#include <netinet/in.h>
#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
#include "asterisk/linkedlists.h"
Go to the source code of this file.
Data Structures | |
struct | ast_rtp_protocol |
struct | ast_rtp_quality |
Defines | |
#define | AST_RTP_CISCO_DTMF (1 << 2) |
#define | AST_RTP_CN (1 << 1) |
#define | AST_RTP_DTMF (1 << 0) |
#define | AST_RTP_MAX AST_RTP_CISCO_DTMF |
#define | FLAG_3389_WARNING (1 << 0) |
#define | MAX_RTP_PT 256 |
Typedefs | |
typedef int(*) | ast_rtp_callback (struct ast_rtp *rtp, struct ast_frame *f, void *data) |
Enumerations | |
enum | ast_rtp_get_result { AST_RTP_GET_FAILED = 0, AST_RTP_TRY_PARTIAL, AST_RTP_TRY_NATIVE } |
enum | ast_rtp_options { AST_RTP_OPT_G726_NONSTANDARD = (1 << 0) } |
Functions | |
int | ast_rtcp_fd (struct ast_rtp *rtp) |
ast_frame * | ast_rtcp_read (struct ast_rtp *rtp) |
int | ast_rtcp_send_h261fur (void *data) |
Send an H.261 fast update request. Some devices need this rather than the XML message in SIP. | |
size_t | ast_rtp_alloc_size (void) |
Get the amount of space required to hold an RTP session. | |
int | ast_rtp_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms) |
Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk. | |
int | ast_rtp_codec_getformat (int pt) |
ast_codec_pref * | ast_rtp_codec_getpref (struct ast_rtp *rtp) |
int | ast_rtp_codec_setpref (struct ast_rtp *rtp, struct ast_codec_pref *prefs) |
void | ast_rtp_destroy (struct ast_rtp *rtp) |
int | ast_rtp_early_bridge (struct ast_channel *dest, struct ast_channel *src) |
If possible, create an early bridge directly between the devices without having to send a re-invite later. | |
int | ast_rtp_fd (struct ast_rtp *rtp) |
ast_rtp * | ast_rtp_get_bridged (struct ast_rtp *rtp) |
void | ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats) |
Return the union of all of the codecs that were set by rtp_set...() calls They're returned as two distinct sets: AST_FORMATs, and AST_RTPs. | |
int | ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them) |
char * | ast_rtp_get_quality (struct ast_rtp *rtp, struct ast_rtp_quality *qual) |
Return RTCP quality string. | |
int | ast_rtp_get_rtpholdtimeout (struct ast_rtp *rtp) |
Get rtp hold timeout. | |
int | ast_rtp_get_rtpkeepalive (struct ast_rtp *rtp) |
Get RTP keepalive interval. | |
int | ast_rtp_get_rtptimeout (struct ast_rtp *rtp) |
Get rtp timeout. | |
void | ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us) |
int | ast_rtp_getnat (struct ast_rtp *rtp) |
void | ast_rtp_init (void) |
Initialize the RTP system in Asterisk. | |
int | ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code) |
Looks up an RTP code out of our *static* outbound list. | |
char * | ast_rtp_lookup_mime_multiple (char *buf, size_t size, const int capability, const int isAstFormat, enum ast_rtp_options options) |
Build a string of MIME subtype names from a capability list. | |
const char * | ast_rtp_lookup_mime_subtype (int isAstFormat, int code, enum ast_rtp_options options) |
Mapping an Asterisk code into a MIME subtype (string):. | |
rtpPayloadType | ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt) |
Mapping between RTP payload format codes and Asterisk codes:. | |
int | ast_rtp_make_compatible (struct ast_channel *dest, struct ast_channel *src, int media) |
ast_rtp * | ast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode) |
Initializate a RTP session. | |
void | ast_rtp_new_init (struct ast_rtp *rtp) |
Initialize a new RTP structure. | |
void | ast_rtp_new_source (struct ast_rtp *rtp) |
ast_rtp * | ast_rtp_new_with_bindaddr (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in) |
Initializate a RTP session using an in_addr structure. | |
int | ast_rtp_proto_register (struct ast_rtp_protocol *proto) |
Register interface to channel driver. | |
void | ast_rtp_proto_unregister (struct ast_rtp_protocol *proto) |
Unregister interface to channel driver. | |
void | ast_rtp_pt_clear (struct ast_rtp *rtp) |
Setting RTP payload types from lines in a SDP description:. | |
void | ast_rtp_pt_copy (struct ast_rtp *dest, struct ast_rtp *src) |
Copy payload types between RTP structures. | |
void | ast_rtp_pt_default (struct ast_rtp *rtp) |
Set payload types to defaults. | |
ast_frame * | ast_rtp_read (struct ast_rtp *rtp) |
int | ast_rtp_reload (void) |
void | ast_rtp_reset (struct ast_rtp *rtp) |
int | ast_rtp_sendcng (struct ast_rtp *rtp, int level) |
generate comfort noice (CNG) | |
int | ast_rtp_senddigit_begin (struct ast_rtp *rtp, char digit) |
Send begin frames for DTMF. | |
int | ast_rtp_senddigit_end (struct ast_rtp *rtp, char digit) |
void | ast_rtp_set_alt_peer (struct ast_rtp *rtp, struct sockaddr_in *alt) |
set potential alternate source for RTP media | |
void | ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback) |
void | ast_rtp_set_constantssrc (struct ast_rtp *rtp) |
When changing sources, don't generate a new SSRC. | |
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_a_audio().
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 521 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 869 of file rtp.c.
References ast_rtcp::accumulated_transit, ast_rtcp::altthem, ast_assert, AST_CONTROL_VIDUPDATE, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), ast_frame::datalen, errno, ast_rtp::f, f, ast_frame::frametype, len(), LOG_DEBUG, LOG_WARNING, ast_frame::mallocd, ast_rtcp::maxrtt, ast_rtcp::minrtt, ast_rtp::nat, option_debug, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, rtcp_debug_test_addr(), RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, ast_rtcp::rtt, ast_rtcp::rxlsr, ast_rtp::s, ast_rtcp::s, ast_frame::samples, ast_rtcp::soc, ast_rtcp::spc, ast_frame::src, ast_frame::subclass, ast_rtcp::them, ast_rtcp::themrxlsr, and timeval2ntp().
Referenced by oh323_read(), sip_rtp_read(), and skinny_rtp_read().
