Fri Apr 24 16:26:42 2009

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 517 of file rtp.c.

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

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

00518 {
00519    if (rtp->rtcp)
00520       return rtp->rtcp->s;
00521    return -1;
00522 }

struct ast_frame* ast_rtcp_read ( struct ast_rtp rtp  ) 

Definition at line 825 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().

00826 {
00827    socklen_t len;
00828    int position, i, packetwords;
00829    int res;
00830    struct sockaddr_in sin;
00831    unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
00832    unsigned int *rtcpheader;
00833    int pt;
00834    struct timeval now;
00835    unsigned int length;
00836    int rc;
00837    double rttsec;
00838    uint64_t rtt = 0;
00839    unsigned int dlsr;
00840    unsigned int lsr;
00841    unsigned int msw;
00842    unsigned int lsw;
00843    unsigned int comp;
00844    struct ast_frame *f = &ast_null_frame;
00845    
00846    if (!rtp || !rtp->rtcp)
00847       return &ast_null_frame;
00848 
00849    len = sizeof(sin);
00850    
00851    res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
00852                0, (struct sockaddr *)&sin, &len);
00853    if (option_debug > 2)
00854       ast_log(LOG_DEBUG, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
00855 
00856    rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
00857    
00858    if (res < 0) {
00859       ast_assert(errno != EBADF);
00860       if (errno != EAGAIN) {
00861          ast_log(LOG_WARNING, "RTCP Read error: %s.  Hanging up.\n", strerror(errno));
00862          ast_log(LOG_WARNING, "socket RTCP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
00863          return NULL;
00864       }
00865       return &ast_null_frame;
00866    }
00867 
00868    packetwords = res / 4;
00869    
00870    if (rtp->nat) {
00871       /* Send to whoever sent to us */
00872       if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
00873           (rtp->rtcp->them.sin_port != sin.sin_port)) {
00874          memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
00875          if (option_debug || rtpdebug)
00876             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));
00877       }
00878    }
00879 
00880    if (option_debug)
00881       ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
00882 
00883    /* Process a compound packet */
00884    position = 0;
00885    while (position < packetwords) {
00886       i = position;
00887       length = ntohl(rtcpheader[i]);
00888       pt = (length & 0xff0000) >> 16;
00889       rc = (length & 0x1f000000) >> 24;
00890       length &= 0xffff;
00891     
00892       if ((i + length) > packetwords) {
00893          ast_log(LOG_WARNING, "RTCP Read too short\n");
00894          return &ast_null_frame;
00895       }
00896       
00897       if (rtcp_debug_test_addr(&sin)) {
00898          ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
00899          ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
00900          ast_verbose("Reception reports: %d\n", rc);
00901          ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
00902       }
00903     
00904       i += 2; /* Advance past header and ssrc */
00905       
00906       switch (pt) {
00907       case RTCP_PT_SR:
00908          gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
00909          rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
00910          rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
00911          rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
00912     
00913          if (rtcp_debug_test_addr(&sin)) {
00914             ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
00915             ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
00916             ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
00917          }
00918          i += 5;
00919          if (rc < 1)
00920             break;
00921          /* Intentional fall through */
00922       case RTCP_PT_RR:
00923          /* Don't handle multiple reception reports (rc > 1) yet */
00924          /* Calculate RTT per RFC */
00925          gettimeofday(&now, NULL);
00926          timeval2ntp(now, &msw, &lsw);
00927          if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */
00928             comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
00929             lsr = ntohl(rtcpheader[i + 4]);
00930             dlsr = ntohl(rtcpheader[i + 5]);
00931             rtt = comp - lsr - dlsr;
00932 
00933             /* Convert end to end delay to usec (keeping the calculation in 64bit space)
00934                sess->ee_delay = (eedelay * 1000) / 65536; */
00935             if (rtt < 4294) {
00936                 rtt = (rtt * 1000000) >> 16;
00937             } else {
00938                 rtt = (rtt * 1000) >> 16;
00939                 rtt *= 1000;
00940             }
00941             rtt = rtt / 1000.;
00942             rttsec = rtt / 1000.;
00943 
00944             if (comp - dlsr >= lsr) {
00945                rtp->rtcp->accumulated_transit += rttsec;
00946                rtp->rtcp->rtt = rttsec;
00947                if (rtp->rtcp->maxrtt<rttsec)
00948                   rtp->rtcp->maxrtt = rttsec;
00949                if (rtp->rtcp->minrtt>rttsec)
00950                   rtp->rtcp->minrtt = rttsec;
00951             } else if (rtcp_debug_test_addr(&sin)) {
00952                ast_verbose("Internal RTCP NTP clock skew detected: "
00953                         "lsr=%u, now=%u, dlsr=%u (%d:%03dms), "
00954                         "diff=%d\n",
00955                         lsr, comp, dlsr, dlsr / 65536,
00956                         (dlsr % 65536) * 1000 / 65536,
00957                         dlsr - (comp - lsr));
00958             }
00959          }
00960 
00961          rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
00962          rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
00963          if (rtcp_debug_test_addr(&sin)) {
00964             ast_verbose("  Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
00965             ast_verbose("  Packets lost so far: %d\n", rtp->rtcp->reported_lost);
00966             ast_verbose("  Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
00967             ast_verbose("  Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
00968             ast_verbose("  Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
00969             ast_verbose("  Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
00970             ast_verbose("  DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
00971             if (rtt)
00972                ast_verbose("  RTT: %lu(sec)\n", (unsigned long) rtt);
00973          }
00974          break;
00975       case RTCP_PT_FUR:
00976          if (rtcp_debug_test_addr(&sin))
00977             ast_verbose("Received an RTCP Fast Update Request\n");
00978          rtp->f.frametype = AST_FRAME_CONTROL;
00979          rtp->f.subclass = AST_CONTROL_VIDUPDATE;
00980          rtp->f.datalen = 0;
00981          rtp->f.samples = 0;
00982          rtp->f.mallocd = 0;
00983          rtp->f.src = "RTP";
00984          f = &rtp->f;
00985          break;
00986       case RTCP_PT_SDES:
00987          if (rtcp_debug_test_addr(&sin))
00988             ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00989          break;
00990       case RTCP_PT_BYE:
00991          if (rtcp_debug_test_addr(&sin))
00992             ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00993          break;
00994       default:
00995          if (option_debug)
00996             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));
00997          break;
00998       }
00999       position += (length + 1);
01000    }
01001          
01002    return f;
01003 }

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 2369 of file rtp.c.

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

02370 {
02371    struct ast_rtp *rtp = data;
02372    int res;
02373 
02374    rtp->rtcp->sendfur = 1;
02375    res = ast_rtcp_write(data);
02376    
02377    return res;
02378 }

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 397 of file rtp.c.

Referenced by process_sdp().

00398 {
00399    return sizeof(struct ast_rtp);
00400 }

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

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

Definition at line 3347 of file rtp.c.

