#include <netinet/in.h>
#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
#include "asterisk/linkedlists.h"
Include dependency graph for rtp.h:
This graph shows which files directly or indirectly include this file:
Go to the source code of this file.
Data Structures | |
struct | ast_rtp_protocol |
struct | ast_rtp_quality |
Defines | |
#define | AST_RTP_CISCO_DTMF (1 << 2) |
#define | AST_RTP_CN (1 << 1) |
#define | AST_RTP_DTMF (1 << 0) |
#define | AST_RTP_MAX AST_RTP_CISCO_DTMF |
#define | FLAG_3389_WARNING (1 << 0) |
#define | MAX_RTP_PT 256 |
Typedefs | |
typedef int(*) | ast_rtp_callback (struct ast_rtp *rtp, struct ast_frame *f, void *data) |
Enumerations | |
enum | ast_rtp_get_result { AST_RTP_GET_FAILED = 0, AST_RTP_TRY_PARTIAL, AST_RTP_TRY_NATIVE } |
enum | ast_rtp_options { AST_RTP_OPT_G726_NONSTANDARD = (1 << 0) } |
Functions | |
int | ast_rtcp_fd (struct ast_rtp *rtp) |
ast_frame * | ast_rtcp_read (struct ast_rtp *rtp) |
int | ast_rtcp_send_h261fur (void *data) |
Send an H.261 fast update request. Some devices need this rather than the XML message in SIP. | |
size_t | ast_rtp_alloc_size (void) |
Get the amount of space required to hold an RTP session. | |
int | ast_rtp_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms) |
Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk. | |
int | ast_rtp_codec_getformat (int pt) |
ast_codec_pref * | ast_rtp_codec_getpref (struct ast_rtp *rtp) |
int | ast_rtp_codec_setpref (struct ast_rtp *rtp, struct ast_codec_pref *prefs) |
void | ast_rtp_destroy (struct ast_rtp *rtp) |
int | ast_rtp_early_bridge (struct ast_channel *dest, struct ast_channel *src) |
If possible, create an early bridge directly between the devices without having to send a re-invite later. | |
int | ast_rtp_fd (struct ast_rtp *rtp) |
ast_rtp * | ast_rtp_get_bridged (struct ast_rtp *rtp) |
void | ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats) |
Return the union of all of the codecs that were set by rtp_set...() calls They're returned as two distinct sets: AST_FORMATs, and AST_RTPs. | |
int | ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them) |
char * | ast_rtp_get_quality (struct ast_rtp *rtp, struct ast_rtp_quality *qual) |
Return RTCP quality string. | |
int | ast_rtp_get_rtpholdtimeout (struct ast_rtp *rtp) |
Get rtp hold timeout. | |
int | ast_rtp_get_rtpkeepalive (struct ast_rtp *rtp) |
Get RTP keepalive interval. | |
int | ast_rtp_get_rtptimeout (struct ast_rtp *rtp) |
Get rtp timeout. | |
void | ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us) |
int | ast_rtp_getnat (struct ast_rtp *rtp) |
void | ast_rtp_init (void) |
Initialize the RTP system in Asterisk. | |
int | ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code) |
Looks up an RTP code out of our *static* outbound list. | |
char * | ast_rtp_lookup_mime_multiple (char *buf, size_t size, const int capability, const int isAstFormat, enum ast_rtp_options options) |
Build a string of MIME subtype names from a capability list. | |
const char * | ast_rtp_lookup_mime_subtype (int isAstFormat, int code, enum ast_rtp_options options) |
Mapping an Asterisk code into a MIME subtype (string):. | |
rtpPayloadType | ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt) |
Mapping between RTP payload format codes and Asterisk codes:. | |
int | ast_rtp_make_compatible (struct ast_channel *dest, struct ast_channel *src, int media) |
ast_rtp * | ast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode) |
Initializate a RTP session. | |
void | ast_rtp_new_init (struct ast_rtp *rtp) |
Initialize a new RTP structure. | |
ast_rtp * | ast_rtp_new_with_bindaddr (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in) |
Initializate a RTP session using an in_addr structure. | |
int | ast_rtp_proto_register (struct ast_rtp_protocol *proto) |
Register interface to channel driver. | |
void | ast_rtp_proto_unregister (struct ast_rtp_protocol *proto) |
Unregister interface to channel driver. | |
void | ast_rtp_pt_clear (struct ast_rtp *rtp) |
Setting RTP payload types from lines in a SDP description:. | |
void | ast_rtp_pt_copy (struct ast_rtp *dest, struct ast_rtp *src) |
Copy payload types between RTP structures. | |
void | ast_rtp_pt_default (struct ast_rtp *rtp) |
Set payload types to defaults. | |
ast_frame * | ast_rtp_read (struct ast_rtp *rtp) |
int | ast_rtp_reload (void) |
void | ast_rtp_reset (struct ast_rtp *rtp) |
int | ast_rtp_sendcng (struct ast_rtp *rtp, int level) |
generate comfort noice (CNG) | |
int | ast_rtp_senddigit_begin (struct ast_rtp *rtp, char digit) |
Send begin frames for DTMF. | |
int | ast_rtp_senddigit_end (struct ast_rtp *rtp, char digit) |
void | ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback) |
void | ast_rtp_set_data (struct ast_rtp *rtp, void *data) |
void | ast_rtp_set_m_type (struct ast_rtp *rtp, int pt) |
Make a note of a RTP payload type that was seen in a SDP "m=" line. By default, use the well-known value for this type (although it may still be set to a different value by a subsequent "a=rtpmap:" line). | |
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 | |
void | ast_rtp_set_rtpmap_type (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype, enum ast_rtp_options options) |
Make a note of a RTP payload type (with MIME type) that was seen in an SDP "a=rtpmap:" line. | |
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) |
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(), and ast_rtp_set_rtpmap_type().
typedef int(*) ast_rtp_callback(struct ast_rtp *rtp, struct ast_frame *f, void *data) |
enum ast_rtp_get_result |
Definition at line 57 of file rtp.h.
00057 { 00058 /*! Failed to find the RTP structure */ 00059 AST_RTP_GET_FAILED = 0, 00060 /*! RTP structure exists but true native bridge can not occur so try partial */ 00061 AST_RTP_TRY_PARTIAL, 00062 /*! RTP structure exists and native bridge can occur */ 00063 AST_RTP_TRY_NATIVE, 00064 };
enum ast_rtp_options |
int ast_rtcp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 517 of file rtp.c.
References ast_rtp::rtcp, and ast_rtcp::s.
Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), sip_new(), and start_rtp().
Definition at line 822 of file rtp.c.
References ast_rtcp::accumulated_transit, AST_CONTROL_VIDUPDATE, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), CRASH, ast_frame::datalen, ast_rtp::f, f, ast_frame::frametype, len, LOG_DEBUG, LOG_WARNING, ast_frame::mallocd, ast_rtcp::maxrtt, ast_rtcp::minrtt, ast_rtp::nat, option_debug, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, rtcp_debug_test_addr(), RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, ast_rtcp::rtt, ast_rtcp::rxlsr, ast_rtcp::s, ast_frame::samples, ast_rtcp::soc, ast_rtcp::spc, ast_frame::src, ast_frame::subclass, ast_rtcp::them, ast_rtcp::themrxlsr, and timeval2ntp().
Referenced by oh323_read(), sip_rtp_read(), and skinny_rtp_read().
