Supports RTP and RTCP with Symmetric RTP support for NAT traversal. More...
#include "asterisk/network.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 |
| This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem. More... | |
| struct | ast_rtp_quality |
| RTCP quality report storage. More... | |
| struct | rtpPayloadType |
| The value of each payload format mapping: More... | |
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 |
| #define | RED_MAX_GENERATION 5 |
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) } |
| enum | ast_rtp_qos_vars { AST_RTP_TXCOUNT, AST_RTP_RXCOUNT, AST_RTP_TXJITTER, AST_RTP_RXJITTER, AST_RTP_RXPLOSS, AST_RTP_TXPLOSS, AST_RTP_RTT } |
| Variables used in ast_rtcp_get function. More... | |
| enum | ast_rtp_quality_type { RTPQOS_SUMMARY = 0, RTPQOS_JITTER, RTPQOS_LOSS, RTPQOS_RTT } |
Functions | |
| int | ast_rtcp_fd (struct ast_rtp *rtp) |
| struct 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) |
| The RTP bridge. | |
| void | ast_rtp_change_source (struct ast_rtp *rtp) |
| Indicate that we need to set the marker bit and change the ssrc. | |
| int | ast_rtp_codec_getformat (int pt) |
| get format from predefined dynamic payload format | |
| struct ast_codec_pref * | ast_rtp_codec_getpref (struct ast_rtp *rtp) |
| Get codec preference. | |
| void | ast_rtp_codec_setpref (struct ast_rtp *rtp, struct ast_codec_pref *prefs) |
| Set codec preference. | |
| void | ast_rtp_destroy (struct ast_rtp *rtp) |
| int | ast_rtp_early_bridge (struct ast_channel *c0, struct ast_channel *c1) |
| 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) |
| struct 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) |
| int | ast_rtp_get_qos (struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen) |
| Get QOS stats on a RTP channel. | |
| unsigned int | ast_rtp_get_qosvalue (struct ast_rtp *rtp, enum ast_rtp_qos_vars value) |
| Return RTP and RTCP QoS values. | |
| char * | ast_rtp_get_quality (struct ast_rtp *rtp, struct ast_rtp_quality *qual, enum ast_rtp_quality_type qtype) |
| 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): | |
| struct rtpPayloadType | ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt) |
| Mapping between RTP payload format codes and Asterisk codes: | |
| unsigned int | ast_rtp_lookup_sample_rate (int isAstFormat, int code) |
| Get the sample rate associated with known RTP payload types. | |
| int | ast_rtp_make_compatible (struct ast_channel *dest, struct ast_channel *src, int media) |
| struct 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) |
| Indicate that we need to set the marker bit. | |
| 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. | |
| int | ast_rtp_proto_register (struct ast_rtp_protocol *proto) |
| Register an RTP channel client. | |
| void | ast_rtp_proto_unregister (struct ast_rtp_protocol *proto) |
| Unregister an RTP channel client. | |
| 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. | |
| struct 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) |
| int | ast_rtp_senddigit_end_with_duration (struct ast_rtp *rtp, char digit, unsigned int duration) |
| Send end packets for DTMF. | |
| 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) |
| Set payload type to a known MIME media type for a codec. | |
| int | ast_rtp_set_rtpmap_type_rate (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype, enum ast_rtp_options options, unsigned int sample_rate) |
| Set payload type to a known MIME media type for a codec with a specific sample rate. | |
| 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_set_vars (struct ast_channel *chan, struct ast_rtp *rtp) |
| Set RTPAUDIOQOS(...) variables on a channel when it is being hung up. | |
| 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) |
| int | ast_rtp_setqos (struct ast_rtp *rtp, int tos, int cos, char *desc) |
| void | ast_rtp_setstun (struct ast_rtp *rtp, int stun_enable) |
| Enable STUN capability. | |
| void | ast_rtp_stop (struct ast_rtp *rtp) |
| void | ast_rtp_stun_request (struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username) |
| Send STUN request for an RTP socket Deprecated, this is just a wrapper for ast_rtp_stun_request() | |
| 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) |
| int | ast_stun_request (int s, struct sockaddr_in *dst, const char *username, struct sockaddr_in *answer) |
| Generic STUN request send a generic stun request to the server specified. | |
| void | red_buffer_t140 (struct ast_rtp *rtp, struct ast_frame *f) |
| Buffer t.140 data. | |
| int | rtp_red_init (struct ast_rtp *rtp, int ti, int *pt, int num_gen) |
| Initalize t.140 redudancy. | |
Detailed Description
Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
RTP is defined in RFC 3550.
Definition in file rtp.h.
Define Documentation
| #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(), add_sdp(), ast_rtp_read(), ast_rtp_senddigit_begin(), bridge_p2p_rtp_write(), check_peer_ok(), 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 |
Maxmum number of payload defintions for a RTP session
Definition at line 52 of file rtp.h.
Referenced by ast_rtp_codec_getformat(), 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_rate(), ast_rtp_unset_m_type(), and process_sdp_a_audio().
| #define RED_MAX_GENERATION 5 |
T.140 Redundancy Maxium number of generations
Definition at line 55 of file rtp.h.
Referenced by process_sdp_a_text().
Typedef Documentation
| typedef int(* ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data) |
Enumeration Type Documentation
| enum ast_rtp_get_result |
- Enumerator:
Definition at line 63 of file rtp.h.
{
/*! Failed to find the RTP structure */
AST_RTP_GET_FAILED = 0,
/*! RTP structure exists but true native bridge can not occur so try partial */
AST_RTP_TRY_PARTIAL,
/*! RTP structure exists and native bridge can occur */
AST_RTP_TRY_NATIVE,
};
| enum ast_rtp_options |
Definition at line 59 of file rtp.h.
{
AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
};
| enum ast_rtp_qos_vars |
| enum ast_rtp_quality_type |
Definition at line 109 of file rtp.h.
{
RTPQOS_SUMMARY = 0,
RTPQOS_JITTER,
RTPQOS_LOSS,
RTPQOS_RTT
};
Function Documentation
| int ast_rtcp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 722 of file rtp.c.
References ast_rtp::rtcp, and ast_rtcp::s.
Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), jingle_new(), sip_new(), start_rtp(), and unistim_new().
Definition at line 1182 of file rtp.c.
References ast_rtcp::accumulated_transit, ast_rtcp::altthem, ast_assert, AST_CONTROL_VIDUPDATE, ast_debug, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), ast_frame::datalen, errno, EVENT_FLAG_REPORTING, ast_rtp::f, f, ast_frame::frametype, len(), LOG_DEBUG, LOG_WARNING, ast_frame::mallocd, manager_event, ast_rtcp::maxrtt, ast_rtcp::minrtt, ast_rtp::nat, normdev_compute(), ast_rtcp::normdevrtt, option_debug, ast_rtcp::reported_jitter, ast_rtcp::reported_jitter_count, ast_rtcp::reported_lost, ast_rtcp::reported_maxjitter, ast_rtcp::reported_maxlost, ast_rtcp::reported_minjitter, ast_rtcp::reported_minlost, ast_rtcp::reported_normdev_jitter, ast_rtcp::reported_normdev_lost, ast_rtcp::reported_stdev_jitter, ast_rtcp::reported_stdev_lost, ast_rtp::rtcp, rtcp_debug_test_addr(), ast_rtcp::rtcp_info, RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, ast_rtcp::rtt, ast_rtcp::rtt_count, ast_rtcp::rxlsr, ast_rtcp::s, ast_frame::samples, ast_rtcp::soc, ast_rtcp::spc, ast_frame::src, stddev_compute(), ast_rtcp::stdevrtt, ast_frame::subclass, ast_rtcp::them, ast_rtcp::themrxlsr, and timeval2ntp().
Referenced by oh323_read(), sip_rtp_read(), skinny_rtp_read(), and unistim_rtp_read().
{
socklen_t len;
int position, i, packetwords;
int res;
struct sockaddr_in sock_in;
unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
unsigned int *rtcpheader;
int pt;
struct timeval now;
unsigned int length;
int rc;
double rttsec;
uint64_t rtt = 0;
unsigned int dlsr;
unsigned int lsr;
unsigned int msw;
unsigned int lsw;
unsigned int comp;
struct ast_frame *f = &ast_null_frame;
double reported_jitter;
double reported_normdev_jitter_current;
double normdevrtt_current;
double reported_lost;
double reported_normdev_lost_current;
if (!rtp || !rtp->rtcp)
return &ast_null_frame;
len = sizeof(sock_in);
res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
0, (struct sockaddr *)&sock_in, &len);
rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
if (res < 0) {
ast_assert(errno != EBADF);
if (errno != EAGAIN) {
ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno));
return NULL;
}
return &ast_null_frame;
}
packetwords = res / 4;
if (rtp->nat) {
/* Send to whoever sent to us */
if (((rtp->rtcp->them.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
(rtp->rtcp->them.sin_port != sock_in.sin_port)) &&
((rtp->rtcp->altthem.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
(rtp->rtcp->altthem.sin_port != sock_in.sin_port))) {
memcpy(&rtp->rtcp->them, &sock_in, sizeof(rtp->rtcp->them));
if (option_debug || rtpdebug)
ast_debug(0, "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));
}
}
ast_debug(1, "Got RTCP report of %d bytes\n", res);
/* Process a compound packet */
position = 0;
while (position < packetwords) {
i = position;
length = ntohl(rtcpheader[i]);
pt = (length & 0xff0000) >> 16;
rc = (length & 0x1f000000) >> 24;
length &= 0xffff;
if ((i + length) > packetwords) {
if (option_debug || rtpdebug)
ast_log(LOG_DEBUG, "RTCP Read too short\n");
return &ast_null_frame;
}
if (rtcp_debug_test_addr(&sock_in)) {
ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port));
ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
ast_verbose("Reception reports: %d\n", rc);
ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
}
i += 2; /* Advance past header and ssrc */
switch (pt) {
case RTCP_PT_SR:
gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
if (rtcp_debug_test_addr(&sock_in)) {
ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
}
i += 5;
if (rc < 1)
break;
/* Intentional fall through */
case RTCP_PT_RR:
/* Don't handle multiple reception reports (rc > 1) yet */
/* Calculate RTT per RFC */
gettimeofday(&now, NULL);
timeval2ntp(now, &msw, &lsw);
if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */
comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
lsr = ntohl(rtcpheader[i + 4]);
dlsr = ntohl(rtcpheader[i + 5]);
rtt = comp - lsr - dlsr;
/* Convert end to end delay to usec (keeping the calculation in 64bit space)
sess->ee_delay = (eedelay * 1000) / 65536; */
if (rtt < 4294) {
rtt = (rtt * 1000000) >> 16;
} else {
rtt = (rtt * 1000) >> 16;
rtt *= 1000;
}
rtt = rtt / 1000.;
rttsec = rtt / 1000.;
rtp->rtcp->rtt = rttsec;
if (comp - dlsr >= lsr) {
rtp->rtcp->accumulated_transit += rttsec;
if (rtp->rtcp->rtt_count == 0)
rtp->rtcp->minrtt = rttsec;
if (rtp->rtcp->maxrtt<rttsec)
rtp->rtcp->maxrtt = rttsec;
if (rtp->rtcp->minrtt>rttsec)
rtp->rtcp->minrtt = rttsec;
normdevrtt_current = normdev_compute(rtp->rtcp->normdevrtt, rttsec, rtp->rtcp->rtt_count);
rtp->rtcp->stdevrtt = stddev_compute(rtp->rtcp->stdevrtt, rttsec, rtp->rtcp->normdevrtt, normdevrtt_current, rtp->rtcp->rtt_count);
rtp->rtcp->normdevrtt = normdevrtt_current;
rtp->rtcp->rtt_count++;
} else if (rtcp_debug_test_addr(&sock_in)) {
ast_verbose("Internal RTCP NTP clock skew detected: "
"lsr=%u, now=%u, dlsr=%u (%d:%03dms), "
"diff=%d\n",
lsr, comp, dlsr, dlsr / 65536,
