#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_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 520 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 2426 of file rtp.c.
References ast_rtcp_write(), ast_rtp::rtcp, and ast_rtcp::sendfur.
02427 { 02428 struct ast_rtp *rtp = data; 02429 int res; 02430 02431 rtp->rtcp->sendfur = 1; 02432 res = ast_rtcp_write(data); 02433 02434 return res; 02435 }
size_t ast_rtp_alloc_size | ( | void | ) |
Get the amount of space required to hold an RTP session.
Definition at line 400 of file rtp.c.
Referenced by process_sdp().
00401 { 00402 return sizeof(struct ast_rtp); 00403 }
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 3402 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.
03403 { 03404 struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ 03405 struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */ 03406 struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL; 03407 enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED; 03408 enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED; 03409 enum ast_bridge_result res = AST_BRIDGE_FAILED; 03410 int codec0 = 0, codec1 = 0; 03411 void *pvt0 = NULL, *pvt1 = NULL; 03412 03413 /* Lock channels */ 03414 ast_channel_lock(c0); 03415 while(ast_channel_trylock(c1)) { 03416 ast_channel_unlock(c0); 03417 usleep(1); 03418 ast_channel_lock(c0); 03419 } 03420 03421 /* Ensure neither channel got hungup during lock avoidance */ 03422 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 03423 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name); 03424 ast_channel_unlock(c0); 03425 ast_channel_unlock(c1); 03426 return AST_BRIDGE_FAILED; 03427 } 03428 03429 /* Find channel driver interfaces */ 03430 if (!(pr0 = get_proto(c0))) { 03431 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); 03432 ast_channel_unlock(c0); 03433 ast_channel_unlock(c1); 03434 return AST_BRIDGE_FAILED; 03435 } 03436 if (!(pr1 = get_proto(c1))) { 03437 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); 03438 ast_channel_unlock(c0); 03439 ast_channel_unlock(c1); 03440 return AST_BRIDGE_FAILED; 03441 } 03442 03443 /* Get channel specific interface structures */ 03444 pvt0 = c0->tech_pvt; 03445 pvt1 = c1->tech_pvt; 03446 03447 /* Get audio and video interface (if native bridge is possible) */ 03448 audio_p0_res = pr0->get_rtp_info(c0, &p0); 03449 video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED; 03450 audio_p1_res = pr1->get_rtp_info(c1, &p1); 03451 video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED; 03452 03453 /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */ 03454 if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE)) 03455 audio_p0_res = AST_RTP_GET_FAILED; 03456 if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE)) 03457 audio_p1_res = AST_RTP_GET_FAILED; 03458 03459 /* Check if a bridge is possible (partial/native) */ 03460 if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) { 03461 /* Somebody doesn't want to play... */ 03462 ast_channel_unlock(c0); 03463 ast_channel_unlock(c1); 03464 return AST_BRIDGE_FAILED_NOWARN; 03465 } 03466 03467 /* If we need to feed DTMF frames into the core then only do a partial native bridge */ 03468 if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) { 03469 ast_set_flag(p0, FLAG_P2P_NEED_DTMF); 03470 audio_p0_res = AST_RTP_TRY_PARTIAL; 03471 } 03472 03473 if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) { 03474 ast_set_flag(p1, FLAG_P2P_NEED_DTMF); 03475 audio_p1_res = AST_RTP_TRY_PARTIAL; 03476 } 03477 03478 /* If both sides are not using the same method of DTMF transmission 03479 * (ie: one is RFC2833, other is INFO... then we can not do direct media. 03480 * -------------------------------------------------- 03481 * | DTMF Mode | HAS_DTMF | Accepts Begin Frames | 03482 * |-----------|------------|-----------------------| 03483 * | Inband | False | True | 03484 * | RFC2833 | True | True | 03485 * | SIP INFO | False | False | 03486 * -------------------------------------------------- 03487 * However, if DTMF from both channels is being monitored by the core, then 03488 * we can still do packet-to-packet bridging, because passing through the 03489 * core will handle DTMF mode translation. 03490 */ 03491 if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) || 03492 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) { 03493 if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) { 03494 ast_channel_unlock(c0); 03495 ast_channel_unlock(c1); 03496 return AST_BRIDGE_FAILED_NOWARN; 03497 } 03498 audio_p0_res = AST_RTP_TRY_PARTIAL; 03499 audio_p1_res = AST_RTP_TRY_PARTIAL; 03500 } 03501 03502 /* If we need to feed frames into the core don't do a P2P bridge */ 03503 if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) || 03504 (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) { 03505 ast_channel_unlock(c0); 03506 ast_channel_unlock(c1); 03507 return AST_BRIDGE_FAILED_NOWARN; 03508 } 03509 03510 /* Get codecs from both sides */ 03511 codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0; 03512 codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0; 03513 if (codec0 && codec1 && !(codec0 & codec1)) { 03514 /* Hey, we can't do native bridging if both parties speak different codecs */ 03515 if (option_debug) 03516 ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); 03517 ast_channel_unlock(c0); 03518 ast_channel_unlock(c1); 03519 return AST_BRIDGE_FAILED_NOWARN; 03520 } 03521 03522 /* If either side can only do a partial bridge, then don't try for a true native bridge */ 03523 if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) { 03524 struct ast_format_list fmt0, fmt1; 03525 03526 /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */ 03527 if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) { 03528 if (option_debug) 03529 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n"); 03530 ast_channel_unlock(c0); 03531 ast_channel_unlock(c1); 03532 return AST_BRIDGE_FAILED_NOWARN; 03533 } 03534 /* They must also be using the same packetization */ 03535 fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat); 03536 fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat); 03537 if (fmt0.cur_ms != fmt1.cur_ms) { 03538 if (option_debug) 03539 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n"); 03540 ast_channel_unlock(c0); 03541 ast_channel_unlock(c1); 03542 return AST_BRIDGE_FAILED_NOWARN; 03543 } 03544 03545 if (option_verbose > 2) 03546 ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name); 03547 res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1); 03548 } else { 03549 if (option_verbose > 2) 03550 ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name); 03551 res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1); 03552 } 03553 03554 return res; 03555 }
int ast_rtp_codec_getformat | ( | int | pt | ) |
Definition at line 2854 of file rtp.c.
References rtpPayloadType::code, and static_RTP_PT.
Referenced by process_sdp().
02855 { 02856 if (pt < 0 || pt > MAX_RTP_PT) 02857 return 0; /* bogus payload type */ 02858 02859 if (static_RTP_PT[pt].isAstFormat) 02860 return static_RTP_PT[pt].code; 02861 else 02862 return 0; 02863 }
struct ast_codec_pref* ast_rtp_codec_getpref | ( | struct ast_rtp * | rtp | ) |
Definition at line 2849 of file rtp.c.