00823 { 00824 socklen_t len; 00825 int position, i, packetwords; 00826 int res; 00827 struct sockaddr_in sin; 00828 unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET]; 00829 unsigned int *rtcpheader; 00830 int pt; 00831 struct timeval now; 00832 unsigned int length; 00833 int rc; 00834 double rttsec; 00835 uint64_t rtt = 0; 00836 unsigned int dlsr; 00837 unsigned int lsr; 00838 unsigned int msw; 00839 unsigned int lsw; 00840 unsigned int comp; 00841 struct ast_frame *f = &ast_null_frame; 00842 00843 if (!rtp || !rtp->rtcp) 00844 return &ast_null_frame; 00845 00846 len = sizeof(sin); 00847 00848 res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET, 00849 0, (struct sockaddr *)&sin, &len); 00850 rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET); 00851 00852 if (res < 0) { 00853 if (errno == EBADF) 00854 CRASH; 00855 if (errno != EAGAIN) { 00856 ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno)); 00857 return NULL; 00858 } 00859 return &ast_null_frame; 00860 } 00861 00862 packetwords = res / 4; 00863 00864 if (rtp->nat) { 00865 /* Send to whoever sent to us */ 00866 if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 00867 (rtp->rtcp->them.sin_port != sin.sin_port)) { 00868 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 00869 if (option_debug || rtpdebug) 00870 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)); 00871 } 00872 } 00873 00874 if (option_debug) 00875 ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res); 00876 00877 /* Process a compound packet */ 00878 position = 0; 00879 while (position < packetwords) { 00880 i = position; 00881 length = ntohl(rtcpheader[i]); 00882 pt = (length & 0xff0000) >> 16; 00883 rc = (length & 0x1f000000) >> 24; 00884 length &= 0xffff; 00885 00886 if ((i + length) > packetwords) { 00887 ast_log(LOG_WARNING, "RTCP Read too short\n"); 00888 return &ast_null_frame; 00889 } 00890 00891 if (rtcp_debug_test_addr(&sin)) { 00892 ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); 00893 ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown"); 00894 ast_verbose("Reception reports: %d\n", rc); 00895 ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]); 00896 } 00897 00898 i += 2; /* Advance past header and ssrc */ 00899 00900 switch (pt) { 00901 case RTCP_PT_SR: 00902 gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */ 00903 rtp->rtcp->spc = ntohl(rtcpheader[i+3]); 00904 rtp->rtcp->soc = ntohl(rtcpheader[i + 4]); 00905 rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/ 00906 00907 if (rtcp_debug_test_addr(&sin)) { 00908 ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096); 00909 ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2])); 00910 ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4])); 00911 } 00912 i += 5; 00913 if (rc < 1) 00914 break; 00915 /* Intentional fall through */ 00916 case RTCP_PT_RR: 00917 /* Don't handle multiple reception reports (rc > 1) yet */ 00918 /* Calculate RTT per RFC */ 00919 gettimeofday(&now, NULL); 00920 timeval2ntp(now, &msw, &lsw); 00921 if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */ 00922 comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16); 00923 lsr = ntohl(rtcpheader[i + 4]); 00924 dlsr = ntohl(rtcpheader[i + 5]); 00925 rtt = comp - lsr - dlsr; 00926 00927 /* Convert end to end delay to usec (keeping the calculation in 64bit space) 00928 sess->ee_delay = (eedelay * 1000) / 65536; */ 00929 if (rtt < 4294) { 00930 rtt = (rtt * 1000000) >> 16; 00931 } else { 00932 rtt = (rtt * 1000) >> 16; 00933 rtt *= 1000; 00934 } 00935 rtt = rtt / 1000.; 00936 rttsec = rtt / 1000.; 00937 00938 if (comp - dlsr >= lsr) { 00939 rtp->rtcp->accumulated_transit += rttsec; 00940 rtp->rtcp->rtt = rttsec; 00941 if (rtp->rtcp->maxrtt<rttsec) 00942 rtp->rtcp->maxrtt = rttsec; 00943 if (rtp->rtcp->minrtt>rttsec) 00944 rtp->rtcp->minrtt = rttsec; 00945 } else if (rtcp_debug_test_addr(&sin)) { 00946 ast_verbose("Internal RTCP NTP clock skew detected: " 00947 "lsr=%u, now=%u, dlsr=%u (%d:%03dms), " 00948 "diff=%d\n", 00949 lsr, comp, dlsr, dlsr / 65536, 00950 (dlsr % 65536) * 1000 / 65536, 00951 dlsr - (comp - lsr)); 00952 } 00953 } 00954 00955 rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]); 00956 rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff; 00957 if (rtcp_debug_test_addr(&sin)) { 00958 ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24)); 00959 ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost); 00960 ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff)); 00961 ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16); 00962 ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter); 00963 ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096); 00964 ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0); 00965 if (rtt) 00966 ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt); 00967 } 00968 break; 00969 case RTCP_PT_FUR: 00970 if (rtcp_debug_test_addr(&sin)) 00971 ast_verbose("Received an RTCP Fast Update Request\n"); 00972 rtp->f.frametype = AST_FRAME_CONTROL; 00973 rtp->f.subclass = AST_CONTROL_VIDUPDATE; 00974 rtp->f.datalen = 0; 00975 rtp->f.samples = 0; 00976 rtp->f.mallocd = 0; 00977 rtp->f.src = "RTP"; 00978 f = &rtp->f; 00979 break; 00980 case RTCP_PT_SDES: 00981 if (rtcp_debug_test_addr(&sin)) 00982 ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00983 break; 00984 case RTCP_PT_BYE: 00985 if (rtcp_debug_test_addr(&sin)) 00986 ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); 00987 break; 00988 default: 00989 if (option_debug) 00990 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)); 00991 break; 00992 } 00993 position += (length + 1); 00994 } 00995 00996 return f; 00997 }
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 2314 of file rtp.c.
References ast_rtcp_write(), ast_rtp::rtcp, and ast_rtcp::sendfur.
02315 { 02316 struct ast_rtp *rtp = data; 02317 int res; 02318 02319 rtp->rtcp->sendfur = 1; 02320 res = ast_rtcp_write(data); 02321 02322 return res; 02323 }
size_t ast_rtp_alloc_size | ( | void | ) |
Get the amount of space required to hold an RTP session.
Definition at line 397 of file rtp.c.
Referenced by process_sdp().
00398 { 00399 return sizeof(struct ast_rtp); 00400 }
int ast_rtp_bridge | ( | struct ast_channel * | c0, | |
struct ast_channel * | c1, | |||
int | flags, | |||
struct ast_frame ** | fo, | |||
struct ast_channel ** | rc, | |||
int | timeoutms | |||
) |
Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk.
Definition at line 3222 of file rtp.c.
References AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_codec_pref_getsize(), ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_set_flag, ast_test_flag, ast_verbose(), bridge_native_loop(), bridge_p2p_loop(), ast_format_list::cur_ms, FLAG_DTMF_COMPENSATE, FLAG_HAS_DTMF, FLAG_P2P_NEED_DTMF, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, option_debug, option_verbose, ast_rtp::pref, ast_channel::rawreadformat, ast_channel::rawwriteformat, ast_channel_tech::send_digit_begin, ast_channel::tech, ast_channel::tech_pvt, and VERBOSE_PREFIX_3.
03223 { 03224 struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ 03225 struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */ 03226 struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL; 03227 enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED; 03228 enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED; 03229 enum ast_bridge_result res = AST_BRIDGE_FAILED; 03230 int codec0 = 0, codec1 = 0; 03231 void *pvt0 = NULL, *pvt1 = NULL; 03232 03233 /* Lock channels */ 03234 ast_channel_lock(c0); 03235 while(ast_channel_trylock(c1)) { 03236 ast_channel_unlock(c0); 03237 usleep(1); 03238 ast_channel_lock(c0); 03239 } 03240 03241 /* Find channel driver interfaces */ 03242 if (!(pr0 = get_proto(c0))) { 03243 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); 03244 ast_channel_unlock(c0); 03245 ast_channel_unlock(c1); 03246 return AST_BRIDGE_FAILED; 03247 } 03248 if (!(pr1 = get_proto(c1))) { 03249 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); 03250 ast_channel_unlock(c0); 03251 ast_channel_unlock(c1); 03252 return AST_BRIDGE_FAILED; 03253 } 03254 03255 /* Get channel specific interface structures */ 03256 pvt0 = c0->tech_pvt; 03257 pvt1 = c1->tech_pvt; 03258 03259 /* Get audio and video interface (if native bridge is possible) */ 03260 audio_p0_res = pr0->get_rtp_info(c0, &p0); 03261 video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED; 03262 audio_p1_res = pr1->get_rtp_info(c1, &p1); 03263 video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED; 03264 03265 /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */ 03266 if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE)) 03267 audio_p0_res = AST_RTP_GET_FAILED; 03268 if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE)) 03269 audio_p1_res = AST_RTP_GET_FAILED; 03270 03271 /* Check if a bridge is possible (partial/native) */ 03272 if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) { 03273 /* Somebody doesn't want to play... */ 03274 ast_channel_unlock(c0); 03275 ast_channel_unlock(c1); 03276 return AST_BRIDGE_FAILED_NOWARN; 03277 } 03278 03279 /* If we need to feed DTMF frames into the core then only do a partial native bridge */ 03280 if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) { 03281 ast_set_flag(p0, FLAG_P2P_NEED_DTMF); 03282 audio_p0_res = AST_RTP_TRY_PARTIAL; 03283 } 03284 03285 if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) { 03286 ast_set_flag(p1, FLAG_P2P_NEED_DTMF); 03287 audio_p1_res = AST_RTP_TRY_PARTIAL; 03288 } 03289 03290 /* If both sides are not using the same method of DTMF transmission 03291 * (ie: one is RFC2833, other is INFO... then we can not do direct media. 03292 * -------------------------------------------------- 03293 * | DTMF Mode | HAS_DTMF | Accepts Begin Frames | 03294 * |-----------|------------|-----------------------| 03295 * | Inband | False | True | 03296 * | RFC2833 | True | True | 03297 * | SIP INFO | False | False | 03298 * -------------------------------------------------- 03299 * However, if DTMF from both channels is being monitored by the core, then 03300 * we can still do packet-to-packet bridging, because passing through the 03301 * core will handle DTMF mode translation. 03302 */ 03303 if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) || 03304 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) { 03305 if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) { 03306 ast_channel_unlock(c0); 03307 ast_channel_unlock(c1); 03308 return AST_BRIDGE_FAILED_NOWARN; 03309 } 03310 audio_p0_res = AST_RTP_TRY_PARTIAL; 03311 audio_p1_res = AST_RTP_TRY_PARTIAL; 03312 } 03313 03314 /* If the core will need to compensate and the P2P bridge will need to feed up DTMF frames then we can not reliably do so yet, so do not P2P bridge */ 03315 if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF) && ast_test_flag(p0, FLAG_DTMF_COMPENSATE)) || 03316 (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF) && ast_test_flag(p1, FLAG_DTMF_COMPENSATE))) { 03317 ast_channel_unlock(c0); 03318 ast_channel_unlock(c1); 03319 return AST_BRIDGE_FAILED_NOWARN; 03320 } 03321 03322 /* Get codecs from both sides */ 03323 codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0; 03324 codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0; 03325 if (codec0 && codec1 && !(codec0 & codec1)) { 03326 /* Hey, we can't do native bridging if both parties speak different codecs */ 03327 if (option_debug) 03328 ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); 03329 ast_channel_unlock(c0); 03330 ast_channel_unlock(c1); 03331 return AST_BRIDGE_FAILED_NOWARN; 03332 } 03333 03334 /* If either side can only do a partial bridge, then don't try for a true native bridge */ 03335 if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) { 03336 struct ast_format_list fmt0, fmt1; 03337 03338 /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */ 03339 if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) { 03340 if (option_debug) 03341 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n"); 03342 ast_channel_unlock(c0); 03343 ast_channel_unlock(c1); 03344 return AST_BRIDGE_FAILED_NOWARN; 03345 } 03346 /* They must also be using the same packetization */ 03347 fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat); 03348 fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat); 03349 if (fmt0.cur_ms != fmt1.cur_ms) { 03350 if (option_debug) 03351 ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n"); 03352 ast_channel_unlock(c0); 03353 ast_channel_unlock(c1); 03354 return AST_BRIDGE_FAILED_NOWARN; 03355 } 03356 03357 if (option_verbose > 2) 03358 ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name); 03359 res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1); 03360 } else { 03361 if (option_verbose > 2) 03362 ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name); 03363 res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1); 03364 } 03365 03366 return res; 03367 }
int ast_rtp_codec_getformat | ( | int | pt | ) |
Definition at line 2697 of file rtp.c.
