Sat Jul 26 06:14:29 2008

Asterisk developer's documentation


rtp.h File Reference

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

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

Include dependency graph for rtp.h:

This graph shows which files directly or indirectly include this file:

Go to the source code of this file.

Data Structures

struct  ast_rtp_protocol
struct  ast_rtp_quality

Defines

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

Typedefs

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

Enumerations

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

Functions

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


Detailed Description

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

RTP is defined in RFC 3550.

Definition in file rtp.h.


Define Documentation

#define AST_RTP_CISCO_DTMF   (1 << 2)

DTMF (Cisco Proprietary)

Definition at line 47 of file rtp.h.

Referenced by ast_rtp_read().

#define AST_RTP_CN   (1 << 1)

'Comfort Noise' (RFC3389)

Definition at line 45 of file rtp.h.

Referenced by ast_rtp_read(), and ast_rtp_sendcng().

#define AST_RTP_DTMF   (1 << 0)

DTMF (RFC2833)

Definition at line 43 of file rtp.h.

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

#define AST_RTP_MAX   AST_RTP_CISCO_DTMF

Maximum RTP-specific code

Definition at line 49 of file rtp.h.

Referenced by add_sdp(), and ast_rtp_lookup_mime_multiple().

#define FLAG_3389_WARNING   (1 << 0)

Definition at line 93 of file rtp.h.

#define MAX_RTP_PT   256

Definition at line 51 of file rtp.h.

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


Typedef Documentation

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

Definition at line 95 of file rtp.h.


Enumeration Type Documentation

enum ast_rtp_get_result

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

Definition at line 57 of file rtp.h.

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

enum ast_rtp_options

Enumerator:
AST_RTP_OPT_G726_NONSTANDARD 

Definition at line 53 of file rtp.h.

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


Function Documentation

int ast_rtcp_fd ( struct ast_rtp rtp  ) 

Definition at line 518 of file rtp.c.

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

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

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

struct ast_frame* ast_rtcp_read ( struct ast_rtp rtp  ) 

Definition at line 827 of file rtp.c.

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

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

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

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 2341 of file rtp.c.

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

02342 {
02343    struct ast_rtp *rtp = data;
02344    int res;
02345 
02346    rtp->rtcp->sendfur = 1;
02347    res = ast_rtcp_write(data);
02348    
02349    return res;
02350 }

size_t ast_rtp_alloc_size ( void   ) 

Get the amount of space required to hold an RTP session.

Returns:
number of bytes required

Definition at line 398 of file rtp.c.

Referenced by process_sdp().

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

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

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

Definition at line 3279 of file rtp.c.

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

03280 {
03281    struct ast_rtp *p0 = NULL, *p1 = NULL;    /* Audio RTP Channels */
03282    struct ast_rtp *vp0 = NULL, *vp1 = NULL;  /* Video RTP channels */
03283    struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
03284    enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
03285    enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
03286    enum ast_bridge_result res = AST_BRIDGE_FAILED;
03287    int codec0 = 0, codec1 = 0;
03288    void *pvt0 = NULL, *pvt1 = NULL;
03289 
03290    /* Lock channels */
03291    ast_channel_lock(c0);
03292    while(ast_channel_trylock(c1)) {
03293       ast_channel_unlock(c0);
03294       usleep(1);
03295       ast_channel_lock(c0);
03296    }
03297 
03298    /* Ensure neither channel got hungup during lock avoidance */
03299    if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
03300       ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
03301       ast_channel_unlock(c0);
03302       ast_channel_unlock(c1);
03303       return AST_BRIDGE_FAILED;
03304    }
03305       
03306    /* Find channel driver interfaces */
03307    if (!(pr0 = get_proto(c0))) {
03308       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
03309       ast_channel_unlock(c0);
03310       ast_channel_unlock(c1);
03311       return AST_BRIDGE_FAILED;
03312    }
03313    if (!(pr1 = get_proto(c1))) {
03314       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
03315       ast_channel_unlock(c0);
03316       ast_channel_unlock(c1);
03317       return AST_BRIDGE_FAILED;
03318    }
03319 
03320    /* Get channel specific interface structures */
03321    pvt0 = c0->tech_pvt;
03322    pvt1 = c1->tech_pvt;
03323 
03324    /* Get audio and video interface (if native bridge is possible) */
03325    audio_p0_res = pr0->get_rtp_info(c0, &p0);
03326    video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
03327    audio_p1_res = pr1->get_rtp_info(c1, &p1);
03328    video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
03329 
03330    /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
03331    if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
03332       audio_p0_res = AST_RTP_GET_FAILED;
03333    if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
03334       audio_p1_res = AST_RTP_GET_FAILED;
03335 
03336    /* Check if a bridge is possible (partial/native) */
03337    if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
03338       /* Somebody doesn't want to play... */
03339       ast_channel_unlock(c0);
03340       ast_channel_unlock(c1);
03341       return AST_BRIDGE_FAILED_NOWARN;
03342    }
03343 
03344    /* If we need to feed DTMF frames into the core then only do a partial native bridge */
03345    if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
03346       ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
03347       audio_p0_res = AST_RTP_TRY_PARTIAL;
03348    }
03349 
03350    if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
03351       ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
03352       audio_p1_res = AST_RTP_TRY_PARTIAL;
03353    }
03354 
03355    /* If both sides are not using the same method of DTMF transmission 
03356     * (ie: one is RFC2833, other is INFO... then we can not do direct media. 
03357     * --------------------------------------------------
03358     * | DTMF Mode |  HAS_DTMF  |  Accepts Begin Frames |
03359     * |-----------|------------|-----------------------|
03360     * | Inband    | False      | True                  |
03361     * | RFC2833   | True       | True                  |
03362     * | SIP INFO  | False      | False                 |
03363     * --------------------------------------------------
03364     * However, if DTMF from both channels is being monitored by the core, then
03365     * we can still do packet-to-packet bridging, because passing through the 
03366     * core will handle DTMF mode translation.
03367     */
03368    if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
03369        (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
03370       if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
03371          ast_channel_unlock(c0);
03372          ast_channel_unlock(c1);
03373          return AST_BRIDGE_FAILED_NOWARN;
03374       }
03375       audio_p0_res = AST_RTP_TRY_PARTIAL;
03376       audio_p1_res = AST_RTP_TRY_PARTIAL;
03377    }
03378 
03379    /* If we need to feed frames into the core don't do a P2P bridge */
03380    if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) ||
03381        (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) {
03382       ast_channel_unlock(c0);
03383       ast_channel_unlock(c1);
03384       return AST_BRIDGE_FAILED_NOWARN;
03385    }
03386 
03387    /* Get codecs from both sides */
03388    codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
03389    codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
03390    if (codec0 && codec1 && !(codec0 & codec1)) {
03391       /* Hey, we can't do native bridging if both parties speak different codecs */
03392       if (option_debug)
03393          ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
03394       ast_channel_unlock(c0);
03395       ast_channel_unlock(c1);
03396       return AST_BRIDGE_FAILED_NOWARN;
03397    }
03398 
03399    /* If either side can only do a partial bridge, then don't try for a true native bridge */
03400    if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
03401       struct ast_format_list fmt0, fmt1;
03402 
03403       /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
03404       if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
03405          if (option_debug)
03406             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n");
03407          ast_channel_unlock(c0);
03408          ast_channel_unlock(c1);
03409          return AST_BRIDGE_FAILED_NOWARN;
03410       }
03411       /* They must also be using the same packetization */
03412       fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
03413       fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
03414       if (fmt0.cur_ms != fmt1.cur_ms) {
03415          if (option_debug)
03416             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n");
03417          ast_channel_unlock(c0);
03418          ast_channel_unlock(c1);
03419          return AST_BRIDGE_FAILED_NOWARN;
03420       }
03421 
03422       if (option_verbose > 2)
03423          ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
03424       res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
03425    } else {
03426       if (option_verbose > 2) 
03427          ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
03428       res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
03429    }
03430 
03431    return res;
03432 }