(dlsr % 65536) * 1000 / 65536,
dlsr - (comp - lsr));
}
}
rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
reported_jitter = (double) rtp->rtcp->reported_jitter;
if (rtp->rtcp->reported_jitter_count == 0)
rtp->rtcp->reported_minjitter = reported_jitter;
if (reported_jitter < rtp->rtcp->reported_minjitter)
rtp->rtcp->reported_minjitter = reported_jitter;
if (reported_jitter > rtp->rtcp->reported_maxjitter)
rtp->rtcp->reported_maxjitter = reported_jitter;
reported_normdev_jitter_current = normdev_compute(rtp->rtcp->reported_normdev_jitter, reported_jitter, rtp->rtcp->reported_jitter_count);
rtp->rtcp->reported_stdev_jitter = stddev_compute(rtp->rtcp->reported_stdev_jitter, reported_jitter, rtp->rtcp->reported_normdev_jitter, reported_normdev_jitter_current, rtp->rtcp->reported_jitter_count);
rtp->rtcp->reported_normdev_jitter = reported_normdev_jitter_current;
rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
reported_lost = (double) rtp->rtcp->reported_lost;
/* using same counter as for jitter */
if (rtp->rtcp->reported_jitter_count == 0)
rtp->rtcp->reported_minlost = reported_lost;
if (reported_lost < rtp->rtcp->reported_minlost)
rtp->rtcp->reported_minlost = reported_lost;
if (reported_lost > rtp->rtcp->reported_maxlost)
rtp->rtcp->reported_maxlost = reported_lost;
reported_normdev_lost_current = normdev_compute(rtp->rtcp->reported_normdev_lost, reported_lost, rtp->rtcp->reported_jitter_count);
rtp->rtcp->reported_stdev_lost = stddev_compute(rtp->rtcp->reported_stdev_lost, reported_lost, rtp->rtcp->reported_normdev_lost, reported_normdev_lost_current, rtp->rtcp->reported_jitter_count);
rtp->rtcp->reported_normdev_lost = reported_normdev_lost_current;
rtp->rtcp->reported_jitter_count++;
if (rtcp_debug_test_addr(&sock_in)) {
ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost);
ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
if (rtt)
ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt);
}
if (rtt) {
manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From: %s:%d\r\n"
"PT: %d(%s)\r\n"
"ReceptionReports: %d\r\n"
"SenderSSRC: %u\r\n"
"FractionLost: %ld\r\n"
"PacketsLost: %d\r\n"
"HighestSequence: %ld\r\n"
"SequenceNumberCycles: %ld\r\n"
"IAJitter: %u\r\n"
"LastSR: %lu.%010lu\r\n"
"DLSR: %4.4f(sec)\r\n"
"RTT: %llu(sec)\r\n",
ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port),
pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
rc,
rtcpheader[i + 1],
(((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
rtp->rtcp->reported_lost,
(long) (ntohl(rtcpheader[i + 2]) & 0xffff),
(long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
rtp->rtcp->reported_jitter,
(unsigned long) ntohl(rtcpheader[i + 4]) >> 16, ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
ntohl(rtcpheader[i + 5])/65536.0,
(unsigned long long)rtt);
} else {
manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From: %s:%d\r\n"
"PT: %d(%s)\r\n"
"ReceptionReports: %d\r\n"
"SenderSSRC: %u\r\n"
"FractionLost: %ld\r\n"
"PacketsLost: %d\r\n"
"HighestSequence: %ld\r\n"
"SequenceNumberCycles: %ld\r\n"
"IAJitter: %u\r\n"
"LastSR: %lu.%010lu\r\n"
"DLSR: %4.4f(sec)\r\n",
ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port),
pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
rc,
rtcpheader[i + 1],
(((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
rtp->rtcp->reported_lost,
(long) (ntohl(rtcpheader[i + 2]) & 0xffff),
(long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
rtp->rtcp->reported_jitter,
(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,
((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
ntohl(rtcpheader[i + 5])/65536.0);
}
break;
case RTCP_PT_FUR:
if (rtcp_debug_test_addr(&sock_in))
ast_verbose("Received an RTCP Fast Update Request\n");
rtp->f.frametype = AST_FRAME_CONTROL;
rtp->f.subclass = AST_CONTROL_VIDUPDATE;
rtp->f.datalen = 0;
rtp->f.samples = 0;
rtp->f.mallocd = 0;
rtp->f.src = "RTP";
f = &rtp->f;
break;
case RTCP_PT_SDES:
if (rtcp_debug_test_addr(&sock_in))
ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
break;
case RTCP_PT_BYE:
if (rtcp_debug_test_addr(&sock_in))
ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
break;
default:
ast_debug(1, "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));
break;
}
position += (length + 1);
}
rtp->rtcp->rtcp_info = 1;
return f;
}
| 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 3357 of file rtp.c.
References ast_rtcp_write(), ast_rtp::data, ast_rtp::rtcp, and ast_rtcp::sendfur.
| size_t ast_rtp_alloc_size | ( | void | ) |
Get the amount of space required to hold an RTP session.
- Returns:
- number of bytes required
Definition at line 496 of file rtp.c.
Referenced by process_sdp().
{
return sizeof(struct ast_rtp);
}
| int ast_rtp_bridge | ( | struct ast_channel * | c0, |
| struct ast_channel * | c1, | ||
| int | flags, | ||
| struct ast_frame ** | fo, | ||
| struct ast_channel ** | rc, | ||
| int | timeoutms | ||
| ) |
The RTP bridge.
Definition at line 4456 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_check_hangup(), ast_codec_pref_getsize(), ast_debug, ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_set_flag, ast_test_flag, ast_verb, 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_trtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, ast_rtp::pref, ast_channel::rawreadformat, ast_channel::rawwriteformat, ast_channel_tech::send_digit_begin, ast_channel::tech, and ast_channel::tech_pvt.
{
struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */
struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */
struct ast_rtp *tp0 = NULL, *tp1 = NULL; /* Text RTP channels */
struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED, text_p0_res = AST_RTP_GET_FAILED;
enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED, text_p1_res = AST_RTP_GET_FAILED;
enum ast_bridge_result res = AST_BRIDGE_FAILED;
int codec0 = 0, codec1 = 0;
void *pvt0 = NULL, *pvt1 = NULL;
/* Lock channels */
ast_channel_lock(c0);
while (ast_channel_trylock(c1)) {
ast_channel_unlock(c0);
usleep(1);
ast_channel_lock(c0);
}
/* Ensure neither channel got hungup during lock avoidance */
if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
/* Find channel driver interfaces */
if (!(pr0 = get_proto(c0))) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
if (!(pr1 = get_proto(c1))) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
/* Get channel specific interface structures */
pvt0 = c0->tech_pvt;
pvt1 = c1->tech_pvt;
/* Get audio and video interface (if native bridge is possible) */
audio_p0_res = pr0->get_rtp_info(c0, &p0);
video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
text_p0_res = pr0->get_trtp_info ? pr0->get_trtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
audio_p1_res = pr1->get_rtp_info(c1, &p1);
video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
text_p1_res = pr1->get_trtp_info ? pr1->get_trtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
/* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
audio_p0_res = AST_RTP_GET_FAILED;
if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
audio_p1_res = AST_RTP_GET_FAILED;
/* Check if a bridge is possible (partial/native) */
if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
/* Somebody doesn't want to play... */
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
/* If we need to feed DTMF frames into the core then only do a partial native bridge */
if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
audio_p0_res = AST_RTP_TRY_PARTIAL;
}
if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
audio_p1_res = AST_RTP_TRY_PARTIAL;
}
/* If both sides are not using the same method of DTMF transmission
* (ie: one is RFC2833, other is INFO... then we can not do direct media.
* --------------------------------------------------
* | DTMF Mode | HAS_DTMF | Accepts Begin Frames |
* |-----------|------------|-----------------------|
* | Inband | False | True |
* | RFC2833 | True | True |
* | SIP INFO | False | False |
* --------------------------------------------------
* However, if DTMF from both channels is being monitored by the core, then
* we can still do packet-to-packet bridging, because passing through the
* core will handle DTMF mode translation.
*/
if ((ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
(!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
audio_p0_res = AST_RTP_TRY_PARTIAL;
audio_p1_res = AST_RTP_TRY_PARTIAL;
}
/* If we need to feed frames into the core don't do a P2P bridge */
if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) ||
(audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) {
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
/* Get codecs from both sides */
codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
if (codec0 && codec1 && !(codec0 & codec1)) {
/* Hey, we can't do native bridging if both parties speak different codecs */
ast_debug(3, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
/* If either side can only do a partial bridge, then don't try for a true native bridge */
if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
struct ast_format_list fmt0, fmt1;
/* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
ast_debug(1, "Cannot packet2packet bridge - raw formats are incompatible\n");
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
/* They must also be using the same packetization */
fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
if (fmt0.cur_ms != fmt1.cur_ms) {
ast_debug(1, "Cannot packet2packet bridge - packetization settings prevent it\n");
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
ast_verb(3, "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
} else {
ast_verb(3, "Native bridging %s and %s\n", c0->name, c1->name);
res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, tp0, tp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
}
return res;
}
| void ast_rtp_change_source | ( | struct ast_rtp * | rtp | ) |
Indicate that we need to set the marker bit and change the ssrc.
Definition at line 2692 of file rtp.c.
References ast_debug, ast_random(), ast_rtp::set_marker_bit, and ast_rtp::ssrc.
Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), and skinny_indicate().
{
if (rtp) {
unsigned int ssrc = ast_random();
rtp->set_marker_bit = 1;
ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
rtp->ssrc = ssrc;
}
}
| int ast_rtp_codec_getformat | ( | int | pt | ) |
get format from predefined dynamic payload format
Definition at line 3837 of file rtp.c.
References rtpPayloadType::code, and MAX_RTP_PT.
Referenced by process_sdp_a_audio().
{
if (pt < 0 || pt >= MAX_RTP_PT)
return 0; /* bogus payload type */
if (static_RTP_PT[pt].isAstFormat)
return static_RTP_PT[pt].code;
else
return 0;
}
| struct ast_codec_pref* ast_rtp_codec_getpref | ( | struct ast_rtp * | rtp | ) | [read] |
Get codec preference.
Definition at line 3832 of file rtp.c.
References ast_rtp::pref.
Referenced by add_codec_to_sdp(), and process_sdp_a_audio().
{
return &rtp->pref;
}
| void ast_rtp_codec_setpref | ( | struct ast_rtp * | rtp, |
| struct ast_codec_pref * | prefs | ||
| ) |
Set codec preference.
Definition at line 3786 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_peer_ok(), create_addr_from_peer(), gtalk_new(), jingle_new(), process_sdp_a_audio(), register_verify(), set_peer_capabilities(), sip_alloc(), start_rtp(), and transmit_response_with_sdp().