References rtpPayloadType::code, and static_RTP_PT.
Referenced by process_sdp().
02698 { 02699 if (pt < 0 || pt > MAX_RTP_PT) 02700 return 0; /* bogus payload type */ 02701 02702 if (static_RTP_PT[pt].isAstFormat) 02703 return static_RTP_PT[pt].code; 02704 else 02705 return 0; 02706 }
struct ast_codec_pref* ast_rtp_codec_getpref | ( | struct ast_rtp * | rtp | ) |
Definition at line 2692 of file rtp.c.
References ast_rtp::pref.
Referenced by add_codec_to_sdp(), and process_sdp().
02693 { 02694 return &rtp->pref; 02695 }
int ast_rtp_codec_setpref | ( | struct ast_rtp * | rtp, | |
struct ast_codec_pref * | prefs | |||
) |
Definition at line 2679 of file rtp.c.
References ast_smoother_free(), ast_codec_pref::framing, ast_codec_pref::order, ast_rtp::pref, prefs, and ast_rtp::smoother.
Referenced by __oh323_rtp_create(), check_user_full(), create_addr_from_peer(), process_sdp(), register_verify(), set_peer_capabilities(), start_rtp(), and transmit_response_with_sdp().
02680 { 02681 int x; 02682 for (x = 0; x < 32; x++) { /* Ugly way */ 02683 rtp->pref.order[x] = prefs->order[x]; 02684 rtp->pref.framing[x] = prefs->framing[x]; 02685 } 02686 if (rtp->smoother) 02687 ast_smoother_free(rtp->smoother); 02688 rtp->smoother = NULL; 02689 return 0; 02690 }
void ast_rtp_destroy | ( | struct ast_rtp * | rtp | ) |
Definition at line 2096 of file rtp.c.
References ast_io_remove(), ast_mutex_destroy(), ast_sched_del(), ast_smoother_free(), ast_verbose(), ast_rtp::bridge_lock, ast_rtcp::expected_prior, free, ast_rtp::io, ast_rtp::ioid, ast_rtcp::received_prior, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtcp::rr_count, ast_rtp::rtcp, rtcp_debug_test_addr(), ast_rtcp::rtt, ast_rtp::rxcount, ast_rtp::rxjitter, ast_rtp::rxtransit, ast_rtcp::s, ast_rtp::s, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::smoother, ast_rtcp::sr_count, ast_rtp::ssrc, ast_rtp::them, ast_rtp::themssrc, and ast_rtp::txcount.
Referenced by __oh323_destroy(), __sip_destroy(), check_user_full(), cleanup_connection(), create_addr_from_peer(), destroy_endpoint(), gtalk_free_pvt(), mgcp_hangup(), oh323_alloc(), skinny_hangup(), start_rtp(), and unalloc_sub().
02097 { 02098 if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) { 02099 /*Print some info on the call here */ 02100 ast_verbose(" RTP-stats\n"); 02101 ast_verbose("* Our Receiver:\n"); 02102 ast_verbose(" SSRC: %u\n", rtp->themssrc); 02103 ast_verbose(" Received packets: %u\n", rtp->rxcount); 02104 ast_verbose(" Lost packets: %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior); 02105 ast_verbose(" Jitter: %.4f\n", rtp->rxjitter); 02106 ast_verbose(" Transit: %.4f\n", rtp->rxtransit); 02107 ast_verbose(" RR-count: %u\n", rtp->rtcp->rr_count); 02108 ast_verbose("* Our Sender:\n"); 02109 ast_verbose(" SSRC: %u\n", rtp->ssrc); 02110 ast_verbose(" Sent packets: %u\n", rtp->txcount); 02111 ast_verbose(" Lost packets: %u\n", rtp->rtcp->reported_lost); 02112 ast_verbose(" Jitter: %u\n", rtp->rtcp->reported_jitter); 02113 ast_verbose(" SR-count: %u\n", rtp->rtcp->sr_count); 02114 ast_verbose(" RTT: %f\n", rtp->rtcp->rtt); 02115 } 02116 02117 if (rtp->smoother) 02118 ast_smoother_free(rtp->smoother); 02119 if (rtp->ioid) 02120 ast_io_remove(rtp->io, rtp->ioid); 02121 if (rtp->s > -1) 02122 close(rtp->s); 02123 if (rtp->rtcp) { 02124 if (rtp->rtcp->schedid > 0) 02125 ast_sched_del(rtp->sched, rtp->rtcp->schedid); 02126 close(rtp->rtcp->s); 02127 free(rtp->rtcp); 02128 rtp->rtcp=NULL; 02129 } 02130 02131 ast_mutex_destroy(&rtp->bridge_lock); 02132 02133 free(rtp); 02134 }
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 1482 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, option_debug, and ast_rtp_protocol::set_rtp_peer.
Referenced by wait_for_answer().
01483 { 01484 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01485 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01486 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01487 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01488 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01489 int srccodec, destcodec, nat_active = 0; 01490 01491 /* Lock channels */ 01492 ast_channel_lock(dest); 01493 if (src) { 01494 while(ast_channel_trylock(src)) { 01495 ast_channel_unlock(dest); 01496 usleep(1); 01497 ast_channel_lock(dest); 01498 } 01499 } 01500 01501 /* Find channel driver interfaces */ 01502 destpr = get_proto(dest); 01503 if (src) 01504 srcpr = get_proto(src); 01505 if (!destpr) { 01506 if (option_debug) 01507 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01508 ast_channel_unlock(dest); 01509 if (src) 01510 ast_channel_unlock(src); 01511 return 0; 01512 } 01513 if (!srcpr) { 01514 if (option_debug) 01515 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>"); 01516 ast_channel_unlock(dest); 01517 if (src) 01518 ast_channel_unlock(src); 01519 return 0; 01520 } 01521 01522 /* Get audio and video interface (if native bridge is possible) */ 01523 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01524 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01525 if (srcpr) { 01526 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01527 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01528 } 01529 01530 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01531 if (audio_dest_res != AST_RTP_TRY_NATIVE) { 01532 /* Somebody doesn't want to play... */ 01533 ast_channel_unlock(dest); 01534 if (src) 01535 ast_channel_unlock(src); 01536 return 0; 01537 } 01538 if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec) 01539 srccodec = srcpr->get_codec(src); 01540 else 01541 srccodec = 0; 01542 if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec) 01543 destcodec = destpr->get_codec(dest); 01544 else 01545 destcodec = 0; 01546 /* Ensure we have at least one matching codec */ 01547 if (!(srccodec & destcodec)) { 01548 ast_channel_unlock(dest); 01549 if (src) 01550 ast_channel_unlock(src); 01551 return 0; 01552 } 01553 /* Consider empty media as non-existant */ 01554 if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) 01555 srcp = NULL; 01556 /* If the client has NAT stuff turned on then just safe NAT is active */ 01557 if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01558 nat_active = 1; 01559 /* Bridge media early */ 01560 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active)) 01561 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01562 ast_channel_unlock(dest); 01563 if (src) 01564 ast_channel_unlock(src); 01565 if (option_debug) 01566 ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>"); 01567 return 1; 01568 }
int ast_rtp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 512 of file rtp.c.
References ast_rtp::s.
Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), mgcp_new(), sip_new(), skinny_new(), and start_rtp().
