Mon Nov 24 15:34:50 2008

Asterisk developer's documentation


rtp.h File Reference

Supports RTP and RTCP with Symmetric RTP support for NAT traversal. More...

#include <netinet/in.h>
#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
#include "asterisk/linkedlists.h"

Go to the source code of this file.

Data Structures

struct  ast_rtp_protocol
struct  ast_rtp_quality

Defines

#define AST_RTP_CISCO_DTMF   (1 << 2)
#define AST_RTP_CN   (1 << 1)
#define AST_RTP_DTMF   (1 << 0)
#define AST_RTP_MAX   AST_RTP_CISCO_DTMF
#define FLAG_3389_WARNING   (1 << 0)
#define MAX_RTP_PT   256

Typedefs

typedef int(*) ast_rtp_callback (struct ast_rtp *rtp, struct ast_frame *f, void *data)

Enumerations

enum  ast_rtp_get_result { AST_RTP_GET_FAILED = 0, AST_RTP_TRY_PARTIAL, AST_RTP_TRY_NATIVE }
enum  ast_rtp_options { AST_RTP_OPT_G726_NONSTANDARD = (1 << 0) }

Functions

int ast_rtcp_fd (struct ast_rtp *rtp)
ast_frameast_rtcp_read (struct ast_rtp *rtp)
int ast_rtcp_send_h261fur (void *data)
 Send an H.261 fast update request. Some devices need this rather than the XML message in SIP.
size_t ast_rtp_alloc_size (void)
 Get the amount of space required to hold an RTP session.
int ast_rtp_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
 Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk.
int ast_rtp_codec_getformat (int pt)
ast_codec_prefast_rtp_codec_getpref (struct ast_rtp *rtp)
int ast_rtp_codec_setpref (struct ast_rtp *rtp, struct ast_codec_pref *prefs)
void ast_rtp_destroy (struct ast_rtp *rtp)
int ast_rtp_early_bridge (struct ast_channel *dest, struct ast_channel *src)
 If possible, create an early bridge directly between the devices without having to send a re-invite later.
int ast_rtp_fd (struct ast_rtp *rtp)
ast_rtpast_rtp_get_bridged (struct ast_rtp *rtp)
void ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats)
 Return the union of all of the codecs that were set by rtp_set...() calls They're returned as two distinct sets: AST_FORMATs, and AST_RTPs.
int ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them)
char * ast_rtp_get_quality (struct ast_rtp *rtp, struct ast_rtp_quality *qual)
 Return RTCP quality string.
int ast_rtp_get_rtpholdtimeout (struct ast_rtp *rtp)
 Get rtp hold timeout.
int ast_rtp_get_rtpkeepalive (struct ast_rtp *rtp)
 Get RTP keepalive interval.
int ast_rtp_get_rtptimeout (struct ast_rtp *rtp)
 Get rtp timeout.
void ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us)
int ast_rtp_getnat (struct ast_rtp *rtp)
void ast_rtp_init (void)
 Initialize the RTP system in Asterisk.
int ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code)
 Looks up an RTP code out of our *static* outbound list.
char * ast_rtp_lookup_mime_multiple (char *buf, size_t size, const int capability, const int isAstFormat, enum ast_rtp_options options)
 Build a string of MIME subtype names from a capability list.
const char * ast_rtp_lookup_mime_subtype (int isAstFormat, int code, enum ast_rtp_options options)
 Mapping an Asterisk code into a MIME subtype (string):.
rtpPayloadType ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt)
 Mapping between RTP payload format codes and Asterisk codes:.
int ast_rtp_make_compatible (struct ast_channel *dest, struct ast_channel *src, int media)
ast_rtpast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
 Initializate a RTP session.
void ast_rtp_new_init (struct ast_rtp *rtp)
 Initialize a new RTP structure.
void ast_rtp_new_source (struct ast_rtp *rtp)
ast_rtpast_rtp_new_with_bindaddr (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in)
 Initializate a RTP session using an in_addr structure.
int ast_rtp_proto_register (struct ast_rtp_protocol *proto)
 Register interface to channel driver.
void ast_rtp_proto_unregister (struct ast_rtp_protocol *proto)
 Unregister interface to channel driver.
void ast_rtp_pt_clear (struct ast_rtp *rtp)
 Setting RTP payload types from lines in a SDP description:.
void ast_rtp_pt_copy (struct ast_rtp *dest, struct ast_rtp *src)
 Copy payload types between RTP structures.
void ast_rtp_pt_default (struct ast_rtp *rtp)
 Set payload types to defaults.
ast_frameast_rtp_read (struct ast_rtp *rtp)
int ast_rtp_reload (void)
void ast_rtp_reset (struct ast_rtp *rtp)
int ast_rtp_sendcng (struct ast_rtp *rtp, int level)
 generate comfort noice (CNG)
int ast_rtp_senddigit_begin (struct ast_rtp *rtp, char digit)
 Send begin frames for DTMF.
int ast_rtp_senddigit_end (struct ast_rtp *rtp, char digit)
void ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback)
void ast_rtp_set_data (struct ast_rtp *rtp, void *data)
void ast_rtp_set_m_type (struct ast_rtp *rtp, int pt)
 Activate payload type.
void ast_rtp_set_peer (struct ast_rtp *rtp, struct sockaddr_in *them)
void ast_rtp_set_rtpholdtimeout (struct ast_rtp *rtp, int timeout)
 Set rtp hold timeout.
void ast_rtp_set_rtpkeepalive (struct ast_rtp *rtp, int period)
 set RTP keepalive interval
int ast_rtp_set_rtpmap_type (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype, enum ast_rtp_options options)
 Initiate payload type to a known MIME media type for a codec.
void ast_rtp_set_rtptimeout (struct ast_rtp *rtp, int timeout)
 Set rtp timeout.
void ast_rtp_set_rtptimers_onhold (struct ast_rtp *rtp)
void ast_rtp_setdtmf (struct ast_rtp *rtp, int dtmf)
 Indicate whether this RTP session is carrying DTMF or not.
void ast_rtp_setdtmfcompensate (struct ast_rtp *rtp, int compensate)
 Compensate for devices that send RFC2833 packets all at once.
void ast_rtp_setnat (struct ast_rtp *rtp, int nat)
void ast_rtp_setstun (struct ast_rtp *rtp, int stun_enable)
 Enable STUN capability.
int ast_rtp_settos (struct ast_rtp *rtp, int tos)
void ast_rtp_stop (struct ast_rtp *rtp)
void ast_rtp_stun_request (struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username)
void ast_rtp_unset_m_type (struct ast_rtp *rtp, int pt)
 clear payload type
int ast_rtp_write (struct ast_rtp *rtp, struct ast_frame *f)


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)

DTMF (Cisco Proprietary)

Definition at line 47 of file rtp.h.

Referenced by ast_rtp_read().

#define AST_RTP_CN   (1 << 1)

'Comfort Noise' (RFC3389)

Definition at line 45 of file rtp.h.

Referenced by ast_rtp_read(), and ast_rtp_sendcng().

#define AST_RTP_DTMF   (1 << 0)

DTMF (RFC2833)

Definition at line 43 of file rtp.h.

Referenced by add_noncodec_to_sdp(), ast_rtp_read(), ast_rtp_senddigit_begin(), bridge_p2p_rtp_write(), check_user_full(), create_addr(), create_addr_from_peer(), oh323_alloc(), oh323_request(), process_sdp(), sip_alloc(), and sip_dtmfmode().

#define AST_RTP_MAX   AST_RTP_CISCO_DTMF

Maximum RTP-specific code

Definition at line 49 of file rtp.h.

Referenced by add_sdp(), and ast_rtp_lookup_mime_multiple().

#define FLAG_3389_WARNING   (1 << 0)

Definition at line 93 of file rtp.h.

#define MAX_RTP_PT   256

Definition at line 51 of file rtp.h.

Referenced by ast_rtp_get_current_formats(), ast_rtp_lookup_code(), ast_rtp_lookup_pt(), ast_rtp_pt_clear(), ast_rtp_pt_copy(), ast_rtp_pt_default(), ast_rtp_set_m_type(), ast_rtp_set_rtpmap_type(), ast_rtp_unset_m_type(), and process_sdp().


Typedef Documentation

typedef int(*) ast_rtp_callback(struct ast_rtp *rtp, struct ast_frame *f, void *data)

Definition at line 95 of file rtp.h.


Enumeration Type Documentation

enum ast_rtp_get_result

Enumerator:
AST_RTP_GET_FAILED  Failed to find the RTP structure
AST_RTP_TRY_PARTIAL  RTP structure exists but true native bridge can not occur so try partial
AST_RTP_TRY_NATIVE  RTP structure exists and native bridge can occur

Definition at line 57 of file rtp.h.

00057                         {
00058    /*! Failed to find the RTP structure */
00059    AST_RTP_GET_FAILED = 0,
00060    /*! RTP structure exists but true native bridge can not occur so try partial */
00061    AST_RTP_TRY_PARTIAL,
00062    /*! RTP structure exists and native bridge can occur */
00063    AST_RTP_TRY_NATIVE,
00064 };

enum ast_rtp_options

Enumerator:
AST_RTP_OPT_G726_NONSTANDARD 

Definition at line 53 of file rtp.h.

00053                      {
00054    AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
00055 };


Function Documentation

int ast_rtcp_fd ( struct ast_rtp rtp  ) 

Definition at line 518 of file rtp.c.

References ast_rtp::rtcp, and ast_rtcp::s.

Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), sip_new(), and start_rtp().

00519 {
00520    if (rtp->rtcp)
00521       return rtp->rtcp->s;
00522    return -1;
00523 }

struct ast_frame* ast_rtcp_read ( struct ast_rtp rtp  ) 

Definition at line 827 of file rtp.c.

References ast_rtcp::accumulated_transit, ast_assert, AST_CONTROL_VIDUPDATE, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), ast_frame::datalen, errno, ast_rtp::f, f, ast_frame::frametype, len, LOG_DEBUG, LOG_WARNING, ast_frame::mallocd, ast_rtcp::maxrtt, ast_rtcp::minrtt, ast_rtp::nat, option_debug, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, rtcp_debug_test_addr(), RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, ast_rtcp::rtt, ast_rtcp::rxlsr, ast_rtp::s, ast_rtcp::s, ast_frame::samples, ast_rtcp::soc, ast_rtcp::spc, ast_frame::src, ast_frame::subclass, ast_rtcp::them, ast_rtcp::themrxlsr, and timeval2ntp().