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

03348 {
03349    struct ast_rtp *p0 = NULL, *p1 = NULL;    /* Audio RTP Channels */
03350    struct ast_rtp *vp0 = NULL, *vp1 = NULL;  /* Video RTP channels */
03351    struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
03352    enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
03353    enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
03354    enum ast_bridge_result res = AST_BRIDGE_FAILED;
03355    int codec0 = 0, codec1 = 0;
03356    void *pvt0 = NULL, *pvt1 = NULL;
03357 
03358    /* Lock channels */
03359    ast_channel_lock(c0);
03360    while(ast_channel_trylock(c1)) {
03361       ast_channel_unlock(c0);
03362       usleep(1);
03363       ast_channel_lock(c0);
03364    }
03365 
03366    /* Ensure neither channel got hungup during lock avoidance */
03367    if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
03368       ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
03369       ast_channel_unlock(c0);
03370       ast_channel_unlock(c1);
03371       return AST_BRIDGE_FAILED;
03372    }
03373       
03374    /* Find channel driver interfaces */
03375    if (!(pr0 = get_proto(c0))) {
03376       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
03377       ast_channel_unlock(c0);
03378       ast_channel_unlock(c1);
03379       return AST_BRIDGE_FAILED;
03380    }
03381    if (!(pr1 = get_proto(c1))) {
03382       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
03383       ast_channel_unlock(c0);
03384       ast_channel_unlock(c1);
03385       return AST_BRIDGE_FAILED;
03386    }
03387 
03388    /* Get channel specific interface structures */
03389    pvt0 = c0->tech_pvt;
03390    pvt1 = c1->tech_pvt;
03391 
03392    /* Get audio and video interface (if native bridge is possible) */
03393    audio_p0_res = pr0->get_rtp_info(c0, &p0);
03394    video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
03395    audio_p1_res = pr1->get_rtp_info(c1, &p1);
03396    video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
03397 
03398    /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
03399    if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
03400       audio_p0_res = AST_RTP_GET_FAILED;
03401    if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
03402       audio_p1_res = AST_RTP_GET_FAILED;
03403 
03404    /* Check if a bridge is possible (partial/native) */
03405    if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
03406       /* Somebody doesn't want to play... */
03407       ast_channel_unlock(c0);
03408       ast_channel_unlock(c1);
03409       return AST_BRIDGE_FAILED_NOWARN;
03410    }
03411 
03412    /* If we need to feed DTMF frames into the core then only do a partial native bridge */
03413    if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
03414       ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
03415       audio_p0_res = AST_RTP_TRY_PARTIAL;
03416    }
03417 
03418    if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
03419       ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
03420       audio_p1_res = AST_RTP_TRY_PARTIAL;
03421    }
03422 
03423    /* If both sides are not using the same method of DTMF transmission 
03424     * (ie: one is RFC2833, other is INFO... then we can not do direct media. 
03425     * --------------------------------------------------
03426     * | DTMF Mode |  HAS_DTMF  |  Accepts Begin Frames |
03427     * |-----------|------------|-----------------------|
03428     * | Inband    | False      | True                  |
03429     * | RFC2833   | True       | True                  |
03430     * | SIP INFO  | False      | False                 |
03431     * --------------------------------------------------
03432     * However, if DTMF from both channels is being monitored by the core, then
03433     * we can still do packet-to-packet bridging, because passing through the 
03434     * core will handle DTMF mode translation.
03435     */
03436    if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
03437        (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
03438       if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
03439          ast_channel_unlock(c0);
03440          ast_channel_unlock(c1);
03441          return AST_BRIDGE_FAILED_NOWARN;
03442       }
03443       audio_p0_res = AST_RTP_TRY_PARTIAL;
03444       audio_p1_res = AST_RTP_TRY_PARTIAL;
03445    }
03446 
03447    /* If we need to feed frames into the core don't do a P2P bridge */
03448    if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) ||
03449        (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) {
03450       ast_channel_unlock(c0);
03451       ast_channel_unlock(c1);
03452       return AST_BRIDGE_FAILED_NOWARN;
03453    }
03454 
03455    /* Get codecs from both sides */
03456    codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
03457    codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
03458    if (codec0 && codec1 && !(codec0 & codec1)) {
03459       /* Hey, we can't do native bridging if both parties speak different codecs */
03460       if (option_debug)
03461          ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
03462       ast_channel_unlock(c0);
03463       ast_channel_unlock(c1);
03464       return AST_BRIDGE_FAILED_NOWARN;
03465    }
03466 
03467    /* If either side can only do a partial bridge, then don't try for a true native bridge */
03468    if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
03469       struct ast_format_list fmt0, fmt1;
03470 
03471       /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
03472       if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
03473          if (option_debug)
03474             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n");
03475          ast_channel_unlock(c0);
03476          ast_channel_unlock(c1);
03477          return AST_BRIDGE_FAILED_NOWARN;
03478       }
03479       /* They must also be using the same packetization */
03480       fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
03481       fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
03482       if (fmt0.cur_ms != fmt1.cur_ms) {
03483          if (option_debug)
03484             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n");
03485          ast_channel_unlock(c0);
03486          ast_channel_unlock(c1);
03487          return AST_BRIDGE_FAILED_NOWARN;
03488       }
03489 
03490       if (option_verbose > 2)
03491          ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
03492       res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
03493    } else {
03494       if (option_verbose > 2) 
03495          ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
03496       res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
03497    }
03498 
03499    return res;
03500 }

int ast_rtp_codec_getformat ( int  pt  ) 

Definition at line 2791 of file rtp.c.

References rtpPayloadType::code, and static_RTP_PT.

Referenced by process_sdp().

02792 {
02793    if (pt < 0 || pt > MAX_RTP_PT)
02794       return 0; /* bogus payload type */
02795 
02796    if (static_RTP_PT[pt].isAstFormat)
02797       return static_RTP_PT[pt].code;
02798    else
02799       return 0;
02800 }

struct ast_codec_pref* ast_rtp_codec_getpref ( struct ast_rtp rtp  ) 

Definition at line 2786 of file rtp.c.

References ast_rtp::pref.

Referenced by add_codec_to_sdp(), and process_sdp().

02787 {
02788    return &rtp->pref;
02789 }

int ast_rtp_codec_setpref ( struct ast_rtp rtp,
struct ast_codec_pref prefs 
)

Definition at line 2739 of file rtp.c.

References ast_codec_pref_getsize(), ast_log(), ast_smoother_new(), ast_smoother_reconfigure(), ast_smoother_set_flags(), ast_format_list::cur_ms, ast_format_list::flags, ast_format_list::fr_len, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, option_debug, ast_rtp::pref, prefs, and ast_rtp::smoother.

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

02740 {
02741    struct ast_format_list current_format_old, current_format_new;
02742 
02743    /* if no packets have been sent through this session yet, then
02744     *  changing preferences does not require any extra work
02745     */
02746    if (rtp->lasttxformat == 0) {
02747       rtp->pref = *prefs;
02748       return 0;
02749    }
02750 
02751    current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
02752 
02753    rtp->pref = *prefs;
02754 
02755    current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
02756 
02757    /* if the framing desired for the current format has changed, we may have to create
02758     * or adjust the smoother for this session
02759     */
02760    if ((current_format_new.inc_ms != 0) &&
02761        (current_format_new.cur_ms != current_format_old.cur_ms)) {
02762       int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms;
02763 
02764       if (rtp->smoother) {
02765          ast_smoother_reconfigure(rtp->smoother, new_size);
02766          if (option_debug) {
02767             ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size);
02768          }
02769       } else {
02770          if (!(rtp->smoother = ast_smoother_new(new_size))) {
02771             ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
02772             return -1;
02773          }
02774          if (current_format_new.flags) {
02775             ast_smoother_set_flags(rtp->smoother, current_format_new.flags);
02776          }
02777          if (option_debug) {
02778             ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
02779          }
02780       }
02781    }
02782 
02783    return 0;
02784 }

void ast_rtp_destroy ( struct ast_rtp rtp  ) 

Definition at line 2152 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().