00513 { 00514 return rtp->s; 00515 }
Definition at line 2018 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, and ast_rtp::bridged.
Referenced by ast_rtp_read(), and do_monitor().
02019 { 02020 struct ast_rtp *bridged = NULL; 02021 02022 ast_mutex_lock(&rtp->bridge_lock); 02023 bridged = rtp->bridged; 02024 ast_mutex_unlock(&rtp->bridge_lock); 02025 02026 return bridged; 02027 }
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 1688 of file rtp.c.
References ast_mutex_lock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, and MAX_RTP_PT.
Referenced by process_sdp().
01690 { 01691 int pt; 01692 01693 ast_mutex_lock(&rtp->bridge_lock); 01694 01695 *astFormats = *nonAstFormats = 0; 01696 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01697 if (rtp->current_RTP_PT[pt].isAstFormat) { 01698 *astFormats |= rtp->current_RTP_PT[pt].code; 01699 } else { 01700 *nonAstFormats |= rtp->current_RTP_PT[pt].code; 01701 } 01702 } 01703 01704 ast_mutex_unlock(&rtp->bridge_lock); 01705 01706 return; 01707 }
int ast_rtp_get_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 2000 of file rtp.c.
References ast_rtp::them.
Referenced by add_sdp(), bridge_native_loop(), do_monitor(), gtalk_update_stun(), oh323_set_rtp_peer(), sip_set_rtp_peer(), and transmit_modify_with_sdp().
02001 { 02002 if ((them->sin_family != AF_INET) || 02003 (them->sin_port != rtp->them.sin_port) || 02004 (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) { 02005 them->sin_family = AF_INET; 02006 them->sin_port = rtp->them.sin_port; 02007 them->sin_addr = rtp->them.sin_addr; 02008 return 1; 02009 } 02010 return 0; 02011 }
char* ast_rtp_get_quality | ( | struct ast_rtp * | rtp, | |
struct ast_rtp_quality * | qual | |||
) |
Return RTCP quality string.
Definition at line 2066 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().
02067 { 02068 /* 02069 *ssrc our ssrc 02070 *themssrc their ssrc 02071 *lp lost packets 02072 *rxjitter our calculated jitter(rx) 02073 *rxcount no. received packets 02074 *txjitter reported jitter of the other end 02075 *txcount transmitted packets 02076 *rlp remote lost packets 02077 *rtt round trip time 02078 */ 02079 02080 if (qual) { 02081 qual->local_ssrc = rtp->ssrc; 02082 qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior; 02083 qual->local_jitter = rtp->rxjitter; 02084 qual->local_count = rtp->rxcount; 02085 qual->remote_ssrc = rtp->themssrc; 02086 qual->remote_lostpackets = rtp->rtcp->reported_lost; 02087 qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0; 02088 qual->remote_count = rtp->txcount; 02089 qual->rtt = rtp->rtcp->rtt; 02090 } 02091 snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", rtp->ssrc, rtp->themssrc, rtp->rtcp->expected_prior - rtp->rtcp->received_prior, rtp->rxjitter, rtp->rxcount, (double)rtp->rtcp->reported_jitter/65536., rtp->txcount, rtp->rtcp->reported_lost, rtp->rtcp->rtt); 02092 02093 return rtp->rtcp->quality; 02094 }
int ast_rtp_get_rtpholdtimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp hold timeout.
Definition at line 567 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by do_monitor().
00568 { 00569 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00570 return 0; 00571 return rtp->rtpholdtimeout; 00572 }
int ast_rtp_get_rtpkeepalive | ( | struct ast_rtp * | rtp | ) |
Get RTP keepalive interval.
Definition at line 575 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by do_monitor().
00576 { 00577 return rtp->rtpkeepalive; 00578 }
int ast_rtp_get_rtptimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp timeout.
Definition at line 559 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by do_monitor().
00560 { 00561 if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ 00562 return 0; 00563 return rtp->rtptimeout; 00564 }
void ast_rtp_get_us | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | us | |||
) |
Definition at line 2013 of file rtp.c.
References ast_rtp::us.
Referenced by add_sdp(), external_rtp_create(), gtalk_create_candidates(), handle_open_receive_channel_ack_message(), and oh323_set_rtp_peer().
int ast_rtp_getnat | ( | struct ast_rtp * | rtp | ) |
Definition at line 595 of file rtp.c.
References ast_test_flag, and FLAG_NAT_ACTIVE.
Referenced by sip_get_rtp_peer().
00596 { 00597 return ast_test_flag(rtp, FLAG_NAT_ACTIVE); 00598 }
void ast_rtp_init | ( | void | ) |
Initialize the RTP system in Asterisk.
Definition at line 3752 of file rtp.c.
References ast_cli_register_multiple(), ast_rtp_reload(), and cli_rtp.
Referenced by main().
03753 { 03754 ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry)); 03755 ast_rtp_reload(); 03756 }
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 1731 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().
01732 { 01733 int pt = 0; 01734 01735 ast_mutex_lock(&rtp->bridge_lock); 01736 01737 if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && 01738 code == rtp->rtp_lookup_code_cache_code) { 01739 /* Use our cached mapping, to avoid the overhead of the loop below */ 01740 pt = rtp->rtp_lookup_code_cache_result; 01741 ast_mutex_unlock(&rtp->bridge_lock); 01742 return pt; 01743 } 01744 01745 /* Check the dynamic list first */ 01746 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01747 if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) { 01748 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01749 rtp->rtp_lookup_code_cache_code = code; 01750 rtp->rtp_lookup_code_cache_result = pt; 01751 ast_mutex_unlock(&rtp->bridge_lock); 01752 return pt; 01753 } 01754 } 01755 01756 /* Then the static list */ 01757 for (pt = 0; pt < MAX_RTP_PT; ++pt) { 01758 if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) { 01759 rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; 01760 rtp->rtp_lookup_code_cache_code = code; 01761 rtp->rtp_lookup_code_cache_result = pt; 01762 ast_mutex_unlock(&rtp->bridge_lock); 01763 return pt; 01764 } 01765 } 01766 01767 ast_mutex_unlock(&rtp->bridge_lock); 01768 01769 return -1; 01770 }
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 1791 of file rtp.c.
References ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, format, len, and name.
Referenced by process_sdp().
01793 { 01794 int format; 01795 unsigned len; 01796 char *end = buf; 01797 char *start = buf; 01798 01799 if (!buf || !size) 01800 return NULL; 01801 01802 snprintf(end, size, "0x%x (", capability); 01803 01804 len = strlen(end); 01805 end += len; 01806 size -= len; 01807 start = end; 01808 01809 for (format = 1; format < AST_RTP_MAX; format <<= 1) { 01810 if (capability & format) { 01811 const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options); 01812 01813 snprintf(end, size, "%s|", name); 01814 len = strlen(end); 01815 end += len; 01816 size -= len; 01817 } 01818 } 01819 01820 if (start == end) 01821 snprintf(start, size, "nothing)"); 01822 else if (size > 1) 01823 *(end -1) = ')'; 01824 01825 return buf; 01826 }
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 1772 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().
01774 { 01775 unsigned int i; 01776 01777 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01778 if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) { 01779 if (isAstFormat && 01780 (code == AST_FORMAT_G726_AAL2) && 01781 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01782 return "G726-32"; 01783 else 01784 return mimeTypes[i].subtype; 01785 } 01786 } 01787 01788 return ""; 01789 }
struct rtpPayloadType ast_rtp_lookup_pt | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Mapping between RTP payload format codes and Asterisk codes:.
Definition at line 1709 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), MAX_RTP_PT, result, and static_RTP_PT.
Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and setup_rtp_connection().
01710 { 01711 struct rtpPayloadType result; 01712 01713 result.isAstFormat = result.code = 0; 01714 01715 if (pt < 0 || pt > MAX_RTP_PT) 01716 return result; /* bogus payload type */ 01717 01718 /* Start with negotiated codecs */ 01719 ast_mutex_lock(&rtp->bridge_lock); 01720 result = rtp->current_RTP_PT[pt]; 01721 ast_mutex_unlock(&rtp->bridge_lock); 01722 01723 /* If it doesn't exist, check our static RTP type list, just in case */ 01724 if (!result.code) 01725 result = static_RTP_PT[pt]; 01726 01727 return result; 01728 }
int ast_rtp_make_compatible | ( | struct ast_channel * | dest, | |
struct ast_channel * | src, | |||
int | media | |||
) |
Definition at line 1570 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, ast_rtp_pt_copy(), AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, option_debug, and ast_rtp_protocol::set_rtp_peer.
Referenced by wait_for_answer().