00870 { 00871 socklen_t len; 00872 int position, i, packetwords; 00873 int res; 00874 struct sockaddr_in sin; 00875 unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET]; 00876 unsigned int *rtcpheader; 00877 int pt; 00878 struct timeval now; 00879 unsigned int length; 00880 int rc; 00881 double rttsec; 00882 uint64_t rtt = 0; 00883 unsigned int dlsr; 00884 unsigned int lsr; 00885 unsigned int msw; 00886 unsigned int lsw; 00887 unsigned int comp; 00888 struct ast_frame *f = &ast_null_frame; 00889 00890 if (!rtp || !rtp->rtcp) 00891 return &ast_null_frame; 00892 00893 len = sizeof(sin); 00894 00895 res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET, 00896 0, (struct sockaddr *)&sin, &len); 00897 if (option_debug > 2) 00898 ast_log(LOG_DEBUG, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 00899 00900 rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET); 00901 00902 if (res < 0) { 00903 ast_assert(errno != EBADF); 00904 if (errno != EAGAIN) { 00905 ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno)); 00906 ast_log(LOG_WARNING, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 00907 return NULL; 00908 } 00909 return &ast_null_frame; 00910 } 00911 00912 packetwords = res / 4; 00913 00914 if (rtp->nat) { 00915 /* Send to whoever sent to us */ 00916 if (((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 00917 (rtp->rtcp->them.sin_port != sin.sin_port)) && 00918 ((rtp->rtcp->altthem.sin_addr.s_addr != sin.sin_addr.s_addr) || 00919 (rtp->rtcp->altthem.sin_port != sin.sin_port))) { 00920 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 00921 if (option_debug || rtpdebug) 00922 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)); 00923 } 00924 } 00925 00926 if (option_debug) 00927 ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res); 00928 00929 /* Process a compound packet */ 00930 position = 0; 00931 while (position < packetwords) { 00932 i = position; 00933 length = ntohl(rtcpheader[i]); 00934 pt = (length & 0xff0000) >> 16; 00935 rc = (length & 0x1f000000) >> 24; 00936 length &= 0xffff; 00937 00938 if ((i + length) > packetwords) { 00939 ast_log(LOG_WARNING, "RTCP Read too short\n"); 00940 return &ast_null_frame; 00941 } 00942 00943 if (rtcp_debug_test_addr(&sin)) { 00944 ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); 00945 ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown"); 00946 ast_verbose("Reception reports: %d\n", rc); 00947 ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]); 00948 } 00949 00950 i += 2; /* Advance past header and ssrc */ 00951 00952 switch (pt) { 00953 case RTCP_PT_SR: 00954 gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */ 00955 rtp->rtcp->spc = ntohl(rtcpheader[i+3]); 00956 rtp->rtcp->soc = ntohl(rtcpheader[i + 4]); 00957 rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/ 00958 00959 if (rtcp_debug_test_addr(&sin)) { 00960 ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096); 00961 ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2])); 00962 ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4])); 00963 } 00964 i += 5; 00965 if (rc < 1) 00966 break; 00967 /* Intentional fall through */ 00968 case RTCP_PT_RR: 00969 /* Don't handle multiple reception reports (rc > 1) yet */ 00970 /* Calculate RTT per RFC */ 00971 gettimeofday(&now, NULL); 00972 timeval2ntp(now, &msw, &lsw); 00973 if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */ 00974 comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16); 00975 lsr = ntohl(rtcpheader[i + 4]); 00976 dlsr = ntohl(rtcpheader[i + 5]); 00977 rtt = comp - lsr - dlsr; 00978 00979 /* Convert end to end delay to usec (keeping the calculation in 64bit space) 00980 sess->ee_delay = (eedelay * 1000) / 65536; */ 00981 if (rtt < 4294) { 00982 rtt = (rtt * 1000000) >> 16; 00983 } else { 00984 rtt = (rtt * 1000) >> 16; 00985 rtt *= 1000; 00986 } 00987 rtt = rtt / 1000.; 00988 rttsec = rtt / 1000.; 00989 00990 if (comp - dlsr >= lsr) { 00991 rtp->rtcp->accumulated_transit += rttsec; 00992 rtp->rtcp->rtt = rttsec; 00993 if (rtp->rtcp->maxrtt<rttsec) 00994 rtp->rtcp->maxrtt = rttsec; 00995 if (rtp->rtcp->minrtt>rttsec) 00996 rtp->rtcp->minrtt = rttsec; 00997 } else if (rtcp_debug_test_addr(&sin)) { 00998 ast_verbose("Internal RTCP NTP clock skew detected: " 00999 "lsr=%u, now=%u, dlsr=%u (%d:%03dms), " 01000 "diff=%d\n", 01001 lsr, comp, dlsr, dlsr / 65536, 01002 (dlsr % 65536) * 1000 / 65536, 01003 dlsr - (comp - lsr)); 01004 } 01005 } 01006 01007 rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]); 01008 rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff; 01009 if (rtcp_debug_test_addr(&sin)) { 01010 ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24)); 01011 ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost); 01012 ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff)); 01013 ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16); 01014 ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter); 01015 ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096); 01016 ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0); 01017 if (rtt) 01018 ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt); 01019 } 01020 break; 01021 case RTCP_PT_FUR: 01022 if (rtcp_debug_test_addr(&sin)) 01023 ast_verbose("Received an RTCP Fast Update Request\n"); 01024 rtp->f.frametype = AST_FRAME_CONTROL; 01025 rtp->f.subclass = AST_CONTROL_VIDUPDATE; 01026 rtp->f.datalen = 0; 01027 rtp->f.samples = 0; 01028 rtp->f.mallocd = 0; 01029 rtp->f.src = "RTP"; 01030 f = &rtp->f; 01031 break; 01032 case RTCP_PT_SDES: 01033 if (rtcp_debug_test_addr(&sin)) 01034 ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 01035 break; 01036 case RTCP_PT_BYE: 01037 if (rtcp_debug_test_addr(&sin)) 01038 ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 01039 break; 01040 default: 01041 if (option_debug) 01042 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)); 01043 break; 01044 } 01045 position += (length + 1); 01046 } 01047 01048 return f; 01049 }
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 2438 of file rtp.c.
References ast_rtcp_write(), ast_rtp::rtcp, and ast_rtcp::sendfur.
02439 { 02440 struct ast_rtp *rtp = data; 02441 int res; 02442 02443 rtp->rtcp->sendfur = 1; 02444 res = ast_rtcp_write(data); 02445 02446 return res; 02447 }
size_t ast_rtp_alloc_size | ( | void | ) |
Get the amount of space required to hold an RTP session.
Definition at line 401 of file rtp.c.
Referenced by process_sdp().
00402 { 00403 return sizeof(struct ast_rtp); 00404 }
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 3414 of file rtp.c.
References AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_check_hangup(), ast_codec_pref_getsize(), ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_set_flag, ast_test_flag, ast_verbose(), bridge_native_loop(), bridge_p2p_loop(), ast_format_list::cur_ms, FLAG_HAS_DTMF, FLAG_P2P_NEED_DTMF, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, option_debug, option_verbose, ast_rtp::pref, ast_channel::rawreadformat, ast_channel::rawwriteformat, ast_channel_tech::send_digit_begin, ast_channel::tech, ast_channel::tech_pvt, and VERBOSE_PREFIX_3.
03415 { 03416 struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ 03417 struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */ 03418 struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL; 03419 enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED; 03420 enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED; 03421 enum ast_bridge_result res = AST_BRIDGE_FAILED; 03422 int codec0 = 0, codec1 = 0; 03423 void *pvt0 = NULL, *pvt1 = NULL; 03424 03425 /* Lock channels */ 03426 ast_channel_lock(c0); 03427 while(ast_channel_trylock(c1)) { 03428 ast_channel_unlock(c0); 03429 usleep(1); 03430 ast_channel_lock(c0); 03431 } 03432 03433 /* Ensure neither channel got hungup during lock avoidance */ 03434 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 03435 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name); 03436 ast_channel_unlock(c0); 03437 ast_channel_unlock(c1); 03438 return AST_BRIDGE_FAILED; 03439 } 03440 03441 /* Find channel driver interfaces */ 03442 if (!(pr0 = get_proto(c0))) { 03443 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); 03444 ast_channel_unlock(c0); 03445 ast_channel_unlock(c1); 03446 return AST_BRIDGE_FAILED; 03447 } 03448 if (!(pr1 = get_proto(c1))) { 03449 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); 03450 ast_channel_unlock(c0); 03451 ast_channel_unlock(c1); 03452 return AST_BRIDGE_FAILED; 03453 } 03454 03455 /* Get channel specific interface structures */ 03456 pvt0 = c0->tech_pvt; 03457 pvt1 = c1->tech_pvt; 03458 03459 /* Get audio and video interface (if native bridge is possible) */ 03460 audio_p0_res = pr0->get_rtp_info(c0, &p0); 03461 video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED; 03462 audio_p1_res = pr1->get_rtp_info(c1, &p1); 03463 video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED; 03464 03465 /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */ 03466 if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE)) 03467 audio_p0_res = AST_RTP_GET_FAILED; 03468 if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE)) 03469 audio_p1_res = AST_RTP_GET_FAILED; 03470 03471 /* Check if a bridge is possible (partial/native) */ 03472 if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) { 03473 /* Somebody doesn't want to play... */ 03474 ast_channel_unlock(c0); 03475 ast_channel_unlock(c1); 03476 return AST_BRIDGE_FAILED_NOWARN; 03477 } 03478 03479 /* If we need to feed DTMF frames into the core then only do a partial native bridge */ 03480 if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) { 03481 ast_set_flag(p0, FLAG_P2P_NEED_DTMF); 03482 audio_p0_res = AST_RTP_TRY_PARTIAL; 03483 } 03484 03485 if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) { 03486 ast_set_flag(p1, FLAG_P2P_NEED_DTMF); 03487 audio_p1_res = AST_RTP_TRY_PARTIAL; 03488 } 03489 03490 /* If both sides are not using the same method of DTMF transmission 03491 * (ie: one is RFC2833, other is INFO... then we can not do direct media. 03492 * -------------------------------------------------- 03493 * | DTMF Mode | HAS_DTMF | Accepts Begin Frames | 03494 * |-----------|------------|-----------------------| 03495 * | Inband | False | True | 03496 * | RFC2833 | True | True | 03497 * | SIP INFO | False | False | 03498 * -------------------------------------------------- 03499 * However, if DTMF from both channels is being monitored by the core, then 03500 * we can still do packet-to-packet bridging, because passing through the 03501 * core will handle DTMF mode translation. 03502 */ 03503 if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) || 03504 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) { 03505 if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) { 03506 ast_channel_unlock(c0); 03507 ast_channel_unlock(c1); 03508 return AST_BRIDGE_FAILED_NOWARN; 03509 } 03510 audio_p0_res = AST_RTP_TRY_PARTIAL; 03511 audio_p1_res = AST_RTP_TRY_PARTIAL; 03512 } 03513 03514 /* If we need to feed frames into the core don't do a P2P bridge */ 03515 if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) || 03516 (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) { 03517 ast_channel_unlock(c0); 03518 ast_channel_unlock(c1); 03519 return AST_BRIDGE_FAILED_NOWARN; 03520 } 03521 03522 /* Get codecs from both sides */ 03523 codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0; 03524 codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0; 03525 if (codec0 && codec1 && !(codec0 & codec1)) { 03526 /* Hey, we can't do native bridging if both parties speak different codecs */ 03527 if (option_debug) 03528 ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); 03529 ast_channel_unlock(c0); 03530 ast_channel_unlock(c1); 03531 return AST_BRIDGE_FAILED_NOWARN; 03532 } 03533 03534 /* If either side can only do a partial bridge, then don't try for a true native bridge */ 03535 if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) { 03536 struct ast_format_list fmt0, fmt1; 03537 03538 /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */ 03539 if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) { 03540 if (option_debug) 03541 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n"); 03542 ast_channel_unlock(c0); 03543 ast_channel_unlock(c1); 03544 return AST_BRIDGE_FAILED_NOWARN; 03545 } 03546 /* They must also be using the same packetization */ 03547 fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat); 03548 fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat); 03549 if (fmt0.cur_ms != fmt1.cur_ms) { 03550 if (option_debug) 03551 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n"); 03552 ast_channel_unlock(c0); 03553 ast_channel_unlock(c1); 03554 return AST_BRIDGE_FAILED_NOWARN; 03555 } 03556 03557 if (option_verbose > 2) 03558 ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name); 03559 res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1); 03560 } else { 03561 if (option_verbose > 2) 03562 ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name); 03563 res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1); 03564 } 03565 03566 return res; 03567 }
int ast_rtp_codec_getformat | ( | int | pt | ) |
Definition at line 2866 of file rtp.c.