int ast_rtp_codec_getformat ( int  pt  ) 

Definition at line 2723 of file rtp.c.

References rtpPayloadType::code, and static_RTP_PT.

Referenced by process_sdp().

02724 {
02725    if (pt < 0 || pt > MAX_RTP_PT)
02726       return 0; /* bogus payload type */
02727 
02728    if (static_RTP_PT[pt].isAstFormat)
02729       return static_RTP_PT[pt].code;
02730    else
02731       return 0;
02732 }

struct ast_codec_pref* ast_rtp_codec_getpref ( struct ast_rtp rtp  ) 

Definition at line 2718 of file rtp.c.

References ast_rtp::pref.

Referenced by add_codec_to_sdp(), and process_sdp().

02719 {
02720    return &rtp->pref;
02721 }

int ast_rtp_codec_setpref ( struct ast_rtp rtp,
struct ast_codec_pref prefs 
)

Definition at line 2705 of file rtp.c.

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

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

02706 {
02707    int x;
02708    for (x = 0; x < 32; x++) {  /* Ugly way */
02709       rtp->pref.order[x] = prefs->order[x];
02710       rtp->pref.framing[x] = prefs->framing[x];
02711    }
02712    if (rtp->smoother)
02713       ast_smoother_free(rtp->smoother);
02714    rtp->smoother = NULL;
02715    return 0;
02716 }

void ast_rtp_destroy ( struct ast_rtp rtp  ) 

Definition at line 2124 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().

02125 {
02126    if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
02127       /*Print some info on the call here */
02128       ast_verbose("  RTP-stats\n");
02129       ast_verbose("* Our Receiver:\n");
02130       ast_verbose("  SSRC:     %u\n", rtp->themssrc);
02131       ast_verbose("  Received packets: %u\n", rtp->rxcount);
02132       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
02133       ast_verbose("  Jitter:      %.4f\n", rtp->rxjitter);
02134       ast_verbose("  Transit:     %.4f\n", rtp->rxtransit);
02135       ast_verbose("  RR-count:    %u\n", rtp->rtcp->rr_count);
02136       ast_verbose("* Our Sender:\n");
02137       ast_verbose("  SSRC:     %u\n", rtp->ssrc);
02138       ast_verbose("  Sent packets:   %u\n", rtp->txcount);
02139       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->reported_lost);
02140       ast_verbose("  Jitter:      %u\n", rtp->rtcp->reported_jitter / (unsigned int)65536.0);
02141       ast_verbose("  SR-count:    %u\n", rtp->rtcp->sr_count);
02142       ast_verbose("  RTT:      %f\n", rtp->rtcp->rtt);
02143    }
02144 
02145    if (rtp->smoother)
02146       ast_smoother_free(rtp->smoother);
02147    if (rtp->ioid)
02148       ast_io_remove(rtp->io, rtp->ioid);
02149    if (rtp->s > -1)
02150       close(rtp->s);
02151    if (rtp->rtcp) {
02152       AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
02153       close(rtp->rtcp->s);
02154       free(rtp->rtcp);
02155       rtp->rtcp=NULL;
02156    }
02157 
02158    ast_mutex_destroy(&rtp->bridge_lock);
02159 
02160    free(rtp);
02161 }

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 1476 of file rtp.c.

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

Referenced by wait_for_answer().