01571 { 01572 struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ 01573 struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ 01574 struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; 01575 enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED; 01576 enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 01577 int srccodec, destcodec; 01578 01579 /* Lock channels */ 01580 ast_channel_lock(dest); 01581 while(ast_channel_trylock(src)) { 01582 ast_channel_unlock(dest); 01583 usleep(1); 01584 ast_channel_lock(dest); 01585 } 01586 01587 /* Find channel driver interfaces */ 01588 if (!(destpr = get_proto(dest))) { 01589 if (option_debug) 01590 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name); 01591 ast_channel_unlock(dest); 01592 ast_channel_unlock(src); 01593 return 0; 01594 } 01595 if (!(srcpr = get_proto(src))) { 01596 if (option_debug) 01597 ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name); 01598 ast_channel_unlock(dest); 01599 ast_channel_unlock(src); 01600 return 0; 01601 } 01602 01603 /* Get audio and video interface (if native bridge is possible) */ 01604 audio_dest_res = destpr->get_rtp_info(dest, &destp); 01605 video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; 01606 audio_src_res = srcpr->get_rtp_info(src, &srcp); 01607 video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; 01608 01609 /* Ensure we have at least one matching codec */ 01610 if (srcpr->get_codec) 01611 srccodec = srcpr->get_codec(src); 01612 else 01613 srccodec = 0; 01614 if (destpr->get_codec) 01615 destcodec = destpr->get_codec(dest); 01616 else 01617 destcodec = 0; 01618 01619 /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ 01620 if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) { 01621 /* Somebody doesn't want to play... */ 01622 ast_channel_unlock(dest); 01623 ast_channel_unlock(src); 01624 return 0; 01625 } 01626 ast_rtp_pt_copy(destp, srcp); 01627 if (vdestp && vsrcp) 01628 ast_rtp_pt_copy(vdestp, vsrcp); 01629 if (media) { 01630 /* Bridge early */ 01631 if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) 01632 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name); 01633 } 01634 ast_channel_unlock(dest); 01635 ast_channel_unlock(src); 01636 if (option_debug) 01637 ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name); 01638 return 1; 01639 }
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 1972 of file rtp.c.
References ast_rtp_new_with_bindaddr(), io, and sched.
01973 { 01974 struct in_addr ia; 01975 01976 memset(&ia, 0, sizeof(ia)); 01977 return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia); 01978 }
void ast_rtp_new_init | ( | struct ast_rtp * | rtp | ) |
Initialize a new RTP structure.
Definition at line 1872 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().
01873 { 01874 ast_mutex_init(&rtp->bridge_lock); 01875 01876 rtp->them.sin_family = AF_INET; 01877 rtp->us.sin_family = AF_INET; 01878 rtp->ssrc = ast_random(); 01879 rtp->seqno = ast_random() & 0xffff; 01880 ast_set_flag(rtp, FLAG_HAS_DTMF); 01881 01882 return; 01883 }
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 1885 of file rtp.c.
References ast_calloc, ast_io_add(), AST_IO_IN, ast_log(), ast_random(), ast_rtcp_new(), ast_rtp_new_init(), ast_rtp_pt_default(), ast_set_flag, FLAG_CALLBACK_MODE, free, io, LOG_ERROR, rtp_socket(), rtpread(), and sched.
Referenced by __oh323_rtp_create(), ast_rtp_new(), gtalk_alloc(), sip_alloc(), and start_rtp().
01886 { 01887 struct ast_rtp *rtp; 01888 int x; 01889 int first; 01890 int startplace; 01891 01892 if (!(rtp = ast_calloc(1, sizeof(*rtp)))) 01893 return NULL; 01894 01895 ast_rtp_new_init(rtp); 01896 01897 rtp->s = rtp_socket(); 01898 if (rtp->s < 0) { 01899 free(rtp); 01900 ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno)); 01901 return NULL; 01902 } 01903 if (sched && rtcpenable) { 01904 rtp->sched = sched; 01905 rtp->rtcp = ast_rtcp_new(); 01906 } 01907 01908 /* Select a random port number in the range of possible RTP */ 01909 x = (ast_random() % (rtpend-rtpstart)) + rtpstart; 01910 x = x & ~1; 01911 /* Save it for future references. */ 01912 startplace = x; 01913 /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */ 01914 for (;;) { 01915 /* Must be an even port number by RTP spec */ 01916 rtp->us.sin_port = htons(x); 01917 rtp->us.sin_addr = addr; 01918 /* If there's rtcp, initialize it as well. */ 01919 if (rtp->rtcp) { 01920 rtp->rtcp->us.sin_port = htons(x + 1); 01921 rtp->rtcp->us.sin_addr = addr; 01922 } 01923 /* Try to bind it/them. */ 01924 if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) && 01925 (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) 01926 break; 01927 if (!first) { 01928 /* Primary bind succeeded! Gotta recreate it */ 01929 close(rtp->s); 01930 rtp->s = rtp_socket(); 01931 } 01932 if (errno != EADDRINUSE) { 01933 /* We got an error that wasn't expected, abort! */ 01934 ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); 01935 close(rtp->s); 01936 if (rtp->rtcp) { 01937 close(rtp->rtcp->s); 01938 free(rtp->rtcp); 01939 } 01940 free(rtp); 01941 return NULL; 01942 } 01943 /* The port was used, increment it (by two). */ 01944 x += 2; 01945 /* Did we go over the limit ? */ 01946 if (x > rtpend) 01947 /* then, start from the begingig. */ 01948 x = (rtpstart + 1) & ~1; 01949 /* Check if we reached the place were we started. */ 01950 if (x == startplace) { 01951 /* If so, there's no ports available. */ 01952 ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); 01953 close(rtp->s); 01954 if (rtp->rtcp) { 01955 close(rtp->rtcp->s); 01956 free(rtp->rtcp); 01957 } 01958 free(rtp); 01959 return NULL; 01960 } 01961 } 01962 rtp->sched = sched; 01963 rtp->io = io; 01964 if (callbackmode) { 01965 rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); 01966 ast_set_flag(rtp, FLAG_CALLBACK_MODE); 01967 } 01968 ast_rtp_pt_default(rtp); 01969 return rtp; 01970 }
int ast_rtp_proto_register | ( | struct ast_rtp_protocol * | proto | ) |
Register interface to channel driver.
Definition at line 2797 of file rtp.c.
References AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_log(), LOG_WARNING, protos, and ast_rtp_protocol::type.
Referenced by load_module().
02798 { 02799 struct ast_rtp_protocol *cur; 02800 02801 AST_LIST_LOCK(&protos); 02802 AST_LIST_TRAVERSE(&protos, cur, list) { 02803 if (!strcmp(cur->type, proto->type)) { 02804 ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); 02805 AST_LIST_UNLOCK(&protos); 02806 return -1; 02807 } 02808 } 02809 AST_LIST_INSERT_HEAD(&protos, proto, list); 02810 AST_LIST_UNLOCK(&protos); 02811 02812 return 0; 02813 }
void ast_rtp_proto_unregister | ( | struct ast_rtp_protocol * | proto | ) |
Unregister interface to channel driver.
Definition at line 2789 of file rtp.c.
References AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, and protos.
Referenced by load_module(), and unload_module().
02790 { 02791 AST_LIST_LOCK(&protos); 02792 AST_LIST_REMOVE(&protos, proto, list); 02793 AST_LIST_UNLOCK(&protos); 02794 }
void ast_rtp_pt_clear | ( | struct ast_rtp * | rtp | ) |
Setting RTP payload types from lines in a SDP description:.
Definition at line 1406 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.
Referenced by process_sdp().
01407 { 01408 int i; 01409 01410 if (!rtp) 01411 return; 01412 01413 ast_mutex_lock(&rtp->bridge_lock); 01414 01415 for (i = 0; i < MAX_RTP_PT; ++i) { 01416 rtp->current_RTP_PT[i].isAstFormat = 0; 01417 rtp->current_RTP_PT[i].code = 0; 01418 } 01419 01420 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01421 rtp->rtp_lookup_code_cache_code = 0; 01422 rtp->rtp_lookup_code_cache_result = 0; 01423 01424 ast_mutex_unlock(&rtp->bridge_lock); 01425 }
Copy payload types between RTP structures.
Definition at line 1446 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().
01447 { 01448 unsigned int i; 01449 01450 ast_mutex_lock(&dest->bridge_lock); 01451 ast_mutex_lock(&src->bridge_lock); 01452 01453 for (i=0; i < MAX_RTP_PT; ++i) { 01454 dest->current_RTP_PT[i].isAstFormat = 01455 src->current_RTP_PT[i].isAstFormat; 01456 dest->current_RTP_PT[i].code = 01457 src->current_RTP_PT[i].code; 01458 } 01459 dest->rtp_lookup_code_cache_isAstFormat = 0; 01460 dest->rtp_lookup_code_cache_code = 0; 01461 dest->rtp_lookup_code_cache_result = 0; 01462 01463 ast_mutex_unlock(&src->bridge_lock); 01464 ast_mutex_unlock(&dest->bridge_lock); 01465 }
void ast_rtp_pt_default | ( | struct ast_rtp * | rtp | ) |
Set payload types to defaults.
Definition at line 1427 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, ast_rtp::rtp_lookup_code_cache_result, and static_RTP_PT.
Referenced by ast_rtp_new_with_bindaddr().
01428 { 01429 int i; 01430 01431 ast_mutex_lock(&rtp->bridge_lock); 01432 01433 /* Initialize to default payload types */ 01434 for (i = 0; i < MAX_RTP_PT; ++i) { 01435 rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat; 01436 rtp->current_RTP_PT[i].code = static_RTP_PT[i].code; 01437 } 01438 01439 rtp->rtp_lookup_code_cache_isAstFormat = 0; 01440 rtp->rtp_lookup_code_cache_code = 0; 01441 rtp->rtp_lookup_code_cache_result = 0; 01442 01443 ast_mutex_unlock(&rtp->bridge_lock); 01444 }
Definition at line 1097 of file rtp.c.