References rtpPayloadType::code, and static_RTP_PT.
Referenced by process_sdp_a_audio().
02867 { 02868 if (pt < 0 || pt >= MAX_RTP_PT) 02869 return 0; /* bogus payload type */ 02870 02871 if (static_RTP_PT[pt].isAstFormat) 02872 return static_RTP_PT[pt].code; 02873 else 02874 return 0; 02875 }
struct ast_codec_pref* ast_rtp_codec_getpref | ( | struct ast_rtp * | rtp | ) |
Definition at line 2861 of file rtp.c.
References ast_rtp::pref.
Referenced by add_codec_to_sdp(), and process_sdp_a_audio().
02862 { 02863 return &rtp->pref; 02864 }
int ast_rtp_codec_setpref | ( | struct ast_rtp * | rtp, | |
struct ast_codec_pref * | prefs | |||
) |
Definition at line 2814 of file rtp.c.
References ast_codec_pref_getsize(), ast_log(), ast_smoother_new(), ast_smoother_reconfigure(), ast_smoother_set_flags(), ast_format_list::cur_ms, ast_format_list::flags, ast_format_list::fr_len, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, option_debug, ast_rtp::pref, prefs, and ast_rtp::smoother.
Referenced by __oh323_rtp_create(), check_user_full(), create_addr_from_peer(), process_sdp_a_audio(), register_verify(), set_peer_capabilities(), sip_alloc(), start_rtp(), and transmit_response_with_sdp().
02815 { 02816 struct ast_format_list current_format_old, current_format_new; 02817 02818 /* if no packets have been sent through this session yet, then 02819 * changing preferences does not require any extra work 02820 */ 02821 if (rtp->lasttxformat == 0) { 02822 rtp->pref = *prefs; 02823 return 0; 02824 } 02825 02826 current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat); 02827 02828 rtp->pref = *prefs; 02829 02830 current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat); 02831 02832 /* if the framing desired for the current format has changed, we may have to create 02833 * or adjust the smoother for this session 02834 */ 02835 if ((current_format_new.inc_ms != 0) && 02836 (current_format_new.cur_ms != current_format_old.cur_ms)) { 02837 int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms; 02838 02839 if (rtp->smoother) { 02840 ast_smoother_reconfigure(rtp->smoother, new_size); 02841 if (option_debug) { 02842 ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size); 02843 } 02844 } else { 02845 if (!(rtp->smoother = ast_smoother_new(new_size))) { 02846 ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size); 02847 return -1; 02848 } 02849 if (current_format_new.flags) { 02850 ast_smoother_set_flags(rtp->smoother, current_format_new.flags); 02851 } 02852 if (option_debug) { 02853 ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size); 02854 } 02855 } 02856 } 02857 02858 return 0; 02859 }
void ast_rtp_destroy | ( | struct ast_rtp * | rtp | ) |
Definition at line 2221 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(), sip_alloc(), skinny_hangup(), start_rtp(), and unalloc_sub().
02222 { 02223 if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) { 02224 /*Print some info on the call here */ 02225 ast_verbose(" RTP-stats\n"); 02226 ast_verbose("* Our Receiver:\n"); 02227 ast_verbose(" SSRC: %u\n", rtp->themssrc); 02228 ast_verbose(" Received packets: %u\n", rtp->rxcount); 02229 ast_verbose(" Lost packets: %u\n", rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0); 02230 ast_verbose(" Jitter: %.4f\n", rtp->rxjitter); 02231 ast_verbose(" Transit: %.4f\n", rtp->rxtransit); 02232 ast_verbose(" RR-count: %u\n", rtp->rtcp ? rtp->rtcp->rr_count : 0); 02233 ast_verbose("* Our Sender:\n"); 02234 ast_verbose(" SSRC: %u\n", rtp->ssrc); 02235 ast_verbose(" Sent packets: %u\n", rtp->txcount); 02236 ast_verbose(" Lost packets: %u\n", rtp->rtcp ? rtp->rtcp->reported_lost : 0); 02237 ast_verbose(" Jitter: %u\n", rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int)65536.0) : 0); 02238 ast_verbose(" SR-count: %u\n", rtp->rtcp ? rtp->rtcp->sr_count : 0); 02239 ast_verbose(" RTT: %f\n", rtp->rtcp ? rtp->rtcp->rtt : 0); 02240 } 02241 02242 if (rtp->smoother) 02243 ast_smoother_free(rtp->smoother); 02244 if (rtp->ioid) 02245 ast_io_remove(rtp->io, rtp->ioid); 02246 if (rtp->s > -1) 02247 close(rtp->s); 02248 if (rtp->rtcp) { 02249 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02250 close(rtp->rtcp->s); 02251 free(rtp->rtcp); 02252 rtp->rtcp=NULL; 02253 } 02254 02255 ast_mutex_destroy(&rtp->bridge_lock); 02256 02257 free(rtp); 02258 }
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 1546 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, ast_channel::name, option_debug, and ast_rtp_protocol::set_rtp_peer.
Referenced by wait_for_answer().
01547 { 01548 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01549 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01550 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01551 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01552 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01553 int srccodec, destcodec, nat_active = 0; 01554 01555 /* Lock channels */ 01556 ast_channel_lock(dest); 01557 if (src) { 01558 while(ast_channel_trylock(src)) { 01559 ast_channel_unlock(dest); 01560 usleep(1); 01561 ast_channel_lock(dest); 01562 } 01563 } 01564 01565 /* Find channel driver interfaces */ 01566 destpr = get_proto(dest); 01567 if (src) 01568 srcpr = get_proto(src); 01569 if (!destpr) { 01570 if (option_debug) 01571 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01572 ast_channel_unlock(dest); 01573 if (src) 01574 ast_channel_unlock(src); 01575 return 0; 01576 } 01577 if (!srcpr) { 01578 if (option_debug) 01579 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>"); 01580 ast_channel_unlock(dest); 01581 if (src) 01582 ast_channel_unlock(src); 01583 return 0; 01584 } 01585 01586 /* Get audio and video interface (if native bridge is possible) */ 01587 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01588 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01589 if (srcpr) { 01590 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01591 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01592 } 01593 01594 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01595 if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) { 01596 /* Somebody doesn't want to play... */ 01597 ast_channel_unlock(dest); 01598 if (src) 01599 ast_channel_unlock(src); 01600 return 0; 01601 } 01602 if (audio_src_res == AST_RTP_TRY_NATIVE && (video_src_res == AST_RTP_GET_FAILED || video_src_res == AST_RTP_TRY_NATIVE) && srcpr->get_codec) 01603 srccodec = srcpr->get_codec(src); 01604 else 01605 srccodec = 0; 01606 if (audio_dest_res == AST_RTP_TRY_NATIVE && (video_dest_res == AST_RTP_GET_FAILED || video_dest_res == AST_RTP_TRY_NATIVE) && destpr->get_codec) 01607 destcodec = destpr->get_codec(dest); 01608 else 01609 destcodec = 0; 01610 /* Ensure we have at least one matching codec */ 01611 if (srcp && !(srccodec & destcodec)) { 01612 ast_channel_unlock(dest); 01613 ast_channel_unlock(src); 01614 return 0; 01615 } 01616 /* Consider empty media as non-existant */ 01617 if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) 01618 srcp = NULL; 01619 /* If the client has NAT stuff turned on then just safe NAT is active */ 01620 if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01621 nat_active = 1; 01622 /* Bridge media early */ 01623 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active)) 01624 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01625 ast_channel_unlock(dest); 01626 if (src) 01627 ast_channel_unlock(src); 01628 if (option_debug) 01629 ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01630 return 1; 01631 }
int ast_rtp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 516 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().