Referenced by oh323_read(), sip_rtp_read(), and skinny_rtp_read().

00828 {
00829    socklen_t len;
00830    int position, i, packetwords;
00831    int res;
00832    struct sockaddr_in sin;
00833    unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
00834    unsigned int *rtcpheader;
00835    int pt;
00836    struct timeval now;
00837    unsigned int length;
00838    int rc;
00839    double rttsec;
00840    uint64_t rtt = 0;
00841    unsigned int dlsr;
00842    unsigned int lsr;
00843    unsigned int msw;
00844    unsigned int lsw;
00845    unsigned int comp;
00846    struct ast_frame *f = &ast_null_frame;
00847    
00848    if (!rtp || !rtp->rtcp)
00849       return &ast_null_frame;
00850 
00851    len = sizeof(sin);
00852    
00853    res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
00854                0, (struct sockaddr *)&sin, &len);
00855    if (option_debug)
00856       ast_log(LOG_DEBUG, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
00857 
00858    rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
00859    
00860    if (res < 0) {
00861       ast_assert(errno != EBADF);
00862       if (errno != EAGAIN) {
00863          ast_log(LOG_WARNING, "RTCP Read error: %s.  Hanging up.\n", strerror(errno));
00864          ast_log(LOG_WARNING, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
00865          return NULL;
00866       }
00867       return &ast_null_frame;
00868    }
00869 
00870    packetwords = res / 4;
00871    
00872    if (rtp->nat) {
00873       /* Send to whoever sent to us */
00874       if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
00875           (rtp->rtcp->them.sin_port != sin.sin_port)) {
00876          memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
00877          if (option_debug || rtpdebug)
00878             ast_log(LOG_DEBUG, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00879       }
00880    }
00881 
00882    if (option_debug)
00883       ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
00884 
00885    /* Process a compound packet */
00886    position = 0;
00887    while (position < packetwords) {
00888       i = position;
00889       length = ntohl(rtcpheader[i]);
00890       pt = (length & 0xff0000) >> 16;
00891       rc = (length & 0x1f000000) >> 24;
00892       length &= 0xffff;
00893     
00894       if ((i + length) > packetwords) {
00895          ast_log(LOG_WARNING, "RTCP Read too short\n");
00896          return &ast_null_frame;
00897       }
00898       
00899       if (rtcp_debug_test_addr(&sin)) {
00900          ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
00901          ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
00902          ast_verbose("Reception reports: %d\n", rc);
00903          ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
00904       }
00905     
00906       i += 2; /* Advance past header and ssrc */
00907       
00908       switch (pt) {
00909       case RTCP_PT_SR:
00910          gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
00911          rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
00912          rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
00913          rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
00914     
00915          if (rtcp_debug_test_addr(&sin)) {
00916             ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
00917             ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
00918             ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
00919          }
00920          i += 5;
00921          if (rc < 1)
00922             break;
00923          /* Intentional fall through */
00924       case RTCP_PT_RR:
00925          /* Don't handle multiple reception reports (rc > 1) yet */
00926          /* Calculate RTT per RFC */
00927          gettimeofday(&now, NULL);
00928          timeval2ntp(now, &msw, &lsw);
00929          if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */
00930             comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
00931             lsr = ntohl(rtcpheader[i + 4]);
00932             dlsr = ntohl(rtcpheader[i + 5]);
00933             rtt = comp - lsr - dlsr;
00934 
00935             /* Convert end to end delay to usec (keeping the calculation in 64bit space)
00936                sess->ee_delay = (eedelay * 1000) / 65536; */
00937             if (rtt < 4294) {
00938                 rtt = (rtt * 1000000) >> 16;
00939             } else {
00940                 rtt = (rtt * 1000) >> 16;
00941                 rtt *= 1000;
00942             }
00943             rtt = rtt / 1000.;
00944             rttsec = rtt / 1000.;
00945 
00946             if (comp - dlsr >= lsr) {
00947                rtp->rtcp->accumulated_transit += rttsec;
00948                rtp->rtcp->rtt = rttsec;
00949                if (rtp->rtcp->maxrtt<rttsec)
00950                   rtp->rtcp->maxrtt = rttsec;
00951                if (rtp->rtcp->minrtt>rttsec)
00952                   rtp->rtcp->minrtt = rttsec;
00953             } else if (rtcp_debug_test_addr(&sin)) {
00954                ast_verbose("Internal RTCP NTP clock skew detected: "
00955                         "lsr=%u, now=%u, dlsr=%u (%d:%03dms), "
00956                         "diff=%d\n",
00957                         lsr, comp, dlsr, dlsr / 65536,
00958                         (dlsr % 65536) * 1000 / 65536,
00959                         dlsr - (comp - lsr));
00960             }
00961          }
00962 
00963          rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
00964          rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
00965          if (rtcp_debug_test_addr(&sin)) {
00966             ast_verbose("  Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
00967             ast_verbose("  Packets lost so far: %d\n", rtp->rtcp->reported_lost);
00968             ast_verbose("  Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
00969             ast_verbose("  Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
00970             ast_verbose("  Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
00971             ast_verbose("  Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
00972             ast_verbose("  DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
00973             if (rtt)
00974                ast_verbose("  RTT: %lu(sec)\n", (unsigned long) rtt);
00975          }
00976          break;
00977       case RTCP_PT_FUR:
00978          if (rtcp_debug_test_addr(&sin))
00979             ast_verbose("Received an RTCP Fast Update Request\n");
00980          rtp->f.frametype = AST_FRAME_CONTROL;
00981          rtp->f.subclass = AST_CONTROL_VIDUPDATE;
00982          rtp->f.datalen = 0;
00983          rtp->f.samples = 0;
00984          rtp->f.mallocd = 0;
00985          rtp->f.src = "RTP";
00986          f = &rtp->f;
00987          break;
00988       case RTCP_PT_SDES:
00989          if (rtcp_debug_test_addr(&sin))
00990             ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00991          break;
00992       case RTCP_PT_BYE:
00993          if (rtcp_debug_test_addr(&sin))
00994             ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00995          break;
00996       default:
00997          if (option_debug)
00998             ast_log(LOG_DEBUG, "Unknown RTCP packet (pt=%d) received from %s:%d\n", pt, ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00999          break;
01000       }
01001       position += (length + 1);
01002    }
01003          
01004    return f;
01005 }

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 2357 of file rtp.c.

References ast_rtcp_write(), ast_rtp::rtcp, and ast_rtcp::sendfur.

02358 {
02359    struct ast_rtp *rtp = data;
02360    int res;
02361 
02362    rtp->rtcp->sendfur = 1;
02363    res = ast_rtcp_write(data);
02364    
02365    return res;
02366 }

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 398 of file rtp.c.

Referenced by process_sdp().

00399 {
00400    return sizeof(struct ast_rtp);
00401 }

int ast_rtp_bridge ( struct ast_channel c0,
struct ast_channel c1,
int  flags,
struct ast_frame **  fo,
struct ast_channel **  rc,
int  timeoutms 
)

Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk.

Definition at line 3295 of file rtp.c.

References AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_check_hangup(), ast_codec_pref_getsize(), ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_set_flag, ast_test_flag, ast_verbose(), bridge_native_loop(), bridge_p2p_loop(), ast_format_list::cur_ms, FLAG_HAS_DTMF, FLAG_P2P_NEED_DTMF, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, option_debug, option_verbose, ast_rtp::pref, ast_channel::rawreadformat, ast_channel::rawwriteformat, ast_channel_tech::send_digit_begin, ast_channel::tech, ast_channel::tech_pvt, and VERBOSE_PREFIX_3.