References ast_backtrace(), ast_codec_get_samples(), AST_FORMAT_MAX_AUDIO, AST_FORMAT_SLINEAR, ast_frame_byteswap_be, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_get_bridged(), ast_rtp_lookup_pt(), ast_rtp_senddigit_continuation(), ast_sched_add(), ast_set_flag, ast_verbose(), bridge_p2p_rtp_write(), ast_rtp::bridged, calc_rxstamp(), rtpPayloadType::code, CRASH, ast_rtp::cycles, ast_frame::data, ast_frame::datalen, ast_frame::delivery, event, ext, ast_rtp::f, f, FLAG_NAT_ACTIVE, ast_frame::frametype, ast_frame::has_timing_info, rtpPayloadType::isAstFormat, ast_rtp::lastevent, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len, LOG_DEBUG, LOG_ERROR, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_rtp::rawdata, ast_rtp::rtcp, rtp_debug_test_addr(), RTP_SEQ_MOD, ast_rtp::rxcount, ast_rtp::rxseqno, ast_rtp::rxssrc, ast_rtp::s, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, ast_rtp::sending_digit, ast_frame::seqno, ast_frame::src, STUN_ACCEPT, stun_handle_packet(), ast_frame::subclass, ast_rtcp::them, ast_rtp::them, ast_rtp::themssrc, and ast_frame::ts.
Referenced by gtalk_rtp_read(), mgcp_rtp_read(), oh323_rtp_read(), rtpread(), sip_rtp_read(), and skinny_rtp_read().
01098 { 01099 int res; 01100 struct sockaddr_in sin; 01101 socklen_t len; 01102 unsigned int seqno; 01103 int version; 01104 int payloadtype; 01105 int hdrlen = 12; 01106 int padding; 01107 int mark; 01108 int ext; 01109 int cc; 01110 unsigned int ssrc; 01111 unsigned int timestamp; 01112 unsigned int *rtpheader; 01113 struct rtpPayloadType rtpPT; 01114 struct ast_rtp *bridged = NULL; 01115 01116 if( !rtp ) { 01117 ast_log(LOG_ERROR, "ast_rtp_read(): called with rtp == NULL\n"); 01118 ast_backtrace(); 01119 return &ast_null_frame; 01120 } 01121 01122 /* If time is up, kill it */ 01123 if (rtp->sending_digit) 01124 ast_rtp_senddigit_continuation(rtp); 01125 01126 len = sizeof(sin); 01127 01128 /* Cache where the header will go */ 01129 res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 01130 0, (struct sockaddr *)&sin, &len); 01131 01132 rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); 01133 if (res < 0) { 01134 if (errno == EBADF) 01135 CRASH; 01136 if (errno != EAGAIN) { 01137 ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno)); 01138 return NULL; 01139 } 01140 return &ast_null_frame; 01141 } 01142 01143 if (res < hdrlen) { 01144 ast_log(LOG_WARNING, "RTP Read too short\n"); 01145 return &ast_null_frame; 01146 } 01147 01148 /* Get fields */ 01149 seqno = ntohl(rtpheader[0]); 01150 01151 /* Check RTP version */ 01152 version = (seqno & 0xC0000000) >> 30; 01153 if (!version) { 01154 if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) && 01155 (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) { 01156 memcpy(&rtp->them, &sin, sizeof(rtp->them)); 01157 } 01158 return &ast_null_frame; 01159 } 01160 01161 #if 0 /* Allow to receive RTP stream with closed transmission path */ 01162 /* If we don't have the other side's address, then ignore this */ 01163 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 01164 return &ast_null_frame; 01165 #endif 01166 01167 /* Send to whoever send to us if NAT is turned on */ 01168 if (rtp->nat) { 01169 if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || 01170 (rtp->them.sin_port != sin.sin_port)) { 01171 rtp->them = sin; 01172 if (rtp->rtcp) { 01173 memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); 01174 rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1); 01175 } 01176 rtp->rxseqno = 0; 01177 ast_set_flag(rtp, FLAG_NAT_ACTIVE); 01178 if (option_debug || rtpdebug) 01179 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)); 01180 } 01181 } 01182 01183 /* If we are bridged to another RTP stream, send direct */ 01184 if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen)) 01185 return &ast_null_frame; 01186 01187 if (version != 2) 01188 return &ast_null_frame; 01189 01190 payloadtype = (seqno & 0x7f0000) >> 16; 01191 padding = seqno & (1 << 29); 01192 mark = seqno & (1 << 23); 01193 ext = seqno & (1 << 28); 01194 cc = (seqno & 0xF000000) >> 24; 01195 seqno &= 0xffff; 01196 timestamp = ntohl(rtpheader[1]); 01197 ssrc = ntohl(rtpheader[2]); 01198 01199 if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) { 01200 if (option_debug || rtpdebug) 01201 ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n"); 01202 mark = 1; 01203 } 01204 01205 rtp->rxssrc = ssrc; 01206 01207 if (padding) { 01208 /* Remove padding bytes */ 01209 res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; 01210 } 01211 01212 if (cc) { 01213 /* CSRC fields present */ 01214 hdrlen += cc*4; 01215 } 01216 01217 if (ext) { 01218 /* RTP Extension present */ 01219 hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2; 01220 hdrlen += 4; 01221 if (option_debug) { 01222 int profile; 01223 profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16; 01224 if (profile == 0x505a) 01225 ast_log(LOG_DEBUG, "Found Zfone extension in RTP stream - zrtp - not supported.\n"); 01226 else 01227 ast_log(LOG_DEBUG, "Found unknown RTP Extensions %x\n", profile); 01228 } 01229 } 01230 01231 if (res < hdrlen) { 01232 ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); 01233 return &ast_null_frame; 01234 } 01235 01236 rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */ 01237 01238 if (rtp->rxcount==1) { 01239 /* This is the first RTP packet successfully received from source */ 01240 rtp->seedrxseqno = seqno; 01241 } 01242 01243 /* Do not schedule RR if RTCP isn't run */ 01244 if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) { 01245 /* Schedule transmission of Receiver Report */ 01246 rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); 01247 } 01248 if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */ 01249 rtp->cycles += RTP_SEQ_MOD; 01250 01251 rtp->lastrxseqno = seqno; 01252 01253 if (rtp->themssrc==0) 01254 rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */ 01255 01256 if (rtp_debug_test_addr(&sin)) 01257 ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 01258 ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); 01259 01260 rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); 01261 if (!rtpPT.isAstFormat) { 01262 struct ast_frame *f = NULL; 01263 01264 /* This is special in-band data that's not one of our codecs */ 01265 if (rtpPT.code == AST_RTP_DTMF) { 01266 /* It's special -- rfc2833 process it */ 01267 if (rtp_debug_test_addr(&sin)) { 01268 unsigned char *data; 01269 unsigned int event; 01270 unsigned int event_end; 01271 unsigned int duration; 01272 data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen; 01273 event = ntohl(*((unsigned int *)(data))); 01274 event >>= 24; 01275 event_end = ntohl(*((unsigned int *)(data))); 01276 event_end <<= 8; 01277 event_end >>= 24; 01278 duration = ntohl(*((unsigned int *)(data))); 01279 duration &= 0xFFFF; 01280 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); 01281 } 01282 f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp); 01283 } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { 01284 /* It's really special -- process it the Cisco way */ 01285 if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) { 01286 f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01287 rtp->lastevent = seqno; 01288 } 01289 } else if (rtpPT.code == AST_RTP_CN) { 01290 /* Comfort Noise */ 01291 f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); 01292 } else { 01293 ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr)); 01294 } 01295 return f ? f : &ast_null_frame; 01296 } 01297 rtp->lastrxformat = rtp->f.subclass = rtpPT.code; 01298 rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO; 01299 01300 if (!rtp->lastrxts) 01301 rtp->lastrxts = timestamp; 01302 01303 rtp->rxseqno = seqno; 01304 01305 /* Record received timestamp as last received now */ 01306 rtp->lastrxts = timestamp; 01307 01308 rtp->f.mallocd = 0; 01309 rtp->f.datalen = res - hdrlen; 01310 rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; 01311 rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; 01312 rtp->f.seqno = seqno; 01313 if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { 01314 rtp->f.samples = ast_codec_get_samples(&rtp->f); 01315 if (rtp->f.subclass == AST_FORMAT_SLINEAR) 01316 ast_frame_byteswap_be(&rtp->f); 01317 calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark); 01318 /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */ 01319 rtp->f.has_timing_info = 1; 01320 rtp->f.ts = timestamp / 8; 01321 rtp->f.len = rtp->f.samples / 8; 01322 } else { 01323 /* Video -- samples is # of samples vs. 90000 */ 01324 if (!rtp->lastividtimestamp) 01325 rtp->lastividtimestamp = timestamp; 01326 rtp->f.samples = timestamp - rtp->lastividtimestamp; 01327 rtp->lastividtimestamp = timestamp; 01328 rtp->f.delivery.tv_sec = 0; 01329 rtp->f.delivery.tv_usec = 0; 01330 if (mark) 01331 rtp->f.subclass |= 0x1; 01332 01333 } 01334 rtp->f.src = "RTP"; 01335 return &rtp->f; 01336 }
int ast_rtp_reload | ( | void | ) |
Definition at line 3687 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().