01477 {
01478    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01479    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01480    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01481    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01482    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
01483    int srccodec, destcodec, nat_active = 0;
01484 
01485    /* Lock channels */
01486    ast_channel_lock(dest);
01487    if (src) {
01488       while(ast_channel_trylock(src)) {
01489          ast_channel_unlock(dest);
01490          usleep(1);
01491          ast_channel_lock(dest);
01492       }
01493    }
01494 
01495    /* Find channel driver interfaces */
01496    destpr = get_proto(dest);
01497    if (src)
01498       srcpr = get_proto(src);
01499    if (!destpr) {
01500       if (option_debug)
01501          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01502       ast_channel_unlock(dest);
01503       if (src)
01504          ast_channel_unlock(src);
01505       return 0;
01506    }
01507    if (!srcpr) {
01508       if (option_debug)
01509          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
01510       ast_channel_unlock(dest);
01511       if (src)
01512          ast_channel_unlock(src);
01513       return 0;
01514    }
01515 
01516    /* Get audio and video interface (if native bridge is possible) */
01517    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01518    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01519    if (srcpr) {
01520       audio_src_res = srcpr->get_rtp_info(src, &srcp);
01521       video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01522    }
01523 
01524    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01525    if (audio_dest_res != AST_RTP_TRY_NATIVE) {
01526       /* Somebody doesn't want to play... */
01527       ast_channel_unlock(dest);
01528       if (src)
01529          ast_channel_unlock(src);
01530       return 0;
01531    }
01532    if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
01533       srccodec = srcpr->get_codec(src);
01534    else
01535       srccodec = 0;
01536    if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
01537       destcodec = destpr->get_codec(dest);
01538    else
01539       destcodec = 0;
01540    /* Ensure we have at least one matching codec */
01541    if (!(srccodec & destcodec)) {
01542       ast_channel_unlock(dest);
01543       if (src)
01544          ast_channel_unlock(src);
01545       return 0;
01546    }
01547    /* Consider empty media as non-existant */
01548    if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
01549       srcp = NULL;
01550    /* If the client has NAT stuff turned on then just safe NAT is active */
01551    if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01552       nat_active = 1;
01553    /* Bridge media early */
01554    if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active))
01555       ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>");
01556    ast_channel_unlock(dest);
01557    if (src)
01558       ast_channel_unlock(src);
01559    if (option_debug)
01560       ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
01561    return 1;
01562 }

int ast_rtp_fd ( struct ast_rtp rtp  ) 

Definition at line 513 of file rtp.c.

References ast_rtp::s.

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

00514 {
00515    return rtp->s;
00516 }

struct ast_rtp* ast_rtp_get_bridged ( struct ast_rtp rtp  ) 

Definition at line 2035 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().

02036 {
02037    struct ast_rtp *bridged = NULL;
02038 
02039    ast_mutex_lock(&rtp->bridge_lock);
02040    bridged = rtp->bridged;
02041    ast_mutex_unlock(&rtp->bridge_lock);
02042 
02043    return bridged;
02044 }

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 1698 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().

01700 {
01701    int pt;
01702    
01703    ast_mutex_lock(&rtp->bridge_lock);
01704    
01705    *astFormats = *nonAstFormats = 0;
01706    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01707       if (rtp->current_RTP_PT[pt].isAstFormat) {
01708          *astFormats |= rtp->current_RTP_PT[pt].code;
01709       } else {
01710          *nonAstFormats |= rtp->current_RTP_PT[pt].code;
01711       }
01712    }
01713    
01714    ast_mutex_unlock(&rtp->bridge_lock);
01715    
01716    return;
01717 }

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

Definition at line 2017 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().

02018 {
02019    if ((them->sin_family != AF_INET) ||
02020       (them->sin_port != rtp->them.sin_port) ||
02021       (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
02022       them->sin_family = AF_INET;
02023       them->sin_port = rtp->them.sin_port;
02024       them->sin_addr = rtp->them.sin_addr;
02025       return 1;
02026    }
02027    return 0;
02028 }

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

Return RTCP quality string.

Definition at line 2080 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().

02081 {
02082    /*
02083    *ssrc          our ssrc
02084    *themssrc      their ssrc
02085    *lp            lost packets
02086    *rxjitter      our calculated jitter(rx)
02087    *rxcount       no. received packets
02088    *txjitter      reported jitter of the other end
02089    *txcount       transmitted packets
02090    *rlp           remote lost packets
02091    *rtt           round trip time
02092    */
02093 
02094    if (qual && rtp) {
02095       qual->local_ssrc = rtp->ssrc;
02096       qual->local_jitter = rtp->rxjitter;
02097       qual->local_count = rtp->rxcount;
02098       qual->remote_ssrc = rtp->themssrc;
02099       qual->remote_count = rtp->txcount;
02100       if (rtp->rtcp) {
02101          qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
02102          qual->remote_lostpackets = rtp->rtcp->reported_lost;
02103          qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
02104          qual->rtt = rtp->rtcp->rtt;
02105       }
02106    }
02107    if (rtp->rtcp) {
02108       snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality),
02109          "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
02110          rtp->ssrc,
02111          rtp->themssrc,
02112          rtp->rtcp->expected_prior - rtp->rtcp->received_prior,
02113          rtp->rxjitter,
02114          rtp->rxcount,
02115          (double)rtp->rtcp->reported_jitter / 65536.0,
02116          rtp->txcount,
02117          rtp->rtcp->reported_lost,
02118          rtp->rtcp->rtt);
02119       return rtp->rtcp->quality;
02120    } else
02121       return "<Unknown> - RTP/RTCP has already been destroyed";
02122 }

int ast_rtp_get_rtpholdtimeout ( struct ast_rtp rtp  ) 

Get rtp hold timeout.

Definition at line 568 of file rtp.c.

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

Referenced by do_monitor().

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

int ast_rtp_get_rtpkeepalive ( struct ast_rtp rtp  ) 

Get RTP keepalive interval.

Definition at line 576 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by do_monitor().

00577 {
00578    return rtp->rtpkeepalive;
00579 }

int ast_rtp_get_rtptimeout ( struct ast_rtp rtp  ) 

Get rtp timeout.

Definition at line 560 of file rtp.c.

References ast_rtp::rtptimeout.

Referenced by do_monitor().

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

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

Definition at line 2030 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().

02031 {
02032    *us = rtp->us;
02033 }

int ast_rtp_getnat ( struct ast_rtp rtp  ) 

Definition at line 596 of file rtp.c.

References ast_test_flag, and FLAG_NAT_ACTIVE.

Referenced by sip_get_rtp_peer().

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

void ast_rtp_init ( void   ) 

Initialize the RTP system in Asterisk.

Definition at line 3817 of file rtp.c.

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

Referenced by main().

03818 {
03819    ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
03820    ast_rtp_reload();
03821 }

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 1741 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().