00517 { 00518 return rtp->s; 00519 }
Definition at line 2131 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().
02132 { 02133 struct ast_rtp *bridged = NULL; 02134 02135 ast_mutex_lock(&rtp->bridge_lock); 02136 bridged = rtp->bridged; 02137 ast_mutex_unlock(&rtp->bridge_lock); 02138 02139 return bridged; 02140 }
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 1767 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().
01769 { 01770 int pt; 01771 01772 ast_mutex_lock(&rtp->bridge_lock); 01773 01774 *astFormats = *nonAstFormats = 0; 01775 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01776 if (rtp->current_RTP_PT[pt].isAstFormat) { 01777 *astFormats |= rtp->current_RTP_PT[pt].code; 01778 } else { 01779 *nonAstFormats |= rtp->current_RTP_PT[pt].code; 01780 } 01781 } 01782 01783 ast_mutex_unlock(&rtp->bridge_lock); 01784 01785 return; 01786 }
int ast_rtp_get_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2113 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().
02114 { 02115 if ((them->sin_family != AF_INET) || 02116 (them->sin_port != rtp->them.sin_port) || 02117 (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) { 02118 them->sin_family = AF_INET; 02119 them->sin_port = rtp->them.sin_port; 02120 them->sin_addr = rtp->them.sin_addr; 02121 return 1; 02122 } 02123 return 0; 02124 }
char* ast_rtp_get_quality | ( | struct ast_rtp * | rtp, | |
struct ast_rtp_quality * | qual | |||
) |
Return RTCP quality string.
Definition at line 2177 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().
02178 { 02179 /* 02180 *ssrc our ssrc 02181 *themssrc their ssrc 02182 *lp lost packets 02183 *rxjitter our calculated jitter(rx) 02184 *rxcount no. received packets 02185 *txjitter reported jitter of the other end 02186 *txcount transmitted packets 02187 *rlp remote lost packets 02188 *rtt round trip time 02189 */ 02190 02191 if (qual && rtp) { 02192 qual->local_ssrc = rtp->ssrc; 02193 qual->local_jitter = rtp->rxjitter; 02194 qual->local_count = rtp->rxcount; 02195 qual->remote_ssrc = rtp->themssrc; 02196 qual->remote_count = rtp->txcount; 02197 if (rtp->rtcp) { 02198 qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior; 02199 qual->remote_lostpackets = rtp->rtcp->reported_lost; 02200 qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0; 02201 qual->rtt = rtp->rtcp->rtt; 02202 } 02203 } 02204 if (rtp->rtcp) { 02205 snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), 02206 "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", 02207 rtp->ssrc, 02208 rtp->themssrc, 02209 rtp->rtcp->expected_prior - rtp->rtcp->received_prior, 02210 rtp->rxjitter, 02211 rtp->rxcount, 02212 (double)rtp->rtcp->reported_jitter / 65536.0, 02213 rtp->txcount, 02214 rtp->rtcp->reported_lost, 02215 rtp->rtcp->rtt); 02216 return rtp->rtcp->quality; 02217 } else 02218 return "<Unknown> - RTP/RTCP has already been destroyed"; 02219 }
int ast_rtp_get_rtpholdtimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp hold timeout.
Definition at line 576 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by do_monitor().
00577 { 00578 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00579 return 0; 00580 return rtp->rtpholdtimeout; 00581 }
int ast_rtp_get_rtpkeepalive | ( | struct ast_rtp * | rtp | ) |
Get RTP keepalive interval.
Definition at line 584 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by do_monitor().
00585 { 00586 return rtp->rtpkeepalive; 00587 }
int ast_rtp_get_rtptimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp timeout.
Definition at line 568 of file rtp.c.
References 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->rtptimeout; 00573 }
void ast_rtp_get_us | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | us | |||
) |
Definition at line 2126 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 604 of file rtp.c.
References ast_test_flag, and FLAG_NAT_ACTIVE.
Referenced by sip_get_rtp_peer().
00605 { 00606 return ast_test_flag(rtp, FLAG_NAT_ACTIVE); 00607 }
void ast_rtp_init | ( | void | ) |
Initialize the RTP system in Asterisk.
Definition at line 3952 of file rtp.c.
References ast_cli_register_multiple(), ast_rtp_reload(), and cli_rtp.
Referenced by main().
03953 { 03954 ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry)); 03955 ast_rtp_reload(); 03956 }
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 1810 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().
01811 { 01812 int pt = 0; 01813 01814 ast_mutex_lock(&rtp->bridge_lock); 01815 01816 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && 01817 code == rtp->rtp_lookup_code_cache_code) { 01818 /* Use our cached mapping, to avoid the overhead of the loop below */ 01819 pt = rtp->rtp_lookup_code_cache_result; 01820 ast_mutex_unlock(&rtp->bridge_lock); 01821 return pt; 01822 } 01823 01824 /* Check the dynamic list first */ 01825 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01826 if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) { 01827 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01828 rtp->rtp_lookup_code_cache_code = code; 01829 rtp->rtp_lookup_code_cache_result = pt; 01830 ast_mutex_unlock(&rtp->bridge_lock); 01831 return pt; 01832 } 01833 } 01834 01835 /* Then the static list */ 01836 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01837 if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) { 01838 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01839 rtp->rtp_lookup_code_cache_code = code; 01840 rtp->rtp_lookup_code_cache_result = pt; 01841 ast_mutex_unlock(&rtp->bridge_lock); 01842 return pt; 01843 } 01844 } 01845 01846 ast_mutex_unlock(&rtp->bridge_lock); 01847 01848 return -1; 01849 }
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 1870 of file rtp.c.
References ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, format, len(), and name.
Referenced by process_sdp().
01872 { 01873 int format; 01874 unsigned len; 01875 char *end = buf; 01876 char *start = buf; 01877 01878 if (!buf || !size) 01879 return NULL; 01880 01881 snprintf(end, size, "0x%x (", capability); 01882 01883 len = strlen(end); 01884 end += len; 01885 size -= len; 01886 start = end; 01887 01888 for (format = 1; format < AST_RTP_MAX; format <<= 1) { 01889 if (capability & format) { 01890 const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options); 01891 01892 snprintf(end, size, "%s|", name); 01893 len = strlen(end); 01894 end += len; 01895 size -= len; 01896 } 01897 } 01898 01899 if (start == end) 01900 snprintf(start, size, "nothing)"); 01901 else if (size > 1) 01902 *(end -1) = ')'; 01903 01904 return buf; 01905 }
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 1851 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().
01853 { 01854 unsigned int i; 01855 01856 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01857 if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) { 01858 if (isAstFormat && 01859 (code == AST_FORMAT_G726_AAL2) && 01860 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01861 return "G726-32"; 01862 else 01863 return mimeTypes[i].subtype; 01864 } 01865 } 01866 01867 return ""; 01868 }
struct rtpPayloadType ast_rtp_lookup_pt | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Mapping between RTP payload format codes and Asterisk codes:.
Definition at line 1788 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().
01789 { 01790 struct rtpPayloadType result; 01791 01792 result.isAstFormat = result.code = 0; 01793 01794 if (pt < 0 || pt >= MAX_RTP_PT) 01795 return result; /* bogus payload type */ 01796 01797 /* Start with negotiated codecs */ 01798 ast_mutex_lock(&rtp->bridge_lock); 01799 result = rtp->current_RTP_PT[pt]; 01800 ast_mutex_unlock(&rtp->bridge_lock); 01801 01802 /* If it doesn't exist, check our static RTP type list, just in case */ 01803 if (!result.code) 01804 result = static_RTP_PT[pt]; 01805 01806 return result; 01807 }
int ast_rtp_make_compatible | ( | struct ast_channel * | dest, | |
struct ast_channel * | src, | |||
int | media | |||
) |
Definition at line 1633 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, ast_rtp_pt_copy(), AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, ast_channel::name, option_debug, and ast_rtp_protocol::set_rtp_peer.
Referenced by wait_for_answer().