References ast_rtp::pref.
Referenced by add_codec_to_sdp(), and process_sdp().
02850 { 02851 return &rtp->pref; 02852 }
int ast_rtp_codec_setpref | ( | struct ast_rtp * | rtp, | |
struct ast_codec_pref * | prefs | |||
) |
Definition at line 2802 of file rtp.c.
References ast_codec_pref_getsize(), ast_log(), ast_smoother_new(), ast_smoother_reconfigure(), ast_smoother_set_flags(), ast_format_list::cur_ms, ast_format_list::flags, ast_format_list::fr_len, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, option_debug, ast_rtp::pref, prefs, and ast_rtp::smoother.
Referenced by __oh323_rtp_create(), check_user_full(), create_addr_from_peer(), process_sdp(), register_verify(), set_peer_capabilities(), sip_alloc(), start_rtp(), and transmit_response_with_sdp().
02803 { 02804 struct ast_format_list current_format_old, current_format_new; 02805 02806 /* if no packets have been sent through this session yet, then 02807 * changing preferences does not require any extra work 02808 */ 02809 if (rtp->lasttxformat == 0) { 02810 rtp->pref = *prefs; 02811 return 0; 02812 } 02813 02814 current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat); 02815 02816 rtp->pref = *prefs; 02817 02818 current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat); 02819 02820 /* if the framing desired for the current format has changed, we may have to create 02821 * or adjust the smoother for this session 02822 */ 02823 if ((current_format_new.inc_ms != 0) && 02824 (current_format_new.cur_ms != current_format_old.cur_ms)) { 02825 int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms; 02826 02827 if (rtp->smoother) { 02828 ast_smoother_reconfigure(rtp->smoother, new_size); 02829 if (option_debug) { 02830 ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size); 02831 } 02832 } else { 02833 if (!(rtp->smoother = ast_smoother_new(new_size))) { 02834 ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size); 02835 return -1; 02836 } 02837 if (current_format_new.flags) { 02838 ast_smoother_set_flags(rtp->smoother, current_format_new.flags); 02839 } 02840 if (option_debug) { 02841 ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size); 02842 } 02843 } 02844 } 02845 02846 return 0; 02847 }
void ast_rtp_destroy | ( | struct ast_rtp * | rtp | ) |
Definition at line 2209 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().
02210 { 02211 if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) { 02212 /*Print some info on the call here */ 02213 ast_verbose(" RTP-stats\n"); 02214 ast_verbose("* Our Receiver:\n"); 02215 ast_verbose(" SSRC: %u\n", rtp->themssrc); 02216 ast_verbose(" Received packets: %u\n", rtp->rxcount); 02217 ast_verbose(" Lost packets: %u\n", rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0); 02218 ast_verbose(" Jitter: %.4f\n", rtp->rxjitter); 02219 ast_verbose(" Transit: %.4f\n", rtp->rxtransit); 02220 ast_verbose(" RR-count: %u\n", rtp->rtcp ? rtp->rtcp->rr_count : 0); 02221 ast_verbose("* Our Sender:\n"); 02222 ast_verbose(" SSRC: %u\n", rtp->ssrc); 02223 ast_verbose(" Sent packets: %u\n", rtp->txcount); 02224 ast_verbose(" Lost packets: %u\n", rtp->rtcp ? rtp->rtcp->reported_lost : 0); 02225 ast_verbose(" Jitter: %u\n", rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int)65536.0) : 0); 02226 ast_verbose(" SR-count: %u\n", rtp->rtcp ? rtp->rtcp->sr_count : 0); 02227 ast_verbose(" RTT: %f\n", rtp->rtcp ? rtp->rtcp->rtt : 0); 02228 } 02229 02230 if (rtp->smoother) 02231 ast_smoother_free(rtp->smoother); 02232 if (rtp->ioid) 02233 ast_io_remove(rtp->io, rtp->ioid); 02234 if (rtp->s > -1) 02235 close(rtp->s); 02236 if (rtp->rtcp) { 02237 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02238 close(rtp->rtcp->s); 02239 free(rtp->rtcp); 02240 rtp->rtcp=NULL; 02241 } 02242 02243 ast_mutex_destroy(&rtp->bridge_lock); 02244 02245 free(rtp); 02246 }
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 1541 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().
01542 { 01543 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01544 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01545 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01546 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01547 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01548 int srccodec, destcodec, nat_active = 0; 01549 01550 /* Lock channels */ 01551 ast_channel_lock(dest); 01552 if (src) { 01553 while(ast_channel_trylock(src)) { 01554 ast_channel_unlock(dest); 01555 usleep(1); 01556 ast_channel_lock(dest); 01557 } 01558 } 01559 01560 /* Find channel driver interfaces */ 01561 destpr = get_proto(dest); 01562 if (src) 01563 srcpr = get_proto(src); 01564 if (!destpr) { 01565 if (option_debug) 01566 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01567 ast_channel_unlock(dest); 01568 if (src) 01569 ast_channel_unlock(src); 01570 return 0; 01571 } 01572 if (!srcpr) { 01573 if (option_debug) 01574 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>"); 01575 ast_channel_unlock(dest); 01576 if (src) 01577 ast_channel_unlock(src); 01578 return 0; 01579 } 01580 01581 /* Get audio and video interface (if native bridge is possible) */ 01582 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01583 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01584 if (srcpr) { 01585 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01586 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01587 } 01588 01589 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01590 if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) { 01591 /* Somebody doesn't want to play... */ 01592 ast_channel_unlock(dest); 01593 if (src) 01594 ast_channel_unlock(src); 01595 return 0; 01596 } 01597 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) 01598 srccodec = srcpr->get_codec(src); 01599 else 01600 srccodec = 0; 01601 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) 01602 destcodec = destpr->get_codec(dest); 01603 else 01604 destcodec = 0; 01605 /* Ensure we have at least one matching codec */ 01606 if (srcp && !(srccodec & destcodec)) { 01607 ast_channel_unlock(dest); 01608 ast_channel_unlock(src); 01609 return 0; 01610 } 01611 /* Consider empty media as non-existant */ 01612 if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) 01613 srcp = NULL; 01614 /* If the client has NAT stuff turned on then just safe NAT is active */ 01615 if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01616 nat_active = 1; 01617 /* Bridge media early */ 01618 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active)) 01619 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01620 ast_channel_unlock(dest); 01621 if (src) 01622 ast_channel_unlock(src); 01623 if (option_debug) 01624 ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01625 return 1; 01626 }
int ast_rtp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 515 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().
00516 { 00517 return rtp->s; 00518 }
Definition at line 2119 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().