01742 {
01743    int pt = 0;
01744 
01745    ast_mutex_lock(&rtp->bridge_lock);
01746 
01747    if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
01748       code == rtp->rtp_lookup_code_cache_code) {
01749       /* Use our cached mapping, to avoid the overhead of the loop below */
01750       pt = rtp->rtp_lookup_code_cache_result;
01751       ast_mutex_unlock(&rtp->bridge_lock);
01752       return pt;
01753    }
01754 
01755    /* Check the dynamic list first */
01756    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01757       if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
01758          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01759          rtp->rtp_lookup_code_cache_code = code;
01760          rtp->rtp_lookup_code_cache_result = pt;
01761          ast_mutex_unlock(&rtp->bridge_lock);
01762          return pt;
01763       }
01764    }
01765 
01766    /* Then the static list */
01767    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01768       if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
01769          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01770          rtp->rtp_lookup_code_cache_code = code;
01771          rtp->rtp_lookup_code_cache_result = pt;
01772          ast_mutex_unlock(&rtp->bridge_lock);
01773          return pt;
01774       }
01775    }
01776 
01777    ast_mutex_unlock(&rtp->bridge_lock);
01778 
01779    return -1;
01780 }

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 1801 of file rtp.c.

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

Referenced by process_sdp().

01803 {
01804    int format;
01805    unsigned len;
01806    char *end = buf;
01807    char *start = buf;
01808 
01809    if (!buf || !size)
01810       return NULL;
01811 
01812    snprintf(end, size, "0x%x (", capability);
01813 
01814    len = strlen(end);
01815    end += len;
01816    size -= len;
01817    start = end;
01818 
01819    for (format = 1; format < AST_RTP_MAX; format <<= 1) {
01820       if (capability & format) {
01821          const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
01822 
01823          snprintf(end, size, "%s|", name);
01824          len = strlen(end);
01825          end += len;
01826          size -= len;
01827       }
01828    }
01829 
01830    if (start == end)
01831       snprintf(start, size, "nothing)"); 
01832    else if (size > 1)
01833       *(end -1) = ')';
01834    
01835    return buf;
01836 }

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 1782 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().

01784 {
01785    unsigned int i;
01786 
01787    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01788       if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
01789          if (isAstFormat &&
01790              (code == AST_FORMAT_G726_AAL2) &&
01791              (options & AST_RTP_OPT_G726_NONSTANDARD))
01792             return "G726-32";
01793          else
01794             return mimeTypes[i].subtype;
01795       }
01796    }
01797 
01798    return "";
01799 }

struct rtpPayloadType ast_rtp_lookup_pt ( struct ast_rtp rtp,
int  pt 
)

Mapping between RTP payload format codes and Asterisk codes:.

Definition at line 1719 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().

01720 {
01721    struct rtpPayloadType result;
01722 
01723    result.isAstFormat = result.code = 0;
01724 
01725    if (pt < 0 || pt > MAX_RTP_PT) 
01726       return result; /* bogus payload type */
01727 
01728    /* Start with negotiated codecs */
01729    ast_mutex_lock(&rtp->bridge_lock);
01730    result = rtp->current_RTP_PT[pt];
01731    ast_mutex_unlock(&rtp->bridge_lock);
01732 
01733    /* If it doesn't exist, check our static RTP type list, just in case */
01734    if (!result.code) 
01735       result = static_RTP_PT[pt];
01736 
01737    return result;
01738 }

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

Definition at line 1564 of file rtp.c.

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

Referenced by wait_for_answer().

01565 {
01566    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01567    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01568    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01569    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01570    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 
01571    int srccodec, destcodec;
01572 
01573    /* Lock channels */
01574    ast_channel_lock(dest);
01575    while(ast_channel_trylock(src)) {
01576       ast_channel_unlock(dest);
01577       usleep(1);
01578       ast_channel_lock(dest);
01579    }
01580 
01581    /* Find channel driver interfaces */
01582    if (!(destpr = get_proto(dest))) {
01583       if (option_debug)
01584          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01585       ast_channel_unlock(dest);
01586       ast_channel_unlock(src);
01587       return 0;
01588    }
01589    if (!(srcpr = get_proto(src))) {
01590       if (option_debug)
01591          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
01592       ast_channel_unlock(dest);
01593       ast_channel_unlock(src);
01594       return 0;
01595    }
01596 
01597    /* Get audio and video interface (if native bridge is possible) */
01598    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01599    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01600    audio_src_res = srcpr->get_rtp_info(src, &srcp);
01601    video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01602 
01603    /* Ensure we have at least one matching codec */
01604    if (srcpr->get_codec)
01605       srccodec = srcpr->get_codec(src);
01606    else
01607       srccodec = 0;
01608    if (destpr->get_codec)
01609       destcodec = destpr->get_codec(dest);
01610    else
01611       destcodec = 0;
01612 
01613    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01614    if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
01615       /* Somebody doesn't want to play... */
01616       ast_channel_unlock(dest);
01617       ast_channel_unlock(src);
01618       return 0;
01619    }
01620    ast_rtp_pt_copy(destp, srcp);
01621    if (vdestp && vsrcp)
01622       ast_rtp_pt_copy(vdestp, vsrcp);
01623    if (media) {
01624       /* Bridge early */
01625       if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01626          ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
01627    }
01628    ast_channel_unlock(dest);
01629    ast_channel_unlock(src);
01630    if (option_debug)
01631       ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
01632    return 1;
01633 }

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 1983 of file rtp.c.

References ast_rtp_new_with_bindaddr(), io, and sched.

01984 {
01985    struct in_addr ia;
01986 
01987    memset(&ia, 0, sizeof(ia));
01988    return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
01989 }

void ast_rtp_new_init ( struct ast_rtp rtp  ) 

Initialize a new RTP structure.

Definition at line 1883 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().

