of file rtp.c.
References ast_log(), LOG_WARNING, and ast_rtp::s.
Referenced by __oh323_rtp_create(), and sip_alloc().
01992 { 01993 int res; 01994 01995 if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 01996 ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); 01997 return res; 01998 }
void ast_rtp_stop | ( | struct ast_rtp * | rtp | ) |
Definition at line 2046 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().
02047 { 02048 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02049 02050 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 02051 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 02052 if (rtp->rtcp) { 02053 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr)); 02054 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port)); 02055 } 02056 02057 ast_clear_flag(rtp, FLAG_P2P_SENT_MARK); 02058 }
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 1651 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().
01652 { 01653 if (pt < 0 || pt > MAX_RTP_PT) 01654 return; /* bogus payload type */ 01655 01656 ast_mutex_lock(&rtp->bridge_lock); 01657 rtp->current_RTP_PT[pt].isAstFormat = 0; 01658 rtp->current_RTP_PT[pt].code = 0; 01659 ast_mutex_unlock(&rtp->bridge_lock); 01660 }
Definition at line 2734 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_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().
02735 { 02736 struct ast_frame *f; 02737 int codec; 02738 int hdrlen = 12; 02739 int subclass; 02740 02741 02742 /* If we have no peer, return immediately */ 02743 if (!rtp->them.sin_addr.s_addr) 02744 return 0; 02745 02746 /* If there is no data length, return immediately */ 02747 if (!_f->datalen) 02748 return 0; 02749 02750 /* Make sure we have enough space for RTP header */ 02751 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { 02752 ast_log(LOG_WARNING, "RTP can only send voice and video\n"); 02753 return -1; 02754 } 02755 02756 subclass = _f->subclass; 02757 if (_f->frametype == AST_FRAME_VIDEO) 02758 subclass &= ~0x1; 02759 02760 codec = ast_rtp_lookup_code(rtp, 1, subclass); 02761 if (codec < 0) { 02762 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); 02763 return -1; 02764 } 02765 02766 if (rtp->lasttxformat != subclass) { 02767 /* New format, reset the smoother */ 02768 if (option_debug) 02769 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); 02770 rtp->lasttxformat = subclass; 02771 if (rtp->smoother) 02772 ast_smoother_free(rtp->smoother); 02773 rtp->smoother = NULL; 02774 } 02775 02776 if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) { 02777 struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass); 02778 if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */ 02779 if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) { 02780 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)); 02781 return -1; 02782 } 02783 if (fmt.flags) 02784 ast_smoother_set_flags(rtp->smoother, fmt.flags); 02785 if (option_debug) 02786 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)); 02787 } 02788 } 02789 if (rtp->smoother) { 02790 if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) { 02791 ast_smoother_feed_be(rtp->smoother, _f); 02792 } else { 02793 ast_smoother_feed(rtp->smoother, _f); 02794 } 02795 02796 while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) { 02797 if (f->subclass == AST_FORMAT_G722) { 02798 /* G.722 is silllllllllllllly */ 02799 f->samples /= 2; 02800 } 02801 02802 ast_rtp_raw_write(rtp, f, codec); 02803 } 02804 } else { 02805 /* Don't buffer outgoing frames; send them one-per-packet: */ 02806 if (_f->offset < hdrlen) { 02807 f = ast_frdup(_f); 02808 } else { 02809 f = _f; 02810 } 02811 if (f->data) { 02812 if (f->subclass == AST_FORMAT_G722) { 02813 /* G.722 is silllllllllllllly */ 02814 f->samples /= 2; 02815 } 02816 ast_rtp_raw_write(rtp, f, codec); 02817 } 02818 if (f != _f) 02819 ast_frfree(f); 02820 } 02821 02822 return 0; 02823 }