03296 {
03297    struct ast_rtp *p0 = NULL, *p1 = NULL;    /* Audio RTP Channels */
03298    struct ast_rtp *vp0 = NULL, *vp1 = NULL;  /* Video RTP channels */
03299    struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
03300    enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
03301    enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
03302    enum ast_bridge_result res = AST_BRIDGE_FAILED;
03303    int codec0 = 0, codec1 = 0;
03304    void *pvt0 = NULL, *pvt1 = NULL;
03305 
03306    /* Lock channels */
03307    ast_channel_lock(c0);
03308    while(ast_channel_trylock(c1)) {
03309       ast_channel_unlock(c0);
03310       usleep(1);
03311       ast_channel_lock(c0);
03312    }
03313 
03314    /* Ensure neither channel got hungup during lock avoidance */
03315    if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
03316       ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
03317       ast_channel_unlock(c0);
03318       ast_channel_unlock(c1);
03319       return AST_BRIDGE_FAILED;
03320    }
03321       
03322    /* Find channel driver interfaces */
03323    if (!(pr0 = get_proto(c0))) {
03324       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
03325       ast_channel_unlock(c0);
03326       ast_channel_unlock(c1);
03327       return AST_BRIDGE_FAILED;
03328    }
03329    if (!(pr1 = get_proto(c1))) {
03330       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
03331       ast_channel_unlock(c0);
03332       ast_channel_unlock(c1);
03333       return AST_BRIDGE_FAILED;
03334    }
03335 
03336    /* Get channel specific interface structures */
03337    pvt0 = c0->tech_pvt;
03338    pvt1 = c1->tech_pvt;
03339 
03340    /* Get audio and video interface (if native bridge is possible) */
03341    audio_p0_res = pr0->get_rtp_info(c0, &p0);
03342    video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
03343    audio_p1_res = pr1->get_rtp_info(c1, &p1);
03344    video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
03345 
03346    /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
03347    if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
03348       audio_p0_res = AST_RTP_GET_FAILED;
03349    if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
03350       audio_p1_res = AST_RTP_GET_FAILED;
03351 
03352    /* Check if a bridge is possible (partial/native) */
03353    if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
03354       /* Somebody doesn't want to play... */
03355       ast_channel_unlock(c0);
03356       ast_channel_unlock(c1);
03357       return AST_BRIDGE_FAILED_NOWARN;
03358    }
03359 
03360    /* If we need to feed DTMF frames into the core then only do a partial native bridge */
03361    if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
03362       ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
03363       audio_p0_res = AST_RTP_TRY_PARTIAL;
03364    }
03365 
03366    if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
03367       ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
03368       audio_p1_res = AST_RTP_TRY_PARTIAL;
03369    }
03370 
03371    /* If both sides are not using the same method of DTMF transmission 
03372     * (ie: one is RFC2833, other is INFO... then we can not do direct media. 
03373     * --------------------------------------------------
03374     * | DTMF Mode |  HAS_DTMF  |  Accepts Begin Frames |
03375     * |-----------|------------|-----------------------|
03376     * | Inband    | False      | True                  |
03377     * | RFC2833   | True       | True                  |
03378     * | SIP INFO  | False      | False                 |
03379     * --------------------------------------------------
03380     * However, if DTMF from both channels is being monitored by the core, then
03381     * we can still do packet-to-packet bridging, because passing through the 
03382     * core will handle DTMF mode translation.
03383     */
03384    if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
03385        (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
03386       if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
03387          ast_channel_unlock(c0);
03388          ast_channel_unlock(c1);
03389          return AST_BRIDGE_FAILED_NOWARN;
03390       }
03391       audio_p0_res = AST_RTP_TRY_PARTIAL;
03392       audio_p1_res = AST_RTP_TRY_PARTIAL;
03393    }
03394 
03395    /* If we need to feed frames into the core don't do a P2P bridge */
03396    if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) ||
03397        (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) {
03398       ast_channel_unlock(c0);
03399       ast_channel_unlock(c1);
03400       return AST_BRIDGE_FAILED_NOWARN;
03401    }
03402 
03403    /* Get codecs from both sides */
03404    codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
03405    codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
03406    if (codec0 && codec1 && !(codec0 & codec1)) {
03407       /* Hey, we can't do native bridging if both parties speak different codecs */
03408       if (option_debug)
03409          ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
03410       ast_channel_unlock(c0);
03411       ast_channel_unlock(c1);
03412       return AST_BRIDGE_FAILED_NOWARN;
03413    }
03414 
03415    /* If either side can only do a partial bridge, then don't try for a true native bridge */
03416    if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
03417       struct ast_format_list fmt0, fmt1;
03418 
03419       /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
03420       if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
03421          if (option_debug)
03422             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n");
03423          ast_channel_unlock(c0);
03424          ast_channel_unlock(c1);
03425          return AST_BRIDGE_FAILED_NOWARN;
03426       }
03427       /* They must also be using the same packetization */
03428       fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
03429       fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
03430       if (fmt0.cur_ms != fmt1.cur_ms) {
03431          if (option_debug)
03432             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n");
03433          ast_channel_unlock(c0);
03434          ast_channel_unlock(c1);
03435          return AST_BRIDGE_FAILED_NOWARN;
03436       }
03437 
03438       if (option_verbose > 2)
03439          ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
03440       res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
03441    } else {
03442       if (option_verbose > 2) 
03443          ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
03444       res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
03445    }
03446 
03447    return res;
03448 }

int ast_rtp_codec_getformat ( int  pt  ) 

Definition at line 2739 of file rtp.c.

References rtpPayloadType::code, and static_RTP_PT.

Referenced by process_sdp().

02740 {
02741    if (pt < 0 || pt > MAX_RTP_PT)
02742       return 0; /* bogus payload type */
02743 
02744    if (static_RTP_PT[pt].isAstFormat)
02745       return static_RTP_PT[pt].code;
02746    else
02747       return 0;
02748 }

struct ast_codec_pref* ast_rtp_codec_getpref ( struct ast_rtp rtp  ) 

Definition at line 2734 of file rtp.c.

References ast_rtp::pref.

Referenced by add_codec_to_sdp(), and process_sdp().

02735 {
02736    return &rtp->pref;
02737 }

int ast_rtp_codec_setpref ( struct ast_rtp rtp,
struct ast_codec_pref prefs 
)

Definition at line 2721 of file rtp.c.

References ast_smoother_free(), ast_codec_pref::framing, ast_codec_pref::order, ast_rtp::pref, prefs, and ast_rtp::smoother.

Referenced by __oh323_rtp_create(), check_user_full(), create_addr_from_peer(), process_sdp(), register_verify(), set_peer_capabilities(), sip_alloc(), start_rtp(), and transmit_response_with_sdp().

02722 {
02723    int x;
02724    for (x = 0; x < 32; x++) {  /* Ugly way */
02725       rtp->pref.order[x] = prefs->order[x];
02726       rtp->pref.framing[x] = prefs->framing[x];
02727    }
02728    if (rtp->smoother)
02729       ast_smoother_free(rtp->smoother);
02730    rtp->smoother = NULL;
02731    return 0;
02732 }

void ast_rtp_destroy ( struct ast_rtp rtp  ) 

Definition at line 2140 of file rtp.c.

References ast_io_remove(), ast_mutex_destroy(), AST_SCHED_DEL, ast_smoother_free(), ast_verbose(), ast_rtp::bridge_lock, ast_rtcp::expected_prior, free, ast_rtp::io, ast_rtp::ioid, ast_rtcp::received_prior, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtcp::rr_count, ast_rtp::rtcp, rtcp_debug_test_addr(), ast_rtcp::rtt, ast_rtp::rxcount, ast_rtp::rxjitter, ast_rtp::rxtransit, ast_rtcp::s, ast_rtp::s, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::smoother, ast_rtcp::sr_count, ast_rtp::ssrc, ast_rtp::them, ast_rtp::themssrc, and ast_rtp::txcount.

Referenced by __oh323_destroy(), __sip_destroy(), check_user_full(), cleanup_connection(), create_addr_from_peer(), destroy_endpoint(), gtalk_free_pvt(), mgcp_hangup(), oh323_alloc(), skinny_hangup(), start_rtp(), and unalloc_sub().

02141 {
02142    if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
02143       /*Print some info on the call here */
02144       ast_verbose("  RTP-stats\n");
02145       ast_verbose("* Our Receiver:\n");
02146       ast_verbose("  SSRC:     %u\n", rtp->themssrc);
02147       ast_verbose("  Received packets: %u\n", rtp->rxcount);
02148       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
02149       ast_verbose("  Jitter:      %.4f\n", rtp->rxjitter);
02150       ast_verbose("  Transit:     %.4f\n", rtp->rxtransit);
02151       ast_verbose("  RR-count:    %u\n", rtp->rtcp->rr_count);
02152       ast_verbose("* Our Sender:\n");
02153       ast_verbose("  SSRC:     %u\n", rtp->ssrc);
02154       ast_verbose("  Sent packets:   %u\n", rtp->txcount);
02155       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->reported_lost);
02156       ast_verbose("  Jitter:      %u\n", rtp->rtcp->reported_jitter / (unsigned int)65536.0);
02157       ast_verbose("  SR-count:    %u\n", rtp->rtcp->sr_count);
02158       ast_verbose("  RTT:      %f\n", rtp->rtcp->rtt);
02159    }
02160 
02161    if (rtp->smoother)
02162       ast_smoother_free(rtp->smoother);
02163    if (rtp->ioid)
02164       ast_io_remove(rtp->io, rtp->ioid);
02165    if (rtp->s > -1)
02166       close(rtp->s);
02167    if (rtp->rtcp) {
02168       AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
02169       close(rtp->rtcp->s);
02170       free(rtp->rtcp);
02171       rtp->rtcp=NULL;
02172    }
02173 
02174    ast_mutex_destroy(&rtp->bridge_lock);
02175 
02176    free(rtp);
02177 }

int ast_rtp_early_bridge ( struct ast_channel dest,
struct ast_channel src 
)

If possible, create an early bridge directly between the devices without having to send a re-invite later.

Definition at line 1484 of file rtp.c.

References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, option_debug, and ast_rtp_protocol::set_rtp_peer.

Referenced by wait_for_answer().

01485 {
01486    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01487    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01488    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01489    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01490    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
01491    int srccodec, destcodec, nat_active = 0;
01492 
01493    /* Lock channels */
01494    ast_channel_lock(dest);
01495    if (src) {
01496       while(ast_channel_trylock(src)) {
01497          ast_channel_unlock(dest);
01498          usleep(1);
01499          ast_channel_lock(dest);
01500       }
01501    }
01502 
01503    /* Find channel driver interfaces */
01504    destpr = get_proto(dest);
01505    if (src)
01506       srcpr = get_proto(src);
01507    if (!destpr) {
01508       if (option_debug)
01509          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01510       ast_channel_unlock(dest);
01511       if (src)
01512          ast_channel_unlock(src);
01513       return 0;
01514    }
01515    if (!srcpr) {
01516       if (option_debug)
01517          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
01518       ast_channel_unlock(dest);
01519       if (src)
01520          ast_channel_unlock(src);
01521       return 0;
01522    }
01523 
01524    /* Get audio and video interface (if native bridge is possible) */
01525    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01526    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01527    if (srcpr) {
01528       audio_src_res = srcpr->get_rtp_info(src, &srcp);
01529       video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01530    }
01531 
01532    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01533    if (audio_dest_res != AST_RTP_TRY_NATIVE) {
01534       /* Somebody doesn't want to play... */
01535       ast_channel_unlock(dest);
01536       if (src)
01537          ast_channel_unlock(src);
01538       return 0;
01539    }
01540    if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
01541       srccodec = srcpr->get_codec(src);
01542    else
01543       srccodec = 0;
01544    if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
01545       destcodec = destpr->get_codec(dest);
01546    else
01547       destcodec = 0;
01548    /* Ensure we have at least one matching codec */
01549    if (!(srccodec & destcodec)) {
01550       ast_channel_unlock(dest);
01551       if (src)
01552          ast_channel_unlock(src);
01553       return 0;
01554    }
01555    /* Consider empty media as non-existant */
01556    if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
01557       srcp = NULL;
01558    /* If the client has NAT stuff turned on then just safe NAT is active */
01559    if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01560       nat_active = 1;
01561    /* Bridge media early */
01562    if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active))
01563       ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>");
01564    ast_channel_unlock(dest);
01565    if (src)
01566       ast_channel_unlock(src);
01567    if (option_debug)
01568       ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
01569    return 1;
01570 }

int ast_rtp_fd ( struct ast_rtp rtp  ) 

Definition at line 513 of file rtp.c.