01884 {
01885    ast_mutex_init(&rtp->bridge_lock);
01886 
01887    rtp->them.sin_family = AF_INET;
01888    rtp->us.sin_family = AF_INET;
01889    rtp->ssrc = ast_random();
01890    rtp->seqno = ast_random() & 0xffff;
01891    ast_set_flag(rtp, FLAG_HAS_DTMF);
01892 
01893    return;
01894 }

void ast_rtp_new_source ( struct ast_rtp rtp  ) 

Definition at line 2000 of file rtp.c.

References ast_rtp::set_marker_bit.

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

02001 {
02002    rtp->set_marker_bit = 1;
02003    return;
02004 }

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 1896 of file rtp.c.

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

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

01897 {
01898    struct ast_rtp *rtp;
01899    int x;
01900    int first;
01901    int startplace;
01902    
01903    if (!(rtp = ast_calloc(1, sizeof(*rtp))))
01904       return NULL;
01905 
01906    ast_rtp_new_init(rtp);
01907 
01908    rtp->s = rtp_socket();
01909    if (rtp->s < 0) {
01910       free(rtp);
01911       ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
01912       return NULL;
01913    }
01914    if (sched && rtcpenable) {
01915       rtp->sched = sched;
01916       rtp->rtcp = ast_rtcp_new();
01917    }
01918    
01919    /* Select a random port number in the range of possible RTP */
01920    x = (ast_random() % (rtpend-rtpstart)) + rtpstart;
01921    x = x & ~1;
01922    /* Save it for future references. */
01923    startplace = x;
01924    /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
01925    for (;;) {
01926       /* Must be an even port number by RTP spec */
01927       rtp->us.sin_port = htons(x);
01928       rtp->us.sin_addr = addr;
01929       /* If there's rtcp, initialize it as well. */
01930       if (rtp->rtcp) {
01931          rtp->rtcp->us.sin_port = htons(x + 1);
01932          rtp->rtcp->us.sin_addr = addr;
01933       }
01934       /* Try to bind it/them. */
01935       if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
01936          (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
01937          break;
01938       if (!first) {
01939          /* Primary bind succeeded! Gotta recreate it */
01940          close(rtp->s);
01941          rtp->s = rtp_socket();
01942       }
01943       if (errno != EADDRINUSE) {
01944          /* We got an error that wasn't expected, abort! */
01945          ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
01946          close(rtp->s);
01947          if (rtp->rtcp) {
01948             close(rtp->rtcp->s);
01949             free(rtp->rtcp);
01950          }
01951          free(rtp);
01952          return NULL;
01953       }
01954       /* The port was used, increment it (by two). */
01955       x += 2;
01956       /* Did we go over the limit ? */
01957       if (x > rtpend)
01958          /* then, start from the begingig. */
01959          x = (rtpstart + 1) & ~1;
01960       /* Check if we reached the place were we started. */
01961       if (x == startplace) {
01962          /* If so, there's no ports available. */
01963          ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
01964          close(rtp->s);
01965          if (rtp->rtcp) {
01966             close(rtp->rtcp->s);
01967             free(rtp->rtcp);
01968          }
01969          free(rtp);
01970          return NULL;
01971       }
01972    }
01973    rtp->sched = sched;
01974    rtp->io = io;
01975    if (callbackmode) {
01976       rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
01977       ast_set_flag(rtp, FLAG_CALLBACK_MODE);
01978    }
01979    ast_rtp_pt_default(rtp);
01980    return rtp;
01981 }

int ast_rtp_proto_register ( struct ast_rtp_protocol proto  ) 

Register interface to channel driver.

Definition at line 2834 of file rtp.c.

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

Referenced by load_module().

02835 {
02836    struct ast_rtp_protocol *cur;
02837 
02838    AST_LIST_LOCK(&protos);
02839    AST_LIST_TRAVERSE(&protos, cur, list) {   
02840       if (!strcmp(cur->type, proto->type)) {
02841          ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
02842          AST_LIST_UNLOCK(&protos);
02843          return -1;
02844       }
02845    }
02846    AST_LIST_INSERT_HEAD(&protos, proto, list);
02847    AST_LIST_UNLOCK(&protos);
02848    
02849    return 0;
02850 }

void ast_rtp_proto_unregister ( struct ast_rtp_protocol proto  ) 

Unregister interface to channel driver.

Definition at line 2826 of file rtp.c.

References AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, and protos.

Referenced by load_module(), and unload_module().

02827 {
02828    AST_LIST_LOCK(&protos);
02829    AST_LIST_REMOVE(&protos, proto, list);
02830    AST_LIST_UNLOCK(&protos);
02831 }

void ast_rtp_pt_clear ( struct ast_rtp rtp  ) 

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

Definition at line 1400 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().

01401 {
01402    int i;
01403 
01404    if (!rtp)
01405       return;
01406 
01407    ast_mutex_lock(&rtp->bridge_lock);
01408 
01409    for (i = 0; i < MAX_RTP_PT; ++i) {
01410       rtp->current_RTP_PT[i].isAstFormat = 0;
01411       rtp->current_RTP_PT[i].code = 0;
01412    }
01413 
01414    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01415    rtp->rtp_lookup_code_cache_code = 0;
01416    rtp->rtp_lookup_code_cache_result = 0;
01417 
01418    ast_mutex_unlock(&rtp->bridge_lock);
01419 }

void ast_rtp_pt_copy ( struct ast_rtp dest,
struct ast_rtp src 
)

Copy payload types between RTP structures.

Definition at line 1440 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().

01441 {
01442    unsigned int i;
01443 
01444    ast_mutex_lock(&dest->bridge_lock);
01445    ast_mutex_lock(&src->bridge_lock);
01446 
01447    for (i=0; i < MAX_RTP_PT; ++i) {
01448       dest->current_RTP_PT[i].isAstFormat = 
01449          src->current_RTP_PT[i].isAstFormat;
01450       dest->current_RTP_PT[i].code = 
01451          src->current_RTP_PT[i].code; 
01452    }
01453    dest->rtp_lookup_code_cache_isAstFormat = 0;
01454    dest->rtp_lookup_code_cache_code = 0;
01455    dest->rtp_lookup_code_cache_result = 0;
01456 
01457    ast_mutex_unlock(&src->bridge_lock);
01458    ast_mutex_unlock(&dest->bridge_lock);
01459 }

void ast_rtp_pt_default ( struct ast_rtp rtp  ) 

Set payload types to defaults.