01634 { 01635 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01636 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01637 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01638 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01639 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01640 int srccodec, destcodec; 01641 01642 /* Lock channels */ 01643 ast_channel_lock(dest); 01644 while(ast_channel_trylock(src)) { 01645 ast_channel_unlock(dest); 01646 usleep(1); 01647 ast_channel_lock(dest); 01648 } 01649 01650 /* Find channel driver interfaces */ 01651 if (!(destpr = get_proto(dest))) { 01652 if (option_debug) 01653 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01654 ast_channel_unlock(dest); 01655 ast_channel_unlock(src); 01656 return 0; 01657 } 01658 if (!(srcpr = get_proto(src))) { 01659 if (option_debug) 01660 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name); 01661 ast_channel_unlock(dest); 01662 ast_channel_unlock(src); 01663 return 0; 01664 } 01665 01666 /* Get audio and video interface (if native bridge is possible) */ 01667 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01668 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01669 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01670 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01671 01672 /* Ensure we have at least one matching codec */ 01673 if (srcpr->get_codec) 01674 srccodec = srcpr->get_codec(src); 01675 else 01676 srccodec = 0; 01677 if (destpr->get_codec) 01678 destcodec = destpr->get_codec(dest); 01679 else 01680 destcodec = 0; 01681 01682 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01683 if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) { 01684 /* Somebody doesn't want to play... */ 01685 ast_channel_unlock(dest); 01686 ast_channel_unlock(src); 01687 return 0; 01688 } 01689 ast_rtp_pt_copy(destp, srcp); 01690 if (vdestp && vsrcp) 01691 ast_rtp_pt_copy(vdestp, vsrcp); 01692 if (media) { 01693 /* Bridge early */ 01694 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01695 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name); 01696 } 01697 ast_channel_unlock(dest); 01698 ast_channel_unlock(src); 01699 if (option_debug) 01700 ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name); 01701 return 1; 01702 }
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 2060 of file rtp.c.
References ast_rtp_new_with_bindaddr(), io, and sched.
02061 { 02062 struct in_addr ia; 02063 02064 memset(&ia, 0, sizeof(ia)); 02065 return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia); 02066 }
void ast_rtp_new_init | ( | struct ast_rtp * | rtp | ) |
Initialize a new RTP structure.
Definition at line 1954 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().
01955 { 01956 ast_mutex_init(&rtp->bridge_lock); 01957 01958 rtp->them.sin_family = AF_INET; 01959 rtp->us.sin_family = AF_INET; 01960 rtp->ssrc = ast_random(); 01961 rtp->seqno = ast_random() & 0xffff; 01962 ast_set_flag(rtp, FLAG_HAS_DTMF); 01963 01964 return; 01965 }
void ast_rtp_new_source | ( | struct ast_rtp * | rtp | ) |
Definition at line 2082 of file rtp.c.
References ast_random(), ast_rtp::constantssrc, ast_rtp::set_marker_bit, and ast_rtp::ssrc.
Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), sip_write(), and skinny_indicate().
02083 { 02084 if (rtp) { 02085 rtp->set_marker_bit = 1; 02086 if (!rtp->constantssrc) { 02087 rtp->ssrc = ast_random(); 02088 } 02089 } 02090 }
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 1967 of file rtp.c.
References ast_calloc, ast_log(), ast_random(), ast_rtcp_new(), ast_rtp_new_init(), errno, first, free, LOG_DEBUG, LOG_ERROR, option_debug, rtp_socket(), and sched.
Referenced by __oh323_rtp_create(), ast_rtp_new(), gtalk_alloc(), sip_alloc(), and start_rtp().
01968 { 01969 struct ast_rtp *rtp; 01970 int x; 01971 int first; 01972 int startplace; 01973 01974 if (!(rtp = ast_calloc(1, sizeof(*rtp)))) 01975 return NULL; 01976 01977 ast_rtp_new_init(rtp); 01978 01979 rtp->s = rtp_socket(); 01980 if (option_debug > 2) 01981 ast_log(LOG_DEBUG, "socket RTP fd: %i\n", rtp->s); 01982 if (rtp->s < 0) { 01983 free(rtp); 01984 ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno)); 01985 return NULL; 01986 } 01987 if (sched && rtcpenable) { 01988 rtp->sched = sched; 01989 rtp->rtcp = ast_rtcp_new(); 01990 if (option_debug > 2) 01991 ast_log(LOG_DEBUG, "socket RTCP fd: %i\n", rtp->rtcp->s); 01992 } 01993 01994 /* Select a random port number in the range of possible RTP */ 01995 x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart; 01996 x = x & ~1; 01997 /* Save it for future references. */ 01998 startplace = x; 01999 /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */ 02000 for (;;) { 02001 /* Must be an even port number by RTP spec */ 02002 rtp->us.sin_port = htons(x); 02003 rtp->us.sin_addr = addr; 02004 /* If there's rtcp, initialize it as well. */ 02005 if (rtp->rtcp) { 02006 rtp->rtcp->us.sin_port = htons(x + 1); 02007 rtp->rtcp->us.sin_addr = addr; 02008 } 02009 /* Try to bind it/them. */ 02010 if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) && 02011 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) 02012 break; 02013 if (!first) { 02014 /* Primary bind succeeded! Gotta recreate it */ 02015 close(rtp->s); 02016 rtp->s = rtp_socket(); 02017 if (option_debug > 2) 02018 ast_log(LOG_DEBUG, "socket RTP2 fd: %i\n", rtp->s); 02019 } 02020 if (errno != EADDRINUSE) { 02021 /* We got an error that wasn't expected, abort! */ 02022 ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); 02023 close(rtp->s); 02024 if (rtp->rtcp) { 02025 close(rtp->rtcp->s); 02026 free(rtp->rtcp); 02027 } 02028 free(rtp); 02029 return NULL; 02030 } 02031 /* The port was used, increment it (by two). */ 02032 x += 2; 02033 /* Did we go over the limit ? */ 02034 if (x > rtpend) 02035 /* then, start from the begingig. */ 02036 x = (rtpstart + 1) & ~1; 02037 /* Check if we reached the place were we started. */ 02038 if (x == startplace) { 02039 /* If so, there's no ports available. */ 02040 ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); 02041 close(rtp->s); 02042 if (rtp->rtcp) { 02043 close(rtp->rtcp->s); 02044 free(rtp->rtcp); 02045 } 02046 free(rtp); 02047 return NULL; 02048 } 02049 } 02050 rtp->sched = sched; 02051 rtp->io = io; 02052 if (callbackmode) { 02053 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); 02054 ast_set_flag(rtp, FLAG_CALLBACK_MODE); 02055 } 02056 ast_rtp_pt_default(rtp); 02057 return rtp; 02058 }
int ast_rtp_proto_register | ( | struct ast_rtp_protocol * | proto | ) |
Register interface to channel driver.
Definition at line 2968 of file rtp.c.
References AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_log(), ast_rtp_protocol::list, LOG_WARNING, and ast_rtp_protocol::type.
Referenced by load_module().
02969 { 02970 struct ast_rtp_protocol *cur; 02971 02972 AST_LIST_LOCK(&protos); 02973 AST_LIST_TRAVERSE(&protos, cur, list) { 02974 if (!strcmp(cur->type, proto->type)) { 02975 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); 02976 AST_LIST_UNLOCK(&protos); 02977 return -1; 02978 } 02979 } 02980 AST_LIST_INSERT_HEAD(&protos, proto, list); 02981 AST_LIST_UNLOCK(&protos); 02982 02983 return 0; 02984 }
void ast_rtp_proto_unregister | ( | struct ast_rtp_protocol * | proto | ) |
Unregister interface to channel driver.
Definition at line 2960 of file rtp.c.
References AST_LIST_LOCK, AST_LIST_REMOVE, and AST_LIST_UNLOCK.
Referenced by load_module(), and unload_module().
02961 { 02962 AST_LIST_LOCK(&protos); 02963 AST_LIST_REMOVE(&protos, proto, list); 02964 AST_LIST_UNLOCK(&protos); 02965 }
void ast_rtp_pt_clear | ( | struct ast_rtp * | rtp | ) |
Setting RTP payload types from lines in a SDP description:.
Definition at line 1470 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().
01471 { 01472 int i; 01473 01474 if (!rtp) 01475 return; 01476 01477 ast_mutex_lock(&rtp->bridge_lock); 01478 01479 for (i = 0; i < MAX_RTP_PT; ++i) { 01480 rtp->current_RTP_PT[i].isAstFormat = 0; 01481 rtp->current_RTP_PT[i].code = 0; 01482 } 01483 01484 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01485 rtp->rtp_lookup_code_cache_code = 0; 01486 rtp->rtp_lookup_code_cache_result = 0; 01487 01488 ast_mutex_unlock(&rtp->bridge_lock); 01489 }
Copy payload types between RTP structures.
Definition at line 1510 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().
01511 { 01512 unsigned int i; 01513 01514 ast_mutex_lock(&dest->bridge_lock); 01515 ast_mutex_lock(&src->bridge_lock); 01516 01517 for (i=0; i < MAX_RTP_PT; ++i) { 01518 dest->current_RTP_PT[i].isAstFormat = 01519 src->current_RTP_PT[i].isAstFormat; 01520 dest->current_RTP_PT[i].code = 01521 src->current_RTP_PT[i].code; 01522 } 01523 dest->rtp_lookup_code_cache_isAstFormat = 0; 01524 dest->rtp_lookup_code_cache_code = 0; 01525 dest->rtp_lookup_code_cache_result = 0; 01526 01527 ast_mutex_unlock(&src->bridge_lock); 01528 ast_mutex_unlock(&dest->bridge_lock); 01529 }
void ast_rtp_pt_default | ( | struct ast_rtp * | rtp | ) |
Set payload types to defaults.