02120 { 02121 struct ast_rtp *bridged = NULL; 02122 02123 ast_mutex_lock(&rtp->bridge_lock); 02124 bridged = rtp->bridged; 02125 ast_mutex_unlock(&rtp->bridge_lock); 02126 02127 return bridged; 02128 }
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 1762 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().
01764 { 01765 int pt; 01766 01767 ast_mutex_lock(&rtp->bridge_lock); 01768 01769 *astFormats = *nonAstFormats = 0; 01770 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01771 if (rtp->current_RTP_PT[pt].isAstFormat) { 01772 *astFormats |= rtp->current_RTP_PT[pt].code; 01773 } else { 01774 *nonAstFormats |= rtp->current_RTP_PT[pt].code; 01775 } 01776 } 01777 01778 ast_mutex_unlock(&rtp->bridge_lock); 01779 01780 return; 01781 }
int ast_rtp_get_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2101 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().
02102 { 02103 if ((them->sin_family != AF_INET) || 02104 (them->sin_port != rtp->them.sin_port) || 02105 (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) { 02106 them->sin_family = AF_INET; 02107 them->sin_port = rtp->them.sin_port; 02108 them->sin_addr = rtp->them.sin_addr; 02109 return 1; 02110 } 02111 return 0; 02112 }
char* ast_rtp_get_quality | ( | struct ast_rtp * | rtp, | |
struct ast_rtp_quality * | qual | |||
) |
Return RTCP quality string.
Definition at line 2165 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().
02166 { 02167 /* 02168 *ssrc our ssrc 02169 *themssrc their ssrc 02170 *lp lost packets 02171 *rxjitter our calculated jitter(rx) 02172 *rxcount no. received packets 02173 *txjitter reported jitter of the other end 02174 *txcount transmitted packets 02175 *rlp remote lost packets 02176 *rtt round trip time 02177 */ 02178 02179 if (qual && rtp) { 02180 qual->local_ssrc = rtp->ssrc; 02181 qual->local_jitter = rtp->rxjitter; 02182 qual->local_count = rtp->rxcount; 02183 qual->remote_ssrc = rtp->themssrc; 02184 qual->remote_count = rtp->txcount; 02185 if (rtp->rtcp) { 02186 qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior; 02187 qual->remote_lostpackets = rtp->rtcp->reported_lost; 02188 qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0; 02189 qual->rtt = rtp->rtcp->rtt; 02190 } 02191 } 02192 if (rtp->rtcp) { 02193 snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), 02194 "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", 02195 rtp->ssrc, 02196 rtp->themssrc, 02197 rtp->rtcp->expected_prior - rtp->rtcp->received_prior, 02198 rtp->rxjitter, 02199 rtp->rxcount, 02200 (double)rtp->rtcp->reported_jitter / 65536.0, 02201 rtp->txcount, 02202 rtp->rtcp->reported_lost, 02203 rtp->rtcp->rtt); 02204 return rtp->rtcp->quality; 02205 } else 02206 return "<Unknown> - RTP/RTCP has already been destroyed"; 02207 }
int ast_rtp_get_rtpholdtimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp hold timeout.
Definition at line 575 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by do_monitor().
00576 { 00577 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00578 return 0; 00579 return rtp->rtpholdtimeout; 00580 }
int ast_rtp_get_rtpkeepalive | ( | struct ast_rtp * | rtp | ) |
Get RTP keepalive interval.
Definition at line 583 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by do_monitor().
00584 { 00585 return rtp->rtpkeepalive; 00586 }
int ast_rtp_get_rtptimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp timeout.
Definition at line 567 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by do_monitor().
00568 { 00569 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00570 return 0; 00571 return rtp->rtptimeout; 00572 }
void ast_rtp_get_us | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | us | |||
) |
Definition at line 2114 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 603 of file rtp.c.
References ast_test_flag, and FLAG_NAT_ACTIVE.
Referenced by sip_get_rtp_peer().
00604 { 00605 return ast_test_flag(rtp, FLAG_NAT_ACTIVE); 00606 }
void ast_rtp_init | ( | void | ) |
Initialize the RTP system in Asterisk.
Definition at line 3940 of file rtp.c.
References ast_cli_register_multiple(), ast_rtp_reload(), and cli_rtp.
Referenced by main().
03941 { 03942 ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry)); 03943 ast_rtp_reload(); 03944 }
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 1805 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().
01806 { 01807 int pt = 0; 01808 01809 ast_mutex_lock(&rtp->bridge_lock); 01810 01811 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && 01812 code == rtp->rtp_lookup_code_cache_code) { 01813 /* Use our cached mapping, to avoid the overhead of the loop below */ 01814 pt = rtp->rtp_lookup_code_cache_result; 01815 ast_mutex_unlock(&rtp->bridge_lock); 01816 return pt; 01817 } 01818 01819 /* Check the dynamic list first */ 01820 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01821 if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) { 01822 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01823 rtp->rtp_lookup_code_cache_code = code; 01824 rtp->rtp_lookup_code_cache_result = pt; 01825 ast_mutex_unlock(&rtp->bridge_lock); 01826 return pt; 01827 } 01828 } 01829 01830 /* Then the static list */ 01831 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01832 if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) { 01833 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01834 rtp->rtp_lookup_code_cache_code = code; 01835 rtp->rtp_lookup_code_cache_result = pt; 01836 ast_mutex_unlock(&rtp->bridge_lock); 01837 return pt; 01838 } 01839 } 01840 01841 ast_mutex_unlock(&rtp->bridge_lock); 01842 01843 return -1; 01844 }
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 1865 of file rtp.c.
References ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, format, len(), and name.
Referenced by process_sdp().
01867 { 01868 int format; 01869 unsigned len; 01870 char *end = buf; 01871 char *start = buf; 01872 01873 if (!buf || !size) 01874 return NULL; 01875 01876 snprintf(end, size, "0x%x (", capability); 01877 01878 len = strlen(end); 01879 end += len; 01880 size -= len; 01881 start = end; 01882 01883 for (format = 1; format < AST_RTP_MAX; format <<= 1) { 01884 if (capability & format) { 01885 const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options); 01886 01887 snprintf(end, size, "%s|", name); 01888 len = strlen(end); 01889 end += len; 01890 size -= len; 01891 } 01892 } 01893 01894 if (start == end) 01895 snprintf(start, size, "nothing)"); 01896 else if (size > 1) 01897 *(end -1) = ')'; 01898 01899 return buf; 01900 }
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 1846 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().
01848 { 01849 unsigned int i; 01850 01851 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01852 if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) { 01853 if (isAstFormat && 01854 (code == AST_FORMAT_G726_AAL2) && 01855 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01856 return "G726-32"; 01857 else 01858 return mimeTypes[i].subtype; 01859 } 01860 } 01861 01862 return ""; 01863 }
struct rtpPayloadType ast_rtp_lookup_pt | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Mapping between RTP payload format codes and Asterisk codes:.