Definition at line 1421 of file rtp.c.

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

Referenced by ast_rtp_new_with_bindaddr().

01422 {
01423    int i;
01424 
01425    ast_mutex_lock(&rtp->bridge_lock);
01426 
01427    /* Initialize to default payload types */
01428    for (i = 0; i < MAX_RTP_PT; ++i) {
01429       rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
01430       rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
01431    }
01432 
01433    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01434    rtp->rtp_lookup_code_cache_code = 0;
01435    rtp->rtp_lookup_code_cache_result = 0;
01436 
01437    ast_mutex_unlock(&rtp->bridge_lock);
01438 }

struct ast_frame* ast_rtp_read ( struct ast_rtp rtp  ) 

Definition at line 1105 of file rtp.c.

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

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

01106 {
01107    int res;
01108    struct sockaddr_in sin;
01109    socklen_t len;
01110    unsigned int seqno;
01111    int version;
01112    int payloadtype;
01113    int hdrlen = 12;
01114    int padding;
01115    int mark;
01116    int ext;
01117    int cc;
01118    unsigned int ssrc;
01119    unsigned int timestamp;
01120    unsigned int *rtpheader;
01121    struct rtpPayloadType rtpPT;
01122    struct ast_rtp *bridged = NULL;
01123    
01124    /* If time is up, kill it */
01125    if (rtp->sending_digit)
01126       ast_rtp_senddigit_continuation(rtp);
01127 
01128    len = sizeof(sin);
01129    
01130    /* Cache where the header will go */
01131    res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
01132                0, (struct sockaddr *)&sin, &len);
01133 
01134    rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
01135    if (res < 0) {
01136       ast_assert(errno != EBADF);
01137       if (errno != EAGAIN) {
01138          ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up.\n", strerror(errno));
01139          return NULL;
01140       }
01141       return &ast_null_frame;
01142    }
01143    
01144    if (res < hdrlen) {
01145       ast_log(LOG_WARNING, "RTP Read too short\n");
01146       return &ast_null_frame;
01147    }
01148 
01149    /* Get fields */
01150    seqno = ntohl(rtpheader[0]);
01151 
01152    /* Check RTP version */
01153    version = (seqno & 0xC0000000) >> 30;
01154    if (!version) {
01155       if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) &&
01156          (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
01157          memcpy(&rtp->them, &sin, sizeof(rtp->them));
01158       }
01159       return &ast_null_frame;
01160    }
01161 
01162 #if 0 /* Allow to receive RTP stream with closed transmission path */
01163    /* If we don't have the other side's address, then ignore this */
01164    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
01165       return &ast_null_frame;
01166 #endif
01167 
01168    /* Send to whoever send to us if NAT is turned on */
01169    if (rtp->nat) {
01170       if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
01171           (rtp->them.sin_port != sin.sin_port)) {
01172          rtp->them = sin;
01173          if (rtp->rtcp) {
01174             memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
01175             rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
01176          }
01177          rtp->rxseqno = 0;
01178          ast_set_flag(rtp, FLAG_NAT_ACTIVE);
01179          if (option_debug || rtpdebug)
01180             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));
01181       }
01182    }
01183 
01184    /* If we are bridged to another RTP stream, send direct */
01185    if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
01186       return &ast_null_frame;
01187 
01188    if (version != 2)
01189       return &ast_null_frame;
01190 
01191    payloadtype = (seqno & 0x7f0000) >> 16;
01192    padding = seqno & (1 << 29);
01193    mark = seqno & (1 << 23);
01194    ext = seqno & (1 << 28);
01195    cc = (seqno & 0xF000000) >> 24;
01196    seqno &= 0xffff;
01197    timestamp = ntohl(rtpheader[1]);
01198    ssrc = ntohl(rtpheader[2]);
01199    
01200    if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
01201       if (option_debug || rtpdebug)
01202          ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
01203       mark = 1;
01204    }
01205 
01206    rtp->rxssrc = ssrc;
01207    
01208    if (padding) {
01209       /* Remove padding bytes */
01210       res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
01211    }
01212    
01213    if (cc) {
01214       /* CSRC fields present */
01215       hdrlen += cc*4;
01216    }
01217 
01218    if (ext) {
01219       /* RTP Extension present */
01220       hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
01221       hdrlen += 4;
01222    }
01223 
01224    if (res < hdrlen) {
01225       ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
01226       return &ast_null_frame;
01227    }
01228 
01229    rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
01230 
01231    if (rtp->rxcount==1) {
01232       /* This is the first RTP packet successfully received from source */
01233       rtp->seedrxseqno = seqno;
01234    }
01235 
01236    /* Do not schedule RR if RTCP isn't run */
01237    if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
01238       /* Schedule transmission of Receiver Report */
01239       rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
01240    }
01241    if ( (int)rtp->lastrxseqno - (int)seqno  > 100) /* if so it would indicate that the sender cycled; allow for misordering */
01242       rtp->cycles += RTP_SEQ_MOD;
01243 
01244    rtp->lastrxseqno = seqno;
01245    
01246    if (rtp->themssrc==0)
01247       rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
01248    
01249    if (rtp_debug_test_addr(&sin))
01250       ast_verbose("Got  RTP packet from    %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
01251          ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
01252 
01253    rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
01254    if (!rtpPT.isAstFormat) {
01255       struct ast_frame *f = NULL;
01256 
01257       /* This is special in-band data that's not one of our codecs */
01258       if (rtpPT.code == AST_RTP_DTMF) {
01259          /* It's special -- rfc2833 process it */
01260          if (rtp_debug_test_addr(&sin)) {
01261             unsigned char *data;
01262             unsigned int event;
01263             unsigned int event_end;
01264             unsigned int duration;
01265             data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
01266             event = ntohl(*((unsigned int *)(data)));
01267             event >>= 24;
01268             event_end = ntohl(*((unsigned int *)(data)));
01269             event_end <<= 8;
01270             event_end >>= 24;
01271             duration = ntohl(*((unsigned int *)(data)));
01272             duration &= 0xFFFF;
01273             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);
01274          }
01275          f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
01276       } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
01277          /* It's really special -- process it the Cisco way */
01278          if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
01279             f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01280             rtp->lastevent = seqno;
01281          }
01282       } else if (rtpPT.code == AST_RTP_CN) {
01283          /* Comfort Noise */
01284          f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01285       } else {
01286          ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
01287       }
01288       return f ? f : &ast_null_frame;
01289    }
01290    rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
01291    rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
01292 
01293    if (!rtp->lastrxts)
01294       rtp->lastrxts = timestamp;
01295 
01296    rtp->rxseqno = seqno;
01297 
01298    /* Record received timestamp as last received now */
01299    rtp->lastrxts = timestamp;
01300 
01301    rtp->f.mallocd = 0;
01302    rtp->f.datalen = res - hdrlen;
01303    rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
01304    rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
01305    rtp->f.seqno = seqno;
01306    if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
01307       rtp->f.samples = ast_codec_get_samples(&rtp->f);
01308       if (rtp->f.subclass == AST_FORMAT_SLINEAR) 
01309          ast_frame_byteswap_be(&rtp->f);
01310       calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
01311       /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
01312       ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
01313       rtp->f.ts = timestamp / 8;
01314       rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000);
01315    } else {
01316       /* Video -- samples is # of samples vs. 90000 */
01317       if (!rtp->lastividtimestamp)
01318          rtp->lastividtimestamp = timestamp;
01319       rtp->f.samples = timestamp - rtp->lastividtimestamp;
01320       rtp->lastividtimestamp = timestamp;
01321       rtp->f.delivery.tv_sec = 0;
01322       rtp->f.delivery.tv_usec = 0;
01323       if (mark)
01324          rtp->f.subclass |= 0x1;
01325       
01326    }
01327    rtp->f.src = "RTP";
01328    return &rtp->f;
01329 }