Definition at line 1491 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.
01492 { 01493 int i; 01494 01495 ast_mutex_lock(&rtp->bridge_lock); 01496 01497 /* Initialize to default payload types */ 01498 for (i = 0; i < MAX_RTP_PT; ++i) { 01499 rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat; 01500 rtp->current_RTP_PT[i].code = static_RTP_PT[i].code; 01501 } 01502 01503 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01504 rtp->rtp_lookup_code_cache_code = 0; 01505 rtp->rtp_lookup_code_cache_result = 0; 01506 01507 ast_mutex_unlock(&rtp->bridge_lock); 01508 }
Definition at line 1160 of file rtp.c.
References ast_rtp::altthem, ast_assert, ast_codec_get_samples(), AST_FORMAT_MAX_AUDIO, ast_format_rate(), AST_FORMAT_SLINEAR, ast_frame_byteswap_be, AST_FRAME_DTMF_END, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_get_bridged(), ast_rtp_lookup_pt(), ast_rtp_senddigit_continuation(), ast_samp2tv(), ast_sched_add(), ast_set_flag, ast_tv(), ast_tvdiff_ms(), ast_verbose(), bridge_p2p_rtp_write(), ast_rtp::bridged, calc_rxstamp(), rtpPayloadType::code, ast_rtp::cycles, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp::dtmf_duration, ast_rtp::dtmf_timeout, errno, ext, ast_rtp::f, f, FLAG_NAT_ACTIVE, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lastevent, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len(), LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_rtp::rawdata, ast_rtp::resp, ast_rtp::rtcp, rtp_debug_test_addr(), rtp_get_rate(), RTP_SEQ_MOD, ast_rtp::rxcount, ast_rtp::rxseqno, ast_rtp::rxssrc, ast_rtcp::s, ast_rtp::s, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, send_dtmf(), ast_rtp::sending_digit, ast_frame::seqno, ast_frame::src, STUN_ACCEPT, stun_handle_packet(), ast_frame::subclass, ast_rtcp::them, ast_rtp::them, ast_rtp::themssrc, and ast_frame::ts.
Referenced by gtalk_rtp_read(), mgcp_rtp_read(), oh323_rtp_read(), rtpread(), sip_rtp_read(), and skinny_rtp_read().
01161 { 01162 int res; 01163 struct sockaddr_in sin; 01164 socklen_t len; 01165 unsigned int seqno; 01166 int version; 01167 int payloadtype; 01168 int hdrlen = 12; 01169 int padding; 01170 int mark; 01171 int ext; 01172 int cc; 01173 unsigned int ssrc; 01174 unsigned int timestamp; 01175 unsigned int *rtpheader; 01176 struct rtpPayloadType rtpPT; 01177 struct ast_rtp *bridged = NULL; 01178 01179 /* If time is up, kill it */ 01180 if (rtp->sending_digit) 01181 ast_rtp_senddigit_continuation(rtp); 01182 01183 len = sizeof(sin); 01184 01185 /* Cache where the header will go */ 01186 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 01187 0, (struct sockaddr *)&sin, &len); 01188 if (option_debug > 3) 01189 ast_log(LOG_DEBUG, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 01190 01191 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); 01192 if (res < 0) { 01193 ast_assert(errno != EBADF); 01194 if (errno != EAGAIN) { 01195 ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno)); 01196 ast_log(LOG_WARNING, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 01197 return NULL; 01198 } 01199 return &ast_null_frame; 01200 } 01201 01202 if (res < hdrlen) { 01203 ast_log(LOG_WARNING, "RTP Read too short\n"); 01204 return &ast_null_frame; 01205 } 01206 01207 /* Get fields */ 01208 seqno = ntohl(rtpheader[0]); 01209 01210 /* Check RTP version */ 01211 version = (seqno & 0xC0000000) >> 30; 01212 if (!version) { 01213 if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) && 01214 (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) { 01215 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 01216 } 01217 return &ast_null_frame; 01218 } 01219 01220 #if 0 /* Allow to receive RTP stream with closed transmission path */ 01221 /* If we don't have the other side's address, then ignore this */ 01222 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 01223 return &ast_null_frame; 01224 #endif 01225 01226 /* Send to whoever send to us if NAT is turned on */ 01227 if (rtp->nat) { 01228 if (((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 01229 (rtp->them.sin_port != sin.sin_port)) && 01230 ((rtp->altthem.sin_addr.s_addr != sin.sin_addr.s_addr) || 01231 (rtp->altthem.sin_port != sin.sin_port))) { 01232 rtp->them = sin; 01233 if (rtp->rtcp) { 01234 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 01235 rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1); 01236 } 01237 rtp->rxseqno = 0; 01238 ast_set_flag(rtp, FLAG_NAT_ACTIVE); 01239 if (option_debug || rtpdebug) 01240 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)); 01241 } 01242 } 01243 01244 /* If we are bridged to another RTP stream, send direct */ 01245 if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen)) 01246 return &ast_null_frame; 01247 01248 if (version != 2) 01249 return &ast_null_frame; 01250 01251 payloadtype = (seqno & 0x7f0000) >> 16; 01252 padding = seqno & (1 << 29); 01253 mark = seqno & (1 << 23); 01254 ext = seqno & (1 << 28); 01255 cc = (seqno & 0xF000000) >> 24; 01256 seqno &= 0xffff; 01257 timestamp = ntohl(rtpheader[1]); 01258 ssrc = ntohl(rtpheader[2]); 01259 01260 if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) { 01261 if (option_debug || rtpdebug) 01262 ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n"); 01263 mark = 1; 01264 } 01265 01266 rtp->rxssrc = ssrc; 01267 01268 if (padding) { 01269 /* Remove padding bytes */ 01270 res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; 01271 } 01272 01273 if (cc) { 01274 /* CSRC fields present */ 01275 hdrlen += cc*4; 01276 } 01277 01278 if (ext) { 01279 /* RTP Extension present */ 01280 hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2; 01281 hdrlen += 4; 01282 } 01283 01284 if (res < hdrlen) { 01285 ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); 01286 return &ast_null_frame; 01287 } 01288 01289 rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */ 01290 01291 if (rtp->rxcount==1) { 01292 /* This is the first RTP packet successfully received from source */ 01293 rtp->seedrxseqno = seqno; 01294 } 01295 01296 /* Do not schedule RR if RTCP isn't run */ 01297 if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) { 01298 /* Schedule transmission of Receiver Report */ 01299 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); 01300 } 01301 if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */ 01302 rtp->cycles += RTP_SEQ_MOD; 01303 01304 rtp->lastrxseqno = seqno; 01305 01306 if (rtp->themssrc==0) 01307 rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */ 01308 01309 if (rtp_debug_test_addr(&sin)) 01310 ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 01311 ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); 01312 01313 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); 01314 if (!rtpPT.isAstFormat) { 01315 struct ast_frame *f = NULL; 01316 01317 /* This is special in-band data that's not one of our codecs */ 01318 if (rtpPT.code == AST_RTP_DTMF) { 01319 /* It's special -- rfc2833 process it */ 01320 if (rtp_debug_test_addr(&sin)) { 01321 unsigned char *data; 01322 unsigned int event; 01323 unsigned int event_end; 01324 unsigned int duration; 01325 data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen; 01326 event = ntohl(*((unsigned int *)(data))); 01327 event >>= 24; 01328 event_end = ntohl(*((unsigned int *)(data))); 01329 event_end <<= 8; 01330 event_end >>= 24; 01331 duration = ntohl(*((unsigned int *)(data))); 01332 duration &= 0xFFFF; 01333 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); 01334 } 01335 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp); 01336 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { 01337 /* It's really special -- process it the Cisco way */ 01338 if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) { 01339 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01340 rtp->lastevent = seqno; 01341 } 01342 } else if (rtpPT.code == AST_RTP_CN) { 01343 /* Comfort Noise */ 01344 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01345 } else { 01346 ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr)); 01347 } 01348 return f ? f : &ast_null_frame; 01349 } 01350 rtp->lastrxformat = rtp->f.subclass = rtpPT.code; 01351 rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO; 01352 01353 rtp->rxseqno = seqno; 01354 01355 if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) { 01356 rtp->dtmf_timeout = 0; 01357 01358 if (rtp->resp) { 01359 struct ast_frame *f; 01360 f = send_dtmf(rtp, AST_FRAME_DTMF_END); 01361 f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0)); 01362 rtp->resp = 0; 01363 rtp->dtmf_timeout = rtp->dtmf_duration = 0; 01364 return f; 01365 } 01366 } 01367 01368 /* Record received timestamp as last received now */ 01369 rtp->lastrxts = timestamp; 01370 01371 rtp->f.mallocd = 0; 01372 rtp->f.datalen = res - hdrlen; 01373 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; 01374 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; 01375 rtp->f.seqno = seqno; 01376 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { 01377 rtp->f.samples = ast_codec_get_samples(&rtp->f); 01378 if (rtp->f.subclass == AST_FORMAT_SLINEAR) 01379 ast_frame_byteswap_be(&rtp->f); 01380 calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark); 01381 /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */ 01382 ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO); 01383 rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass) / 1000); 01384 rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000); 01385 } else { 01386 /* Video -- samples is # of samples vs. 90000 */ 01387 if (!rtp->lastividtimestamp) 01388 rtp->lastividtimestamp = timestamp; 01389 rtp->f.samples = timestamp - rtp->lastividtimestamp; 01390 rtp->lastividtimestamp = timestamp; 01391 rtp->f.delivery.tv_sec = 0; 01392 rtp->f.delivery.tv_usec = 0; 01393 if (mark) 01394 rtp->f.subclass |= 0x1; 01395 } 01396 rtp->f.src = "RTP"; 01397 return &rtp->f; 01398 }
int ast_rtp_reload | ( | void | ) |
Definition at line 3887 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().