{
struct ast_format_list current_format_old, current_format_new;
/* if no packets have been sent through this session yet, then
* changing preferences does not require any extra work
*/
if (rtp->lasttxformat == 0) {
rtp->pref = *prefs;
return;
}
current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
rtp->pref = *prefs;
current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
/* if the framing desired for the current format has changed, we may have to create
* or adjust the smoother for this session
*/
if ((current_format_new.inc_ms != 0) &&
(current_format_new.cur_ms != current_format_old.cur_ms)) {
int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms;
if (rtp->smoother) {
ast_smoother_reconfigure(rtp->smoother, new_size);
if (option_debug) {
ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size);
}
} else {
if (!(rtp->smoother = ast_smoother_new(new_size))) {
ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
return;
}
if (current_format_new.flags) {
ast_smoother_set_flags(rtp->smoother, current_format_new.flags);
}
if (option_debug) {
ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
}
}
}
}
| void ast_rtp_destroy | ( | struct ast_rtp * | rtp | ) |
Destroy RTP session
Definition at line 3105 of file rtp.c.
References ast_free, ast_io_remove(), ast_mutex_destroy(), AST_SCHED_DEL, ast_smoother_free(), ast_verbose(), EVENT_FLAG_REPORTING, ast_rtcp::expected_prior, ast_rtp::io, ast_rtp::ioid, manager_event, 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_peer_ok(), cleanup_connection(), create_addr_from_peer(), destroy_endpoint(), gtalk_free_pvt(), jingle_free_pvt(), mgcp_hangup(), oh323_alloc(), skinny_hangup(), start_rtp(), unalloc_sub(), and unistim_hangup().
{
if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
/*Print some info on the call here */
ast_verbose(" RTP-stats\n");
ast_verbose("* Our Receiver:\n");
ast_verbose(" SSRC: %u\n", rtp->themssrc);
ast_verbose(" Received packets: %u\n", rtp->rxcount);
ast_verbose(" Lost packets: %u\n", rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0);
ast_verbose(" Jitter: %.4f\n", rtp->rxjitter);
ast_verbose(" Transit: %.4f\n", rtp->rxtransit);
ast_verbose(" RR-count: %u\n", rtp->rtcp ? rtp->rtcp->rr_count : 0);
ast_verbose("* Our Sender:\n");
ast_verbose(" SSRC: %u\n", rtp->ssrc);
ast_verbose(" Sent packets: %u\n", rtp->txcount);
ast_verbose(" Lost packets: %u\n", rtp->rtcp ? rtp->rtcp->reported_lost : 0);
ast_verbose(" Jitter: %u\n", rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int)65536.0) : 0);
ast_verbose(" SR-count: %u\n", rtp->rtcp ? rtp->rtcp->sr_count : 0);
ast_verbose(" RTT: %f\n", rtp->rtcp ? rtp->rtcp->rtt : 0);
}
manager_event(EVENT_FLAG_REPORTING, "RTPReceiverStat", "SSRC: %u\r\n"
"ReceivedPackets: %u\r\n"
"LostPackets: %u\r\n"
"Jitter: %.4f\r\n"
"Transit: %.4f\r\n"
"RRCount: %u\r\n",
rtp->themssrc,
rtp->rxcount,
rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0,
rtp->rxjitter,
rtp->rxtransit,
rtp->rtcp ? rtp->rtcp->rr_count : 0);
manager_event(EVENT_FLAG_REPORTING, "RTPSenderStat", "SSRC: %u\r\n"
"SentPackets: %u\r\n"
"LostPackets: %u\r\n"
"Jitter: %u\r\n"
"SRCount: %u\r\n"
"RTT: %f\r\n",
rtp->ssrc,
rtp->txcount,
rtp->rtcp ? rtp->rtcp->reported_lost : 0,
rtp->rtcp ? rtp->rtcp->reported_jitter : 0,
rtp->rtcp ? rtp->rtcp->sr_count : 0,
rtp->rtcp ? rtp->rtcp->rtt : 0);
if (rtp->smoother)
ast_smoother_free(rtp->smoother);
if (rtp->ioid)
ast_io_remove(rtp->io, rtp->ioid);
if (rtp->s > -1)
close(rtp->s);
if (rtp->rtcp) {
AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
close(rtp->rtcp->s);
ast_free(rtp->rtcp);
rtp->rtcp=NULL;
}
#ifdef P2P_INTENSE
ast_mutex_destroy(&rtp->bridge_lock);
#endif
ast_free(rtp);
}
| int ast_rtp_early_bridge | ( | struct ast_channel * | c0, |
| struct ast_channel * | c1 | ||
| ) |
If possible, create an early bridge directly between the devices without having to send a re-invite later.
Definition at line 2114 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_debug, 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_trtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, and ast_rtp_protocol::set_rtp_peer.
{
struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */
struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED;
int srccodec, destcodec, nat_active = 0;
/* Lock channels */
ast_channel_lock(c0);
if (c1) {
while (ast_channel_trylock(c1)) {
ast_channel_unlock(c0);
usleep(1);
ast_channel_lock(c0);
}
}
/* Find channel driver interfaces */
destpr = get_proto(c0);
if (c1)
srcpr = get_proto(c1);
if (!destpr) {
ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c0->name);
ast_channel_unlock(c0);
if (c1)
ast_channel_unlock(c1);
return -1;
}
if (!srcpr) {
ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c1 ? c1->name : "<unspecified>");
ast_channel_unlock(c0);
if (c1)
ast_channel_unlock(c1);
return -1;
}
/* Get audio, video and text interface (if native bridge is possible) */
audio_dest_res = destpr->get_rtp_info(c0, &destp);
video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(c0, &vdestp) : AST_RTP_GET_FAILED;
text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(c0, &tdestp) : AST_RTP_GET_FAILED;
if (srcpr) {
audio_src_res = srcpr->get_rtp_info(c1, &srcp);
video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(c1, &vsrcp) : AST_RTP_GET_FAILED;
text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED;
}
/* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {
/* Somebody doesn't want to play... */
ast_channel_unlock(c0);
if (c1)
ast_channel_unlock(c1);
return -1;
}
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)
srccodec = srcpr->get_codec(c1);
else
srccodec = 0;
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)
destcodec = destpr->get_codec(c0);
else
destcodec = 0;
/* Ensure we have at least one matching codec */
if (srcp && !(srccodec & destcodec)) {
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return 0;
}
/* Consider empty media as non-existent */
if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
srcp = NULL;
if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
nat_active = 1;
/* Bridge media early */
if (destpr->set_rtp_peer(c0, srcp, vsrcp, tsrcp, srccodec, nat_active))
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
ast_channel_unlock(c0);
if (c1)
ast_channel_unlock(c1);
ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
return 0;
}
| int ast_rtp_fd | ( | struct ast_rtp * | rtp | ) |
Definition at line 717 of file rtp.c.
References ast_rtp::s.
Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), jingle_new(), mgcp_new(), p2p_callback_disable(), sip_new(), skinny_new(), start_rtp(), and unistim_new().
{
return rtp->s;
}
Definition at line 2746 of file rtp.c.
References ast_rtp::bridged, rtp_bridge_lock(), and rtp_bridge_unlock().
Referenced by __sip_destroy(), ast_rtp_read(), and dialog_needdestroy().
{
struct ast_rtp *bridged = NULL;
rtp_bridge_lock(rtp);
bridged = rtp->bridged;
rtp_bridge_unlock(rtp);
return bridged;
}
| 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 2362 of file rtp.c.
References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().
Referenced by gtalk_is_answered(), gtalk_newcall(), and process_sdp().
{
int pt;
rtp_bridge_lock(rtp);
*astFormats = *nonAstFormats = 0;
for (pt = 0; pt < MAX_RTP_PT; ++pt) {
if (rtp->current_RTP_PT[pt].isAstFormat) {
*astFormats |= rtp->current_RTP_PT[pt].code;
} else {
*nonAstFormats |= rtp->current_RTP_PT[pt].code;
}
}
rtp_bridge_unlock(rtp);
}
| int ast_rtp_get_peer | ( | struct ast_rtp * | rtp, |
| struct sockaddr_in * | them | ||
| ) |
Definition at line 2728 of file rtp.c.
References ast_rtp::them.
Referenced by acf_channel_read(), add_sdp(), bridge_native_loop(), check_rtp_timeout(), gtalk_update_stun(), oh323_set_rtp_peer(), process_sdp(), sip_set_rtp_peer(), skinny_set_rtp_peer(), and transmit_modify_with_sdp().
| int ast_rtp_get_qos | ( | struct ast_rtp * | rtp, |
| const char * | qos, | ||
| char * | buf, | ||
| unsigned int | buflen | ||
| ) |
Get QOS stats on a RTP channel.
- Since:
- 1.6.1
Definition at line 2867 of file rtp.c.
References __ast_rtp_get_qos().
Referenced by acf_channel_read().
{
double value;
int found;
value = __ast_rtp_get_qos(rtp, qos, &found);
if (!found)
return -1;
snprintf(buf, buflen, "%.0lf", value);
return 0;
}
| unsigned int ast_rtp_get_qosvalue | ( | struct ast_rtp * | rtp, |
| enum ast_rtp_qos_vars | value | ||
| ) |
Return RTP and RTCP QoS values.
- Since:
- 1.6.1
Get QoS values from RTP and RTCP data (used in "sip show channelstats")
Definition at line 2801 of file rtp.c.
References ast_log(), AST_RTP_RTT, AST_RTP_RXCOUNT, AST_RTP_RXJITTER, AST_RTP_RXPLOSS, AST_RTP_TXCOUNT, AST_RTP_TXJITTER, AST_RTP_TXPLOSS, ast_rtcp::expected_prior, LOG_DEBUG, option_debug, ast_rtcp::received_prior, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, ast_rtcp::rtt, ast_rtp::rxcount, ast_rtp::rxjitter, and ast_rtp::txcount.
Referenced by show_chanstats_cb().
{
if (rtp == NULL) {
if (option_debug > 1)
ast_log(LOG_DEBUG, "NO RTP Structure? Kidding me? \n");
return 0;
}
if (option_debug > 1 && rtp->rtcp == NULL) {
ast_log(LOG_DEBUG, "NO RTCP structure. Maybe in RTP p2p bridging mode? \n");
}
switch (value) {
case AST_RTP_TXCOUNT:
return (unsigned int) rtp->txcount;
case AST_RTP_RXCOUNT:
return (unsigned int) rtp->rxcount;
case AST_RTP_TXJITTER:
return (unsigned int) (rtp->rxjitter * 1000.0);
case AST_RTP_RXJITTER:
return (unsigned int) (rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int) 65536.0) : 0);
case AST_RTP_RXPLOSS:
return rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0;
case AST_RTP_TXPLOSS:
return rtp->rtcp ? rtp->rtcp->reported_lost : 0;
case AST_RTP_RTT:
return (unsigned int) (rtp->rtcp ? (rtp->rtcp->rtt * 100) : 0);
}
return 0; /* To make the compiler happy */
}
| char* ast_rtp_get_quality | ( | struct ast_rtp * | rtp, |
| struct ast_rtp_quality * | qual, | ||
| enum ast_rtp_quality_type | qtype | ||
| ) |
Return RTCP quality string.