Definition at line 1783 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().
01784 { 01785 struct rtpPayloadType result; 01786 01787 result.isAstFormat = result.code = 0; 01788 01789 if (pt < 0 || pt > MAX_RTP_PT) 01790 return result; /* bogus payload type */ 01791 01792 /* Start with negotiated codecs */ 01793 ast_mutex_lock(&rtp->bridge_lock); 01794 result = rtp->current_RTP_PT[pt]; 01795 ast_mutex_unlock(&rtp->bridge_lock); 01796 01797 /* If it doesn't exist, check our static RTP type list, just in case */ 01798 if (!result.code) 01799 result = static_RTP_PT[pt]; 01800 01801 return result; 01802 }
int ast_rtp_make_compatible | ( | struct ast_channel * | dest, | |
struct ast_channel * | src, | |||
int | media | |||
) |
Definition at line 1628 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().
01629 { 01630 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01631 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01632 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01633 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01634 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01635 int srccodec, destcodec; 01636 01637 /* Lock channels */ 01638 ast_channel_lock(dest); 01639 while(ast_channel_trylock(src)) { 01640 ast_channel_unlock(dest); 01641 usleep(1); 01642 ast_channel_lock(dest); 01643 } 01644 01645 /* Find channel driver interfaces */ 01646 if (!(destpr = get_proto(dest))) { 01647 if (option_debug) 01648 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01649 ast_channel_unlock(dest); 01650 ast_channel_unlock(src); 01651 return 0; 01652 } 01653 if (!(srcpr = get_proto(src))) { 01654 if (option_debug) 01655 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name); 01656 ast_channel_unlock(dest); 01657 ast_channel_unlock(src); 01658 return 0; 01659 } 01660 01661 /* Get audio and video interface (if native bridge is possible) */ 01662 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01663 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01664 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01665 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01666 01667 /* Ensure we have at least one matching codec */ 01668 if (srcpr->get_codec) 01669 srccodec = srcpr->get_codec(src); 01670 else 01671 srccodec = 0; 01672 if (destpr->get_codec) 01673 destcodec = destpr->get_codec(dest); 01674 else 01675 destcodec = 0; 01676 01677 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01678 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)) { 01679 /* Somebody doesn't want to play... */ 01680 ast_channel_unlock(dest); 01681 ast_channel_unlock(src); 01682 return 0; 01683 } 01684 ast_rtp_pt_copy(destp, srcp); 01685 if (vdestp && vsrcp) 01686 ast_rtp_pt_copy(vdestp, vsrcp); 01687 if (media) { 01688 /* Bridge early */ 01689 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01690 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name); 01691 } 01692 ast_channel_unlock(dest); 01693 ast_channel_unlock(src); 01694 if (option_debug) 01695 ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name); 01696 return 1; 01697 }
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 2055 of file rtp.c.
References ast_rtp_new_with_bindaddr(), io, and sched.
02056 { 02057 struct in_addr ia; 02058 02059 memset(&ia, 0, sizeof(ia)); 02060 return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia); 02061 }
void ast_rtp_new_init | ( | struct ast_rtp * | rtp | ) |
Initialize a new RTP structure.
Definition at line 1949 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().
01950 { 01951 ast_mutex_init(&rtp->bridge_lock); 01952 01953 rtp->them.sin_family = AF_INET; 01954 rtp->us.sin_family = AF_INET; 01955 rtp->ssrc = ast_random(); 01956 rtp->seqno = ast_random() & 0xffff; 01957 ast_set_flag(rtp, FLAG_HAS_DTMF); 01958 01959 return; 01960 }
void ast_rtp_new_source | ( | struct ast_rtp * | rtp | ) |
Definition at line 2072 of file rtp.c.
References ast_rtp::set_marker_bit.
Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), sip_write(), and skinny_indicate().
struct ast_rtp* ast_rtp_new_with_bindaddr | ( | struct sched_context * | sched, | |
struct io_context * | io, | |||
int | rtcpenable, | |||
int | callbackmode, | |||
struct in_addr | in | |||
) |
Initializate a RTP session using an in_addr structure.
This fuction gets called by ast_rtp_new().
sched | ||
io | ||
rtcpenable | ||
callbackmode | ||
in |
Definition at line 1962 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().
01963 { 01964 struct ast_rtp *rtp; 01965 int x; 01966 int first; 01967 int startplace; 01968 01969 if (!(rtp = ast_calloc(1, sizeof(*rtp)))) 01970 return NULL; 01971 01972 ast_rtp_new_init(rtp); 01973 01974 rtp->s = rtp_socket(); 01975 if (option_debug > 2) 01976 ast_log(LOG_DEBUG, "socket RTP fd: %i\n", rtp->s); 01977 if (rtp->s < 0) { 01978 free(rtp); 01979 ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno)); 01980 return NULL; 01981 } 01982 if (sched && rtcpenable) { 01983 rtp->sched = sched; 01984 rtp->rtcp = ast_rtcp_new(); 01985 if (option_debug > 2) 01986 ast_log(LOG_DEBUG, "socket RTCP fd: %i\n", rtp->rtcp->s); 01987 } 01988 01989 /* Select a random port number in the range of possible RTP */ 01990 x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart; 01991 x = x & ~1; 01992 /* Save it for future references. */ 01993 startplace = x; 01994 /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */ 01995 for (;;) { 01996 /* Must be an even port number by RTP spec */ 01997 rtp->us.sin_port = htons(x); 01998 rtp->us.sin_addr = addr; 01999 /* If there's rtcp, initialize it as well. */ 02000 if (rtp->rtcp) { 02001 rtp->rtcp->us.sin_port = htons(x + 1); 02002 rtp->rtcp->us.sin_addr = addr; 02003 } 02004 /* Try to bind it/them. */ 02005 if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) && 02006 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) 02007 break; 02008 if (!first) { 02009 /* Primary bind succeeded! Gotta recreate it */ 02010 close(rtp->s); 02011 rtp->s = rtp_socket(); 02012 if (option_debug > 2) 02013 ast_log(LOG_DEBUG, "socket RTP2 fd: %i\n", rtp->s); 02014 } 02015 if (errno != EADDRINUSE) { 02016 /* We got an error that wasn't expected, abort! */ 02017 ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); 02018 close(rtp->s); 02019 if (rtp->rtcp) { 02020 close(rtp->rtcp->s); 02021 free(rtp->rtcp); 02022 } 02023 free(rtp); 02024 return NULL; 02025 } 02026 /* The port was used, increment it (by two). */ 02027 x += 2; 02028 /* Did we go over the limit ? */ 02029 if (x > rtpend) 02030 /* then, start from the begingig. */ 02031 x = (rtpstart + 1) & ~1; 02032 /* Check if we reached the place were we started. */ 02033 if (x == startplace) { 02034 /* If so, there's no ports available. */ 02035 ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); 02036 close(rtp->s); 02037 if (rtp->rtcp) { 02038 close(rtp->rtcp->s); 02039 free(rtp->rtcp); 02040 } 02041 free(rtp); 02042 return NULL; 02043 } 02044 } 02045 rtp->sched = sched; 02046 rtp->io = io; 02047 if (callbackmode) { 02048 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); 02049 ast_set_flag(rtp, FLAG_CALLBACK_MODE); 02050 } 02051 ast_rtp_pt_default(rtp); 02052 return rtp; 02053 }
int ast_rtp_proto_register | ( | struct ast_rtp_protocol * | proto | ) |
Register interface to channel driver.