int ast_rtp_reload ( void   ) 

Definition at line 3752 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().

03753 {
03754    struct ast_config *cfg;
03755    const char *s;
03756 
03757    rtpstart = 5000;
03758    rtpend = 31000;
03759    dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03760    cfg = ast_config_load("rtp.conf");
03761    if (cfg) {
03762       if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
03763          rtpstart = atoi(s);
03764          if (rtpstart < 1024)
03765             rtpstart = 1024;
03766          if (rtpstart > 65535)
03767             rtpstart = 65535;
03768       }
03769       if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
03770          rtpend = atoi(s);
03771          if (rtpend < 1024)
03772             rtpend = 1024;
03773          if (rtpend > 65535)
03774             rtpend = 65535;
03775       }
03776       if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
03777          rtcpinterval = atoi(s);
03778          if (rtcpinterval == 0)
03779             rtcpinterval = 0; /* Just so we're clear... it's zero */
03780          if (rtcpinterval < RTCP_MIN_INTERVALMS)
03781             rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
03782          if (rtcpinterval > RTCP_MAX_INTERVALMS)
03783             rtcpinterval = RTCP_MAX_INTERVALMS;
03784       }
03785       if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
03786 #ifdef SO_NO_CHECK
03787          if (ast_false(s))
03788             nochecksums = 1;
03789          else
03790             nochecksums = 0;
03791 #else
03792          if (ast_false(s))
03793             ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
03794 #endif
03795       }
03796       if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
03797          dtmftimeout = atoi(s);
03798          if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
03799             ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
03800                dtmftimeout, DEFAULT_DTMF_TIMEOUT);
03801             dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03802          };
03803       }
03804       ast_config_destroy(cfg);
03805    }
03806    if (rtpstart >= rtpend) {
03807       ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
03808       rtpstart = 5000;
03809       rtpend = 31000;
03810    }
03811    if (option_verbose > 1)
03812       ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
03813    return 0;
03814 }

void ast_rtp_reset ( struct ast_rtp rtp  ) 

Definition at line 2060 of file rtp.c.

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

02061 {
02062    memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
02063    memset(&rtp->txcore, 0, sizeof(rtp->txcore));
02064    memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
02065    rtp->lastts = 0;
02066    rtp->lastdigitts = 0;
02067    rtp->lastrxts = 0;
02068    rtp->lastividtimestamp = 0;
02069    rtp->lastovidtimestamp = 0;
02070    rtp->lasteventseqn = 0;
02071    rtp->lastevent = 0;
02072    rtp->lasttxformat = 0;
02073    rtp->lastrxformat = 0;
02074    rtp->dtmfcount = 0;
02075    rtp->dtmfsamples = 0;
02076    rtp->seqno = 0;
02077    rtp->rxseqno = 0;
02078 }

int ast_rtp_sendcng ( struct ast_rtp rtp,
int  level 
)

generate comfort noice (CNG)

Definition at line 2575 of file rtp.c.

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

Referenced by do_monitor().

02576 {
02577    unsigned int *rtpheader;
02578    int hdrlen = 12;
02579    int res;
02580    int payload;
02581    char data[256];
02582    level = 127 - (level & 0x7f);
02583    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
02584 
02585    /* If we have no peer, return immediately */ 
02586    if (!rtp->them.sin_addr.s_addr)
02587       return 0;
02588 
02589    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02590 
02591    /* Get a pointer to the header */
02592    rtpheader = (unsigned int *)data;
02593    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
02594    rtpheader[1] = htonl(rtp->lastts);
02595    rtpheader[2] = htonl(rtp->ssrc); 
02596    data[12] = level;
02597    if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
02598       res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
02599       if (res <0) 
02600          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));
02601       if (rtp_debug_test_addr(&rtp->them))
02602          ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
02603                , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);         
02604          
02605    }
02606    return 0;
02607 }

int ast_rtp_senddigit_begin ( struct ast_rtp rtp,
char  digit 
)

Send begin frames for DTMF.

Definition at line 2183 of file rtp.c.