References ast_rtp::s.

Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), mgcp_new(), sip_new(), skinny_new(), and start_rtp().

00514 {
00515    return rtp->s;
00516 }

struct ast_rtp* ast_rtp_get_bridged ( struct ast_rtp rtp  ) 

Definition at line 2049 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, and ast_rtp::bridged.

Referenced by __sip_destroy(), and ast_rtp_read().

02050 {
02051    struct ast_rtp *bridged = NULL;
02052 
02053    ast_mutex_lock(&rtp->bridge_lock);
02054    bridged = rtp->bridged;
02055    ast_mutex_unlock(&rtp->bridge_lock);
02056 
02057    return bridged;
02058 }

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 1706 of file rtp.c.

References ast_mutex_lock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, and MAX_RTP_PT.

Referenced by gtalk_is_answered(), gtalk_newcall(), and process_sdp().

01708 {
01709    int pt;
01710    
01711    ast_mutex_lock(&rtp->bridge_lock);
01712    
01713    *astFormats = *nonAstFormats = 0;
01714    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01715       if (rtp->current_RTP_PT[pt].isAstFormat) {
01716          *astFormats |= rtp->current_RTP_PT[pt].code;
01717       } else {
01718          *nonAstFormats |= rtp->current_RTP_PT[pt].code;
01719       }
01720    }
01721    
01722    ast_mutex_unlock(&rtp->bridge_lock);
01723    
01724    return;
01725 }

int ast_rtp_get_peer ( struct ast_rtp rtp,
struct sockaddr_in *  them 
)

Definition at line 2031 of file rtp.c.

References ast_rtp::them.

Referenced by add_sdp(), bridge_native_loop(), do_monitor(), gtalk_update_stun(), oh323_set_rtp_peer(), process_sdp(), sip_set_rtp_peer(), and transmit_modify_with_sdp().

02032 {
02033    if ((them->sin_family != AF_INET) ||
02034       (them->sin_port != rtp->them.sin_port) ||
02035       (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
02036       them->sin_family = AF_INET;
02037       them->sin_port = rtp->them.sin_port;
02038       them->sin_addr = rtp->them.sin_addr;
02039       return 1;
02040    }
02041    return 0;
02042 }

char* ast_rtp_get_quality ( struct ast_rtp rtp,
struct ast_rtp_quality qual 
)

Return RTCP quality string.

Definition at line 2096 of file rtp.c.

References ast_rtcp::expected_prior, ast_rtp_quality::local_count, ast_rtp_quality::local_jitter, ast_rtp_quality::local_lostpackets, ast_rtp_quality::local_ssrc, ast_rtcp::quality, ast_rtcp::received_prior, ast_rtp_quality::remote_count, ast_rtp_quality::remote_jitter, ast_rtp_quality::remote_lostpackets, ast_rtp_quality::remote_ssrc, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, ast_rtcp::rtt, ast_rtp_quality::rtt, ast_rtp::rxcount, ast_rtp::rxjitter, ast_rtp::ssrc, ast_rtp::themssrc, and ast_rtp::txcount.

Referenced by acf_channel_read(), handle_request_bye(), and sip_hangup().

02097 {
02098    /*
02099    *ssrc          our ssrc
02100    *themssrc      their ssrc
02101    *lp            lost packets
02102    *rxjitter      our calculated jitter(rx)
02103    *rxcount       no. received packets
02104    *txjitter      reported jitter of the other end
02105    *txcount       transmitted packets
02106    *rlp           remote lost packets
02107    *rtt           round trip time
02108    */
02109 
02110    if (qual && rtp) {
02111       qual->local_ssrc = rtp->ssrc;
02112       qual->local_jitter = rtp->rxjitter;
02113       qual->local_count = rtp->rxcount;
02114       qual->remote_ssrc = rtp->themssrc;
02115       qual->remote_count = rtp->txcount;
02116       if (rtp->rtcp) {
02117          qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
02118          qual->remote_lostpackets = rtp->rtcp->reported_lost;
02119          qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
02120          qual->rtt = rtp->rtcp->rtt;
02121       }
02122    }
02123    if (rtp->rtcp) {
02124       snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality),
02125          "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
02126          rtp->ssrc,
02127          rtp->themssrc,
02128          rtp->rtcp->expected_prior - rtp->rtcp->received_prior,
02129          rtp->rxjitter,
02130          rtp->rxcount,
02131          (double)rtp->rtcp->reported_jitter / 65536.0,
02132          rtp->txcount,
02133          rtp->rtcp->reported_lost,
02134          rtp->rtcp->rtt);
02135       return rtp->rtcp->quality;
02136    } else
02137       return "<Unknown> - RTP/RTCP has already been destroyed";
02138 }

int ast_rtp_get_rtpholdtimeout ( struct ast_rtp rtp  ) 

Get rtp hold timeout.

Definition at line 568 of file rtp.c.

References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.

Referenced by do_monitor().

00569 {
00570    if (rtp->rtptimeout < 0)   /* We're not checking, but remembering the setting (during T.38 transmission) */
00571       return 0;
00572    return rtp->rtpholdtimeout;
00573 }

int ast_rtp_get_rtpkeepalive ( struct ast_rtp rtp  ) 

Get RTP keepalive interval.

Definition at line 576 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by do_monitor().

00577 {
00578    return rtp->rtpkeepalive;
00579 }

int ast_rtp_get_rtptimeout ( struct ast_rtp rtp  ) 

Get rtp timeout.

Definition at line 560 of file rtp.c.

References ast_rtp::rtptimeout.

Referenced by do_monitor().

00561 {
00562    if (rtp->rtptimeout < 0)   /* We're not checking, but remembering the setting (during T.38 transmission) */
00563       return 0;
00564    return rtp->rtptimeout;
00565 }

void ast_rtp_get_us ( struct ast_rtp rtp,
struct sockaddr_in *  us 
)

Definition at line 2044 of file rtp.c.

References ast_rtp::us.

Referenced by add_sdp(), external_rtp_create(), gtalk_create_candidates(), handle_open_receive_channel_ack_message(), and oh323_set_rtp_peer().

02045 {
02046    *us = rtp->us;
02047 }

int ast_rtp_getnat ( struct ast_rtp rtp  ) 

Definition at line 596 of file rtp.c.

References ast_test_flag, and FLAG_NAT_ACTIVE.

Referenced by sip_get_rtp_peer().

00597 {
00598    return ast_test_flag(rtp, FLAG_NAT_ACTIVE);
00599 }

void ast_rtp_init ( void   ) 

Initialize the RTP system in Asterisk.

Definition at line 3833 of file rtp.c.

References ast_cli_register_multiple(), ast_rtp_reload(), and cli_rtp.

Referenced by main().

03834 {
03835    ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
03836    ast_rtp_reload();
03837 }

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 1749 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by add_codec_to_answer(), add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), ast_rtp_sendcng(), ast_rtp_senddigit_begin(), ast_rtp_write(), and bridge_p2p_rtp_write().

01750 {
01751    int pt = 0;
01752 
01753    ast_mutex_lock(&rtp->bridge_lock);
01754 
01755    if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
01756       code == rtp->rtp_lookup_code_cache_code) {
01757       /* Use our cached mapping, to avoid the overhead of the loop below */
01758       pt = rtp->rtp_lookup_code_cache_result;
01759       ast_mutex_unlock(&rtp->bridge_lock);
01760       return pt;
01761    }
01762 
01763    /* Check the dynamic list first */
01764    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01765       if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
01766          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01767          rtp->rtp_lookup_code_cache_code = code;
01768          rtp->rtp_lookup_code_cache_result = pt;
01769          ast_mutex_unlock(&rtp->bridge_lock);
01770          return pt;
01771       }
01772    }
01773 
01774    /* Then the static list */
01775    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01776       if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
01777          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01778          rtp->rtp_lookup_code_cache_code = code;
01779          rtp->rtp_lookup_code_cache_result = pt;
01780          ast_mutex_unlock(&rtp->bridge_lock);
01781          return pt;
01782       }
01783    }
01784 
01785    ast_mutex_unlock(&rtp->bridge_lock);
01786 
01787    return -1;
01788 }

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 1809 of file rtp.c.

References ast_rtp_lookup_mime_subtype(), AST_RTP_MAX, format, len, and name.

Referenced by process_sdp().

01811 {
01812    int format;
01813    unsigned len;
01814    char *end = buf;
01815    char *start = buf;
01816 
01817    if (!buf || !size)
01818       return NULL;
01819 
01820    snprintf(end, size, "0x%x (", capability);
01821 
01822    len = strlen(end);
01823    end += len;
01824    size -= len;
01825    start = end;
01826 
01827    for (format = 1; format < AST_RTP_MAX; format <<= 1) {
01828       if (capability & format) {
01829          const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
01830 
01831          snprintf(end, size, "%s|", name);
01832          len = strlen(end);
01833          end += len;
01834          size -= len;
01835       }
01836    }
01837 
01838    if (start == end)
01839       snprintf(start, size, "nothing)"); 
01840    else if (size > 1)
01841       *(end -1) = ')';
01842    
01843    return buf;
01844 }

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 1790 of file rtp.c.

References AST_FORMAT_G726_AAL2, AST_RTP_OPT_G726_NONSTANDARD, rtpPayloadType::code, mimeTypes, and payloadType.

Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), ast_rtp_lookup_mime_multiple(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().

01792 {
01793    unsigned int i;
01794 
01795    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01796       if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
01797          if (isAstFormat &&
01798              (code == AST_FORMAT_G726_AAL2) &&
01799              (options & AST_RTP_OPT_G726_NONSTANDARD))
01800             return "G726-32";
01801          else
01802             return mimeTypes[i].subtype;
01803       }
01804    }
01805 
01806    return "";
01807 }

struct rtpPayloadType ast_rtp_lookup_pt ( struct ast_rtp rtp,
int  pt 
)

Mapping between RTP payload format codes and Asterisk codes:.

Definition at line 1727 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), rtpPayloadType::isAstFormat, MAX_RTP_PT, and static_RTP_PT.

Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and setup_rtp_connection().

01728 {
01729    struct rtpPayloadType result;
01730 
01731    result.isAstFormat = result.code = 0;
01732 
01733    if (pt < 0 || pt > MAX_RTP_PT) 
01734       return result; /* bogus payload type */
01735 
01736    /* Start with negotiated codecs */
01737    ast_mutex_lock(&rtp->bridge_lock);
01738    result = rtp->current_RTP_PT[pt];
01739    ast_mutex_unlock(&rtp->bridge_lock);
01740 
01741    /* If it doesn't exist, check our static RTP type list, just in case */
01742    if (!result.code) 
01743       result = static_RTP_PT[pt];
01744 
01745    return result;
01746 }

int ast_rtp_make_compatible ( struct ast_channel dest,
struct ast_channel src,
int  media 
)

Definition at line 1572 of file rtp.c.

References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_log(), AST_RTP_GET_FAILED, ast_rtp_pt_copy(), AST_RTP_TRY_NATIVE, ast_test_flag, FLAG_NAT_ACTIVE, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, LOG_WARNING, option_debug, and ast_rtp_protocol::set_rtp_peer.

Referenced by wait_for_answer().

01573 {
01574    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01575    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01576    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01577    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01578    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 
01579    int srccodec, destcodec;
01580 
01581    /* Lock channels */
01582    ast_channel_lock(dest);
01583    while(ast_channel_trylock(src)) {
01584       ast_channel_unlock(dest);
01585       usleep(1);
01586       ast_channel_lock(dest);
01587    }
01588 
01589    /* Find channel driver interfaces */
01590    if (!(destpr = get_proto(dest))) {
01591       if (option_debug)
01592          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01593       ast_channel_unlock(dest);
01594       ast_channel_unlock(src);
01595       return 0;
01596    }
01597    if (!(srcpr = get_proto(src))) {
01598       if (option_debug)
01599          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
01600       ast_channel_unlock(dest);
01601       ast_channel_unlock(src);
01602       return 0;
01603    }
01604 
01605    /* Get audio and video interface (if native bridge is possible) */
01606    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01607    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01608    audio_src_res = srcpr->get_rtp_info(src, &srcp);
01609    video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01610 
01611    /* Ensure we have at least one matching codec */
01612    if (srcpr->get_codec)
01613       srccodec = srcpr->get_codec(src);
01614    else
01615       srccodec = 0;
01616    if (destpr->get_codec)
01617       destcodec = destpr->get_codec(dest);
01618    else
01619       destcodec = 0;
01620 
01621    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01622    if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
01623       /* Somebody doesn't want to play... */
01624       ast_channel_unlock(dest);
01625       ast_channel_unlock(src);
01626       return 0;
01627    }
01628    ast_rtp_pt_copy(destp, srcp);
01629    if (vdestp && vsrcp)
01630       ast_rtp_pt_copy(vdestp, vsrcp);
01631    if (media) {
01632       /* Bridge early */
01633       if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01634          ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
01635    }
01636    ast_channel_unlock(dest);
01637    ast_channel_unlock(src);
01638    if (option_debug)
01639       ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
01640    return 1;
01641 }

struct ast_rtp* ast_rtp_new ( struct sched_context sched,
struct io_context io,
int  rtcpenable,
int  callbackmode 
)

Initializate a RTP session.

Parameters:
sched 
io 
rtcpenable 
callbackmode 
Returns:
A representation (structure) of an RTP session.

Definition at line 1995 of file rtp.c.

References ast_rtp_new_with_bindaddr(), io, and sched.

01996 {
01997    struct in_addr ia;
01998 
01999    memset(&ia, 0, sizeof(ia));
02000    return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
02001 }

void ast_rtp_new_init ( struct ast_rtp rtp  ) 

Initialize a new RTP structure.

Definition at line 1892 of file rtp.c.

References ast_mutex_init(), ast_random(), ast_set_flag, ast_rtp::bridge_lock, FLAG_HAS_DTMF, ast_rtp::seqno, ast_rtp::ssrc, ast_rtp::them, and ast_rtp::us.

Referenced by ast_rtp_new_with_bindaddr(), and process_sdp().

01893 {
01894    ast_mutex_init(&rtp->bridge_lock);
01895 
01896    rtp->them.sin_family = AF_INET;
01897    rtp->us.sin_family = AF_INET;
01898    rtp->ssrc = ast_random();
01899    rtp->seqno = ast_random() & 0xffff;
01900    ast_set_flag(rtp, FLAG_HAS_DTMF);
01901 
01902    return;
01903 }

void ast_rtp_new_source ( struct ast_rtp rtp  ) 

Definition at line 2012 of file rtp.c.

References ast_rtp::set_marker_bit.

Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), sip_write(), and skinny_indicate().

02013 {
02014    if (rtp) {
02015       rtp->set_marker_bit = 1;
02016    }
02017    return;
02018 }

struct ast_rtp* ast_rtp_new_with_bindaddr ( struct sched_context sched,
struct io_context io,
int  rtcpenable,
int  callbackmode,
struct in_addr  in 
)

Initializate a RTP session using an in_addr structure.

This fuction gets called by ast_rtp_new().

Parameters:
sched 
io 
rtcpenable 
callbackmode 
in 
Returns:
A representation (structure) of an RTP session.

Definition at line 1905 of file rtp.c.

References ast_calloc, ast_io_add(), AST_IO_IN, ast_log(), ast_random(), ast_rtcp_new(), ast_rtp_new_init(), ast_rtp_pt_default(), ast_set_flag, errno, FLAG_CALLBACK_MODE, free, io, LOG_ERROR, LOG_NOTICE, rtp_socket(), rtpread(), and sched.

Referenced by __oh323_rtp_create(), ast_rtp_new(), gtalk_alloc(), sip_alloc(), and start_rtp().

01906 {
01907    struct ast_rtp *rtp;
01908    int x;
01909    int first;
01910    int startplace;
01911    
01912    if (!(rtp = ast_calloc(1, sizeof(*rtp))))
01913       return NULL;
01914 
01915    ast_rtp_new_init(rtp);
01916 
01917    rtp->s = rtp_socket();
01918    ast_log(LOG_NOTICE, "socket RTP fd: %i\n", rtp->s); 
01919    if (rtp->s < 0) {
01920       free(rtp);
01921       ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
01922       return NULL;
01923    }
01924    if (sched && rtcpenable) {
01925       rtp->sched = sched;
01926       rtp->rtcp = ast_rtcp_new();
01927       ast_log(LOG_NOTICE, "socket RTCP fd: %i\n", rtp->rtcp->s);
01928    }
01929    
01930    /* Select a random port number in the range of possible RTP */
01931    x = (ast_random() % (rtpend-rtpstart)) + rtpstart;
01932    x = x & ~1;
01933    /* Save it for future references. */
01934    startplace = x;
01935    /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
01936    for (;;) {
01937       /* Must be an even port number by RTP spec */
01938       rtp->us.sin_port = htons(x);
01939       rtp->us.sin_addr = addr;
01940       /* If there's rtcp, initialize it as well. */
01941       if (rtp->rtcp) {
01942          rtp->rtcp->us.sin_port = htons(x + 1);
01943          rtp->rtcp->us.sin_addr = addr;
01944       }
01945       /* Try to bind it/them. */
01946       if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
01947          (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
01948          break;
01949       if (!first) {
01950          /* Primary bind succeeded! Gotta recreate it */
01951          close(rtp->s);
01952          rtp->s = rtp_socket();
01953          ast_log(LOG_NOTICE, "socket RTP2 fd: %i\n", rtp->s); 
01954       }
01955       if (errno != EADDRINUSE) {
01956          /* We got an error that wasn't expected, abort! */
01957          ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
01958          close(rtp->s);
01959          if (rtp->rtcp) {
01960             close(rtp->rtcp->s);
01961             free(rtp->rtcp);
01962          }
01963          free(rtp);
01964          return NULL;
01965       }
01966       /* The port was used, increment it (by two). */
01967       x += 2;
01968       /* Did we go over the limit ? */
01969       if (x > rtpend)
01970          /* then, start from the begingig. */
01971          x = (rtpstart + 1) & ~1;
01972       /* Check if we reached the place were we started. */
01973       if (x == startplace) {
01974          /* If so, there's no ports available. */
01975          ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
01976          close(rtp->s);
01977          if (rtp->rtcp) {
01978             close(rtp->rtcp->s);
01979             free(rtp->rtcp);
01980          }
01981          free(rtp);
01982          return NULL;
01983       }
01984    }
01985    rtp->sched = sched;
01986    rtp->io = io;
01987    if (callbackmode) {
01988       rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
01989       ast_set_flag(rtp, FLAG_CALLBACK_MODE);
01990    }
01991    ast_rtp_pt_default(rtp);
01992    return rtp;
01993 }

int ast_rtp_proto_register ( struct ast_rtp_protocol proto  ) 

Register interface to channel driver.

Definition at line 2850 of file rtp.c.

References AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_log(), LOG_WARNING, protos, and ast_rtp_protocol::type.

Referenced by load_module().

02851 {
02852    struct ast_rtp_protocol *cur;
02853 
02854    AST_LIST_LOCK(&protos);
02855    AST_LIST_TRAVERSE(&protos, cur, list) {   
02856       if (!strcmp(cur->type, proto->type)) {
02857          ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
02858          AST_LIST_UNLOCK(&protos);
02859          return -1;
02860       }
02861    }
02862    AST_LIST_INSERT_HEAD(&protos, proto, list);
02863    AST_LIST_UNLOCK(&protos);
02864    
02865    return 0;
02866 }

void ast_rtp_proto_unregister ( struct ast_rtp_protocol proto  ) 

Unregister interface to channel driver.

Definition at line 2842 of file rtp.c.

References AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, and protos.

Referenced by load_module(), and unload_module().

02843 {
02844    AST_LIST_LOCK(&protos);
02845    AST_LIST_REMOVE(&protos, proto, list);
02846    AST_LIST_UNLOCK(&protos);
02847 }

void ast_rtp_pt_clear ( struct ast_rtp rtp  ) 

Setting RTP payload types from lines in a SDP description:.

Definition at line 1408 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by gtalk_alloc(), and process_sdp().