02153 {
02154    if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
02155       /*Print some info on the call here */
02156       ast_verbose("  RTP-stats\n");
02157       ast_verbose("* Our Receiver:\n");
02158       ast_verbose("  SSRC:     %u\n", rtp->themssrc);
02159       ast_verbose("  Received packets: %u\n", rtp->rxcount);
02160       ast_verbose("  Lost packets:   %u\n", rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0);
02161       ast_verbose("  Jitter:      %.4f\n", rtp->rxjitter);
02162       ast_verbose("  Transit:     %.4f\n", rtp->rxtransit);
02163       ast_verbose("  RR-count:    %u\n", rtp->rtcp ? rtp->rtcp->rr_count : 0);
02164       ast_verbose("* Our Sender:\n");
02165       ast_verbose("  SSRC:     %u\n", rtp->ssrc);
02166       ast_verbose("  Sent packets:   %u\n", rtp->txcount);
02167       ast_verbose("  Lost packets:   %u\n", rtp->rtcp ? rtp->rtcp->reported_lost : 0);
02168       ast_verbose("  Jitter:      %u\n", rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int)65536.0) : 0);
02169       ast_verbose("  SR-count:    %u\n", rtp->rtcp ? rtp->rtcp->sr_count : 0);
02170       ast_verbose("  RTT:      %f\n", rtp->rtcp ? rtp->rtcp->rtt : 0);
02171    }
02172 
02173    if (rtp->smoother)
02174       ast_smoother_free(rtp->smoother);
02175    if (rtp->ioid)
02176       ast_io_remove(rtp->io, rtp->ioid);
02177    if (rtp->s > -1)
02178       close(rtp->s);
02179    if (rtp->rtcp) {
02180       AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
02181       close(rtp->rtcp->s);
02182       free(rtp->rtcp);
02183       rtp->rtcp=NULL;
02184    }
02185 
02186    ast_mutex_destroy(&rtp->bridge_lock);
02187 
02188    free(rtp);
02189 }

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 1494 of file rtp.c.

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

Referenced by wait_for_answer().

01495 {
01496    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01497    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01498    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01499    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01500    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
01501    int srccodec, destcodec, nat_active = 0;
01502 
01503    /* Lock channels */
01504    ast_channel_lock(dest);
01505    if (src) {
01506       while(ast_channel_trylock(src)) {
01507          ast_channel_unlock(dest);
01508          usleep(1);
01509          ast_channel_lock(dest);
01510       }
01511    }
01512 
01513    /* Find channel driver interfaces */
01514    destpr = get_proto(dest);
01515    if (src)
01516       srcpr = get_proto(src);
01517    if (!destpr) {
01518       if (option_debug)
01519          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01520       ast_channel_unlock(dest);
01521       if (src)
01522          ast_channel_unlock(src);
01523       return 0;
01524    }
01525    if (!srcpr) {
01526       if (option_debug)
01527          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
01528       ast_channel_unlock(dest);
01529       if (src)
01530          ast_channel_unlock(src);
01531       return 0;
01532    }
01533 
01534    /* Get audio and video interface (if native bridge is possible) */
01535    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01536    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01537    if (srcpr) {
01538       audio_src_res = srcpr->get_rtp_info(src, &srcp);
01539       video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01540    }
01541 
01542    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01543    if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {
01544       /* Somebody doesn't want to play... */
01545       ast_channel_unlock(dest);
01546       if (src)
01547          ast_channel_unlock(src);
01548       return 0;
01549    }
01550    if (audio_src_res == AST_RTP_TRY_NATIVE && (video_src_res == AST_RTP_GET_FAILED || video_src_res == AST_RTP_TRY_NATIVE) && srcpr->get_codec)
01551       srccodec = srcpr->get_codec(src);
01552    else
01553       srccodec = 0;
01554    if (audio_dest_res == AST_RTP_TRY_NATIVE && (video_dest_res == AST_RTP_GET_FAILED || video_dest_res == AST_RTP_TRY_NATIVE) && destpr->get_codec)
01555       destcodec = destpr->get_codec(dest);
01556    else
01557       destcodec = 0;
01558    /* Ensure we have at least one matching codec */
01559    if (srcp && !(srccodec & destcodec)) {
01560       ast_channel_unlock(dest);
01561       ast_channel_unlock(src);
01562       return 0;
01563    }
01564    /* Consider empty media as non-existant */
01565    if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
01566       srcp = NULL;
01567    /* If the client has NAT stuff turned on then just safe NAT is active */
01568    if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01569       nat_active = 1;
01570    /* Bridge media early */
01571    if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active))
01572       ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>");
01573    ast_channel_unlock(dest);
01574    if (src)
01575       ast_channel_unlock(src);
01576    if (option_debug)
01577       ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
01578    return 1;
01579 }

int ast_rtp_fd ( struct ast_rtp rtp  ) 

Definition at line 512 of file rtp.c.

References ast_rtp::s.

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

00513 {
00514    return rtp->s;
00515 }

struct ast_rtp* ast_rtp_get_bridged ( struct ast_rtp rtp  ) 

Definition at line 2062 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().

02063 {
02064    struct ast_rtp *bridged = NULL;
02065 
02066    ast_mutex_lock(&rtp->bridge_lock);
02067    bridged = rtp->bridged;
02068    ast_mutex_unlock(&rtp->bridge_lock);
02069 
02070    return bridged;
02071 }

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 1715 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().

01717 {
01718    int pt;
01719    
01720    ast_mutex_lock(&rtp->bridge_lock);
01721    
01722    *astFormats = *nonAstFormats = 0;
01723    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01724       if (rtp->current_RTP_PT[pt].isAstFormat) {
01725          *astFormats |= rtp->current_RTP_PT[pt].code;
01726       } else {
01727          *nonAstFormats |= rtp->current_RTP_PT[pt].code;
01728       }
01729    }
01730    
01731    ast_mutex_unlock(&rtp->bridge_lock);
01732    
01733    return;
01734 }

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

Definition at line 2044 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().

02045 {
02046    if ((them->sin_family != AF_INET) ||
02047       (them->sin_port != rtp->them.sin_port) ||
02048       (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
02049       them->sin_family = AF_INET;
02050       them->sin_port = rtp->them.sin_port;
02051       them->sin_addr = rtp->them.sin_addr;
02052       return 1;
02053    }
02054    return 0;
02055 }

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

Return RTCP quality string.

Definition at line 2108 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().

02109 {
02110    /*
02111    *ssrc          our ssrc
02112    *themssrc      their ssrc
02113    *lp            lost packets
02114    *rxjitter      our calculated jitter(rx)
02115    *rxcount       no. received packets
02116    *txjitter      reported jitter of the other end
02117    *txcount       transmitted packets
02118    *rlp           remote lost packets
02119    *rtt           round trip time
02120    */
02121 
02122    if (qual && rtp) {
02123       qual->local_ssrc = rtp->ssrc;
02124       qual->local_jitter = rtp->rxjitter;
02125       qual->local_count = rtp->rxcount;
02126       qual->remote_ssrc = rtp->themssrc;
02127       qual->remote_count = rtp->txcount;
02128       if (rtp->rtcp) {
02129          qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
02130          qual->remote_lostpackets = rtp->rtcp->reported_lost;
02131          qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
02132          qual->rtt = rtp->rtcp->rtt;
02133       }
02134    }
02135    if (rtp->rtcp) {
02136       snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality),
02137          "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
02138          rtp->ssrc,
02139          rtp->themssrc,
02140          rtp->rtcp->expected_prior - rtp->rtcp->received_prior,
02141          rtp->rxjitter,
02142          rtp->rxcount,
02143          (double)rtp->rtcp->reported_jitter / 65536.0,
02144          rtp->txcount,
02145          rtp->rtcp->reported_lost,
02146          rtp->rtcp->rtt);
02147       return rtp->rtcp->quality;
02148    } else
02149       return "<Unknown> - RTP/RTCP has already been destroyed";
02150 }

int ast_rtp_get_rtpholdtimeout ( struct ast_rtp rtp  ) 

Get rtp hold timeout.

Definition at line 567 of file rtp.c.

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

Referenced by do_monitor().