- Parameters:
-
rtp An rtp structure to get qos information about. qual An (optional) rtp quality structure that will be filled with the quality information described in the ast_rtp_quality structure. This structure is not dependent on any qtype, so a call for any type of information would yield the same results because ast_rtp_quality is not a data type specific to any qos type. qtype The quality type you'd like, default should be RTPQOS_SUMMARY which returns basic information about the call. The return from RTPQOS_SUMMARY is basically ast_rtp_quality in a string. The other types are RTPQOS_JITTER, RTPQOS_LOSS and RTPQOS_RTT which will return more specific statistics.
- Version:
- 1.6.1 added qtype parameter
Definition at line 3074 of file rtp.c.
References __ast_rtp_get_quality(), __ast_rtp_get_quality_jitter(), __ast_rtp_get_quality_loss(), __ast_rtp_get_quality_rtt(), 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::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, RTPQOS_JITTER, RTPQOS_LOSS, RTPQOS_RTT, RTPQOS_SUMMARY, 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(), ast_rtp_set_vars(), handle_request_bye(), and sip_hangup().
{
if (qual && rtp) {
qual->local_ssrc = rtp->ssrc;
qual->local_jitter = rtp->rxjitter;
qual->local_count = rtp->rxcount;
qual->remote_ssrc = rtp->themssrc;
qual->remote_count = rtp->txcount;
if (rtp->rtcp) {
qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
qual->remote_lostpackets = rtp->rtcp->reported_lost;
qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
qual->rtt = rtp->rtcp->rtt;
}
}
switch (qtype) {
case RTPQOS_SUMMARY:
return __ast_rtp_get_quality(rtp);
case RTPQOS_JITTER:
return __ast_rtp_get_quality_jitter(rtp);
case RTPQOS_LOSS:
return __ast_rtp_get_quality_loss(rtp);
case RTPQOS_RTT:
return __ast_rtp_get_quality_rtt(rtp);
}
return NULL;
}
| int ast_rtp_get_rtpholdtimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp hold timeout.
Definition at line 777 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by check_rtp_timeout().
{
if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
return 0;
return rtp->rtpholdtimeout;
}
| int ast_rtp_get_rtpkeepalive | ( | struct ast_rtp * | rtp | ) |
Get RTP keepalive interval.
Definition at line 785 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by check_rtp_timeout().
{
return rtp->rtpkeepalive;
}
| int ast_rtp_get_rtptimeout | ( | struct ast_rtp * | rtp | ) |
Get rtp timeout.
Definition at line 769 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by check_rtp_timeout().
{
if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
return 0;
return rtp->rtptimeout;
}
| void ast_rtp_get_us | ( | struct ast_rtp * | rtp, |
| struct sockaddr_in * | us | ||
| ) |
Definition at line 2741 of file rtp.c.
References ast_rtp::us.
Referenced by add_sdp(), external_rtp_create(), get_our_media_address(), gtalk_create_candidates(), handle_open_receive_channel_ack_message(), jingle_create_candidates(), oh323_set_rtp_peer(), skinny_set_rtp_peer(), and start_rtp().
| int ast_rtp_getnat | ( | struct ast_rtp * | rtp | ) |
Definition at line 805 of file rtp.c.
References ast_test_flag, and FLAG_NAT_ACTIVE.
Referenced by sip_get_rtp_peer().
{
return ast_test_flag(rtp, FLAG_NAT_ACTIVE);
}
| void ast_rtp_init | ( | void | ) |
Initialize the RTP system in Asterisk.
Definition at line 4877 of file rtp.c.
References __ast_rtp_reload(), and ast_cli_register_multiple().
Referenced by main().
{
ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
__ast_rtp_reload(0);
}
| 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 2403 of file rtp.c.
References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), rtp_bridge_unlock(), 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(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_rtp_sendcng(), ast_rtp_senddigit_begin(), ast_rtp_write(), bridge_p2p_rtp_write(), and start_rtp().
{
int pt = 0;
rtp_bridge_lock(rtp);
if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
code == rtp->rtp_lookup_code_cache_code) {
/* Use our cached mapping, to avoid the overhead of the loop below */
pt = rtp->rtp_lookup_code_cache_result;
rtp_bridge_unlock(rtp);
return pt;
}
/* Check the dynamic list first */
for (pt = 0; pt < MAX_RTP_PT; ++pt) {
if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
rtp->rtp_lookup_code_cache_code = code;
rtp->rtp_lookup_code_cache_result = pt;
rtp_bridge_unlock(rtp);
return pt;
}
}
/* Then the static list */
for (pt = 0; pt < MAX_RTP_PT; ++pt) {
if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
rtp->rtp_lookup_code_cache_code = code;
rtp->rtp_lookup_code_cache_result = pt;
rtp_bridge_unlock(rtp);
return pt;
}
}
rtp_bridge_unlock(rtp);
return -1;
}
| 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 2476 of file rtp.c.
References ast_copy_string(), ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, buf, format, len(), and name.
Referenced by process_sdp().
{
int format;
unsigned len;
char *end = buf;
char *start = buf;
if (!buf || !size)
return NULL;
snprintf(end, size, "0x%x (", capability);
len = strlen(end);
end += len;
size -= len;
start = end;
for (format = 1; format < AST_RTP_MAX; format <<= 1) {
if (capability & format) {
const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
snprintf(end, size, "%s|", name);
len = strlen(end);
end += len;
size -= len;
}
}
if (start == end)
ast_copy_string(start, "nothing)", size);
else if (size > 1)
*(end -1) = ')';
return buf;
}
| 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 2444 of file rtp.c.
References ARRAY_LEN, AST_FORMAT_G726_AAL2, AST_RTP_OPT_G726_NONSTANDARD, rtpPayloadType::isAstFormat, mimeTypes, mimeType::payloadType, and mimeType::subtype.
Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_rtp_lookup_mime_multiple(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
{
unsigned int i;
for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
if (isAstFormat &&
(code == AST_FORMAT_G726_AAL2) &&
(options & AST_RTP_OPT_G726_NONSTANDARD))
return "G726-32";
else
return mimeTypes[i].subtype;
}
}
return "";
}
| struct rtpPayloadType ast_rtp_lookup_pt | ( | struct ast_rtp * | rtp, |
| int | pt | ||
| ) | [read] |
Mapping between RTP payload format codes and Asterisk codes:
Definition at line 2381 of file rtp.c.
References rtpPayloadType::code, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().
Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), process_sdp_a_audio(), and setup_rtp_connection().
{
struct rtpPayloadType result;
result.isAstFormat = result.code = 0;
if (pt < 0 || pt >= MAX_RTP_PT)
return result; /* bogus payload type */
/* Start with negotiated codecs */
rtp_bridge_lock(rtp);
result = rtp->current_RTP_PT[pt];
rtp_bridge_unlock(rtp);
/* If it doesn't exist, check our static RTP type list, just in case */
if (!result.code)
result = static_RTP_PT[pt];
return result;
}
| unsigned int ast_rtp_lookup_sample_rate | ( | int | isAstFormat, |
| int | code | ||
| ) |
Get the sample rate associated with known RTP payload types.
- Parameters:
-
isAstFormat True if the value in the 'code' parameter is an AST_FORMAT value code Format code, either from AST_FORMAT list or from AST_RTP list
- Returns:
- the sample rate if the format was found, zero if it was not found
Definition at line 2463 of file rtp.c.
References ARRAY_LEN, rtpPayloadType::isAstFormat, mimeTypes, mimeType::payloadType, and mimeType::sample_rate.
Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_tcodec_to_sdp(), and add_vcodec_to_sdp().
{
unsigned int i;
for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
return mimeTypes[i].sample_rate;
}
}
return 0;
}
| int ast_rtp_make_compatible | ( | struct ast_channel * | dest, |
| struct ast_channel * | src, | ||
| int | media | ||
| ) |
Definition at line 2200 of file rtp.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_debug, 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_trtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, and ast_rtp_protocol::set_rtp_peer.
Referenced by dial_exec_full(), and do_forward().
{
struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */
struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED;
int srccodec, destcodec;
/* Lock channels */
ast_channel_lock(dest);
while (ast_channel_trylock(src)) {
ast_channel_unlock(dest);
usleep(1);
ast_channel_lock(dest);
}
/* Find channel driver interfaces */
if (!(destpr = get_proto(dest))) {
ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", dest->name);
ast_channel_unlock(dest);
ast_channel_unlock(src);
return 0;
}
if (!(srcpr = get_proto(src))) {
ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", src->name);
ast_channel_unlock(dest);
ast_channel_unlock(src);
return 0;
}
/* Get audio and video interface (if native bridge is possible) */
audio_dest_res = destpr->get_rtp_info(dest, &destp);
video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(dest, &tdestp) : AST_RTP_GET_FAILED;
audio_src_res = srcpr->get_rtp_info(src, &srcp);
video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(src, &tsrcp) : AST_RTP_GET_FAILED;
/* Ensure we have at least one matching codec */
if (srcpr->get_codec)
srccodec = srcpr->get_codec(src);
else
srccodec = 0;
if (destpr->get_codec)
destcodec = destpr->get_codec(dest);
else
destcodec = 0;
/* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
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)) {
/* Somebody doesn't want to play... */
ast_channel_unlock(dest);
ast_channel_unlock(src);
return 0;
}
ast_rtp_pt_copy(destp, srcp);
if (vdestp && vsrcp)
ast_rtp_pt_copy(vdestp, vsrcp);
if (tdestp && tsrcp)
ast_rtp_pt_copy(tdestp, tsrcp);
if (media) {
/* Bridge early */
if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
}
ast_channel_unlock(dest);
ast_channel_unlock(src);
ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
return 1;
}
| struct ast_rtp* ast_rtp_new | ( | struct sched_context * | sched, |
| struct io_context * | io, | ||
| int | rtcpenable, | ||
| int | callbackmode | ||
| ) | [read] |
Initializate a RTP session.
- Parameters:
-
sched io rtcpenable callbackmode
- Returns:
- A representation (structure) of an RTP session.
Definition at line 2671 of file rtp.c.
References ast_rtp_new_with_bindaddr().
{
struct in_addr ia;
memset(&ia, 0, sizeof(ia));
return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
}
| void ast_rtp_new_init | ( | struct ast_rtp * | rtp | ) |
Initialize a new RTP structure.
reload rtp configuration
Definition at line 2562 of file rtp.c.
References ast_mutex_init(), ast_random(), ast_set_flag, FLAG_HAS_DTMF, ast_rtp::seqno, ast_rtp::ssrc, STRICT_RTP_LEARN, STRICT_RTP_OPEN, ast_rtp::strict_rtp_state, ast_rtp::them, and ast_rtp::us.
Referenced by ast_rtp_new_with_bindaddr(), and process_sdp().
{
#ifdef P2P_INTENSE
ast_mutex_init(&rtp->bridge_lock);
#endif
rtp->them.sin_family = AF_INET;
rtp->us.sin_family = AF_INET;
rtp->ssrc = ast_random();
rtp->seqno = ast_random() & 0xffff;
ast_set_flag(rtp, FLAG_HAS_DTMF);
rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
}
| void ast_rtp_new_source | ( | struct ast_rtp * | rtp | ) |
Indicate that we need to set the marker bit.
Definition at line 2684 of file rtp.c.
References ast_debug, and ast_rtp::set_marker_bit.
Referenced by mgcp_indicate(), oh323_indicate(), sip_answer(), sip_indicate(), sip_write(), and skinny_indicate().