01409 {
01410    int i;
01411 
01412    if (!rtp)
01413       return;
01414 
01415    ast_mutex_lock(&rtp->bridge_lock);
01416 
01417    for (i = 0; i < MAX_RTP_PT; ++i) {
01418       rtp->current_RTP_PT[i].isAstFormat = 0;
01419       rtp->current_RTP_PT[i].code = 0;
01420    }
01421 
01422    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01423    rtp->rtp_lookup_code_cache_code = 0;
01424    rtp->rtp_lookup_code_cache_result = 0;
01425 
01426    ast_mutex_unlock(&rtp->bridge_lock);
01427 }

void ast_rtp_pt_copy ( struct ast_rtp dest,
struct ast_rtp src 
)

Copy payload types between RTP structures.

Definition at line 1448 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by ast_rtp_make_compatible(), and process_sdp().

01449 {
01450    unsigned int i;
01451 
01452    ast_mutex_lock(&dest->bridge_lock);
01453    ast_mutex_lock(&src->bridge_lock);
01454 
01455    for (i=0; i < MAX_RTP_PT; ++i) {
01456       dest->current_RTP_PT[i].isAstFormat = 
01457          src->current_RTP_PT[i].isAstFormat;
01458       dest->current_RTP_PT[i].code = 
01459          src->current_RTP_PT[i].code; 
01460    }
01461    dest->rtp_lookup_code_cache_isAstFormat = 0;
01462    dest->rtp_lookup_code_cache_code = 0;
01463    dest->rtp_lookup_code_cache_result = 0;
01464 
01465    ast_mutex_unlock(&src->bridge_lock);
01466    ast_mutex_unlock(&dest->bridge_lock);
01467 }

void ast_rtp_pt_default ( struct ast_rtp rtp  ) 

Set payload types to defaults.

Definition at line 1429 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, ast_rtp::rtp_lookup_code_cache_result, and static_RTP_PT.

Referenced by ast_rtp_new_with_bindaddr().

01430 {
01431    int i;
01432 
01433    ast_mutex_lock(&rtp->bridge_lock);
01434 
01435    /* Initialize to default payload types */
01436    for (i = 0; i < MAX_RTP_PT; ++i) {
01437       rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
01438       rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
01439    }
01440 
01441    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01442    rtp->rtp_lookup_code_cache_code = 0;
01443    rtp->rtp_lookup_code_cache_result = 0;
01444 
01445    ast_mutex_unlock(&rtp->bridge_lock);
01446 }

struct ast_frame* ast_rtp_read ( struct ast_rtp rtp  ) 

Definition at line 1110 of file rtp.c.

References ast_assert, ast_codec_get_samples(), AST_FORMAT_MAX_AUDIO, ast_format_rate(), AST_FORMAT_SLINEAR, ast_frame_byteswap_be, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_get_bridged(), ast_rtp_lookup_pt(), ast_rtp_senddigit_continuation(), ast_sched_add(), ast_set_flag, ast_verbose(), bridge_p2p_rtp_write(), ast_rtp::bridged, calc_rxstamp(), rtpPayloadType::code, ast_rtp::cycles, ast_frame::data, ast_frame::datalen, ast_frame::delivery, errno, event, ext, ast_rtp::f, f, FLAG_NAT_ACTIVE, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lastevent, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len, LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_rtp::rawdata, ast_rtp::rtcp, rtp_debug_test_addr(), RTP_SEQ_MOD, ast_rtp::rxcount, ast_rtp::rxseqno, ast_rtp::rxssrc, ast_rtcp::s, ast_rtp::s, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, ast_rtp::sending_digit, ast_frame::seqno, ast_frame::src, STUN_ACCEPT, stun_handle_packet(), ast_frame::subclass, ast_rtcp::them, ast_rtp::them, ast_rtp::themssrc, and ast_frame::ts.

Referenced by gtalk_rtp_read(), mgcp_rtp_read(), oh323_rtp_read(), rtpread(), sip_rtp_read(), and skinny_rtp_read().

01111 {
01112    int res;
01113    struct sockaddr_in sin;
01114    socklen_t len;
01115    unsigned int seqno;
01116    int version;
01117    int payloadtype;
01118    int hdrlen = 12;
01119    int padding;
01120    int mark;
01121    int ext;
01122    int cc;
01123    unsigned int ssrc;
01124    unsigned int timestamp;
01125    unsigned int *rtpheader;
01126    struct rtpPayloadType rtpPT;
01127    struct ast_rtp *bridged = NULL;
01128    
01129    /* If time is up, kill it */
01130    if (rtp->sending_digit)
01131       ast_rtp_senddigit_continuation(rtp);
01132 
01133    len = sizeof(sin);
01134    
01135    /* Cache where the header will go */
01136    res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
01137                0, (struct sockaddr *)&sin, &len);
01138    if (option_debug > 3)
01139       ast_log(LOG_DEBUG, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
01140 
01141    rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
01142    if (res < 0) {
01143       ast_assert(errno != EBADF);
01144       if (errno != EAGAIN) {
01145          ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up.\n", strerror(errno));
01146          ast_log(LOG_WARNING, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
01147          return NULL;
01148       }
01149       return &ast_null_frame;
01150    }
01151    
01152    if (res < hdrlen) {
01153       ast_log(LOG_WARNING, "RTP Read too short\n");
01154       return &ast_null_frame;
01155    }
01156 
01157    /* Get fields */
01158    seqno = ntohl(rtpheader[0]);
01159 
01160    /* Check RTP version */
01161    version = (seqno & 0xC0000000) >> 30;
01162    if (!version) {
01163       if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) &&
01164          (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
01165          memcpy(&rtp->them, &sin, sizeof(rtp->them));
01166       }
01167       return &ast_null_frame;
01168    }
01169 
01170 #if 0 /* Allow to receive RTP stream with closed transmission path */
01171    /* If we don't have the other side's address, then ignore this */
01172    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
01173       return &ast_null_frame;
01174 #endif
01175 
01176    /* Send to whoever send to us if NAT is turned on */
01177    if (rtp->nat) {
01178       if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
01179           (rtp->them.sin_port != sin.sin_port)) {
01180          rtp->them = sin;
01181          if (rtp->rtcp) {
01182             memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
01183             rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
01184          }
01185          rtp->rxseqno = 0;
01186          ast_set_flag(rtp, FLAG_NAT_ACTIVE);
01187          if (option_debug || rtpdebug)
01188             ast_log(LOG_DEBUG, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
01189       }
01190    }
01191 
01192    /* If we are bridged to another RTP stream, send direct */
01193    if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
01194       return &ast_null_frame;
01195 
01196    if (version != 2)
01197       return &ast_null_frame;
01198 
01199    payloadtype = (seqno & 0x7f0000) >> 16;
01200    padding = seqno & (1 << 29);
01201    mark = seqno & (1 << 23);
01202    ext = seqno & (1 << 28);
01203    cc = (seqno & 0xF000000) >> 24;
01204    seqno &= 0xffff;
01205    timestamp = ntohl(rtpheader[1]);
01206    ssrc = ntohl(rtpheader[2]);
01207    
01208    if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
01209       if (option_debug || rtpdebug)
01210          ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
01211       mark = 1;
01212    }
01213 
01214    rtp->rxssrc = ssrc;
01215    
01216    if (padding) {
01217       /* Remove padding bytes */
01218       res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
01219    }
01220    
01221    if (cc) {
01222       /* CSRC fields present */
01223       hdrlen += cc*4;
01224    }
01225 
01226    if (ext) {
01227       /* RTP Extension present */
01228       hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
01229       hdrlen += 4;
01230    }
01231 
01232    if (res < hdrlen) {
01233       ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
01234       return &ast_null_frame;
01235    }
01236 
01237    rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
01238 
01239    if (rtp->rxcount==1) {
01240       /* This is the first RTP packet successfully received from source */
01241       rtp->seedrxseqno = seqno;
01242    }
01243 
01244    /* Do not schedule RR if RTCP isn't run */
01245    if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
01246       /* Schedule transmission of Receiver Report */
01247       rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
01248    }
01249    if ( (int)rtp->lastrxseqno - (int)seqno  > 100) /* if so it would indicate that the sender cycled; allow for misordering */
01250       rtp->cycles += RTP_SEQ_MOD;
01251 
01252    rtp->lastrxseqno = seqno;
01253    
01254    if (rtp->themssrc==0)
01255       rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
01256    
01257    if (rtp_debug_test_addr(&sin))
01258       ast_verbose("Got  RTP packet from    %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
01259          ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
01260 
01261    rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
01262    if (!rtpPT.isAstFormat) {
01263       struct ast_frame *f = NULL;
01264 
01265       /* This is special in-band data that's not one of our codecs */
01266       if (rtpPT.code == AST_RTP_DTMF) {
01267          /* It's special -- rfc2833 process it */
01268          if (rtp_debug_test_addr(&sin)) {
01269             unsigned char *data;
01270             unsigned int event;
01271             unsigned int event_end;
01272             unsigned int duration;
01273             data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
01274             event = ntohl(*((unsigned int *)(data)));
01275             event >>= 24;
01276             event_end = ntohl(*((unsigned int *)(data)));
01277             event_end <<= 8;
01278             event_end >>= 24;
01279             duration = ntohl(*((unsigned int *)(data)));
01280             duration &= 0xFFFF;
01281             ast_verbose("Got  RTP RFC2833 from   %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
01282          }
01283          f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
01284       } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
01285          /* It's really special -- process it the Cisco way */
01286          if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
01287             f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01288             rtp->lastevent = seqno;
01289          }
01290       } else if (rtpPT.code == AST_RTP_CN) {
01291          /* Comfort Noise */
01292          f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01293       } else {
01294          ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
01295       }
01296       return f ? f : &ast_null_frame;
01297    }
01298    rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
01299    rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
01300 
01301    if (!rtp->lastrxts)
01302       rtp->lastrxts = timestamp;
01303 
01304    rtp->rxseqno = seqno;
01305 
01306    /* Record received timestamp as last received now */
01307    rtp->lastrxts = timestamp;
01308 
01309    rtp->f.mallocd = 0;
01310    rtp->f.datalen = res - hdrlen;
01311    rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
01312    rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
01313    rtp->f.seqno = seqno;
01314    if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
01315       rtp->f.samples = ast_codec_get_samples(&rtp->f);
01316       if (rtp->f.subclass == AST_FORMAT_SLINEAR) 
01317          ast_frame_byteswap_be(&rtp->f);
01318       calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
01319       /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
01320       ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
01321       rtp->f.ts = timestamp / 8;
01322       rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000);
01323    } else {
01324       /* Video -- samples is # of samples vs. 90000 */
01325       if (!rtp->lastividtimestamp)
01326          rtp->lastividtimestamp = timestamp;
01327       rtp->f.samples = timestamp - rtp->lastividtimestamp;
01328       rtp->lastividtimestamp = timestamp;
01329       rtp->f.delivery.tv_sec = 0;
01330       rtp->f.delivery.tv_usec = 0;
01331       if (mark)
01332          rtp->f.subclass |= 0x1;
01333       
01334    }
01335    rtp->f.src = "RTP";
01336    return &rtp->f;
01337 }

int ast_rtp_reload ( void   ) 

Definition at line 3768 of file rtp.c.