00568 {
00569    if (rtp->rtptimeout < 0)   /* We're not checking, but remembering the setting (during T.38 transmission) */
00570       return 0;
00571    return rtp->rtpholdtimeout;
00572 }

int ast_rtp_get_rtpkeepalive ( struct ast_rtp rtp  ) 

Get RTP keepalive interval.

Definition at line 575 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by do_monitor().

00576 {
00577    return rtp->rtpkeepalive;
00578 }

int ast_rtp_get_rtptimeout ( struct ast_rtp rtp  ) 

Get rtp timeout.

Definition at line 559 of file rtp.c.

References ast_rtp::rtptimeout.

Referenced by do_monitor().

00560 {
00561    if (rtp->rtptimeout < 0)   /* We're not checking, but remembering the setting (during T.38 transmission) */
00562       return 0;
00563    return rtp->rtptimeout;
00564 }

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

Definition at line 2057 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().

02058 {
02059    *us = rtp->us;
02060 }

int ast_rtp_getnat ( struct ast_rtp rtp  ) 

Definition at line 595 of file rtp.c.

References ast_test_flag, and FLAG_NAT_ACTIVE.

Referenced by sip_get_rtp_peer().

00596 {
00597    return ast_test_flag(rtp, FLAG_NAT_ACTIVE);
00598 }

void ast_rtp_init ( void   ) 

Initialize the RTP system in Asterisk.

Definition at line 3885 of file rtp.c.

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

Referenced by main().

03886 {
03887    ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
03888    ast_rtp_reload();
03889 }

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 1758 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().

01759 {
01760    int pt = 0;
01761 
01762    ast_mutex_lock(&rtp->bridge_lock);
01763 
01764    if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
01765       code == rtp->rtp_lookup_code_cache_code) {
01766       /* Use our cached mapping, to avoid the overhead of the loop below */
01767       pt = rtp->rtp_lookup_code_cache_result;
01768       ast_mutex_unlock(&rtp->bridge_lock);
01769       return pt;
01770    }
01771 
01772    /* Check the dynamic list first */
01773    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01774       if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
01775          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01776          rtp->rtp_lookup_code_cache_code = code;
01777          rtp->rtp_lookup_code_cache_result = pt;
01778          ast_mutex_unlock(&rtp->bridge_lock);
01779          return pt;
01780       }
01781    }
01782 
01783    /* Then the static list */
01784    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01785       if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
01786          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01787          rtp->rtp_lookup_code_cache_code = code;
01788          rtp->rtp_lookup_code_cache_result = pt;
01789          ast_mutex_unlock(&rtp->bridge_lock);
01790          return pt;
01791       }
01792    }
01793 
01794    ast_mutex_unlock(&rtp->bridge_lock);
01795 
01796    return -1;
01797 }

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 1818 of file rtp.c.

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

Referenced by process_sdp().

01820 {
01821    int format;
01822    unsigned len;
01823    char *end = buf;
01824    char *start = buf;
01825 
01826    if (!buf || !size)
01827       return NULL;
01828 
01829    snprintf(end, size, "0x%x (", capability);
01830 
01831    len = strlen(end);
01832    end += len;
01833    size -= len;
01834    start = end;
01835 
01836    for (format = 1; format < AST_RTP_MAX; format <<= 1) {
01837       if (capability & format) {
01838          const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
01839 
01840          snprintf(end, size, "%s|", name);
01841          len = strlen(end);
01842          end += len;
01843          size -= len;
01844       }
01845    }
01846 
01847    if (start == end)
01848       snprintf(start, size, "nothing)"); 
01849    else if (size > 1)
01850       *(end -1) = ')';
01851    
01852    return buf;
01853 }

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 1799 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().

01801 {
01802    unsigned int i;
01803 
01804    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01805       if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
01806          if (isAstFormat &&
01807              (code == AST_FORMAT_G726_AAL2) &&
01808              (options & AST_RTP_OPT_G726_NONSTANDARD))
01809             return "G726-32";
01810          else
01811             return mimeTypes[i].subtype;
01812       }
01813    }
01814 
01815    return "";
01816 }

struct rtpPayloadType ast_rtp_lookup_pt ( struct ast_rtp rtp,
int  pt 
)

Mapping between RTP payload format codes and Asterisk codes:.

Definition at line 1736 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().

01737 {
01738    struct rtpPayloadType result;
01739 
01740    result.isAstFormat = result.code = 0;
01741 
01742    if (pt < 0 || pt > MAX_RTP_PT) 
01743       return result; /* bogus payload type */
01744 
01745    /* Start with negotiated codecs */
01746    ast_mutex_lock(&rtp->bridge_lock);
01747    result = rtp->current_RTP_PT[pt];
01748    ast_mutex_unlock(&rtp->bridge_lock);
01749 
01750    /* If it doesn't exist, check our static RTP type list, just in case */
01751    if (!result.code) 
01752       result = static_RTP_PT[pt];
01753 
01754    return result;
01755 }

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

Definition at line 1581 of file rtp.c.

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

Referenced by wait_for_answer().

01582 {
01583    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01584    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01585    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01586    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01587    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 
01588    int srccodec, destcodec;
01589 
01590    /* Lock channels */
01591    ast_channel_lock(dest);
01592    while(ast_channel_trylock(src)) {
01593       ast_channel_unlock(dest);
01594       usleep(1);
01595       ast_channel_lock(dest);
01596    }
01597 
01598    /* Find channel driver interfaces */
01599    if (!(destpr = get_proto(dest))) {
01600       if (option_debug)
01601          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01602       ast_channel_unlock(dest);
01603       ast_channel_unlock(src);
01604       return 0;
01605    }
01606    if (!(srcpr = get_proto(src))) {
01607       if (option_debug)
01608          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
01609       ast_channel_unlock(dest);
01610       ast_channel_unlock(src);
01611       return 0;
01612    }
01613 
01614    /* Get audio and video interface (if native bridge is possible) */
01615    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01616    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01617    audio_src_res = srcpr->get_rtp_info(src, &srcp);
01618    video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01619 
01620    /* Ensure we have at least one matching codec */
01621    if (srcpr->get_codec)
01622       srccodec = srcpr->get_codec(src);
01623    else
01624       srccodec = 0;
01625    if (destpr->get_codec)
01626       destcodec = destpr->get_codec(dest);
01627    else
01628       destcodec = 0;
01629 
01630    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01631    if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) {
01632       /* Somebody doesn't want to play... */
01633       ast_channel_unlock(dest);
01634       ast_channel_unlock(src);
01635       return 0;
01636    }
01637    ast_rtp_pt_copy(destp, srcp);
01638    if (vdestp && vsrcp)
01639       ast_rtp_pt_copy(vdestp, vsrcp);
01640    if (media) {
01641       /* Bridge early */
01642       if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01643          ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
01644    }
01645    ast_channel_unlock(dest);
01646    ast_channel_unlock(src);
01647    if (option_debug)
01648       ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
01649    return 1;
01650 }

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 2008 of file rtp.c.

References ast_rtp_new_with_bindaddr(), io, and sched.

02009 {
02010    struct in_addr ia;
02011 
02012    memset(&ia, 0, sizeof(ia));
02013    return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
02014 }

void ast_rtp_new_init ( struct ast_rtp rtp  ) 

Initialize a new RTP structure.

Definition at line 1902 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().

01903 {
01904    ast_mutex_init(&rtp->bridge_lock);
01905 
01906    rtp->them.sin_family = AF_INET;
01907    rtp->us.sin_family = AF_INET;
01908    rtp->ssrc = ast_random();
01909    rtp->seqno = ast_random() & 0xffff;
01910    ast_set_flag(rtp, FLAG_HAS_DTMF);
01911 
01912    return;
01913 }

void ast_rtp_new_source ( struct ast_rtp rtp  ) 

Definition at line 2025 of file rtp.c.