{
if (rtp) {
rtp->set_marker_bit = 1;
ast_debug(3, "Setting the marker bit due to a source update\n");
}
}
| struct ast_rtp* ast_rtp_new_with_bindaddr | ( | struct sched_context * | sched, |
| struct io_context * | io, | ||
| int | rtcpenable, | ||
| int | callbackmode, | ||
| struct in_addr | in | ||
| ) | [read] |
Initializate a RTP session using an in_addr structure.
This fuction gets called by ast_rtp_new().
- Parameters:
-
sched io rtcpenable callbackmode in
- Returns:
- A representation (structure) of an RTP session.
Definition at line 2576 of file rtp.c.
References ast_calloc, ast_free, ast_io_add(), AST_IO_IN, ast_log(), ast_random(), ast_rtcp_new(), ast_rtp_new_init(), ast_rtp_pt_default(), ast_set_flag, errno, FLAG_CALLBACK_MODE, io, ast_rtp::io, ast_rtp::ioid, LOG_ERROR, ast_rtp::rtcp, rtp_socket(), rtpread(), rtpstart, ast_rtcp::s, ast_rtp::s, sched, ast_rtp::sched, ast_rtcp::us, and ast_rtp::us.
Referenced by __oh323_rtp_create(), ast_rtp_new(), gtalk_alloc(), jingle_alloc(), sip_alloc(), and start_rtp().
{
struct ast_rtp *rtp;
int x;
int startplace;
if (!(rtp = ast_calloc(1, sizeof(*rtp))))
return NULL;
ast_rtp_new_init(rtp);
rtp->s = rtp_socket("RTP");
if (rtp->s < 0)
goto fail;
if (sched && rtcpenable) {
rtp->sched = sched;
rtp->rtcp = ast_rtcp_new();
}
/*
* Try to bind the RTP port, x, and possibly the RTCP port, x+1 as well.
* Start from a random (even, by RTP spec) port number, and
* iterate until success or no ports are available.
* Note that the requirement of RTP port being even, or RTCP being the
* next one, cannot be enforced in presence of a NAT box because the
* mapping is not under our control.
*/
x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart;
x = x & ~1; /* make it an even number */
startplace = x; /* remember the starting point */
/* this is constant across the loop */
rtp->us.sin_addr = addr;
if (rtp->rtcp)
rtp->rtcp->us.sin_addr = addr;
for (;;) {
rtp->us.sin_port = htons(x);
if (!bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) {
/* bind succeeded, if no rtcp then we are done */
if (!rtp->rtcp)
break;
/* have rtcp, try to bind it */
rtp->rtcp->us.sin_port = htons(x + 1);
if (!bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))
break; /* success again, we are really done */
/*
* RTCP bind failed, so close and recreate the
* already bound RTP socket for the next round.
*/
close(rtp->s);
rtp->s = rtp_socket("RTP");
if (rtp->s < 0)
goto fail;
}
/*
* If we get here, there was an error in one of the bind()
* calls, so make sure it is nothing unexpected.
*/
if (errno != EADDRINUSE) {
/* We got an error that wasn't expected, abort! */
ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
goto fail;
}
/*
* One of the ports is in use. For the next iteration,
* increment by two and handle wraparound.
* If we reach the starting point, then declare failure.
*/
x += 2;
if (x > rtpend)
x = (rtpstart + 1) & ~1;
if (x == startplace) {
ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
goto fail;
}
}
rtp->sched = sched;
rtp->io = io;
if (callbackmode) {
rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
ast_set_flag(rtp, FLAG_CALLBACK_MODE);
}
ast_rtp_pt_default(rtp);
return rtp;
fail:
if (rtp->s >= 0)
close(rtp->s);
if (rtp->rtcp) {
close(rtp->rtcp->s);
ast_free(rtp->rtcp);
}
ast_free(rtp);
return NULL;
}
| int ast_rtp_proto_register | ( | struct ast_rtp_protocol * | proto | ) |
Register an RTP channel client.
Definition at line 3954 of file rtp.c.
References ast_log(), AST_RWLIST_INSERT_HEAD, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_rtp_protocol::list, LOG_WARNING, and ast_rtp_protocol::type.
Referenced by load_module().
{
struct ast_rtp_protocol *cur;
AST_RWLIST_WRLOCK(&protos);
AST_RWLIST_TRAVERSE(&protos, cur, list) {
if (!strcmp(cur->type, proto->type)) {
ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
AST_RWLIST_UNLOCK(&protos);
return -1;
}
}
AST_RWLIST_INSERT_HEAD(&protos, proto, list);
AST_RWLIST_UNLOCK(&protos);
return 0;
}
| void ast_rtp_proto_unregister | ( | struct ast_rtp_protocol * | proto | ) |
Unregister an RTP channel client.
Definition at line 3946 of file rtp.c.
References AST_RWLIST_REMOVE, AST_RWLIST_UNLOCK, and AST_RWLIST_WRLOCK.
Referenced by load_module(), and unload_module().
{
AST_RWLIST_WRLOCK(&protos);
AST_RWLIST_REMOVE(&protos, proto, list);
AST_RWLIST_UNLOCK(&protos);
}
| void ast_rtp_pt_clear | ( | struct ast_rtp * | rtp | ) |
Setting RTP payload types from lines in a SDP description:
Definition at line 2038 of file rtp.c.
References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), rtp_bridge_unlock(), 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().
{
int i;
if (!rtp)
return;
rtp_bridge_lock(rtp);
for (i = 0; i < MAX_RTP_PT; ++i) {
rtp->current_RTP_PT[i].isAstFormat = 0;
rtp->current_RTP_PT[i].code = 0;
}
rtp->rtp_lookup_code_cache_isAstFormat = 0;
rtp->rtp_lookup_code_cache_code = 0;
rtp->rtp_lookup_code_cache_result = 0;
rtp_bridge_unlock(rtp);
}
Copy payload types between RTP structures.
Definition at line 2078 of file rtp.c.
References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), rtp_bridge_unlock(), 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().
{
unsigned int i;
rtp_bridge_lock(dest);
rtp_bridge_lock(src);
for (i = 0; i < MAX_RTP_PT; ++i) {
dest->current_RTP_PT[i].isAstFormat =
src->current_RTP_PT[i].isAstFormat;
dest->current_RTP_PT[i].code =
src->current_RTP_PT[i].code;
}
dest->rtp_lookup_code_cache_isAstFormat = 0;
dest->rtp_lookup_code_cache_code = 0;
dest->rtp_lookup_code_cache_result = 0;
rtp_bridge_unlock(src);
rtp_bridge_unlock(dest);
}
| void ast_rtp_pt_default | ( | struct ast_rtp * | rtp | ) |
Set payload types to defaults.
Definition at line 2059 of file rtp.c.
References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), rtp_bridge_unlock(), 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_new_with_bindaddr().
{
int i;
rtp_bridge_lock(rtp);
/* Initialize to default payload types */
for (i = 0; i < MAX_RTP_PT; ++i) {
rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
}
rtp->rtp_lookup_code_cache_isAstFormat = 0;
rtp->rtp_lookup_code_cache_code = 0;
rtp->rtp_lookup_code_cache_result = 0;
rtp_bridge_unlock(rtp);
}
Definition at line 1576 of file rtp.c.
References ast_rtp::altthem, ast_assert, ast_codec_get_samples(), AST_CONTROL_SRCCHANGE, ast_debug, AST_FORMAT_AUDIO_MASK, ast_format_rate(), AST_FORMAT_SLINEAR, AST_FORMAT_T140, AST_FORMAT_T140RED, AST_FORMAT_VIDEO_MASK, ast_frame_byteswap_be, AST_FRAME_CONTROL, AST_FRAME_DTMF_END, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_frisolate(), ast_inet_ntoa(), AST_LIST_EMPTY, AST_LIST_FIRST, AST_LIST_HEAD_INIT_NOLOCK, AST_LIST_INSERT_TAIL, 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, create_dtmf_frame(), 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::lastitexttimestamp, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len(), LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_frame::ptr, 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_rtp::s, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, ast_rtp::sending_digit, ast_frame::seqno, ast_frame::src, ast_rtp::strict_rtp_address, STRICT_RTP_CLOSED, STRICT_RTP_LEARN, ast_rtp::strict_rtp_state, STUN_ACCEPT, stun_handle_packet(), ast_frame::subclass, ast_rtcp::them, ast_rtp::them, ast_rtp::themssrc, ast_frame::ts, and version.
Referenced by gtalk_rtp_read(), jingle_rtp_read(), mgcp_rtp_read(), oh323_rtp_read(), rtpread(), sip_rtp_read(), skinny_rtp_read(), and unistim_rtp_read().
{
int res;
struct sockaddr_in sock_in;
socklen_t len;
unsigned int seqno;
int version;
int payloadtype;
int hdrlen = 12;
int padding;
int mark;
int ext;
int cc;
unsigned int ssrc;
unsigned int timestamp;
unsigned int *rtpheader;
struct rtpPayloadType rtpPT;
struct ast_rtp *bridged = NULL;
int prev_seqno;
struct frame_list frames;
/* If time is up, kill it */
if (rtp->sending_digit)
ast_rtp_senddigit_continuation(rtp);
len = sizeof(sock_in);
/* Cache where the header will go */
res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
0, (struct sockaddr *)&sock_in, &len);
/* If strict RTP protection is enabled see if we need to learn this address or if the packet should be dropped */
if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
/* Copy over address that this packet was received on */
memcpy(&rtp->strict_rtp_address, &sock_in, sizeof(rtp->strict_rtp_address));
/* Now move over to actually protecting the RTP port */
rtp->strict_rtp_state = STRICT_RTP_CLOSED;
ast_debug(1, "Learned remote address is %s:%d for strict RTP purposes, now protecting the port.\n", ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port));
} else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
/* If the address we previously learned doesn't match the address this packet came in on simply drop it */
if ((rtp->strict_rtp_address.sin_addr.s_addr != sock_in.sin_addr.s_addr) || (rtp->strict_rtp_address.sin_port != sock_in.sin_port)) {
ast_debug(1, "Received RTP packet from %s:%d, dropping due to strict RTP protection. Expected it to be from %s:%d\n", ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port), ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port));
return &ast_null_frame;
}
}
rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
if (res < 0) {
ast_assert(errno != EBADF);
if (errno != EAGAIN) {
ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno));
return NULL;
}
return &ast_null_frame;
}
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short\n");
return &ast_null_frame;
}
/* Get fields */
seqno = ntohl(rtpheader[0]);
/* Check RTP version */
version = (seqno & 0xC0000000) >> 30;
if (!version) {
/* If the two high bits are 0, this might be a
* STUN message, so process it. stun_handle_packet()
* answers to requests, and it returns STUN_ACCEPT
* if the request is valid.