Definition at line 2956 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().
02957 { 02958 struct ast_rtp_protocol *cur; 02959 02960 AST_LIST_LOCK(&protos); 02961 AST_LIST_TRAVERSE(&protos, cur, list) { 02962 if (!strcmp(cur->type, proto->type)) { 02963 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); 02964 AST_LIST_UNLOCK(&protos); 02965 return -1; 02966 } 02967 } 02968 AST_LIST_INSERT_HEAD(&protos, proto, list); 02969 AST_LIST_UNLOCK(&protos); 02970 02971 return 0; 02972 }
void ast_rtp_proto_unregister | ( | struct ast_rtp_protocol * | proto | ) |
Unregister interface to channel driver.
Definition at line 2948 of file rtp.c.
References AST_LIST_LOCK, AST_LIST_REMOVE, and AST_LIST_UNLOCK.
Referenced by load_module(), and unload_module().
02949 { 02950 AST_LIST_LOCK(&protos); 02951 AST_LIST_REMOVE(&protos, proto, list); 02952 AST_LIST_UNLOCK(&protos); 02953 }
void ast_rtp_pt_clear | ( | struct ast_rtp * | rtp | ) |
Setting RTP payload types from lines in a SDP description:.
Definition at line 1465 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().
01466 { 01467 int i; 01468 01469 if (!rtp) 01470 return; 01471 01472 ast_mutex_lock(&rtp->bridge_lock); 01473 01474 for (i = 0; i < MAX_RTP_PT; ++i) { 01475 rtp->current_RTP_PT[i].isAstFormat = 0; 01476 rtp->current_RTP_PT[i].code = 0; 01477 } 01478 01479 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01480 rtp->rtp_lookup_code_cache_code = 0; 01481 rtp->rtp_lookup_code_cache_result = 0; 01482 01483 ast_mutex_unlock(&rtp->bridge_lock); 01484 }
Copy payload types between RTP structures.
Definition at line 1505 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().
01506 { 01507 unsigned int i; 01508 01509 ast_mutex_lock(&dest->bridge_lock); 01510 ast_mutex_lock(&src->bridge_lock); 01511 01512 for (i=0; i < MAX_RTP_PT; ++i) { 01513 dest->current_RTP_PT[i].isAstFormat = 01514 src->current_RTP_PT[i].isAstFormat; 01515 dest->current_RTP_PT[i].code = 01516 src->current_RTP_PT[i].code; 01517 } 01518 dest->rtp_lookup_code_cache_isAstFormat = 0; 01519 dest->rtp_lookup_code_cache_code = 0; 01520 dest->rtp_lookup_code_cache_result = 0; 01521 01522 ast_mutex_unlock(&src->bridge_lock); 01523 ast_mutex_unlock(&dest->bridge_lock); 01524 }
void ast_rtp_pt_default | ( | struct ast_rtp * | rtp | ) |
Set payload types to defaults.
Definition at line 1486 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.
01487 { 01488 int i; 01489 01490 ast_mutex_lock(&rtp->bridge_lock); 01491 01492 /* Initialize to default payload types */ 01493 for (i = 0; i < MAX_RTP_PT; ++i) { 01494 rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat; 01495 rtp->current_RTP_PT[i].code = static_RTP_PT[i].code; 01496 } 01497 01498 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01499 rtp->rtp_lookup_code_cache_code = 0; 01500 rtp->rtp_lookup_code_cache_result = 0; 01501 01502 ast_mutex_unlock(&rtp->bridge_lock); 01503 }
Definition at line 1155 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().
01156 { 01157 int res; 01158 struct sockaddr_in sin; 01159 socklen_t len; 01160 unsigned int seqno; 01161 int version; 01162 int payloadtype; 01163 int hdrlen = 12; 01164 int padding; 01165 int mark; 01166 int ext; 01167 int cc; 01168 unsigned int ssrc; 01169 unsigned int timestamp; 01170 unsigned int *rtpheader; 01171 struct rtpPayloadType rtpPT; 01172 struct ast_rtp *bridged = NULL; 01173 01174 /* If time is up, kill it */ 01175 if (rtp->sending_digit) 01176 ast_rtp_senddigit_continuation(rtp); 01177 01178 len = sizeof(sin); 01179 01180 /* Cache where the header will go */ 01181 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 01182 0, (struct sockaddr *)&sin, &len); 01183 if (option_debug > 3) 01184 ast_log(LOG_DEBUG, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 01185 01186 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); 01187 if (res < 0) { 01188 ast_assert(errno != EBADF); 01189 if (errno != EAGAIN) { 01190 ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno)); 01191 ast_log(LOG_WARNING, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s); 01192 return NULL; 01193 } 01194 return &ast_null_frame; 01195 } 01196 01197 if (res < hdrlen) { 01198 ast_log(LOG_WARNING, "RTP Read too short\n"); 01199 return &ast_null_frame; 01200 } 01201 01202 /* Get fields */ 01203 seqno = ntohl(rtpheader[0]); 01204 01205 /* Check RTP version */ 01206 version = (seqno & 0xC0000000) >> 30; 01207 if (!version) { 01208 if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) && 01209 (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) { 01210 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 01211 } 01212 return &ast_null_frame; 01213 } 01214 01215 #if 0 /* Allow to receive RTP stream with closed transmission path */ 01216 /* If we don't have the other side's address, then ignore this */ 01217 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 01218 return &ast_null_frame; 01219 #endif 01220 01221 /* Send to whoever send to us if NAT is turned on */ 01222 if (rtp->nat) { 01223 if (((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 01224 (rtp->them.sin_port != sin.sin_port)) && 01225 ((rtp->altthem.sin_addr.s_addr != sin.sin_addr.s_addr) || 01226 (rtp->altthem.sin_port != sin.sin_port))) { 01227 rtp->them = sin; 01228 if (rtp->rtcp) { 01229 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 01230 rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1); 01231 } 01232 rtp->rxseqno = 0; 01233 ast_set_flag(rtp, FLAG_NAT_ACTIVE); 01234 if (option_debug || rtpdebug) 01235 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)); 01236 } 01237 } 01238 01239 /* If we are bridged to another RTP stream, send direct */ 01240 if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen)) 01241 return &ast_null_frame; 01242 01243 if (version != 2) 01244 return &ast_null_frame; 01245 01246 payloadtype = (seqno & 0x7f0000) >> 16; 01247 padding = seqno & (1 << 29); 01248 mark = seqno & (1 << 23); 01249 ext = seqno & (1 << 28); 01250 cc = (seqno & 0xF000000) >> 24; 01251 seqno &= 0xffff; 01252 timestamp = ntohl(rtpheader[1]); 01253 ssrc = ntohl(rtpheader[2]); 01254 01255 if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) { 01256 if (option_debug || rtpdebug) 01257 ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n"); 01258 mark = 1; 01259 } 01260 01261 rtp->rxssrc = ssrc; 01262 01263 if (padding) { 01264 /* Remove padding bytes */ 01265 res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; 01266 } 01267 01268 if (cc) { 01269 /* CSRC fields present */ 01270 hdrlen += cc*4; 01271 } 01272 