References ast_rtp::set_marker_bit.

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

02026 {
02027    if (rtp) {
02028       rtp->set_marker_bit = 1;
02029    }
02030    return;
02031 }

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 1915 of file rtp.c.

References ast_calloc, ast_log(), ast_random(), ast_rtcp_new(), ast_rtp_new_init(), errno, first, free, LOG_DEBUG, LOG_ERROR, option_debug, rtp_socket(), and sched.

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

01916 {
01917    struct ast_rtp *rtp;
01918    int x;
01919    int first;
01920    int startplace;
01921    
01922    if (!(rtp = ast_calloc(1, sizeof(*rtp))))
01923       return NULL;
01924 
01925    ast_rtp_new_init(rtp);
01926 
01927    rtp->s = rtp_socket();
01928    if (option_debug > 2)
01929          ast_log(LOG_DEBUG, "socket RTP fd: %i\n", rtp->s); 
01930    if (rtp->s < 0) {
01931       free(rtp);
01932       ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
01933       return NULL;
01934    }
01935    if (sched && rtcpenable) {
01936       rtp->sched = sched;
01937       rtp->rtcp = ast_rtcp_new();
01938       if (option_debug > 2)
01939             ast_log(LOG_DEBUG, "socket RTCP fd: %i\n", rtp->rtcp->s);
01940    }
01941    
01942    /* Select a random port number in the range of possible RTP */
01943    x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart;
01944    x = x & ~1;
01945    /* Save it for future references. */
01946    startplace = x;
01947    /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
01948    for (;;) {
01949       /* Must be an even port number by RTP spec */
01950       rtp->us.sin_port = htons(x);
01951       rtp->us.sin_addr = addr;
01952       /* If there's rtcp, initialize it as well. */
01953       if (rtp->rtcp) {
01954          rtp->rtcp->us.sin_port = htons(x + 1);
01955          rtp->rtcp->us.sin_addr = addr;
01956       }
01957       /* Try to bind it/them. */
01958       if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
01959          (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
01960          break;
01961       if (!first) {
01962          /* Primary bind succeeded! Gotta recreate it */
01963          close(rtp->s);
01964          rtp->s = rtp_socket();
01965          if (option_debug > 2)
01966                ast_log(LOG_DEBUG, "socket RTP2 fd: %i\n", rtp->s); 
01967       }
01968       if (errno != EADDRINUSE) {
01969          /* We got an error that wasn't expected, abort! */
01970          ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
01971          close(rtp->s);
01972          if (rtp->rtcp) {
01973             close(rtp->rtcp->s);
01974             free(rtp->rtcp);
01975          }
01976          free(rtp);
01977          return NULL;
01978       }
01979       /* The port was used, increment it (by two). */
01980       x += 2;
01981       /* Did we go over the limit ? */
01982       if (x > rtpend)
01983          /* then, start from the begingig. */
01984          x = (rtpstart + 1) & ~1;
01985       /* Check if we reached the place were we started. */
01986       if (x == startplace) {
01987          /* If so, there's no ports available. */
01988          ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
01989          close(rtp->s);
01990          if (rtp->rtcp) {
01991             close(rtp->rtcp->s);
01992             free(rtp->rtcp);
01993          }
01994          free(rtp);
01995          return NULL;
01996       }
01997    }
01998    rtp->sched = sched;
01999    rtp->io = io;
02000    if (callbackmode) {
02001       rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
02002       ast_set_flag(rtp, FLAG_CALLBACK_MODE);
02003    }
02004    ast_rtp_pt_default(rtp);
02005    return rtp;
02006 }

int ast_rtp_proto_register ( struct ast_rtp_protocol proto  ) 

Register interface to channel driver.

Definition at line 2902 of file rtp.c.

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

Referenced by load_module().

02903 {
02904    struct ast_rtp_protocol *cur;
02905 
02906    AST_LIST_LOCK(&protos);
02907    AST_LIST_TRAVERSE(&protos, cur, list) {   
02908       if (!strcmp(cur->type, proto->type)) {
02909          ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
02910          AST_LIST_UNLOCK(&protos);
02911          return -1;
02912       }
02913    }
02914    AST_LIST_INSERT_HEAD(&protos, proto, list);
02915    AST_LIST_UNLOCK(&protos);
02916    
02917    return 0;
02918 }

void ast_rtp_proto_unregister ( struct ast_rtp_protocol proto  ) 

Unregister interface to channel driver.

Definition at line 2894 of file rtp.c.

References AST_LIST_LOCK, AST_LIST_REMOVE, and AST_LIST_UNLOCK.

Referenced by load_module(), and unload_module().

02895 {
02896    AST_LIST_LOCK(&protos);
02897    AST_LIST_REMOVE(&protos, proto, list);
02898    AST_LIST_UNLOCK(&protos);
02899 }

void ast_rtp_pt_clear ( struct ast_rtp rtp  ) 

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

Definition at line 1418 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().

01419 {
01420    int i;
01421 
01422    if (!rtp)
01423       return;
01424 
01425    ast_mutex_lock(&rtp->bridge_lock);
01426 
01427    for (i = 0; i < MAX_RTP_PT; ++i) {
01428       rtp->current_RTP_PT[i].isAstFormat = 0;
01429       rtp->current_RTP_PT[i].code = 0;
01430    }
01431 
01432    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01433    rtp->rtp_lookup_code_cache_code = 0;
01434    rtp->rtp_lookup_code_cache_result = 0;
01435 
01436    ast_mutex_unlock(&rtp->bridge_lock);
01437 }

void ast_rtp_pt_copy ( struct ast_rtp dest,
struct ast_rtp src 
)

Copy payload types between RTP structures.

Definition at line 1458 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().

01459 {
01460    unsigned int i;
01461 
01462    ast_mutex_lock(&dest->bridge_lock);
01463    ast_mutex_lock(&src->bridge_lock);
01464 
01465    for (i=0; i < MAX_RTP_PT; ++i) {
01466       dest->current_RTP_PT[i].isAstFormat = 
01467          src->current_RTP_PT[i].isAstFormat;
01468       dest->current_RTP_PT[i].code = 
01469          src->current_RTP_PT[i].code; 
01470    }
01471    dest->rtp_lookup_code_cache_isAstFormat = 0;
01472    dest->rtp_lookup_code_cache_code = 0;
01473    dest->rtp_lookup_code_cache_result = 0;
01474 
01475    ast_mutex_unlock(&src->bridge_lock);
01476    ast_mutex_unlock(&dest->bridge_lock);
01477 }

void ast_rtp_pt_default ( struct ast_rtp rtp  ) 

Set payload types to defaults.

Definition at line 1439 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.

01440 {
01441    int i;
01442 
01443    ast_mutex_lock(&rtp->bridge_lock);
01444 
01445    /* Initialize to default payload types */
01446    for (i = 0; i < MAX_RTP_PT; ++i) {
01447       rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
01448       rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
01449    }
01450 
01451    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01452    rtp->rtp_lookup_code_cache_code = 0;
01453    rtp->rtp_lookup_code_cache_result = 0;
01454 
01455    ast_mutex_unlock(&rtp->bridge_lock);
01456 }

struct ast_frame* ast_rtp_read ( struct ast_rtp rtp  ) 

Definition at line 1108 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_DTMF_END, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_get_bridged(), ast_rtp_lookup_pt(), ast_rtp_senddigit_continuation(), ast_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, ast_rtp::dtmfcount, errno, ext, ast_rtp::f, f, FLAG_NAT_ACTIVE, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lastevent, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len(), LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_rtp::rawdata, ast_rtp::resp, ast_rtp::rtcp, rtp_debug_test_addr(), RTP_SEQ_MOD, ast_rtp::rxcount, ast_rtp::rxseqno, ast_rtp::rxssrc, ast_rtcp::s, ast_rtp::s, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, send_dtmf(), ast_rtp::sending_digit, ast_frame::seqno, ast_frame::src, STUN_ACCEPT, stun_handle_packet(), ast_frame::subclass, ast_rtcp::them, ast_rtp::them, ast_rtp::themssrc, and ast_frame::ts.