*/
if ((stun_handle_packet(rtp->s, &sock_in, rtp->rawdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == STUN_ACCEPT) &&
(!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
memcpy(&rtp->them, &sock_in, sizeof(rtp->them));
}
return &ast_null_frame;
}
#if 0 /* Allow to receive RTP stream with closed transmission path */
/* If we don't have the other side's address, then ignore this */
if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
return &ast_null_frame;
#endif
/* Send to whoever send to us if NAT is turned on */
if (rtp->nat) {
if (((rtp->them.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
(rtp->them.sin_port != sock_in.sin_port)) &&
((rtp->altthem.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
(rtp->altthem.sin_port != sock_in.sin_port))) {
rtp->them = sock_in;
if (rtp->rtcp) {
int h = 0;
memcpy(&rtp->rtcp->them, &sock_in, sizeof(rtp->rtcp->them));
h = ntohs(rtp->them.sin_port);
rtp->rtcp->them.sin_port = htons(h + 1);
}
rtp->rxseqno = 0;
ast_set_flag(rtp, FLAG_NAT_ACTIVE);
if (option_debug || rtpdebug)
ast_debug(0, "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));
}
}
/* If we are bridged to another RTP stream, send direct */
if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
return &ast_null_frame;
if (version != 2)
return &ast_null_frame;
payloadtype = (seqno & 0x7f0000) >> 16;
padding = seqno & (1 << 29);
mark = seqno & (1 << 23);
ext = seqno & (1 << 28);
cc = (seqno & 0xF000000) >> 24;
seqno &= 0xffff;
timestamp = ntohl(rtpheader[1]);
ssrc = ntohl(rtpheader[2]);
AST_LIST_HEAD_INIT_NOLOCK(&frames);
/* Force a marker bit and change SSRC if the SSRC changes */
if (rtp->rxssrc && rtp->rxssrc != ssrc) {
struct ast_frame *f, srcupdate = {
AST_FRAME_CONTROL,
.subclass = AST_CONTROL_SRCCHANGE,
};
if (!mark) {
if (option_debug || rtpdebug) {
ast_debug(0, "Forcing Marker bit, because SSRC has changed\n");
}
mark = 1;
}
f = ast_frisolate(&srcupdate);
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
}
rtp->rxssrc = ssrc;
if (padding) {
/* Remove padding bytes */
res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
}
if (cc) {
/* CSRC fields present */
hdrlen += cc*4;
}
if (ext) {
/* RTP Extension present */
hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
hdrlen += 4;
if (option_debug) {
int profile;
profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
if (profile == 0x505a)
ast_debug(1, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
else
ast_debug(1, "Found unknown RTP Extensions %x\n", profile);
}
}
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
}
rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
if (rtp->rxcount==1) {
/* This is the first RTP packet successfully received from source */
rtp->seedrxseqno = seqno;
}
/* Do not schedule RR if RTCP isn't run */
if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
/* Schedule transmission of Receiver Report */
rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
}
if ((int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
rtp->cycles += RTP_SEQ_MOD;
prev_seqno = rtp->lastrxseqno;
rtp->lastrxseqno = seqno;
if (!rtp->themssrc)
rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
if (rtp_debug_test_addr(&sock_in))
ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
if (!rtpPT.isAstFormat) {
struct ast_frame *f = NULL;
/* This is special in-band data that's not one of our codecs */
if (rtpPT.code == AST_RTP_DTMF) {
/* It's special -- rfc2833 process it */
if (rtp_debug_test_addr(&sock_in)) {
unsigned char *data;
unsigned int event;
unsigned int event_end;
unsigned int duration;
data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
event = ntohl(*((unsigned int *)(data)));
event >>= 24;
event_end = ntohl(*((unsigned int *)(data)));
event_end <<= 8;
event_end >>= 24;
duration = ntohl(*((unsigned int *)(data)));
duration &= 0xFFFF;
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(sock_in.sin_addr), ntohs(sock_in.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
}
/* process_rfc2833 may need to return multiple frames. We do this
* by passing the pointer to the frame list to it so that the method
* can append frames to the list as needed
*/
process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &frames);
} else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
/* It's really special -- process it the Cisco way */
if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
rtp->lastevent = seqno;
}
} else if (rtpPT.code == AST_RTP_CN) {
/* Comfort Noise */
f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
} else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
}
if (f) {
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
}
/* Even if no frame was returned by one of the above methods,
* we may have a frame to return in our frame list
*/
if (!AST_LIST_EMPTY(&frames)) {
return AST_LIST_FIRST(&frames);
}
return &ast_null_frame;
}
rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
rtp->rxseqno = seqno;
if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) {
rtp->dtmf_timeout = 0;
if (rtp->resp) {
struct ast_frame *f;
f = create_dtmf_frame(rtp, AST_FRAME_DTMF_END);
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
return AST_LIST_FIRST(&frames);
}
}
/* Record received timestamp as last received now */
rtp->lastrxts = timestamp;
rtp->f.mallocd = 0;
rtp->f.datalen = res - hdrlen;
rtp->f.data.ptr = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.seqno = seqno;
if (rtp->f.subclass == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
unsigned char *data = rtp->f.data.ptr;
memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
rtp->f.datalen +=3;
*data++ = 0xEF;
*data++ = 0xBF;
*data = 0xBD;
}
if (rtp->f.subclass == AST_FORMAT_T140RED) {
unsigned char *data = rtp->f.data.ptr;
unsigned char *header_end;
int num_generations;
int header_length;
int length;
int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
int x;
rtp->f.subclass = AST_FORMAT_T140;
header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
if (header_end == NULL) {
return &ast_null_frame;
}
header_end++;
header_length = header_end - data;
num_generations = header_length / 4;
length = header_length;
if (!diff) {
for (x = 0; x < num_generations; x++)
length += data[x * 4 + 3];
if (!(rtp->f.datalen - length))
return &ast_null_frame;
rtp->f.data.ptr += length;
rtp->f.datalen -= length;
} else if (diff > num_generations && diff < 10) {
length -= 3;
rtp->f.data.ptr += length;
rtp->f.datalen -= length;
data = rtp->f.data.ptr;
*data++ = 0xEF;
*data++ = 0xBF;
*data = 0xBD;
} else {
for ( x = 0; x < num_generations - diff; x++)
length += data[x * 4 + 3];
rtp->f.data.ptr += length;
rtp->f.datalen -= length;
}
}
if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) {
rtp->f.samples = ast_codec_get_samples(&rtp->f);
if (rtp->f.subclass == AST_FORMAT_SLINEAR)
ast_frame_byteswap_be(&rtp->f);
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass) / 1000);
rtp->f.len = rtp->f.samples / ((ast_format_rate(rtp->f.subclass) / 1000));
} else if (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
/* Video -- samples is # of samples vs. 90000 */
if (!rtp->lastividtimestamp)
rtp->lastividtimestamp = timestamp;
rtp->f.samples = timestamp - rtp->lastividtimestamp;
rtp->lastividtimestamp = timestamp;
rtp->f.delivery.tv_sec = 0;
rtp->f.delivery.tv_usec = 0;
/* Pass the RTP marker bit as bit 0 in the subclass field.
* This is ok because subclass is actually a bitmask, and
* the low bits represent audio formats, that are not
* involved here since we deal with video.
*/
if (mark)
rtp->f.subclass |= 0x1;
} else {
/* TEXT -- samples is # of samples vs. 1000 */
if (!rtp->lastitexttimestamp)
rtp->lastitexttimestamp = timestamp;
rtp->f.samples = timestamp - rtp->lastitexttimestamp;
rtp->lastitexttimestamp = timestamp;
rtp->f.delivery.tv_sec = 0;
rtp->f.delivery.tv_usec = 0;
}
rtp->f.src = "RTP";
AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
return AST_LIST_FIRST(&frames);
}
| int ast_rtp_reload | ( | void | ) |
Initialize RTP subsystem
Definition at line 4871 of file rtp.c.
References __ast_rtp_reload().
{
return __ast_rtp_reload(1);
}
| void ast_rtp_reset | ( | struct ast_rtp * | rtp | ) |
Definition at line 2778 of file rtp.c.
References ast_rtp::dtmf_timeout, ast_rtp::dtmfmute, ast_rtp::dtmfsamples, ast_rtp::lastdigitts, ast_rtp::lastevent, ast_rtp::lasteventseqn, ast_rtp::lastitexttimestamp, ast_rtp::lastividtimestamp, ast_rtp::lastotexttimestamp, 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.
{
memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
memset(&rtp->txcore, 0, sizeof(rtp->txcore));
memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
rtp->lastts = 0;
rtp->lastdigitts = 0;
rtp->lastrxts = 0;
rtp->lastividtimestamp = 0;
rtp->lastovidtimestamp = 0;
rtp->lastitexttimestamp = 0;
rtp->lastotexttimestamp = 0;
rtp->lasteventseqn = 0;
rtp->lastevent = 0;
rtp->lasttxformat = 0;
rtp->lastrxformat = 0;
rtp->dtmf_timeout = 0;
rtp->dtmfsamples = 0;
rtp->seqno = 0;
rtp->rxseqno = 0;
}
| int ast_rtp_sendcng | ( | struct ast_rtp * | rtp, |
| int | level | ||
| ) |
generate comfort noice (CNG)
Definition at line 3632 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 check_rtp_timeout().
{
unsigned int *rtpheader;
int hdrlen = 12;
int res;
int payload;
char data[256];
level = 127 - (level & 0x7f);
payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
/* If we have no peer, return immediately */
if (!rtp->them.sin_addr.s_addr)
return 0;
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
/* Get a pointer to the header */
rtpheader = (unsigned int *)data;
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
rtpheader[1] = htonl(rtp->lastts);
rtpheader[2] = htonl(rtp->ssrc);
data[12] = level;
if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
if (res <0)
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));
if (rtp_debug_test_addr(&rtp->them))
ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
, ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);
}
return 0;
}
| int ast_rtp_senddigit_begin | ( | struct ast_rtp * | rtp, |
| char | digit | ||
| ) |
Send begin frames for DTMF.
Definition at line 3188 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().
{
unsigned int *rtpheader;
int hdrlen = 12, res = 0, i = 0, payload = 0;
char data[256];
if ((digit <= '9') && (digit >= '0'))
digit -= '0';
else if (digit == '*')
digit = 10;
else if (digit == '#')
digit = 11;
else if ((digit >= 'A') && (digit <= 'D'))
digit = digit - 'A' + 12;
else if ((digit >= 'a') && (digit <= 'd'))
digit = digit - 'a' + 12;
else {
ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
return 0;
}
/* If we have no peer, return immediately */
if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
return 0;
payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
rtp->send_duration = 160;
rtp->lastdigitts = rtp->lastts + rtp->send_duration;
/* Get a pointer to the header */
rtpheader = (unsigned int *)data;
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
rtpheader[1] = htonl(rtp->lastdigitts);
rtpheader[2] = htonl(rtp->ssrc);
for (i = 0; i < 2; i++) {
rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
if (res < 0)
ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
ast_inet_ntoa(rtp->them.sin_addr),
ntohs(rtp->them.sin_port), strerror(errno));
if (rtp_debug_test_addr(&rtp->them))
ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
ast_inet_ntoa(rtp->them.sin_addr),
ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
/* Increment sequence number */
rtp->seqno++;
/* Increment duration */
rtp->send_duration += 160;
/* Clear marker bit and set seqno */
rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
}
/* Since we received a begin, we can safely store the digit and disable any compensation */
rtp->sending_digit = 1;
rtp->send_digit = digit;
rtp->send_payload = payload;
return 0;
}
| int ast_rtp_senddigit_end | ( | struct ast_rtp * | rtp, |
| char | digit | ||
| ) |
Definition at line 3289 of file rtp.c.
References ast_rtp_senddigit_end_with_duration().
Referenced by mgcp_senddigit_end(), and oh323_digit_end().
{
return ast_rtp_senddigit_end_with_duration(rtp, digit, 0);
}
| int ast_rtp_senddigit_end_with_duration | ( | struct ast_rtp * | rtp, |
| char | digit, | ||
| unsigned int | duration | ||
| ) |
Send end packets for DTMF.