03888 { 03889 struct ast_config *cfg; 03890 const char *s; 03891 03892 rtpstart = 5000; 03893 rtpend = 31000; 03894 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03895 cfg = ast_config_load("rtp.conf"); 03896 if (cfg) { 03897 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { 03898 rtpstart = atoi(s); 03899 if (rtpstart < 1024) 03900 rtpstart = 1024; 03901 if (rtpstart > 65535) 03902 rtpstart = 65535; 03903 } 03904 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { 03905 rtpend = atoi(s); 03906 if (rtpend < 1024) 03907 rtpend = 1024; 03908 if (rtpend > 65535) 03909 rtpend = 65535; 03910 } 03911 if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) { 03912 rtcpinterval = atoi(s); 03913 if (rtcpinterval == 0) 03914 rtcpinterval = 0; /* Just so we're clear... it's zero */ 03915 if (rtcpinterval < RTCP_MIN_INTERVALMS) 03916 rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */ 03917 if (rtcpinterval > RTCP_MAX_INTERVALMS) 03918 rtcpinterval = RTCP_MAX_INTERVALMS; 03919 } 03920 if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) { 03921 #ifdef SO_NO_CHECK 03922 if (ast_false(s)) 03923 nochecksums = 1; 03924 else 03925 nochecksums = 0; 03926 #else 03927 if (ast_false(s)) 03928 ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n"); 03929 #endif 03930 } 03931 if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { 03932 dtmftimeout = atoi(s); 03933 if ((dtmftimeout < 0) || (dtmftimeout > 64000)) { 03934 ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", 03935 dtmftimeout, DEFAULT_DTMF_TIMEOUT); 03936 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03937 }; 03938 } 03939 ast_config_destroy(cfg); 03940 } 03941 if (rtpstart >= rtpend) { 03942 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n"); 03943 rtpstart = 5000; 03944 rtpend = 31000; 03945 } 03946 if (option_verbose > 1) 03947 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); 03948 return 0; 03949 }
void ast_rtp_reset | ( | struct ast_rtp * | rtp | ) |
Definition at line 2158 of file rtp.c.
References ast_rtp::dtmf_timeout, ast_rtp::dtmfmute, ast_rtp::lastdigitts, ast_rtp::lastevent, ast_rtp::lasteventseqn, ast_rtp::lastividtimestamp, ast_rtp::lastovidtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxts, ast_rtp::lastts, ast_rtp::lasttxformat, ast_rtp::rxcore, ast_rtp::rxseqno, ast_rtp::seqno, and ast_rtp::txcore.
02159 { 02160 memset(&rtp->rxcore, 0, sizeof(rtp->rxcore)); 02161 memset(&rtp->txcore, 0, sizeof(rtp->txcore)); 02162 memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute)); 02163 rtp->lastts = 0; 02164 rtp->lastdigitts = 0; 02165 rtp->lastrxts = 0; 02166 rtp->lastividtimestamp = 0; 02167 rtp->lastovidtimestamp = 0; 02168 rtp->lasteventseqn = 0; 02169 rtp->lastevent = 0; 02170 rtp->lasttxformat = 0; 02171 rtp->lastrxformat = 0; 02172 rtp->dtmf_timeout = 0; 02173 rtp->seqno = 0; 02174 rtp->rxseqno = 0; 02175 }
int ast_rtp_sendcng | ( | struct ast_rtp * | rtp, | |
int | level | |||
) |
generate comfort noice (CNG)
Definition at line 2672 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_CN, ast_rtp_lookup_code(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), ast_rtp::data, ast_rtp::dtmfmute, errno, ast_rtp::lastts, LOG_ERROR, rtp_debug_test_addr(), ast_rtp::s, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.
Referenced by do_monitor().
02673 { 02674 unsigned int *rtpheader; 02675 int hdrlen = 12; 02676 int res; 02677 int payload; 02678 char data[256]; 02679 level = 127 - (level & 0x7f); 02680 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN); 02681 02682 /* If we have no peer, return immediately */ 02683 if (!rtp->them.sin_addr.s_addr) 02684 return 0; 02685 02686 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02687 02688 /* Get a pointer to the header */ 02689 rtpheader = (unsigned int *)data; 02690 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++)); 02691 rtpheader[1] = htonl(rtp->lastts); 02692 rtpheader[2] = htonl(rtp->ssrc); 02693 data[12] = level; 02694 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { 02695 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); 02696 if (res <0) 02697 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)); 02698 if (rtp_debug_test_addr(&rtp->them)) 02699 ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n" 02700 , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); 02701 02702 } 02703 return 0; 02704 }
int ast_rtp_senddigit_begin | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
Send begin frames for DTMF.
Definition at line 2280 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), ast_rtp::dtmfmute, errno, ast_rtp::lastdigitts, ast_rtp::lastts, LOG_ERROR, LOG_WARNING, rtp_debug_test_addr(), ast_rtp::s, ast_rtp::send_digit, ast_rtp::send_duration, ast_rtp::send_payload, ast_rtp::sending_digit, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.
Referenced by mgcp_senddigit_begin(), oh323_digit_begin(), and sip_senddigit_begin().
02281 { 02282 unsigned int *rtpheader; 02283 int hdrlen = 12, res = 0, i = 0, payload = 0; 02284 char data[256]; 02285 02286 if ((digit <= '9') && (digit >= '0')) 02287 digit -= '0'; 02288 else if (digit == '*') 02289 digit = 10; 02290 else if (digit == '#') 02291 digit = 11; 02292 else if ((digit >= 'A') && (digit <= 'D')) 02293 digit = digit - 'A' + 12; 02294 else if ((digit >= 'a') && (digit <= 'd')) 02295 digit = digit - 'a' + 12; 02296 else { 02297 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); 02298 return 0; 02299 } 02300 02301 /* If we have no peer, return immediately */ 02302 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 02303 return 0; 02304 02305 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); 02306 02307 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02308 rtp->send_duration = 160; 02309 rtp->lastdigitts = rtp->lastts + rtp->send_duration; 02310 02311 /* Get a pointer to the header */ 02312 rtpheader = (unsigned int *)data; 02313 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno)); 02314 rtpheader[1] = htonl(rtp->lastdigitts); 02315 rtpheader[2] = htonl(rtp->ssrc); 02316 02317 for (i = 0; i < 2; i++) { 02318 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration)); 02319 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); 02320 if (res < 0) 02321 ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n", 02322 ast_inet_ntoa(rtp->them.sin_addr), 02323 ntohs(rtp->them.sin_port), strerror(errno)); 02324 if (rtp_debug_test_addr(&rtp->them)) 02325 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 02326 ast_inet_ntoa(rtp->them.sin_addr), 02327 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 02328 /* Increment sequence number */ 02329 rtp->seqno++; 02330 /* Increment duration */ 02331 rtp->send_duration += 160; 02332 /* Clear marker bit and set seqno */ 02333 rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); 02334 } 02335 02336 /* Since we received a begin, we can safely store the digit and disable any compensation */ 02337 rtp->sending_digit = 1; 02338 rtp->send_digit = digit; 02339 rtp->send_payload = payload; 02340 02341 return 0; 02342 }
int ast_rtp_senddigit_end | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
void ast_rtp_set_alt_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | alt | |||
) |
set potential alternate source for RTP media
rtp | The RTP structure we wish to set up an alternate host/port on | |
alt | The address information for the alternate media source |
void |
Definition at line 2103 of file rtp.c.
References ast_rtcp::altthem, ast_rtp::altthem, and ast_rtp::rtcp.
Referenced by handle_request_invite().
02104 { 02105 rtp->altthem.sin_port = alt->sin_port; 02106 rtp->altthem.sin_addr = alt->sin_addr; 02107 if (rtp->rtcp) { 02108 rtp->rtcp->altthem.sin_port = htons(ntohs(alt->sin_port) + 1); 02109 rtp->rtcp->altthem.sin_addr = alt->sin_addr; 02110 } 02111 }
void ast_rtp_set_callback | ( | struct ast_rtp * | rtp, | |
ast_rtp_callback | callback | |||
) |
Definition at line 594 of file rtp.c.