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

01109 {
01110    int res;
01111    struct sockaddr_in sin;
01112    socklen_t len;
01113    unsigned int seqno;
01114    int version;
01115    int payloadtype;
01116    int hdrlen = 12;
01117    int padding;
01118    int mark;
01119    int ext;
01120    int cc;
01121    unsigned int ssrc;
01122    unsigned int timestamp;
01123    unsigned int *rtpheader;
01124    struct rtpPayloadType rtpPT;
01125    struct ast_rtp *bridged = NULL;
01126    
01127    /* If time is up, kill it */
01128    if (rtp->sending_digit)
01129       ast_rtp_senddigit_continuation(rtp);
01130 
01131    len = sizeof(sin);
01132    
01133    /* Cache where the header will go */
01134    res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
01135                0, (struct sockaddr *)&sin, &len);
01136    if (option_debug > 3)
01137       ast_log(LOG_DEBUG, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
01138 
01139    rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
01140    if (res < 0) {
01141       ast_assert(errno != EBADF);
01142       if (errno != EAGAIN) {
01143          ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up.\n", strerror(errno));
01144          ast_log(LOG_WARNING, "socket RTP read: rtp %i rtcp %i\n", rtp->s, rtp->rtcp->s);
01145          return NULL;
01146       }
01147       return &ast_null_frame;
01148    }
01149    
01150    if (res < hdrlen) {
01151       ast_log(LOG_WARNING, "RTP Read too short\n");
01152       return &ast_null_frame;
01153    }
01154 
01155    /* Get fields */
01156    seqno = ntohl(rtpheader[0]);
01157 
01158    /* Check RTP version */
01159    version = (seqno & 0xC0000000) >> 30;
01160    if (!version) {
01161       if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) &&
01162          (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
01163          memcpy(&rtp->them, &sin, sizeof(rtp->them));
01164       }
01165       return &ast_null_frame;
01166    }
01167 
01168 #if 0 /* Allow to receive RTP stream with closed transmission path */
01169    /* If we don't have the other side's address, then ignore this */
01170    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
01171       return &ast_null_frame;
01172 #endif
01173 
01174    /* Send to whoever send to us if NAT is turned on */
01175    if (rtp->nat) {
01176       if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
01177           (rtp->them.sin_port != sin.sin_port)) {
01178          rtp->them = sin;
01179          if (rtp->rtcp) {
01180             memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
01181             rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
01182          }
01183          rtp->rxseqno = 0;
01184          ast_set_flag(rtp, FLAG_NAT_ACTIVE);
01185          if (option_debug || rtpdebug)
01186             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));
01187       }
01188    }
01189 
01190    /* If we are bridged to another RTP stream, send direct */
01191    if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
01192       return &ast_null_frame;
01193 
01194    if (version != 2)
01195       return &ast_null_frame;
01196 
01197    payloadtype = (seqno & 0x7f0000) >> 16;
01198    padding = seqno & (1 << 29);
01199    mark = seqno & (1 << 23);
01200    ext = seqno & (1 << 28);
01201    cc = (seqno & 0xF000000) >> 24;
01202    seqno &= 0xffff;
01203    timestamp = ntohl(rtpheader[1]);
01204    ssrc = ntohl(rtpheader[2]);
01205    
01206    if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
01207       if (option_debug || rtpdebug)
01208          ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
01209       mark = 1;
01210    }
01211 
01212    rtp->rxssrc = ssrc;
01213    
01214    if (padding) {
01215       /* Remove padding bytes */
01216       res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
01217    }
01218    
01219    if (cc) {
01220       /* CSRC fields present */
01221       hdrlen += cc*4;
01222    }
01223 
01224    if (ext) {
01225       /* RTP Extension present */
01226       hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
01227       hdrlen += 4;
01228    }
01229 
01230    if (res < hdrlen) {
01231       ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
01232       return &ast_null_frame;
01233    }
01234 
01235    rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
01236 
01237    if (rtp->rxcount==1) {
01238       /* This is the first RTP packet successfully received from source */
01239       rtp->seedrxseqno = seqno;
01240    }
01241 
01242    /* Do not schedule RR if RTCP isn't run */
01243    if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
01244       /* Schedule transmission of Receiver Report */
01245       rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
01246    }
01247    if ( (int)rtp->lastrxseqno - (int)seqno  > 100) /* if so it would indicate that the sender cycled; allow for misordering */
01248       rtp->cycles += RTP_SEQ_MOD;
01249 
01250    rtp->lastrxseqno = seqno;
01251    
01252    if (rtp->themssrc==0)
01253       rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
01254    
01255    if (rtp_debug_test_addr(&sin))
01256       ast_verbose("Got  RTP packet from    %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
01257          ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
01258 
01259    rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
01260    if (!rtpPT.isAstFormat) {
01261       struct ast_frame *f = NULL;
01262 
01263       /* This is special in-band data that's not one of our codecs */
01264       if (rtpPT.code == AST_RTP_DTMF) {
01265          /* It's special -- rfc2833 process it */
01266          if (rtp_debug_test_addr(&sin)) {
01267             unsigned char *data;
01268             unsigned int event;
01269             unsigned int event_end;
01270             unsigned int duration;
01271             data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
01272             event = ntohl(*((unsigned int *)(data)));
01273             event >>= 24;
01274             event_end = ntohl(*((unsigned int *)(data)));
01275             event_end <<= 8;
01276             event_end >>= 24;
01277             duration = ntohl(*((unsigned int *)(data)));
01278             duration &= 0xFFFF;
01279             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);
01280          }
01281          f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
01282       } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
01283          /* It's really special -- process it the Cisco way */
01284          if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
01285             f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01286             rtp->lastevent = seqno;
01287          }
01288       } else if (rtpPT.code == AST_RTP_CN) {
01289          /* Comfort Noise */
01290          f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01291       } else {
01292          ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
01293       }
01294       return f ? f : &ast_null_frame;
01295    }
01296    rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
01297    rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
01298 
01299    rtp->rxseqno = seqno;
01300 
01301    if (rtp->dtmfcount) {
01302       rtp->dtmfcount -= (timestamp - rtp->lastrxts);
01303 
01304       if (rtp->dtmfcount < 0) {
01305          rtp->dtmfcount = 0;
01306       }
01307 
01308       if (rtp->resp && !rtp->dtmfcount) {
01309          struct ast_frame *f;
01310          f = send_dtmf(rtp, AST_FRAME_DTMF_END);
01311          rtp->resp = 0;
01312          return f;
01313       }
01314    }
01315 
01316    /* Record received timestamp as last received now */
01317    rtp->lastrxts = timestamp;
01318 
01319    rtp->f.mallocd = 0;
01320    rtp->f.datalen = res - hdrlen;
01321    rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
01322    rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
01323    rtp->f.seqno = seqno;
01324    if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
01325       rtp->f.samples = ast_codec_get_samples(&rtp->f);
01326       if (rtp->f.subclass == AST_FORMAT_SLINEAR) 
01327          ast_frame_byteswap_be(&rtp->f);
01328       calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
01329       /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
01330       ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
01331       rtp->f.ts = timestamp / 8;
01332       rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000);
01333    } else {
01334       /* Video -- samples is # of samples vs. 90000 */
01335       if (!rtp->lastividtimestamp)
01336          rtp->lastividtimestamp = timestamp;
01337       rtp->f.samples = timestamp - rtp->lastividtimestamp;
01338       rtp->lastividtimestamp = timestamp;
01339       rtp->f.delivery.tv_sec = 0;
01340       rtp->f.delivery.tv_usec = 0;
01341       if (mark)
01342          rtp->f.subclass |= 0x1;
01343    }
01344    rtp->f.src = "RTP";
01345    return &rtp->f;
01346 }

int ast_rtp_reload ( void   ) 

Definition at line 3820 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().