Definition at line 3295 of file rtp.c.
References ast_debug, ast_inet_ntoa(), ast_log(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), ast_rtp::dtmfmute, errno, ast_rtp::f, ast_rtp::lastdigitts, ast_rtp::lastts, LOG_ERROR, LOG_WARNING, rtp_debug_test_addr(), rtp_get_rate(), 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, ast_frame::subclass, and ast_rtp::them.
Referenced by ast_rtp_senddigit_end(), and sip_senddigit_end().
{
unsigned int *rtpheader;
int hdrlen = 12, res = 0, i = 0;
char data[256];
unsigned int measured_samples;
/* If no address, then bail out */
if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
return 0;
if ((digit <= '9') && (digit >= '0'))
digit -= '0';
else if (digit == '*')
digit = 10;
else if (digit == '#')
digit = 11;
else if ((digit >= 'A') && (digit <= 'D'))
digit = digit - 'A' + 12;
else if ((digit >= 'a') && (digit <= 'd'))
digit = digit - 'a' + 12;
else {
ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
return 0;
}
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
if (duration > 0 && (measured_samples = duration * rtp_get_rate(rtp->f.subclass) / 1000) > rtp->send_duration) {
ast_debug(2, "Adjusting final end duration from %u to %u\n", rtp->send_duration, measured_samples);
rtp->send_duration = measured_samples;
}
rtpheader = (unsigned int *)data;
rtpheader[1] = htonl(rtp->lastdigitts);
rtpheader[2] = htonl(rtp->ssrc);
rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
/* Set end bit */
rtpheader[3] |= htonl((1 << 23));
/* Send 3 termination packets */
for (i = 0; i < 3; i++) {
rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
rtp->seqno++;
if (res < 0)
ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
ast_inet_ntoa(rtp->them.sin_addr),
ntohs(rtp->them.sin_port), strerror(errno));
if (rtp_debug_test_addr(&rtp->them))
ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
ast_inet_ntoa(rtp->them.sin_addr),
ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
}
rtp->lastts += rtp->send_duration;
rtp->sending_digit = 0;
rtp->send_digit = 0;
return res;
}
| void ast_rtp_set_alt_peer | ( | struct ast_rtp * | rtp, |
| struct sockaddr_in * | alt | ||
| ) |
set potential alternate source for RTP media
- Since:
- 1.4.26 This function may be used to give the RTP stack a hint that there is a potential second source of media. One case where this is used is when the SIP stack receives a REINVITE to which it will be replying with a 491. In such a scenario, the IP and port information in the SDP of that REINVITE lets us know that we may receive media from that source/those sources even though the SIP transaction was unable to be completed successfully
- Parameters:
-
rtp The RTP structure we wish to set up an alternate host/port on alt The address information for the alternate media source
- Return values:
-
void
Definition at line 2718 of file rtp.c.
References ast_rtcp::altthem, ast_rtp::altthem, and ast_rtp::rtcp.
Referenced by handle_request_invite().
| void ast_rtp_set_callback | ( | struct ast_rtp * | rtp, |
| ast_rtp_callback | callback | ||
| ) |
Definition at line 795 of file rtp.c.
References ast_rtp::callback.
Referenced by start_rtp().
{
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.
Definition at line 2277 of file rtp.c.
References ast_rtp::current_RTP_PT, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().
Referenced by gtalk_is_answered(), gtalk_newcall(), jingle_newcall(), and process_sdp().
{
if (pt < 0 || pt >= MAX_RTP_PT || static_RTP_PT[pt].code == 0)
return; /* bogus payload type */
rtp_bridge_lock(rtp);
rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
rtp_bridge_unlock(rtp);
}
| void ast_rtp_set_peer | ( | struct ast_rtp * | rtp, |
| struct sockaddr_in * | them | ||
| ) |
Definition at line 2703 of file rtp.c.
References ast_rtp::rtcp, ast_rtp::rxseqno, STRICT_RTP_LEARN, ast_rtp::strict_rtp_state, ast_rtcp::them, and ast_rtp::them.
Referenced by handle_open_receive_channel_ack_message(), process_sdp(), setup_rtp_connection(), and start_rtp().
{
rtp->them.sin_port = them->sin_port;
rtp->them.sin_addr = them->sin_addr;
if (rtp->rtcp) {
int h = ntohs(them->sin_port);
rtp->rtcp->them.sin_port = htons(h + 1);
rtp->rtcp->them.sin_addr = them->sin_addr;
}
rtp->rxseqno = 0;
/* If strict RTP protection is enabled switch back to the learn state so we don't drop packets from above */
if (strictrtp)
rtp->strict_rtp_state = STRICT_RTP_LEARN;
}
| void ast_rtp_set_rtpholdtimeout | ( | struct ast_rtp * | rtp, |
| int | timeout | ||
| ) |
Set rtp hold timeout.
Definition at line 757 of file rtp.c.
References ast_rtp::rtpholdtimeout.
Referenced by check_rtp_timeout(), create_addr_from_peer(), and sip_alloc().
{
rtp->rtpholdtimeout = timeout;
}
| void ast_rtp_set_rtpkeepalive | ( | struct ast_rtp * | rtp, |
| int | period | ||
| ) |
set RTP keepalive interval
Definition at line 763 of file rtp.c.
References ast_rtp::rtpkeepalive.
Referenced by create_addr_from_peer(), and sip_alloc().
{
rtp->rtpkeepalive = period;
}
| int ast_rtp_set_rtpmap_type | ( | struct ast_rtp * | rtp, |
| int | pt, | ||
| char * | mimeType, | ||
| char * | mimeSubtype, | ||
| enum ast_rtp_options | options | ||
| ) |
Set payload type to a known MIME media type for a codec.
- Parameters:
-
rtp RTP structure to modify pt Payload type entry to modify mimeType top-level MIME type of media stream (typically "audio", "video", "text", etc.) mimeSubtype MIME subtype of media stream (typically a codec name) options Zero or more flags from the ast_rtp_options enum
This function 'fills in' an entry in the list of possible formats for a media stream associated with an RTP structure.
- Return values:
-
0 on success -1 if the payload type is out of range -2 if the mimeType/mimeSubtype combination was not found
Definition at line 2353 of file rtp.c.
References ast_rtp_set_rtpmap_type_rate().
Referenced by __oh323_rtp_create(), gtalk_is_answered(), gtalk_newcall(), jingle_newcall(), process_sdp(), process_sdp_a_text(), set_dtmf_payload(), and setup_rtp_connection().
{
return ast_rtp_set_rtpmap_type_rate(rtp, pt, mimeType, mimeSubtype, options, 0);
}
| int ast_rtp_set_rtpmap_type_rate | ( | struct ast_rtp * | rtp, |
| int | pt, | ||
| char * | mimeType, | ||
| char * | mimeSubtype, | ||
| enum ast_rtp_options | options, | ||
| unsigned int | sample_rate | ||
| ) |
Set payload type to a known MIME media type for a codec with a specific sample rate.
- Parameters:
-
rtp RTP structure to modify pt Payload type entry to modify mimeType top-level MIME type of media stream (typically "audio", "video", "text", etc.) mimeSubtype MIME subtype of media stream (typically a codec name) options Zero or more flags from the ast_rtp_options enum sample_rate The sample rate of the media stream
This function 'fills in' an entry in the list of possible formats for a media stream associated with an RTP structure.
- Return values:
-
0 on success -1 if the payload type is out of range -2 if the mimeType/mimeSubtype combination was not found
Set payload type to a known MIME media type for a codec with a specific sample rate.
- Returns:
- 0 if the MIME type was found and set, -1 if it wasn't found
Definition at line 2304 of file rtp.c.
References ARRAY_LEN, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_RTP_OPT_G726_NONSTANDARD, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, mimeTypes, mimeType::payloadType, rtp_bridge_lock(), rtp_bridge_unlock(), mimeType::sample_rate, mimeType::subtype, and mimeType::type.
Referenced by ast_rtp_set_rtpmap_type(), process_sdp_a_audio(), and process_sdp_a_video().
{
unsigned int i;
int found = 0;
if (pt < 0 || pt >= MAX_RTP_PT)
return -1; /* bogus payload type */
rtp_bridge_lock(rtp);
for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
const struct mimeType *t = &mimeTypes[i];
if (strcasecmp(mimeSubtype, t->subtype)) {
continue;
}
if (strcasecmp(mimeType, t->type)) {
continue;
}
/* if both sample rates have been supplied, and they don't match,
then this not a match; if one has not been supplied, then the
rates are not compared */
if (sample_rate && t->sample_rate &&
(sample_rate != t->sample_rate)) {
continue;
}
found = 1;
rtp->current_RTP_PT[pt] = t->payloadType;
if ((t->payloadType.code == AST_FORMAT_G726) &&
t->payloadType.isAstFormat &&
(options & AST_RTP_OPT_G726_NONSTANDARD)) {
rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
}
break;
}
rtp_bridge_unlock(rtp);
return (found ? 0 : -2);
}
| void ast_rtp_set_rtptimeout | ( | struct ast_rtp * | rtp, |
| int | timeout | ||
| ) |
Set rtp timeout.
Definition at line 751 of file rtp.c.
References ast_rtp::rtptimeout.
Referenced by check_rtp_timeout(), create_addr_from_peer(), and sip_alloc().
{
rtp->rtptimeout = timeout;
}
| void ast_rtp_set_rtptimers_onhold | ( | struct ast_rtp * | rtp | ) |
Definition at line 744 of file rtp.c.
References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.
Referenced by handle_response_invite().
{
rtp->rtptimeout = (-1) * rtp->rtptimeout;
rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
}
| void ast_rtp_set_vars | ( | struct ast_channel * | chan, |
| struct ast_rtp * | rtp | ||
| ) |
Set RTPAUDIOQOS(...) variables on a channel when it is being hung up.
- Since:
- 1.6.1
Definition at line 2882 of file rtp.c.
References ast_bridged_channel(), ast_rtp_get_quality(), ast_channel::bridge, pbx_builtin_setvar_helper(), RTPQOS_JITTER, RTPQOS_LOSS, RTPQOS_RTT, and RTPQOS_SUMMARY.
Referenced by handle_request_bye(), and sip_hangup().
{
char *audioqos;
char *audioqos_jitter;
char *audioqos_loss;
char *audioqos_rtt;
struct ast_channel *bridge;
if (!rtp || !chan)
return;
bridge = ast_bridged_channel(chan);
audioqos = ast_rtp_get_quality(rtp, NULL, RTPQOS_SUMMARY);
audioqos_jitter = ast_rtp_get_quality(rtp, NULL, RTPQOS_JITTER);
audioqos_loss = ast_rtp_get_quality(rtp, NULL, RTPQOS_LOSS);
audioqos_rtt = ast_rtp_get_quality(rtp, NULL, RTPQOS_RTT);
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", audioqos);
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", audioqos_jitter);
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", audioqos_loss);
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", audioqos_rtt);
if (!bridge)
return;
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", audioqos);
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", audioqos_jitter);
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", audioqos_loss);
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", audioqos_rtt);
}
| void ast_rtp_setdtmf | ( | struct ast_rtp * | rtp, |
| int | dtmf | ||
| ) |
Indicate whether this RTP session is carrying DTMF or not.
Definition at line 810 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().