References ast_config_destroy(), ast_config_load(), ast_false(), ast_log(), ast_variable_retrieve(), ast_verbose(), DEFAULT_DTMF_TIMEOUT, LOG_WARNING, option_verbose, RTCP_MAX_INTERVALMS, RTCP_MIN_INTERVALMS, s, and VERBOSE_PREFIX_2.

Referenced by ast_rtp_init().

03769 {
03770    struct ast_config *cfg;
03771    const char *s;
03772 
03773    rtpstart = 5000;
03774    rtpend = 31000;
03775    dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03776    cfg = ast_config_load("rtp.conf");
03777    if (cfg) {
03778       if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
03779          rtpstart = atoi(s);
03780          if (rtpstart < 1024)
03781             rtpstart = 1024;
03782          if (rtpstart > 65535)
03783             rtpstart = 65535;
03784       }
03785       if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
03786          rtpend = atoi(s);
03787          if (rtpend < 1024)
03788             rtpend = 1024;
03789          if (rtpend > 65535)
03790             rtpend = 65535;
03791       }
03792       if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
03793          rtcpinterval = atoi(s);
03794          if (rtcpinterval == 0)
03795             rtcpinterval = 0; /* Just so we're clear... it's zero */
03796          if (rtcpinterval < RTCP_MIN_INTERVALMS)
03797             rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
03798          if (rtcpinterval > RTCP_MAX_INTERVALMS)
03799             rtcpinterval = RTCP_MAX_INTERVALMS;
03800       }
03801       if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
03802 #ifdef SO_NO_CHECK
03803          if (ast_false(s))
03804             nochecksums = 1;
03805          else
03806             nochecksums = 0;
03807 #else
03808          if (ast_false(s))
03809             ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
03810 #endif
03811       }
03812       if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
03813          dtmftimeout = atoi(s);
03814          if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
03815             ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
03816                dtmftimeout, DEFAULT_DTMF_TIMEOUT);
03817             dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03818          };
03819       }
03820       ast_config_destroy(cfg);
03821    }
03822    if (rtpstart >= rtpend) {
03823       ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
03824       rtpstart = 5000;
03825       rtpend = 31000;
03826    }
03827    if (option_verbose > 1)
03828       ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
03829    return 0;
03830 }

void ast_rtp_reset ( struct ast_rtp rtp  ) 

Definition at line 2076 of file rtp.c.

References ast_rtp::dtmfcount, ast_rtp::dtmfmute, ast_rtp::dtmfsamples, ast_rtp::lastdigitts, ast_rtp::lastevent, ast_rtp::lasteventseqn, ast_rtp::lastividtimestamp, ast_rtp::lastovidtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxts, ast_rtp::lastts, ast_rtp::lasttxformat, ast_rtp::rxcore, ast_rtp::rxseqno, ast_rtp::seqno, and ast_rtp::txcore.

02077 {
02078    memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
02079    memset(&rtp->txcore, 0, sizeof(rtp->txcore));
02080    memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
02081    rtp->lastts = 0;
02082    rtp->lastdigitts = 0;
02083    rtp->lastrxts = 0;
02084    rtp->lastividtimestamp = 0;
02085    rtp->lastovidtimestamp = 0;
02086    rtp->lasteventseqn = 0;
02087    rtp->lastevent = 0;
02088    rtp->lasttxformat = 0;
02089    rtp->lastrxformat = 0;
02090    rtp->dtmfcount = 0;
02091    rtp->dtmfsamples = 0;
02092    rtp->seqno = 0;
02093    rtp->rxseqno = 0;
02094 }

int ast_rtp_sendcng ( struct ast_rtp rtp,
int  level 
)

generate comfort noice (CNG)

Definition at line 2591 of file rtp.c.

References ast_inet_ntoa(), ast_log(), AST_RTP_CN, ast_rtp_lookup_code(), ast_tvadd(), ast_verbose(), ast_rtp::data, ast_rtp::dtmfmute, errno, ast_rtp::lastts, LOG_ERROR, rtp_debug_test_addr(), ast_rtp::s, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.

Referenced by do_monitor().

02592 {
02593    unsigned int *rtpheader;
02594    int hdrlen = 12;
02595    int res;
02596    int payload;
02597    char data[256];
02598    level = 127 - (level & 0x7f);
02599    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
02600 
02601    /* If we have no peer, return immediately */ 
02602    if (!rtp->them.sin_addr.s_addr)
02603       return 0;
02604 
02605    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02606 
02607    /* Get a pointer to the header */
02608    rtpheader = (unsigned int *)data;
02609    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
02610    rtpheader[1] = htonl(rtp->lastts);
02611    rtpheader[2] = htonl(rtp->ssrc); 
02612    data[12] = level;
02613    if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
02614       res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
02615       if (res <0) 
02616          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));
02617       if (rtp_debug_test_addr(&rtp->them))
02618          ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
02619                , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);         
02620          
02621    }
02622    return 0;
02623 }

int ast_rtp_senddigit_begin ( struct ast_rtp rtp,
char  digit 
)

Send begin frames for DTMF.

Definition at line 2199 of file rtp.c.

References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_tvadd(), ast_verbose(), ast_rtp::dtmfmute, errno, ast_rtp::lastdigitts, LOG_ERROR, LOG_WARNING, rtp_debug_test_addr(), ast_rtp::s, ast_rtp::send_digit, ast_rtp::send_duration, ast_rtp::send_payload, ast_rtp::sending_digit, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.

Referenced by mgcp_senddigit_begin(), oh323_digit_begin(), and sip_senddigit_begin().

02200 {
02201    unsigned int *rtpheader;
02202    int hdrlen = 12, res = 0, i = 0, payload = 0;
02203    char data[256];
02204 
02205    if ((digit <= '9') && (digit >= '0'))
02206       digit -= '0';
02207    else if (digit == '*')
02208       digit = 10;
02209    else if (digit == '#')
02210       digit = 11;
02211    else if ((digit >= 'A') && (digit <= 'D'))
02212       digit = digit - 'A' + 12;
02213    else if ((digit >= 'a') && (digit <= 'd'))
02214       digit = digit - 'a' + 12;
02215    else {
02216       ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
02217       return 0;
02218    }
02219 
02220    /* If we have no peer, return immediately */ 
02221    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
02222       return 0;
02223 
02224    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
02225 
02226    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02227    rtp->send_duration = 160;
02228    
02229    /* Get a pointer to the header */
02230    rtpheader = (unsigned int *)data;
02231    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
02232    rtpheader[1] = htonl(rtp->lastdigitts);
02233    rtpheader[2] = htonl(rtp->ssrc); 
02234 
02235    for (i = 0; i < 2; i++) {
02236       rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
02237       res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
02238       if (res < 0) 
02239          ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
02240             ast_inet_ntoa(rtp->them.sin_addr),
02241             ntohs(rtp->them.sin_port), strerror(errno));
02242       if (rtp_debug_test_addr(&rtp->them))
02243          ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
02244                 ast_inet_ntoa(rtp->them.sin_addr),
02245                 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
02246       /* Increment sequence number */
02247       rtp->seqno++;
02248       /* Increment duration */
02249       rtp->send_duration += 160;
02250       /* Clear marker bit and set seqno */
02251       rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
02252    }
02253 
02254    /* Since we received a begin, we can safely store the digit and disable any compensation */
02255    rtp->sending_digit = 1;
02256    rtp->send_digit = digit;
02257    rtp->send_payload = payload;
02258 
02259    return 0;
02260 }

int ast_rtp_senddigit_end ( struct ast_rtp rtp,
char  digit 
)

void ast_rtp_set_callback ( struct ast_rtp rtp,
ast_rtp_callback  callback 
)

Definition at line 586 of file rtp.c.

References ast_rtp::callback.

Referenced by start_rtp().

00587 {
00588    rtp->callback = callback;
00589 }

void ast_rtp_set_data ( struct ast_rtp rtp,
void *  data 
)

Definition at line 581 of file rtp.c.

References ast_rtp::data.

Referenced by start_rtp().

00582 {
00583    rtp->data = data;
00584 }

void ast_rtp_set_m_type ( struct ast_rtp rtp,
int  pt 
)

Activate payload type.

Definition at line 1647 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, ast_rtp::current_RTP_PT, MAX_RTP_PT, and static_RTP_PT.

Referenced by gtalk_is_answered(), gtalk_newcall(), and process_sdp().

01648 {
01649    if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 
01650       return; /* bogus payload type */
01651 
01652    ast_mutex_lock(&rtp->bridge_lock);
01653    rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
01654    ast_mutex_unlock(&rtp->bridge_lock);
01655 } 

void ast_rtp_set_peer ( struct ast_rtp rtp,
struct sockaddr_in *  them 
)

Definition at line 2020 of file rtp.c.

References ast_rtp::rtcp, ast_rtp::rxseqno, ast_rtcp::them, and ast_rtp::them.

Referenced by handle_open_receive_channel_ack_message(), process_sdp(), and setup_rtp_connection().