03821 {
03822    struct ast_config *cfg;
03823    const char *s;
03824 
03825    rtpstart = 5000;
03826    rtpend = 31000;
03827    dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03828    cfg = ast_config_load("rtp.conf");
03829    if (cfg) {
03830       if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
03831          rtpstart = atoi(s);
03832          if (rtpstart < 1024)
03833             rtpstart = 1024;
03834          if (rtpstart > 65535)
03835             rtpstart = 65535;
03836       }
03837       if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
03838          rtpend = atoi(s);
03839          if (rtpend < 1024)
03840             rtpend = 1024;
03841          if (rtpend > 65535)
03842             rtpend = 65535;
03843       }
03844       if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
03845          rtcpinterval = atoi(s);
03846          if (rtcpinterval == 0)
03847             rtcpinterval = 0; /* Just so we're clear... it's zero */
03848          if (rtcpinterval < RTCP_MIN_INTERVALMS)
03849             rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
03850          if (rtcpinterval > RTCP_MAX_INTERVALMS)
03851             rtcpinterval = RTCP_MAX_INTERVALMS;
03852       }
03853       if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
03854 #ifdef SO_NO_CHECK
03855          if (ast_false(s))
03856             nochecksums = 1;
03857          else
03858             nochecksums = 0;
03859 #else
03860          if (ast_false(s))
03861             ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
03862 #endif
03863       }
03864       if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
03865          dtmftimeout = atoi(s);
03866          if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
03867             ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
03868                dtmftimeout, DEFAULT_DTMF_TIMEOUT);
03869             dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03870          };
03871       }
03872       ast_config_destroy(cfg);
03873    }
03874    if (rtpstart >= rtpend) {
03875       ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
03876       rtpstart = 5000;
03877       rtpend = 31000;
03878    }
03879    if (option_verbose > 1)
03880       ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
03881    return 0;
03882 }

void ast_rtp_reset ( struct ast_rtp rtp  ) 

Definition at line 2089 of file rtp.c.

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

02090 {
02091    memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
02092    memset(&rtp->txcore, 0, sizeof(rtp->txcore));
02093    memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
02094    rtp->lastts = 0;
02095    rtp->lastdigitts = 0;
02096    rtp->lastrxts = 0;
02097    rtp->lastividtimestamp = 0;
02098    rtp->lastovidtimestamp = 0;
02099    rtp->lasteventseqn = 0;
02100    rtp->lastevent = 0;
02101    rtp->lasttxformat = 0;
02102    rtp->lastrxformat = 0;
02103    rtp->dtmfcount = 0;
02104    rtp->seqno = 0;
02105    rtp->rxseqno = 0;
02106 }

int ast_rtp_sendcng ( struct ast_rtp rtp,
int  level 
)

generate comfort noice (CNG)

Definition at line 2603 of file rtp.c.

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

Referenced by do_monitor().

02604 {
02605    unsigned int *rtpheader;
02606    int hdrlen = 12;
02607    int res;
02608    int payload;
02609    char data[256];
02610    level = 127 - (level & 0x7f);
02611    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
02612 
02613    /* If we have no peer, return immediately */ 
02614    if (!rtp->them.sin_addr.s_addr)
02615       return 0;
02616 
02617    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02618 
02619    /* Get a pointer to the header */
02620    rtpheader = (unsigned int *)data;
02621    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
02622    rtpheader[1] = htonl(rtp->lastts);
02623    rtpheader[2] = htonl(rtp->ssrc); 
02624    data[12] = level;
02625    if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
02626       res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
02627       if (res <0) 
02628          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));
02629       if (rtp_debug_test_addr(&rtp->them))
02630          ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
02631                , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);         
02632          
02633    }
02634    return 0;
02635 }

int ast_rtp_senddigit_begin ( struct ast_rtp rtp,
char  digit 
)

Send begin frames for DTMF.

Definition at line 2211 of file rtp.c.

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

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

02212 {
02213    unsigned int *rtpheader;
02214    int hdrlen = 12, res = 0, i = 0, payload = 0;
02215    char data[256];
02216 
02217    if ((digit <= '9') && (digit >= '0'))
02218       digit -= '0';
02219    else if (digit == '*')
02220       digit = 10;
02221    else if (digit == '#')
02222       digit = 11;
02223    else if ((digit >= 'A') && (digit <= 'D'))
02224       digit = digit - 'A' + 12;
02225    else if ((digit >= 'a') && (digit <= 'd'))
02226       digit = digit - 'a' + 12;
02227    else {
02228       ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
02229       return 0;
02230    }
02231 
02232    /* If we have no peer, return immediately */ 
02233    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
02234       return 0;
02235 
02236    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
02237 
02238    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02239    rtp->send_duration = 160;
02240    rtp->lastdigitts = rtp->lastts + rtp->send_duration;
02241    
02242    /* Get a pointer to the header */
02243    rtpheader = (unsigned int *)data;
02244    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
02245    rtpheader[1] = htonl(rtp->lastdigitts);
02246    rtpheader[2] = htonl(rtp->ssrc); 
02247 
02248    for (i = 0; i < 2; i++) {
02249       rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
02250       res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
02251       if (res < 0) 
02252          ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
02253             ast_inet_ntoa(rtp->them.sin_addr),
02254             ntohs(rtp->them.sin_port), strerror(errno));
02255       if (rtp_debug_test_addr(&rtp->them))
02256          ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
02257                 ast_inet_ntoa(rtp->them.sin_addr),
02258                 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
02259       /* Increment sequence number */
02260       rtp->seqno++;
02261       /* Increment duration */
02262       rtp->send_duration += 160;
02263       /* Clear marker bit and set seqno */
02264       rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
02265    }
02266 
02267    /* Since we received a begin, we can safely store the digit and disable any compensation */
02268    rtp->sending_digit = 1;
02269    rtp->send_digit = digit;
02270    rtp->send_payload = payload;
02271 
02272    return 0;
02273 }

int ast_rtp_senddigit_end ( struct ast_rtp rtp,
char  digit 
)

void ast_rtp_set_callback ( struct ast_rtp rtp,
ast_rtp_callback  callback 
)

Definition at line 585 of file rtp.c.

References ast_rtp::callback.

Referenced by start_rtp().

00586 {
00587    rtp->callback = callback;
00588 }

void ast_rtp_set_data ( struct ast_rtp rtp,
void *  data 
)

Definition at line 580 of file rtp.c.

References ast_rtp::data.

Referenced by start_rtp().

00581 {
00582    rtp->data = data;
00583 }

void ast_rtp_set_m_type ( struct ast_rtp rtp,
int  pt 
)

Activate payload type.

Definition at line 1656 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().

01657 {
01658    if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 
01659       return; /* bogus payload type */
01660 
01661    ast_mutex_lock(&rtp->bridge_lock);
01662    rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
01663    ast_mutex_unlock(&rtp->bridge_lock);
01664 } 

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

Definition at line 2033 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().

02034 {
02035    rtp->them.sin_port = them->sin_port;
02036    rtp->them.sin_addr = them->sin_addr;
02037    if (rtp->rtcp) {
02038       rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
02039       rtp->rtcp->them.sin_addr = them->sin_addr;
02040    }
02041    rtp->rxseqno = 0;
02042 }

void ast_rtp_set_rtpholdtimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp hold timeout.