03688 { 03689 struct ast_config *cfg; 03690 const char *s; 03691 03692 rtpstart = 5000; 03693 rtpend = 31000; 03694 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03695 cfg = ast_config_load("rtp.conf"); 03696 if (cfg) { 03697 if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { 03698 rtpstart = atoi(s); 03699 if (rtpstart < 1024) 03700 rtpstart = 1024; 03701 if (rtpstart > 65535) 03702 rtpstart = 65535; 03703 } 03704 if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { 03705 rtpend = atoi(s); 03706 if (rtpend < 1024) 03707 rtpend = 1024; 03708 if (rtpend > 65535) 03709 rtpend = 65535; 03710 } 03711 if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) { 03712 rtcpinterval = atoi(s); 03713 if (rtcpinterval == 0) 03714 rtcpinterval = 0; /* Just so we're clear... it's zero */ 03715 if (rtcpinterval < RTCP_MIN_INTERVALMS) 03716 rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */ 03717 if (rtcpinterval > RTCP_MAX_INTERVALMS) 03718 rtcpinterval = RTCP_MAX_INTERVALMS; 03719 } 03720 if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) { 03721 #ifdef SO_NO_CHECK 03722 if (ast_false(s)) 03723 nochecksums = 1; 03724 else 03725 nochecksums = 0; 03726 #else 03727 if (ast_false(s)) 03728 ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n"); 03729 #endif 03730 } 03731 if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { 03732 dtmftimeout = atoi(s); 03733 if ((dtmftimeout < 0) || (dtmftimeout > 20000)) { 03734 ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", 03735 dtmftimeout, DEFAULT_DTMF_TIMEOUT); 03736 dtmftimeout = DEFAULT_DTMF_TIMEOUT; 03737 }; 03738 } 03739 ast_config_destroy(cfg); 03740 } 03741 if (rtpstart >= rtpend) { 03742 ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n"); 03743 rtpstart = 5000; 03744 rtpend = 31000; 03745 } 03746 if (option_verbose > 1) 03747 ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); 03748 return 0; 03749 }
void ast_rtp_reset | ( | struct ast_rtp * | rtp | ) |
Definition at line 2046 of file rtp.c.
References ast_rtp::dtmfcount, ast_rtp::dtmfmute, ast_rtp::dtmfsamples, ast_rtp::lastdigitts, ast_rtp::lastevent, ast_rtp::lasteventseqn, ast_rtp::lastividtimestamp, ast_rtp::lastovidtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxts, ast_rtp::lastts, ast_rtp::lasttxformat, ast_rtp::rxcore, ast_rtp::rxseqno, ast_rtp::seqno, and ast_rtp::txcore.
02047 { 02048 memset(&rtp->rxcore, 0, sizeof(rtp->rxcore)); 02049 memset(&rtp->txcore, 0, sizeof(rtp->txcore)); 02050 memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute)); 02051 rtp->lastts = 0; 02052 rtp->lastdigitts = 0; 02053 rtp->lastrxts = 0; 02054 rtp->lastividtimestamp = 0; 02055 rtp->lastovidtimestamp = 0; 02056 rtp->lasteventseqn = 0; 02057 rtp->lastevent = 0; 02058 rtp->lasttxformat = 0; 02059 rtp->lastrxformat = 0; 02060 rtp->dtmfcount = 0; 02061 rtp->dtmfsamples = 0; 02062 rtp->seqno = 0; 02063 rtp->rxseqno = 0; 02064 }
int ast_rtp_sendcng | ( | struct ast_rtp * | rtp, | |
int | level | |||
) |
generate comfort noice (CNG)
Definition at line 2556 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_CN, ast_rtp_lookup_code(), ast_tvadd(), ast_verbose(), ast_rtp::data, ast_rtp::dtmfmute, 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().
02557 { 02558 unsigned int *rtpheader; 02559 int hdrlen = 12; 02560 int res; 02561 int payload; 02562 char data[256]; 02563 level = 127 - (level & 0x7f); 02564 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN); 02565 02566 /* If we have no peer, return immediately */ 02567 if (!rtp->them.sin_addr.s_addr) 02568 return 0; 02569 02570 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02571 02572 /* Get a pointer to the header */ 02573 rtpheader = (unsigned int *)data; 02574 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++)); 02575 rtpheader[1] = htonl(rtp->lastts); 02576 rtpheader[2] = htonl(rtp->ssrc); 02577 data[12] = level; 02578 if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { 02579 res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); 02580 if (res <0) 02581 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)); 02582 if (rtp_debug_test_addr(&rtp->them)) 02583 ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n" 02584 , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); 02585 02586 } 02587 return 0; 02588 }
int ast_rtp_senddigit_begin | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
Send begin frames for DTMF.
Definition at line 2156 of file rtp.c.
References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_tvadd(), ast_verbose(), ast_rtp::dtmfmute, ast_rtp::lastdigitts, LOG_ERROR, LOG_WARNING, rtp_debug_test_addr(), ast_rtp::s, ast_rtp::send_digit, ast_rtp::send_duration, ast_rtp::send_payload, ast_rtp::sending_digit, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.
Referenced by oh323_digit_begin(), and sip_senddigit_begin().
02157 { 02158 unsigned int *rtpheader; 02159 int hdrlen = 12, res = 0, i = 0, payload = 0; 02160 char data[256]; 02161 02162 if ((digit <= '9') && (digit >= '0')) 02163 digit -= '0'; 02164 else if (digit == '*') 02165 digit = 10; 02166 else if (digit == '#') 02167 digit = 11; 02168 else if ((digit >= 'A') && (digit <= 'D')) 02169 digit = digit - 'A' + 12; 02170 else if ((digit >= 'a') && (digit <= 'd')) 02171 digit = digit - 'a' + 12; 02172 else { 02173 ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); 02174 return 0; 02175 } 02176 02177 /* If we have no peer, return immediately */ 02178 if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) 02179 return 0; 02180 02181 payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); 02182 02183 rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); 02184 rtp->send_duration = 160; 02185 02186 /* Get a pointer to the header */ 02187 rtpheader = (unsigned int *)data; 02188 rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno)); 02189 rtpheader[1] = htonl(rtp->lastdigitts); 02190 rtpheader[2] = htonl(rtp->ssrc); 02191 02192 for (i = 0; i < 2; i++) { 02193 rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration)); 02194 res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); 02195 if (res < 0) 02196 ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n", 02197 ast_inet_ntoa(rtp->them.sin_addr), 02198 ntohs(rtp->them.sin_port), strerror(errno)); 02199 if (rtp_debug_test_addr(&rtp->them)) 02200 ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 02201 ast_inet_ntoa(rtp->them.sin_addr), 02202 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 02203 /* Increment sequence number */ 02204 rtp->seqno++; 02205 /* Increment duration */ 02206 rtp->send_duration += 160; 02207 /* Clear marker bit and set seqno */ 02208 rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); 02209 } 02210 02211 /* Since we received a begin, we can safely store the digit and disable any compensation */ 02212 rtp->sending_digit = 1; 02213 rtp->send_digit = digit; 02214 rtp->send_payload = payload; 02215 02216 return 0; 02217 }
int ast_rtp_senddigit_end | ( | struct ast_rtp * | rtp, | |
char | digit | |||
) |
void ast_rtp_set_callback | ( | struct ast_rtp * | rtp, | |
ast_rtp_callback | callback | |||
) |
Definition at line 585 of file rtp.c.
References ast_rtp::callback.
Referenced by start_rtp().
00586 { 00587 rtp->callback = callback; 00588 }
void ast_rtp_set_data | ( | struct ast_rtp * | rtp, | |
void * | data | |||
) |
Definition at line 580 of file rtp.c.
References ast_rtp::data.
Referenced by start_rtp().
00581 { 00582 rtp->data = data; 00583 }
void ast_rtp_set_m_type | ( | struct ast_rtp * | rtp, | |
int | pt | |||
) |
Make a note of a RTP payload type that was seen in a SDP "m=" line. By default, use the well-known value for this type (although it may still be set to a different value by a subsequent "a=rtpmap:" line).
Definition at line 1645 of file rtp.c.
References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, ast_rtp::current_RTP_PT, MAX_RTP_PT, and static_RTP_PT.
Referenced by gtalk_newcall(), and process_sdp().
01646 { 01647 if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 01648 return; /* bogus payload type */ 01649 01650 ast_mutex_lock(&rtp->bridge_lock); 01651 rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; 01652 ast_mutex_unlock(&rtp->bridge_lock); 01653 }
void ast_rtp_set_peer | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | them | |||
) |
Definition at line 1989 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().