{
ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
}
| void ast_rtp_setdtmfcompensate | ( | struct ast_rtp * | rtp, |
| int | compensate | ||
| ) |
Compensate for devices that send RFC2833 packets all at once.
Definition at line 815 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().
{
ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
}
| void ast_rtp_setnat | ( | struct ast_rtp * | rtp, |
| int | nat | ||
| ) |
Definition at line 800 of file rtp.c.
References nat, and ast_rtp::nat.
Referenced by __oh323_rtp_create(), do_setnat(), oh323_rtp_read(), and start_rtp().
| int ast_rtp_setqos | ( | struct ast_rtp * | rtp, |
| int | tos, | ||
| int | cos, | ||
| char * | desc | ||
| ) |
Definition at line 2679 of file rtp.c.
References ast_netsock_set_qos(), and ast_rtp::s.
Referenced by __oh323_rtp_create(), sip_alloc(), and start_rtp().
{
return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
}
| void ast_rtp_setstun | ( | struct ast_rtp * | rtp, |
| int | stun_enable | ||
| ) |
Enable STUN capability.
Definition at line 820 of file rtp.c.
References ast_set2_flag, and FLAG_HAS_STUN.
Referenced by gtalk_new().
{
ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
}
| void ast_rtp_stop | ( | struct ast_rtp * | rtp | ) |
Stop RTP session, do not destroy structure
Definition at line 2757 of file rtp.c.
References ast_clear_flag, AST_SCHED_DEL, FLAG_P2P_SENT_MARK, free, ast_rtp::red, ast_rtp::rtcp, ast_rtp::sched, rtp_red::schedid, ast_rtcp::schedid, ast_rtcp::them, and ast_rtp::them.
Referenced by process_sdp(), setup_rtp_connection(), and stop_media_flows().
{
if (rtp->rtcp) {
AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
}
if (rtp->red) {
AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
free(rtp->red);
rtp->red = NULL;
}
memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
if (rtp->rtcp) {
memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
}
ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
}
| void ast_rtp_stun_request | ( | struct ast_rtp * | rtp, |
| struct sockaddr_in * | suggestion, | ||
| const char * | username | ||
| ) |
Send STUN request for an RTP socket Deprecated, this is just a wrapper for ast_rtp_stun_request()
Definition at line 699 of file rtp.c.
References ast_stun_request(), and ast_rtp::s.
Referenced by gtalk_update_stun(), and jingle_update_stun().
{
ast_stun_request(rtp->s, suggestion, username, NULL);
}
| void ast_rtp_unset_m_type | ( | struct ast_rtp * | rtp, |
| int | pt | ||
| ) |
clear payload type
Definition at line 2289 of file rtp.c.
References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().
Referenced by process_sdp_a_audio(), and process_sdp_a_video().
{
if (pt < 0 || pt >= MAX_RTP_PT)
return; /* bogus payload type */
rtp_bridge_lock(rtp);
rtp->current_RTP_PT[pt].isAstFormat = 0;
rtp->current_RTP_PT[pt].code = 0;
rtp_bridge_unlock(rtp);
}
- Bug:
- XXX this might never be free'd. Why do we do this?
Definition at line 3848 of file rtp.c.
References ast_codec_pref_getsize(), ast_debug, AST_FORMAT_G723_1, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SPEEX, AST_FRAME_TEXT, 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::data, 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_WARNING, ast_frame::offset, ast_rtp::pref, ast_frame::ptr, ast_rtp::red, red_t140_to_red(), ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.
Referenced by gtalk_write(), jingle_write(), mgcp_write(), oh323_write(), red_write(), sip_write(), skinny_write(), and unistim_write().
{
struct ast_frame *f;
int codec;
int hdrlen = 12;
int subclass;
/* If we have no peer, return immediately */
if (!rtp->them.sin_addr.s_addr)
return 0;
/* If there is no data length, return immediately */
if (!_f->datalen && !rtp->red)
return 0;
/* Make sure we have enough space for RTP header */
if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO) && (_f->frametype != AST_FRAME_TEXT)) {
ast_log(LOG_WARNING, "RTP can only send voice, video and text\n");
return -1;
}
if (rtp->red) {
/* return 0; */
/* no primary data or generations to send */
if ((_f = red_t140_to_red(rtp->red)) == NULL)
return 0;
}
/* The bottom bit of a video subclass contains the marker bit */
subclass = _f->subclass;
if (_f->frametype == AST_FRAME_VIDEO)
subclass &= ~0x1;
codec = ast_rtp_lookup_code(rtp, 1, subclass);
if (codec < 0) {
ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
return -1;
}
if (rtp->lasttxformat != subclass) {
/* New format, reset the smoother */
ast_debug(1, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
rtp->lasttxformat = subclass;
if (rtp->smoother)
ast_smoother_free(rtp->smoother);
rtp->smoother = NULL;
}
if (!rtp->smoother) {
struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
switch (subclass) {
case AST_FORMAT_SPEEX:
case AST_FORMAT_G723_1:
case AST_FORMAT_SIREN7:
case AST_FORMAT_SIREN14:
/* these are all frame-based codecs and cannot be safely run through
a smoother */
break;
default:
if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
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));
return -1;
}
if (fmt.flags)
ast_smoother_set_flags(rtp->smoother, fmt.flags);
ast_debug(1, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
}
}
}
if (rtp->smoother) {
if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
ast_smoother_feed_be(rtp->smoother, _f);
} else {
ast_smoother_feed(rtp->smoother, _f);
}
while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
ast_rtp_raw_write(rtp, f, codec);
}
} else {
/* Don't buffer outgoing frames; send them one-per-packet: */
if (_f->offset < hdrlen)
f = ast_frdup(_f); /*! \bug XXX this might never be free'd. Why do we do this? */
else
f = _f;
if (f->data.ptr)
ast_rtp_raw_write(rtp, f, codec);
if (f != _f)
ast_frfree(f);
}
return 0;
}
| int ast_stun_request | ( | int | s, |
| struct sockaddr_in * | dst, | ||
| const char * | username, | ||
| struct sockaddr_in * | answer | ||
| ) |
Generic STUN request send a generic stun request to the server specified.
- Parameters:
-
s the socket used to send the request dst the address of the STUN server username if non null, add the username in the request answer if non null, the function waits for a response and puts here the externally visible address.
- Returns:
- 0 on success, other values on error. The interface it may change in the future.
Generic STUN request send a generic stun request to the server specified.
- Parameters:
-
s the socket used to send the request dst the address of the STUN server username if non null, add the username in the request answer if non null, the function waits for a response and puts here the externally visible address.
- Returns:
- 0 on success, other values on error.
Definition at line 636 of file rtp.c.
References append_attr_string(), ast_log(), ast_poll, stun_attr::attr, stun_header::ies, LOG_WARNING, stun_header::msglen, stun_header::msgtype, s, STUN_BINDREQ, stun_get_mapped(), stun_handle_packet(), stun_req_id(), stun_send(), and STUN_USERNAME.
Referenced by ast_rtp_stun_request(), ast_sip_ouraddrfor(), and reload_config().
{
struct stun_header *req;
unsigned char reqdata[1024];
int reqlen, reqleft;
struct stun_attr *attr;
int res = 0;
int retry;
req = (struct stun_header *)reqdata;
stun_req_id(req);
reqlen = 0;
reqleft = sizeof(reqdata) - sizeof(struct stun_header);
req->msgtype = 0;
req->msglen = 0;
attr = (struct stun_attr *)req->ies;
if (username)
append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
req->msglen = htons(reqlen);
req->msgtype = htons(STUN_BINDREQ);
for (retry = 0; retry < 3; retry++) { /* XXX make retries configurable */
/* send request, possibly wait for reply */
unsigned char reply_buf[1024];
struct pollfd pfds = { .fd = s, .events = POLLIN, };
struct sockaddr_in src;
socklen_t srclen;
res = stun_send(s, dst, req);
if (res < 0) {
ast_log(LOG_WARNING, "ast_stun_request send #%d failed error %d, retry\n",
retry, res);
continue;
}
if (answer == NULL)
break;
res = ast_poll(&pfds, 1, 3000);
if (res <= 0) /* timeout or error */
continue;
memset(&src, '\0', sizeof(src));
srclen = sizeof(src);
/* XXX pass -1 in the size, because stun_handle_packet might
* write past the end of the buffer.
*/
res = recvfrom(s, reply_buf, sizeof(reply_buf) - 1,
0, (struct sockaddr *)&src, &srclen);
if (res < 0) {
ast_log(LOG_WARNING, "ast_stun_request recvfrom #%d failed error %d, retry\n",
retry, res);
continue;
}
memset(answer, '\0', sizeof(struct sockaddr_in));
stun_handle_packet(s, &src, reply_buf, res,
stun_get_mapped, answer);
res = 0; /* signal regular exit */
break;
}
return res;
}
Buffer t.140 data.
Buffer t.140 data.
- Parameters:
-
rtp f frame
Definition at line 4981 of file rtp.c.
References rtp_red::buf_data, ast_frame::data, ast_frame::datalen, ast_frame::ptr, ast_rtp::red, rtp_red::t140, and ast_frame::ts.
Referenced by sip_write().
| int rtp_red_init | ( | struct ast_rtp * | rtp, |
| int | ti, | ||
| int * | red_data_pt, | ||
| int | num_gen | ||
| ) |
Initalize t.140 redudancy.
- Parameters:
-
ti time between each t140red frame is sent red_pt payloadtype for RTP packet pt payloadtype numbers for each generation including primary data num_gen number of redundant generations, primary data excluded
- Since:
- 1.6.1
Initalize t.140 redudancy.
- Parameters:
-
rtp ti buffer t140 for ti (msecs) before sending redundant frame red_data_pt Payloadtypes for primary- and generation-data num_gen numbers of generations (primary generation not encounted)
Definition at line 4942 of file rtp.c.
References ast_calloc, AST_FORMAT_T140RED, AST_FRAME_TEXT, ast_sched_add(), rtp_red::buf_data, ast_frame::data, ast_frame::datalen, ast_frame::frametype, rtp_red::hdrlen, rtp_red::num_gen, rtp_red::prev_ts, rtp_red::pt, ast_frame::ptr, ast_rtp::red, red_write(), ast_rtp::sched, rtp_red::schedid, ast_frame::subclass, rtp_red::t140, rtp_red::t140red, rtp_red::t140red_data, rtp_red::ti, and ast_frame::ts.
Referenced by process_sdp().
{
struct rtp_red *r;
int x;
if (!(r = ast_calloc(1, sizeof(struct rtp_red))))
return -1;
r->t140.frametype = AST_FRAME_TEXT;
r->t140.subclass = AST_FORMAT_T140RED;
r->t140.data.ptr = &r->buf_data;
r->t140.ts = 0;
r->t140red = r->t140;
r->t140red.data.ptr = &r->t140red_data;
r->t140red.datalen = 0;
r->ti = ti;
r->num_gen = num_gen;
r->hdrlen = num_gen * 4 + 1;
r->prev_ts = 0;
for (x = 0; x < num_gen; x++) {
r->pt[x] = red_data_pt[x];
r->pt[x] |= 1 << 7; /* mark redundant generations pt */
r->t140red_data[x*4] = r->pt[x];
}
r->t140red_data[x*4] = r->pt[x] = red_data_pt[x]; /* primary pt */
r->schedid = ast_sched_add(rtp->sched, ti, red_write, rtp);
rtp->red = r;
r->t140.datalen = 0;
return 0;
}