Definition at line 547 of file rtp.c.

References ast_rtp::rtpholdtimeout.

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

00548 {
00549    rtp->rtpholdtimeout = timeout;
00550 }

void ast_rtp_set_rtpkeepalive ( struct ast_rtp rtp,
int  period 
)

set RTP keepalive interval

Definition at line 553 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by create_addr_from_peer(), and sip_alloc().

00554 {
00555    rtp->rtpkeepalive = period;
00556 }

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 1683 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().

01686 {
01687    unsigned int i;
01688    int found = 0;
01689 
01690    if (pt < 0 || pt > MAX_RTP_PT) 
01691       return -1; /* bogus payload type */
01692    
01693    ast_mutex_lock(&rtp->bridge_lock);
01694 
01695    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01696       if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
01697           strcasecmp(mimeType, mimeTypes[i].type) == 0) {
01698          found = 1;
01699          rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
01700          if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
01701              mimeTypes[i].payloadType.isAstFormat &&
01702              (options & AST_RTP_OPT_G726_NONSTANDARD))
01703             rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
01704          break;
01705       }
01706    }
01707 
01708    ast_mutex_unlock(&rtp->bridge_lock);
01709 
01710    return (found ? 0 : -1);
01711 } 

void ast_rtp_set_rtptimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp timeout.

Definition at line 541 of file rtp.c.

References ast_rtp::rtptimeout.

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

00542 {
00543    rtp->rtptimeout = timeout;
00544 }

void ast_rtp_set_rtptimers_onhold ( struct ast_rtp rtp  ) 

Definition at line 534 of file rtp.c.

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

Referenced by handle_response_invite().

00535 {
00536    rtp->rtptimeout = (-1) * rtp->rtptimeout;
00537    rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
00538 }

void ast_rtp_setdtmf ( struct ast_rtp rtp,
int  dtmf 
)

Indicate whether this RTP session is carrying DTMF or not.

Definition at line 600 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_DTMF.

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

00601 {
00602    ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
00603 }

void ast_rtp_setdtmfcompensate ( struct ast_rtp rtp,
int  compensate 
)

Compensate for devices that send RFC2833 packets all at once.

Definition at line 605 of file rtp.c.

References ast_set2_flag, and FLAG_DTMF_COMPENSATE.

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

00606 {
00607    ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
00608 }

void ast_rtp_setnat ( struct ast_rtp rtp,
int  nat 
)

Definition at line 590 of file rtp.c.

References ast_rtp::nat.

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

00591 {
00592    rtp->nat = nat;
00593 }

void ast_rtp_setstun ( struct ast_rtp rtp,
int  stun_enable 
)

Enable STUN capability.

Definition at line 610 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_STUN.

Referenced by gtalk_new().

00611 {
00612    ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
00613 }

int ast_rtp_settos ( struct ast_rtp rtp,
int  tos 
)

Definition at line 2016 of file rtp.c.

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

Referenced by __oh323_rtp_create(), and sip_alloc().

02017 {
02018    int res;
02019 
02020    if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
02021       ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
02022    return res;
02023 }

void ast_rtp_stop ( struct ast_rtp rtp  ) 

Definition at line 2073 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().

02074 {
02075    if (rtp->rtcp) {
02076       AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
02077    }
02078 
02079    memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
02080    memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
02081    if (rtp->rtcp) {
02082       memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
02083       memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
02084    }
02085    
02086    ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
02087 }

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

Definition at line 402 of file rtp.c.

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

Referenced by gtalk_update_stun().

00403 {
00404    struct stun_header *req;
00405    unsigned char reqdata[1024];
00406    int reqlen, reqleft;
00407    struct stun_attr *attr;
00408 
00409    req = (struct stun_header *)reqdata;
00410    stun_req_id(req);
00411    reqlen = 0;
00412    reqleft = sizeof(reqdata) - sizeof(struct stun_header);
00413    req->msgtype = 0;
00414    req->msglen = 0;
00415    attr = (struct stun_attr *)req->ies;
00416    if (username)
00417       append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
00418    req->msglen = htons(reqlen);
00419    req->msgtype = htons(STUN_BINDREQ);
00420    stun_send(rtp->s, suggestion, req);
00421 }

void ast_rtp_unset_m_type ( struct ast_rtp rtp,
int  pt 
)

clear payload type

Definition at line 1668 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().

01669 {
01670    if (pt < 0 || pt > MAX_RTP_PT)
01671       return; /* bogus payload type */
01672 
01673    ast_mutex_lock(&rtp->bridge_lock);
01674    rtp->current_RTP_PT[pt].isAstFormat = 0;
01675    rtp->current_RTP_PT[pt].code = 0;
01676    ast_mutex_unlock(&rtp->bridge_lock);
01677 }

int ast_rtp_write ( struct ast_rtp rtp,
struct ast_frame f 
)

Definition at line 2802 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_format_list::cur_ms, ast_frame::datalen, f, ast_format_list::flags, ast_format_list::fr_len, ast_frame::frametype, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, option_debug, ast_rtp::pref, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.

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

02803 {
02804    struct ast_frame *f;
02805    int codec;
02806    int hdrlen = 12;
02807    int subclass;
02808    
02809 
02810    /* If we have no peer, return immediately */ 
02811    if (!rtp->them.sin_addr.s_addr)
02812       return 0;
02813 
02814    /* If there is no data length, return immediately */
02815    if (!_f->datalen) 
02816       return 0;
02817    
02818    /* Make sure we have enough space for RTP header */
02819    if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
02820       ast_log(LOG_WARNING, "RTP can only send voice and video\n");
02821       return -1;
02822    }
02823 
02824    subclass = _f->subclass;
02825    if (_f->frametype == AST_FRAME_VIDEO)
02826       subclass &= ~0x1;
02827 
02828    codec = ast_rtp_lookup_code(rtp, 1, subclass);
02829    if (codec < 0) {
02830       ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
02831       return -1;
02832    }
02833 
02834    if (rtp->lasttxformat != subclass) {
02835       /* New format, reset the smoother */
02836       if (option_debug)
02837          ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
02838       rtp->lasttxformat = subclass;
02839       if (rtp->smoother)
02840          ast_smoother_free(rtp->smoother);
02841       rtp->smoother = NULL;
02842    }
02843 
02844    if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) {
02845       struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
02846       if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
02847          if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
02848             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));
02849             return -1;
02850          }
02851          if (fmt.flags)
02852             ast_smoother_set_flags(rtp->smoother, fmt.flags);
02853          if (option_debug)
02854             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));
02855       }
02856    }
02857    if (rtp->smoother) {
02858       if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
02859          ast_smoother_feed_be(rtp->smoother, _f);
02860       } else {
02861          ast_smoother_feed(rtp->smoother, _f);
02862       }
02863 
02864       while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) {
02865          if (f->subclass == AST_FORMAT_G722) {
02866             /* G.722 is silllllllllllllly */
02867             f->samples /= 2;
02868          }
02869 
02870          ast_rtp_raw_write(rtp, f, codec);
02871       }
02872    } else {
02873       /* Don't buffer outgoing frames; send them one-per-packet: */
02874       if (_f->offset < hdrlen) {
02875          f = ast_frdup(_f);
02876       } else {
02877          f = _f;
02878       }
02879       if (f->data) {
02880          if (f->subclass == AST_FORMAT_G722) {
02881             /* G.722 is silllllllllllllly */
02882             f->samples /= 2;
02883          }
02884          ast_rtp_raw_write(rtp, f, codec);
02885       }
02886       if (f != _f)
02887          ast_frfree(f);
02888    }
02889       
02890    return 0;
02891 }


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