01273 if (ext) { 01274 /* RTP Extension present */ 01275 hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2; 01276 hdrlen += 4; 01277 } 01278 01279 if (res < hdrlen) { 01280 ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); 01281 return &ast_null_frame; 01282 } 01283 01284 rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */ 01285 01286 if (rtp->rxcount==1) { 01287 /* This is the first RTP packet successfully received from source */ 01288 rtp->seedrxseqno = seqno; 01289 } 01290 01291 /* Do not schedule RR if RTCP isn't run */ 01292 if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) { 01293 /* Schedule transmission of Receiver Report */ 01294 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); 01295 } 01296 if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */ 01297 rtp->cycles += RTP_SEQ_MOD; 01298 01299 rtp->lastrxseqno = seqno; 01300 01301 if (rtp->themssrc==0) 01302 rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */ 01303 01304 if (rtp_debug_test_addr(&sin)) 01305 ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 01306 ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); 01307 01308 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); 01309 if (!rtpPT.isAstFormat) { 01310 struct ast_frame *f = NULL; 01311 01312 /* This is special in-band data that's not one of our codecs */ 01313 if (rtpPT.code == AST_RTP_DTMF) { 01314 /* It's special -- rfc2833 process it */ 01315 if (rtp_debug_test_addr(&sin)) { 01316 unsigned char *data; 01317 unsigned int event; 01318 unsigned int event_end; 01319 unsigned int duration; 01320 data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen; 01321 event = ntohl(*((unsigned int *)(data))); 01322 event >>= 24; 01323 event_end = ntohl(*((unsigned int *)(data))); 01324 event_end <<= 8; 01325 event_end >>= 24; 01326 duration = ntohl(*((unsigned int *)(data))); 01327 duration &= 0xFFFF; 01328 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); 01329 } 01330 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp); 01331 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { 01332 /* It's really special -- process it the Cisco way */ 01333 if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) { 01334 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01335 rtp->lastevent = seqno; 01336 } 01337 } else if (rtpPT.code == AST_RTP_CN) { 01338 /* Comfort Noise */ 01339 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01340 } else { 01341 ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr)); 01342 } 01343 return f ? f : &ast_null_frame; 01344 } 01345 rtp->lastrxformat = rtp->f.subclass = rtpPT.code; 01346 rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO; 01347 01348 rtp->rxseqno = seqno; 01349 01350 if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) { 01351 rtp->dtmf_timeout = 0; 01352 01353 if (rtp->resp) { 01354 struct ast_frame *f; 01355 f = send_dtmf(rtp, AST_FRAME_DTMF_END); 01356 f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0)); 01357 rtp->resp = 0; 01358 rtp->dtmf_timeout = rtp->dtmf_duration = 0; 01359 return f; 01360 } 01361 } 01362 01363 /* Record received timestamp as last received now */ 01364 rtp->lastrxts = timestamp; 01365 01366 rtp->f.mallocd = 0; 01367 rtp->f.datalen = res - hdrlen; 01368 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; 01369 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; 01370 rtp->f.seqno = seqno; 01371 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { 01372 rtp->f.samples = ast_codec_get_samples(&rtp->f); 01373 if (rtp->f.subclass == AST_FORMAT_SLINEAR) 01374 ast_frame_byteswap_be(&rtp->f); 01375 calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark); 01376 /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */ 01377 ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO); 01378 rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass) / 1000); 01379 rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000); 01380 } else { 01381 /* Video -- samples is # of samples vs. 90000 */ 01382 if (!rtp->lastividtimestamp) 01383 rtp->lastividtimestamp = timestamp; 01384 rtp->f.samples = timestamp - rtp->lastividtimestamp; 01385 rtp->lastividtimestamp = timestamp; 01386 rtp->f.delivery.tv_sec = 0; 01387 rtp->f.delivery.tv_usec = 0; 01388 if (mark) 01389 rtp->f.subclass |= 0x1; 01390 } 01391 rtp->f.src = "RTP"; 01392 return &rtp->f; 01393 }
int ast_rtp_reload | ( | void | ) |
Definition at line 3875 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().
03876 { 03877 struct ast_config *cfg; 03878 const char *s; 03879 03880 rtpstart = 5000; 03881 rtpend = 31000; 03882 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03883 cfg = ast_config_load("rtp.conf"); 03884 if (cfg) { 03885 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { 03886 rtpstart = atoi(s); 03887 if (rtpstart < 1024) 03888 rtpstart = 1024; 03889 if (rtpstart > 65535) 03890 rtpstart = 65535; 03891 } 03892 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { 03893 rtpend = atoi(s); 03894 if (rtpend < 1024) 03895 rtpend = 1024; 03896 if (rtpend > 65535) 03897 rtpend = 65535; 03898 } 03899 if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) { 03900 rtcpinterval = atoi(s); 03901 if (rtcpinterval == 0) 03902 rtcpinterval = 0; /* Just so we're clear... it's zero */ 03903 if (rtcpinterval < RTCP_MIN_INTERVALMS) 03904 rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */ 03905 if (rtcpinterval > RTCP_MAX_INTERVALMS) 03906 rtcpinterval = RTCP_MAX_INTERVALMS; 03907 } 03908 if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) { 03909 #ifdef SO_NO_CHECK 03910 if (ast_false(s)) 03911 nochecksums = 1; 03912 else 03913 nochecksums = 0; 03914 #else 03915 if (ast_false(s)) 03916 ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n"); 03917 #endif 03918 } 03919 if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { 03920 dtmftimeout = atoi(s); 03921 if ((dtmftimeout < 0) || (dtmftimeout > 64000)) { 03922 ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", 03923 dtmftimeout, DEFAULT_DTMF_TIMEOUT); 03924 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03925 }; 03926 } 03927 ast_config_destroy(cfg); 03928 } 03929 if (rtpstart >= rtpend) { 03930 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n"); 03931 rtpstart = 5000; 03932 rtpend = 31000; 03933 } 03934 if (option_verbose > 1) 03935 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); 03936 return 0; 03937 }
void ast_rtp_reset | ( | struct ast_rtp * | rtp | ) |
Definition at line 2146 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.