02021 {
02022    rtp->them.sin_port = them->sin_port;
02023    rtp->them.sin_addr = them->sin_addr;
02024    if (rtp->rtcp) {
02025       rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
02026       rtp->rtcp->them.sin_addr = them->sin_addr;
02027    }
02028    rtp->rxseqno = 0;
02029 }

void ast_rtp_set_rtpholdtimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp hold timeout.

Definition at line 548 of file rtp.c.

References ast_rtp::rtpholdtimeout.

Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().

00549 {
00550    rtp->rtpholdtimeout = timeout;
00551 }

void ast_rtp_set_rtpkeepalive ( struct ast_rtp rtp,
int  period 
)

set RTP keepalive interval

Definition at line 554 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by create_addr_from_peer(), and sip_alloc().

00555 {
00556    rtp->rtpkeepalive = period;
00557 }

int ast_rtp_set_rtpmap_type ( struct ast_rtp rtp,
int  pt,
char *  mimeType,
char *  mimeSubtype,
enum ast_rtp_options  options 
)

Initiate payload type to a known MIME media type for a codec.

Returns:
0 if the MIME type was found and set, -1 if it wasn't found

Definition at line 1674 of file rtp.c.

References AST_FORMAT_G726, AST_FORMAT_G726_AAL2, ast_mutex_lock(), ast_mutex_unlock(), AST_RTP_OPT_G726_NONSTANDARD, ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, MAX_RTP_PT, mimeTypes, payloadType, subtype, and type.

Referenced by __oh323_rtp_create(), gtalk_is_answered(), gtalk_newcall(), process_sdp(), and set_dtmf_payload().

01677 {
01678    unsigned int i;
01679    int found = 0;
01680 
01681    if (pt < 0 || pt > MAX_RTP_PT) 
01682       return -1; /* bogus payload type */
01683    
01684    ast_mutex_lock(&rtp->bridge_lock);
01685 
01686    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01687       if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
01688           strcasecmp(mimeType, mimeTypes[i].type) == 0) {
01689          found = 1;
01690          rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
01691          if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
01692              mimeTypes[i].payloadType.isAstFormat &&
01693              (options & AST_RTP_OPT_G726_NONSTANDARD))
01694             rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
01695          break;
01696       }
01697    }
01698 
01699    ast_mutex_unlock(&rtp->bridge_lock);
01700 
01701    return (found ? 0 : -1);
01702 } 

void ast_rtp_set_rtptimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp timeout.

Definition at line 542 of file rtp.c.

References ast_rtp::rtptimeout.

Referenced by create_addr_from_peer(), do_monitor(), and sip_alloc().

00543 {
00544    rtp->rtptimeout = timeout;
00545 }

void ast_rtp_set_rtptimers_onhold ( struct ast_rtp rtp  ) 

Definition at line 535 of file rtp.c.

References ast_rtp::rtpholdtimeout, and ast_rtp::rtptimeout.

Referenced by handle_response_invite().

00536 {
00537    rtp->rtptimeout = (-1) * rtp->rtptimeout;
00538    rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
00539 }

void ast_rtp_setdtmf ( struct ast_rtp rtp,
int  dtmf 
)

Indicate whether this RTP session is carrying DTMF or not.

Definition at line 601 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_DTMF.

Referenced by create_addr_from_peer(), handle_request_invite(), process_sdp(), sip_alloc(), and sip_dtmfmode().

00602 {
00603    ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
00604 }

void ast_rtp_setdtmfcompensate ( struct ast_rtp rtp,
int  compensate 
)

Compensate for devices that send RFC2833 packets all at once.

Definition at line 606 of file rtp.c.

References ast_set2_flag, and FLAG_DTMF_COMPENSATE.

Referenced by create_addr_from_peer(), handle_request_invite(), process_sdp(), and sip_alloc().

00607 {
00608    ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
00609 }

void ast_rtp_setnat ( struct ast_rtp rtp,
int  nat 
)

Definition at line 591 of file rtp.c.

References ast_rtp::nat.

Referenced by __oh323_rtp_create(), do_setnat(), oh323_rtp_read(), and start_rtp().

00592 {
00593    rtp->nat = nat;
00594 }

void ast_rtp_setstun ( struct ast_rtp rtp,
int  stun_enable 
)

Enable STUN capability.

Definition at line 611 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_STUN.

Referenced by gtalk_new().

00612 {
00613    ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
00614 }

int ast_rtp_settos ( struct ast_rtp rtp,
int  tos 
)

Definition at line 2003 of file rtp.c.

References ast_log(), LOG_WARNING, and ast_rtp::s.

Referenced by __oh323_rtp_create(), and sip_alloc().

02004 {
02005    int res;
02006 
02007    if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
02008       ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
02009    return res;
02010 }

void ast_rtp_stop ( struct ast_rtp rtp  ) 

Definition at line 2060 of file rtp.c.

References ast_clear_flag, AST_SCHED_DEL, FLAG_P2P_SENT_MARK, ast_rtp::rtcp, ast_rtp::sched, ast_rtcp::schedid, ast_rtcp::them, and ast_rtp::them.

Referenced by process_sdp(), setup_rtp_connection(), and stop_media_flows().

02061 {
02062    if (rtp->rtcp) {
02063       AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
02064    }
02065 
02066    memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
02067    memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
02068    if (rtp->rtcp) {
02069       memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
02070       memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
02071    }
02072    
02073    ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
02074 }

void ast_rtp_stun_request ( struct ast_rtp rtp,
struct sockaddr_in *  suggestion,
const char *  username 
)

Definition at line 403 of file rtp.c.

References append_attr_string(), stun_attr::attr, ast_rtp::s, STUN_BINDREQ, stun_req_id(), stun_send(), and STUN_USERNAME.

Referenced by gtalk_update_stun().

00404 {
00405    struct stun_header *req;
00406    unsigned char reqdata[1024];
00407    int reqlen, reqleft;
00408    struct stun_attr *attr;
00409 
00410    req = (struct stun_header *)reqdata;
00411    stun_req_id(req);
00412    reqlen = 0;
00413    reqleft = sizeof(reqdata) - sizeof(struct stun_header);
00414    req->msgtype = 0;
00415    req->msglen = 0;
00416    attr = (struct stun_attr *)req->ies;
00417    if (username)
00418       append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
00419    req->msglen = htons(reqlen);
00420    req->msgtype = htons(STUN_BINDREQ);
00421    stun_send(rtp->s, suggestion, req);
00422 }

void ast_rtp_unset_m_type ( struct ast_rtp rtp,
int  pt 
)

clear payload type

Definition at line 1659 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_rtp::bridge_lock, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, and MAX_RTP_PT.

Referenced by process_sdp().

01660 {
01661    if (pt < 0 || pt > MAX_RTP_PT)
01662       return; /* bogus payload type */
01663 
01664    ast_mutex_lock(&rtp->bridge_lock);
01665    rtp->current_RTP_PT[pt].isAstFormat = 0;
01666    rtp->current_RTP_PT[pt].code = 0;
01667    ast_mutex_unlock(&rtp->bridge_lock);
01668 }

int ast_rtp_write ( struct ast_rtp rtp,
struct ast_frame f 
)

Definition at line 2750 of file rtp.c.

References ast_codec_pref_getsize(), AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_SPEEX, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frdup(), ast_frfree, ast_getformatname(), ast_log(), ast_rtp_lookup_code(), ast_rtp_raw_write(), ast_smoother_feed, ast_smoother_feed_be, AST_SMOOTHER_FLAG_BE, ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_smoother_set_flags(), ast_smoother_test_flag(), ast_frame::datalen, f, fmt, ast_frame::frametype, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, option_debug, ast_rtp::pref, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.

Referenced by gtalk_write(), mgcp_write(), oh323_write(), sip_write(), and skinny_write().

02751 {
02752    struct ast_frame *f;
02753    int codec;
02754    int hdrlen = 12;
02755    int subclass;
02756    
02757 
02758    /* If we have no peer, return immediately */ 
02759    if (!rtp->them.sin_addr.s_addr)
02760       return 0;
02761 
02762    /* If there is no data length, return immediately */
02763    if (!_f->datalen) 
02764       return 0;
02765    
02766    /* Make sure we have enough space for RTP header */
02767    if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
02768       ast_log(LOG_WARNING, "RTP can only send voice and video\n");
02769       return -1;
02770    }
02771 
02772    subclass = _f->subclass;
02773    if (_f->frametype == AST_FRAME_VIDEO)
02774       subclass &= ~0x1;
02775 
02776    codec = ast_rtp_lookup_code(rtp, 1, subclass);
02777    if (codec < 0) {
02778       ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
02779       return -1;
02780    }
02781 
02782    if (rtp->lasttxformat != subclass) {
02783       /* New format, reset the smoother */
02784       if (option_debug)
02785          ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
02786       rtp->lasttxformat = subclass;
02787       if (rtp->smoother)
02788          ast_smoother_free(rtp->smoother);
02789       rtp->smoother = NULL;
02790    }
02791 
02792    if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) {
02793       struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
02794       if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
02795          if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
02796             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));
02797             return -1;
02798          }
02799          if (fmt.flags)
02800             ast_smoother_set_flags(rtp->smoother, fmt.flags);
02801          if (option_debug)
02802             ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
02803       }
02804    }
02805    if (rtp->smoother) {
02806       if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
02807          ast_smoother_feed_be(rtp->smoother, _f);
02808       } else {
02809          ast_smoother_feed(rtp->smoother, _f);
02810       }
02811 
02812       while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) {
02813          if (f->subclass == AST_FORMAT_G722) {
02814             /* G.722 is silllllllllllllly */
02815             f->samples /= 2;
02816          }
02817 
02818          ast_rtp_raw_write(rtp, f, codec);
02819       }
02820    } else {
02821       /* Don't buffer outgoing frames; send them one-per-packet: */
02822       if (_f->offset < hdrlen) {
02823          f = ast_frdup(_f);
02824       } else {
02825          f = _f;
02826       }
02827       if (f->data) {
02828          if (f->subclass == AST_FORMAT_G722) {
02829             /* G.722 is silllllllllllllly */
02830             f->samples /= 2;
02831          }
02832          ast_rtp_raw_write(rtp, f, codec);
02833       }
02834       if (f != _f)
02835          ast_frfree(f);
02836    }
02837       
02838    return 0;
02839 }


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