References ast_rtp::callback.
Referenced by start_rtp().
00595 { 00596 rtp->callback = callback; 00597 }
void ast_rtp_set_constantssrc | ( | struct ast_rtp * | rtp | ) |
When changing sources, don't generate a new SSRC.
Definition at line 2077 of file rtp.c.
References ast_rtp::constantssrc.
Referenced by create_addr_from_peer(), and handle_request_invite().
02078 { 02079 rtp->constantssrc = 1; 02080 }
void ast_rtp_set_data | ( | struct ast_rtp * | rtp, | |
void * | data | |||
) |
Definition at line 589 of file rtp.c.
References ast_rtp::data.
Referenced by start_rtp().
00590 { 00591 rtp->data = data; 00592 }
void ast_rtp_set_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Activate payload type.
Definition at line 1708 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().
01709 { 01710 if (pt < 0 || pt >= MAX_RTP_PT || static_RTP_PT[pt].code == 0) 01711 return; /* bogus payload type */ 01712 01713 ast_mutex_lock(&rtp->bridge_lock); 01714 rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; 01715 ast_mutex_unlock(&rtp->bridge_lock); 01716 }
void ast_rtp_set_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2092 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().
02093 { 02094 rtp->them.sin_port = them->sin_port; 02095 rtp->them.sin_addr = them->sin_addr; 02096 if (rtp->rtcp) { 02097 rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1); 02098 rtp->rtcp->them.sin_addr = them->sin_addr; 02099 } 02100 rtp->rxseqno = 0; 02101 }
void ast_rtp_set_rtpholdtimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp hold timeout.
Definition at line 556 of file rtp.c.
References ast_rtp::rtpholdtimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00557 { 00558 rtp->rtpholdtimeout = timeout; 00559 }
void ast_rtp_set_rtpkeepalive | ( | struct ast_rtp * | rtp, | |
int | period | |||
) |
set RTP keepalive interval
Definition at line 562 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by create_addr_from_peer(), and sip_alloc().
00563 { 00564 rtp->rtpkeepalive = period; 00565 }
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 1735 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(), process_sdp_a_audio(), process_sdp_a_video(), and set_dtmf_payload().
01738 { 01739 unsigned int i; 01740 int found = 0; 01741 01742 if (pt < 0 || pt >= MAX_RTP_PT) 01743 return -1; /* bogus payload type */ 01744 01745 ast_mutex_lock(&rtp->bridge_lock); 01746 01747 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01748 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && 01749 strcasecmp(mimeType, mimeTypes[i].type) == 0) { 01750 found = 1; 01751 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; 01752 if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) && 01753 mimeTypes[i].payloadType.isAstFormat && 01754 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01755 rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2; 01756 break; 01757 } 01758 } 01759 01760 ast_mutex_unlock(&rtp->bridge_lock); 01761 01762 return (found ? 0 : -1); 01763 }
void ast_rtp_set_rtptimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp timeout.
Definition at line 550 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00551 { 00552 rtp->rtptimeout = timeout; 00553 }
void ast_rtp_set_rtptimers_onhold | ( | struct ast_rtp * | rtp | ) |
Definition at line 543 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by handle_response_invite().
00544 { 00545 rtp->rtptimeout = (-1) * rtp->rtptimeout; 00546 rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout; 00547 }
void ast_rtp_setdtmf | ( | struct ast_rtp * | rtp, | |
int | dtmf | |||
) |
Indicate whether this RTP session is carrying DTMF or not.
Definition at line 609 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().
00610 { 00611 ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF); 00612 }
void ast_rtp_setdtmfcompensate | ( | struct ast_rtp * | rtp, | |
int | compensate | |||
) |
Compensate for devices that send RFC2833 packets all at once.
Definition at line 614 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().
00615 { 00616 ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE); 00617 }
void ast_rtp_setnat | ( | struct ast_rtp * | rtp, | |
int | nat | |||
) |
Definition at line 599 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 619 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_STUN.
Referenced by gtalk_new().
00620 { 00621 ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN); 00622 }
int ast_rtp_settos | ( | struct ast_rtp * | rtp, | |
int | tos | |||
) |
Definition at line 2068 of file rtp.c.
References ast_log(), LOG_WARNING, and ast_rtp::s.
Referenced by __oh323_rtp_create(), and sip_alloc().
02069 { 02070 int res; 02071 02072 if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 02073 ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); 02074 return res; 02075 }
void ast_rtp_stop | ( | struct ast_rtp * | rtp | ) |
Definition at line 2142 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().
02143 { 02144 if (rtp->rtcp) { 02145 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02146 } 02147 02148 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 02149 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 02150 if (rtp->rtcp) { 02151 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr)); 02152 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port)); 02153 } 02154 02155 ast_clear_flag(rtp, FLAG_P2P_SENT_MARK); 02156 }
void ast_rtp_stun_request | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | suggestion, | |||
const char * | username | |||
) |
Definition at line 406 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().
00407 { 00408 struct stun_header *req; 00409 unsigned char reqdata[1024]; 00410 int reqlen, reqleft; 00411 struct stun_attr *attr; 00412 00413 req = (struct stun_header *)reqdata; 00414 stun_req_id(req); 00415 reqlen = 0; 00416 reqleft = sizeof(reqdata) - sizeof(struct stun_header); 00417 req->msgtype = 0; 00418 req->msglen = 0; 00419 attr = (struct stun_attr *)req->ies; 00420 if (username) 00421 append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft); 00422 req->msglen = htons(reqlen); 00423 req->msgtype = htons(STUN_BINDREQ); 00424 stun_send(rtp->s, suggestion, req); 00425 }
void ast_rtp_unset_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
clear payload type
Definition at line 1720 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_a_audio(), and process_sdp_a_video().
01721 { 01722 if (pt < 0 || pt >= MAX_RTP_PT) 01723 return; /* bogus payload type */ 01724 01725 ast_mutex_lock(&rtp->bridge_lock); 01726 rtp->current_RTP_PT[pt].isAstFormat = 0; 01727 rtp->current_RTP_PT[pt].code = 0; 01728 ast_mutex_unlock(&rtp->bridge_lock); 01729 }
Definition at line 2877 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_format_list::cur_ms, ast_frame::datalen, f, ast_format_list::flags, ast_format_list::fr_len, ast_frame::frametype, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, option_debug, ast_rtp::pref, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.
Referenced by gtalk_write(), mgcp_write(), oh323_write(), sip_write(), and skinny_write().
02878 { 02879 struct ast_frame *f; 02880 int codec; 02881 int hdrlen = 12; 02882 int subclass; 02883 02884 02885 /* If we have no peer, return immediately */ 02886 if (!rtp->them.sin_addr.s_addr) 02887 return 0; 02888 02889 /* If there is no data length, return immediately */ 02890 if (!_f->datalen) 02891 return 0; 02892 02893 /* Make sure we have enough space for RTP header */ 02894 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { 02895 ast_log(LOG_WARNING, "RTP can only send voice and video\n"); 02896 return -1; 02897 } 02898 02899 subclass = _f->subclass; 02900 if (_f->frametype == AST_FRAME_VIDEO) 02901 subclass &= ~0x1; 02902 02903 codec = ast_rtp_lookup_code(rtp, 1, subclass); 02904 if (codec < 0) { 02905 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); 02906 return -1; 02907 } 02908 02909 if (rtp->lasttxformat != subclass) { 02910 /* New format, reset the smoother */ 02911 if (option_debug) 02912 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); 02913 rtp->lasttxformat = subclass; 02914 if (rtp->smoother) 02915 ast_smoother_free(rtp->smoother); 02916 rtp->smoother = NULL; 02917 } 02918 02919 if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) { 02920 struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass); 02921 if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */ 02922 if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) { 02923 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)); 02924 return -1; 02925 } 02926 if (fmt.flags) 02927 ast_smoother_set_flags(rtp->smoother, fmt.flags); 02928 if (option_debug) 02929 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)); 02930 } 02931 } 02932 if (rtp->smoother) { 02933 if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) { 02934 ast_smoother_feed_be(rtp->smoother, _f); 02935 } else { 02936 ast_smoother_feed(rtp->smoother, _f); 02937 } 02938 02939 while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) { 02940 ast_rtp_raw_write(rtp, f, codec); 02941 } 02942 } else { 02943 /* Don't buffer outgoing frames; send them one-per-packet: */ 02944 if (_f->offset < hdrlen) { 02945 f = ast_frdup(_f); 02946 } else { 02947 f = _f; 02948 } 02949 if (f->data) { 02950 ast_rtp_raw_write(rtp, f, codec); 02951 } 02952 if (f != _f) 02953 ast_frfree(f); 02954 } 02955 02956 return 0; 02957 }