02147 { 02148 memset(&rtp->rxcore, 0, sizeof(rtp->rxcore)); 02149 memset(&rtp->txcore, 0, sizeof(rtp->txcore)); 02150 memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute)); 02151 rtp->lastts = 0; 02152 rtp->lastdigitts = 0; 02153 rtp->lastrxts = 0; 02154 rtp->lastividtimestamp = 0; 02155 rtp->lastovidtimestamp = 0; 02156 rtp->lasteventseqn = 0; 02157 rtp->lastevent = 0; 02158 rtp->lasttxformat = 0; 02159 rtp->lastrxformat = 0; 02160 rtp->dtmf_timeout = 0; 02161 rtp->seqno = 0; 02162 rtp->rxseqno = 0; 02163 }
int ast_rtp_sendcng | ( | struct ast_rtp * | rtp, | |
int | level | |||
) |
generate comfort noice (CNG)
Definition at line 2660 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().
02661 { 02662 unsigned int *rtpheader; 02663 int hdrlen = 12; 02664 int res; 02665 int payload; 02666 char data[256]; 02667 level = 127 - (level & 0x7f); 02668 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN); 02669 02670 /* If we have no peer, return immediately */ 02671 if (!rtp->them.sin_addr.s_addr) 02672 return 0; 02673 02674 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02675 02676 /* Get a pointer to the header */ 02677 rtpheader = (unsigned int *)data; 02678 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++)); 02679 rtpheader[1] = htonl(rtp->lastts); 02680 rtpheader[2] = htonl(rtp->ssrc); 02681 data[12] = level; 02682 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { 02683 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); 02684 if (res <0) 02685 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)); 02686 if (rtp_debug_test_addr(&rtp->them)) 02687 ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n" 02688 , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); 02689 02690 } 02691 return 0; 02692 }
int ast_rtp_senddigit_begin | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
Send begin frames for DTMF.
Definition at line 2268 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().
02269 { 02270 unsigned int *rtpheader; 02271 int hdrlen = 12, res = 0, i = 0, payload = 0; 02272 char data[256]; 02273 02274 if ((digit <= '9') && (digit >= '0')) 02275 digit -= '0'; 02276 else if (digit == '*') 02277 digit = 10; 02278 else if (digit == '#') 02279 digit = 11; 02280 else if ((digit >= 'A') && (digit <= 'D')) 02281 digit = digit - 'A' + 12; 02282 else if ((digit >= 'a') && (digit <= 'd')) 02283 digit = digit - 'a' + 12; 02284 else { 02285 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); 02286 return 0; 02287 } 02288 02289 /* If we have no peer, return immediately */ 02290 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 02291 return 0; 02292 02293 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); 02294 02295 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02296 rtp->send_duration = 160; 02297 rtp->lastdigitts = rtp->lastts + rtp->send_duration; 02298 02299 /* Get a pointer to the header */ 02300 rtpheader = (unsigned int *)data; 02301 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno)); 02302 rtpheader[1] = htonl(rtp->lastdigitts); 02303 rtpheader[2] = htonl(rtp->ssrc); 02304 02305 for (i = 0; i < 2; i++) { 02306 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration)); 02307 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); 02308 if (res < 0) 02309 ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n", 02310 ast_inet_ntoa(rtp->them.sin_addr), 02311 ntohs(rtp->them.sin_port), strerror(errno)); 02312 if (rtp_debug_test_addr(&rtp->them)) 02313 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 02314 ast_inet_ntoa(rtp->them.sin_addr), 02315 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 02316 /* Increment sequence number */ 02317 rtp->seqno++; 02318 /* Increment duration */ 02319 rtp->send_duration += 160; 02320 /* Clear marker bit and set seqno */ 02321 rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); 02322 } 02323 02324 /* Since we received a begin, we can safely store the digit and disable any compensation */ 02325 rtp->sending_digit = 1; 02326 rtp->send_digit = digit; 02327 rtp->send_payload = payload; 02328 02329 return 0; 02330 }
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 2091 of file rtp.c.
References ast_rtcp::altthem, ast_rtp::altthem, and ast_rtp::rtcp.
Referenced by handle_request_invite().
02092 { 02093 rtp->altthem.sin_port = alt->sin_port; 02094 rtp->altthem.sin_addr = alt->sin_addr; 02095 if (rtp->rtcp) { 02096 rtp->rtcp->altthem.sin_port = htons(ntohs(alt->sin_port) + 1); 02097 rtp->rtcp->altthem.sin_addr = alt->sin_addr; 02098 } 02099 }
void ast_rtp_set_callback | ( | struct ast_rtp * | rtp, | |
ast_rtp_callback | callback | |||
) |
Definition at line 593 of file rtp.c.
References ast_rtp::callback.
Referenced by start_rtp().
00594 { 00595 rtp->callback = callback; 00596 }
void ast_rtp_set_data | ( | struct ast_rtp * | rtp, | |
void * | data | |||
) |
Definition at line 588 of file rtp.c.
References ast_rtp::data.
Referenced by start_rtp().
00589 { 00590 rtp->data = data; 00591 }
void ast_rtp_set_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Activate payload type.
Definition at line 1703 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().
01704 { 01705 if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 01706 return; /* bogus payload type */ 01707 01708 ast_mutex_lock(&rtp->bridge_lock); 01709 rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; 01710 ast_mutex_unlock(&rtp->bridge_lock); 01711 }
void ast_rtp_set_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2080 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().
02081 { 02082 rtp->them.sin_port = them->sin_port; 02083 rtp->them.sin_addr = them->sin_addr; 02084 if (rtp->rtcp) { 02085 rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1); 02086 rtp->rtcp->them.sin_addr = them->sin_addr; 02087 } 02088 rtp->rxseqno = 0; 02089 }
void ast_rtp_set_rtpholdtimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp hold timeout.
Definition at line 555 of file rtp.c.