01990 { 01991 rtp->them.sin_port = them->sin_port; 01992 rtp->them.sin_addr = them->sin_addr; 01993 if (rtp->rtcp) { 01994 rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1); 01995 rtp->rtcp->them.sin_addr = them->sin_addr; 01996 } 01997 rtp->rxseqno = 0; 01998 }
void ast_rtp_set_rtpholdtimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp hold timeout.
Definition at line 547 of file rtp.c.
References ast_rtp::rtpholdtimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00548 { 00549 rtp->rtpholdtimeout = timeout; 00550 }
void ast_rtp_set_rtpkeepalive | ( | struct ast_rtp * | rtp, | |
int | period | |||
) |
set RTP keepalive interval
Definition at line 553 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by create_addr_from_peer(), and sip_alloc().
00554 { 00555 rtp->rtpkeepalive = period; 00556 }
void ast_rtp_set_rtpmap_type | ( | struct ast_rtp * | rtp, | |
int | pt, | |||
char * | mimeType, | |||
char * | mimeSubtype, | |||
enum ast_rtp_options | options | |||
) |
Make a note of a RTP payload type (with MIME type) that was seen in an SDP "a=rtpmap:" line.
Definition at line 1658 of file rtp.c.
References AST_FORMAT_G726, AST_FORMAT_G726_AAL2, ast_mutex_lock(), ast_mutex_unlock(), AST_RTP_OPT_G726_NONSTANDARD, ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, MAX_RTP_PT, mimeTypes, payloadType, subtype, and type.
Referenced by __oh323_rtp_create(), gtalk_newcall(), process_sdp(), and set_dtmf_payload().
01661 { 01662 unsigned int i; 01663 01664 if (pt < 0 || pt > MAX_RTP_PT) 01665 return; /* bogus payload type */ 01666 01667 ast_mutex_lock(&rtp->bridge_lock); 01668 01669 for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { 01670 if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && 01671 strcasecmp(mimeType, mimeTypes[i].type) == 0) { 01672 rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; 01673 if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) && 01674 mimeTypes[i].payloadType.isAstFormat && 01675 (options & AST_RTP_OPT_G726_NONSTANDARD)) 01676 rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2; 01677 break; 01678 } 01679 } 01680 01681 ast_mutex_unlock(&rtp->bridge_lock); 01682 01683 return; 01684 }
void ast_rtp_set_rtptimeout | ( | struct ast_rtp * | rtp, | |
int | timeout | |||
) |
Set rtp timeout.
Definition at line 541 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().
00542 { 00543 rtp->rtptimeout = timeout; 00544 }
void ast_rtp_set_rtptimers_onhold | ( | struct ast_rtp * | rtp | ) |
Definition at line 534 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by handle_response_invite().
00535 { 00536 rtp->rtptimeout = (-1) * rtp->rtptimeout; 00537 rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout; 00538 }
void ast_rtp_setdtmf | ( | struct ast_rtp * | rtp, | |
int | dtmf | |||
) |
Indicate whether this RTP session is carrying DTMF or not.
Definition at line 600 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_DTMF.
Referenced by create_addr_from_peer(), handle_request_invite(), process_sdp(), sip_alloc(), and sip_dtmfmode().
00601 { 00602 ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF); 00603 }
void ast_rtp_setdtmfcompensate | ( | struct ast_rtp * | rtp, | |
int | compensate | |||
) |
Compensate for devices that send RFC2833 packets all at once.
Definition at line 605 of file rtp.c.
References ast_set2_flag, and FLAG_DTMF_COMPENSATE.
Referenced by create_addr_from_peer(), handle_request_invite(), process_sdp(), and sip_alloc().
00606 { 00607 ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE); 00608 }
void ast_rtp_setnat | ( | struct ast_rtp * | rtp, | |
int | nat | |||
) |
Definition at line 590 of file rtp.c.
References ast_rtp::nat.
Referenced by __oh323_rtp_create(), do_setnat(), oh323_rtp_read(), and start_rtp().
void ast_rtp_setstun | ( | struct ast_rtp * | rtp, | |
int | stun_enable | |||
) |
Enable STUN capability.
Definition at line 610 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_STUN.
Referenced by gtalk_new().
00611 { 00612 ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN); 00613 }
int ast_rtp_settos | ( | struct ast_rtp * | rtp, | |
int | tos | |||
) |
Definition at line 1980 of file rtp.c.
References ast_log(), LOG_WARNING, and ast_rtp::s.
Referenced by __oh323_rtp_create(), and sip_alloc().
01981 { 01982 int res; 01983 01984 if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 01985 ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); 01986 return res; 01987 }
void ast_rtp_stop | ( | struct ast_rtp * | rtp | ) |
Definition at line 2029 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().
02030 { 02031 if (rtp->rtcp && rtp->rtcp->schedid > 0) { 02032 ast_sched_del(rtp->sched, rtp->rtcp->schedid); 02033 rtp->rtcp->schedid = -1; 02034 } 02035 02036 memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); 02037 memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); 02038 if (rtp->rtcp) { 02039 memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr)); 02040 memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port)); 02041 } 02042 02043 ast_clear_flag(rtp, FLAG_P2P_SENT_MARK); 02044 }
void ast_rtp_stun_request | ( | struct ast_rtp * | rtp, | |
struct sockaddr_in * | suggestion, | |||
const char * | username | |||
) |
Definition at line 402 of file rtp.c.
References append_attr_string(), stun_attr::attr, ast_rtp::s, STUN_BINDREQ, stun_req_id(), stun_send(), and STUN_USERNAME.
Referenced by gtalk_update_stun().
00403 { 00404 struct stun_header *req; 00405 unsigned char reqdata[1024]; 00406 int reqlen, reqleft; 00407 struct stun_attr *attr; 00408 00409 req = (struct stun_header *)reqdata; 00410 stun_req_id(req); 00411 reqlen = 0; 00412 reqleft = sizeof(reqdata) - sizeof(struct stun_header); 00413 req->msgtype = 0; 00414 req->msglen = 0; 00415 attr = (struct stun_attr *)req->ies; 00416 if (username) 00417 append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft); 00418 req->msglen = htons(reqlen); 00419 req->msgtype = htons(STUN_BINDREQ); 00420 stun_send(rtp->s, suggestion, req); 00421 }
Definition at line 2708 of file rtp.c.
References ast_codec_pref_getsize(), AST_FORMAT_SPEEX, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frdup(), ast_frfree(), ast_getformatname(), ast_log(), ast_rtp_lookup_code(), ast_rtp_raw_write(), ast_smoother_feed, ast_smoother_feed_be, AST_SMOOTHER_FLAG_BE, ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_smoother_set_flags(), ast_smoother_test_flag(), ast_frame::datalen, f, fmt, ast_frame::frametype, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, option_debug, ast_rtp::pref, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.
Referenced by gtalk_write(), mgcp_write(), oh323_write(), sip_write(), and skinny_write().
02709 { 02710 struct ast_frame *f; 02711 int codec; 02712 int hdrlen = 12; 02713 int subclass; 02714 02715 02716 /* If we have no peer, return immediately */ 02717 if (!rtp->them.sin_addr.s_addr) 02718 return 0; 02719 02720 /* If there is no data length, return immediately */ 02721 if (!_f->datalen) 02722 return 0; 02723 02724 /* Make sure we have enough space for RTP header */ 02725 if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { 02726 ast_log(LOG_WARNING, "RTP can only send voice and video\n"); 02727 return -1; 02728 } 02729 02730 subclass = _f->subclass; 02731 if (_f->frametype == AST_FRAME_VIDEO) 02732 subclass &= ~0x1; 02733 02734 codec = ast_rtp_lookup_code(rtp, 1, subclass); 02735 if (codec < 0) { 02736 ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); 02737 return -1; 02738 } 02739 02740 if (rtp->lasttxformat != subclass) { 02741 /* New format, reset the smoother */ 02742 if (option_debug) 02743 ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); 02744 rtp->lasttxformat = subclass; 02745 if (rtp->smoother) 02746 ast_smoother_free(rtp->smoother); 02747 rtp->smoother = NULL; 02748 } 02749 02750 if (!rtp->smoother && subclass != AST_FORMAT_SPEEX) { 02751 struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass); 02752 if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */ 02753 if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) { 02754 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)); 02755 return -1; 02756 } 02757 if (fmt.flags) 02758 ast_smoother_set_flags(rtp->smoother, fmt.flags); 02759 if (option_debug) 02760 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)); 02761 } 02762 } 02763 if (rtp->smoother) { 02764 if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) { 02765 ast_smoother_feed_be(rtp->smoother, _f); 02766 } else { 02767 ast_smoother_feed(rtp->smoother, _f); 02768 } 02769 02770 while((f = ast_smoother_read(rtp->smoother)) && (f->data)) 02771 ast_rtp_raw_write(rtp, f, codec); 02772 } else { 02773 /* Don't buffer outgoing frames; send them one-per-packet: */ 02774 if (_f->offset < hdrlen) { 02775 f = ast_frdup(_f); 02776 } else { 02777 f = _f; 02778 } 02779 if (f->data) 02780 ast_rtp_raw_write(rtp, f, codec); 02781 if (f != _f) 02782 ast_frfree(f); 02783 } 02784 02785 return 0; 02786 }