References ast_rtp::rtpholdtimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00556 { 00557 rtp->rtpholdtimeout = timeout; 00558 }
void ast_rtp_set_rtpkeepalive | ( | struct ast_rtp * | rtp, | |
int | period | |||
) |
set RTP keepalive interval
Definition at line 561 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by create_addr_from_peer(), and sip_alloc().
00562 { 00563 rtp->rtpkeepalive = period; 00564 }
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 1730 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().
01733 { 01734 unsigned int i; 01735 int found = 0; 01736 01737 if (pt < 0 || pt > MAX_RTP_PT) 01738 return -1; /* bogus payload type */ 01739 01740 ast_mutex_lock(&rtp->bridge_lock); 01741 01742 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01743 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && 01744 strcasecmp(mimeType, mimeTypes[i].type) == 0) { 01745 found = 1; 01746 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; 01747 if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) && 01748 mimeTypes[i].payloadType.isAstFormat && 01749 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01750 rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2; 01751 break; 01752 } 01753 } 01754 01755 ast_mutex_unlock(&rtp->bridge_lock); 01756 01757 return (found ? 0 : -1); 01758 }
void ast_rtp_set_rtptimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp timeout.
Definition at line 549 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00550 { 00551 rtp->rtptimeout = timeout; 00552 }
void ast_rtp_set_rtptimers_onhold | ( | struct ast_rtp * | rtp | ) |
Definition at line 542 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by handle_response_invite().
00543 { 00544 rtp->rtptimeout = (-1) * rtp->rtptimeout; 00545 rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout; 00546 }
void ast_rtp_setdtmf | ( | struct ast_rtp * | rtp, | |
int | dtmf | |||
) |
Indicate whether this RTP session is carrying DTMF or not.
Definition at line 608 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().
00609 { 00610 ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF); 00611 }
void ast_rtp_setdtmfcompensate | ( | struct ast_rtp * | rtp, | |
int | compensate | |||
) |
Compensate for devices that send RFC2833 packets all at once.
Definition at line 613 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().
00614 { 00615 ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE); 00616 }
void ast_rtp_setnat | ( | struct ast_rtp * | rtp, | |
int | nat | |||
) |
Definition at line 598 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 618 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_STUN.
Referenced by gtalk_new().
00619 { 00620 ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN); 00621 }
int ast_rtp_settos | ( | struct ast_rtp * | rtp, | |
int | tos | |||
) |
Definition at line 2063 of file rtp.c.
References ast_log(), LOG_WARNING, and ast_rtp::s.
Referenced by __oh323_rtp_create(), and sip_alloc().
02064 { 02065 int res; 02066 02067 if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 02068 ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); 02069 return res; 02070 }
void ast_rtp_stop | ( | struct ast_rtp * | rtp | ) |
Definition at line 2130 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().
02131 { 02132 if (rtp->rtcp) { 02133 AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); 02134 } 02135 02136 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 02137 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 02138 if (rtp->rtcp) { 02139 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr)); 02140 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port)); 02141 } 02142 02143 ast_clear_flag(rtp, FLAG_P2P_SENT_MARK); 02144 }
void ast_rtp_stun_request | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | suggestion, | |||
const char * | username | |||
) |
Definition at line 405 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().
00406 { 00407 struct stun_header *req; 00408 unsigned char reqdata[1024]; 00409 int reqlen, reqleft; 00410 struct stun_attr *attr; 00411 00412 req = (struct stun_header *)reqdata; 00413 stun_req_id(req); 00414 reqlen = 0; 00415 reqleft = sizeof(reqdata) - sizeof(struct stun_header); 00416 req->msgtype = 0; 00417 req->msglen = 0; 00418 attr = (struct stun_attr *)req->ies; 00419 if (username) 00420 append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft); 00421 req->msglen = htons(reqlen); 00422 req->msgtype = htons(STUN_BINDREQ); 00423 stun_send(rtp->s, suggestion, req); 00424 }
void ast_rtp_unset_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
clear payload type
Definition at line 1715 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().
01716 { 01717 if (pt < 0 || pt > MAX_RTP_PT) 01718 return; /* bogus payload type */ 01719 01720 ast_mutex_lock(&rtp->bridge_lock); 01721 rtp->current_RTP_PT[pt].isAstFormat = 0; 01722 rtp->current_RTP_PT[pt].code = 0; 01723 ast_mutex_unlock(&rtp->bridge_lock); 01724 }
Definition at line 2865 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().
02866 { 02867 struct ast_frame *f; 02868 int codec; 02869 int hdrlen = 12; 02870 int subclass; 02871 02872 02873 /* If we have no peer, return immediately */ 02874 if (!rtp->them.sin_addr.s_addr) 02875 return 0; 02876 02877 /* If there is no data length, return immediately */ 02878 if (!_f->datalen) 02879 return 0; 02880 02881 /* Make sure we have enough space for RTP header */ 02882 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { 02883 ast_log(LOG_WARNING, "RTP can only send voice and video\n"); 02884 return -1; 02885 } 02886 02887 subclass = _f->subclass; 02888 if (_f->frametype == AST_FRAME_VIDEO) 02889 subclass &= ~0x1; 02890 02891 codec = ast_rtp_lookup_code(rtp, 1, subclass); 02892 if (codec < 0) { 02893 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); 02894 return -1; 02895 } 02896 02897 if (rtp->lasttxformat != subclass) { 02898 /* New format, reset the smoother */ 02899 if (option_debug) 02900 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); 02901 rtp->lasttxformat = subclass; 02902 if (rtp->smoother) 02903 ast_smoother_free(rtp->smoother); 02904 rtp->smoother = NULL; 02905 } 02906 02907 if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) { 02908 struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass); 02909 if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */ 02910 if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) { 02911 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)); 02912 return -1; 02913 } 02914 if (fmt.flags) 02915 ast_smoother_set_flags(rtp->smoother, fmt.flags); 02916 if (option_debug) 02917 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)); 02918 } 02919 } 02920 if (rtp->smoother) { 02921 if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) { 02922 ast_smoother_feed_be(rtp->smoother, _f); 02923 } else { 02924 ast_smoother_feed(rtp->smoother, _f); 02925 } 02926 02927 while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) { 02928 ast_rtp_raw_write(rtp, f, codec); 02929 } 02930 } else { 02931 /* Don't buffer outgoing frames; send them one-per-packet: */ 02932 if (_f->offset < hdrlen) { 02933 f = ast_frdup(_f); 02934 } else { 02935 f = _f; 02936 } 02937 if (f->data) { 02938 ast_rtp_raw_write(rtp, f, codec); 02939 } 02940 if (f != _f) 02941 ast_frfree(f); 02942 } 02943 02944 return 0; 02945 }