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

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

02184 {
02185    unsigned int *rtpheader;
02186    int hdrlen = 12, res = 0, i = 0, payload = 0;
02187    char data[256];
02188 
02189    if ((digit <= '9') && (digit >= '0'))
02190       digit -= '0';
02191    else if (digit == '*')
02192       digit = 10;
02193    else if (digit == '#')
02194       digit = 11;
02195    else if ((digit >= 'A') && (digit <= 'D'))
02196       digit = digit - 'A' + 12;
02197    else if ((digit >= 'a') && (digit <= 'd'))
02198       digit = digit - 'a' + 12;
02199    else {
02200       ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
02201       return 0;
02202    }
02203 
02204    /* If we have no peer, return immediately */ 
02205    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
02206       return 0;
02207 
02208    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
02209 
02210    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02211    rtp->send_duration = 160;
02212    
02213    /* Get a pointer to the header */
02214    rtpheader = (unsigned int *)data;
02215    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
02216    rtpheader[1] = htonl(rtp->lastdigitts);
02217    rtpheader[2] = htonl(rtp->ssrc); 
02218 
02219    for (i = 0; i < 2; i++) {
02220       rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
02221       res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
02222       if (res < 0) 
02223          ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
02224             ast_inet_ntoa(rtp->them.sin_addr),
02225             ntohs(rtp->them.sin_port), strerror(errno));
02226       if (rtp_debug_test_addr(&rtp->them))
02227          ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
02228                 ast_inet_ntoa(rtp->them.sin_addr),
02229                 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
02230       /* Increment sequence number */
02231       rtp->seqno++;
02232       /* Increment duration */
02233       rtp->send_duration += 160;
02234       /* Clear marker bit and set seqno */
02235       rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
02236    }
02237 
02238    /* Since we received a begin, we can safely store the digit and disable any compensation */
02239    rtp->sending_digit = 1;
02240    rtp->send_digit = digit;
02241    rtp->send_payload = payload;
02242 
02243    return 0;
02244 }

int ast_rtp_senddigit_end ( struct ast_rtp rtp,
char  digit 
)

void ast_rtp_set_callback ( struct ast_rtp rtp,
ast_rtp_callback  callback 
)

Definition at line 586 of file rtp.c.

References ast_rtp::callback.

Referenced by start_rtp().

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

void ast_rtp_set_data ( struct ast_rtp rtp,
void *  data 
)

Definition at line 581 of file rtp.c.

References ast_rtp::data.

Referenced by start_rtp().

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

void ast_rtp_set_m_type ( struct ast_rtp rtp,
int  pt 
)

Activate payload type.

Definition at line 1639 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().

01640 {
01641    if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 
01642       return; /* bogus payload type */
01643 
01644    ast_mutex_lock(&rtp->bridge_lock);
01645    rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
01646    ast_mutex_unlock(&rtp->bridge_lock);
01647 } 

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

Definition at line 2006 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().

02007 {
02008    rtp->them.sin_port = them->sin_port;
02009    rtp->them.sin_addr = them->sin_addr;
02010    if (rtp->rtcp) {
02011       rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
02012       rtp->rtcp->them.sin_addr = them->sin_addr;
02013    }
02014    rtp->rxseqno = 0;
02015 }

void ast_rtp_set_rtpholdtimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp hold timeout.

Definition at line 548 of file rtp.c.

References ast_rtp::rtpholdtimeout.

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

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

void ast_rtp_set_rtpkeepalive ( struct ast_rtp rtp,
int  period 
)

set RTP keepalive interval

Definition at line 554 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by create_addr_from_peer(), and sip_alloc().

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

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

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

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

Definition at line 1666 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().

01669 {
01670    unsigned int i;
01671    int found = 0;
01672 
01673    if (pt < 0 || pt > MAX_RTP_PT) 
01674       return -1; /* bogus payload type */
01675    
01676    ast_mutex_lock(&rtp->bridge_lock);
01677 
01678    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01679       if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
01680           strcasecmp(mimeType, mimeTypes[i].type) == 0) {
01681          found = 1;
01682          rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
01683          if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
01684              mimeTypes[i].payloadType.isAstFormat &&
01685              (options & AST_RTP_OPT_G726_NONSTANDARD))
01686             rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
01687          break;
01688       }
01689    }
01690 
01691    ast_mutex_unlock(&rtp->bridge_lock);
01692 
01693    return (found ? 0 : -1);
01694 } 

void ast_rtp_set_rtptimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp timeout.

Definition at line 542 of file rtp.c.

References ast_rtp::rtptimeout.

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

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

void ast_rtp_set_rtptimers_onhold ( struct ast_rtp rtp  ) 

Definition at line 535 of file rtp.c.

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

Referenced by handle_response_invite().

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

void ast_rtp_setdtmf ( struct ast_rtp rtp,
int  dtmf 
)

Indicate whether this RTP session is carrying DTMF or not.

Definition at line 601 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_DTMF.

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

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

void ast_rtp_setdtmfcompensate ( struct ast_rtp rtp,
int  compensate 
)

Compensate for devices that send RFC2833 packets all at once.

Definition at line 606 of file rtp.c.

References ast_set2_flag, and FLAG_DTMF_COMPENSATE.

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

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

void ast_rtp_setnat ( struct ast_rtp rtp,
int  nat 
)

Definition at line 591 of file rtp.c.

References ast_rtp::nat.

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

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

void ast_rtp_setstun ( struct ast_rtp rtp,
int  stun_enable 
)

Enable STUN capability.

Definition at line 611 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_STUN.

Referenced by gtalk_new().

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

int ast_rtp_settos ( struct ast_rtp rtp,
int  tos 
)

Definition at line 1991 of file rtp.c.

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

Referenced by __oh323_rtp_create(), and sip_alloc().

01992 {
01993    int res;
01994 
01995    if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
01996       ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
01997    return res;
01998 }

void ast_rtp_stop ( struct ast_rtp rtp  ) 

Definition at line 2046 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().

02047 {
02048    AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
02049 
02050    memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
02051    memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
02052    if (rtp->rtcp) {
02053       memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
02054       memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
02055    }
02056    
02057    ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
02058 }

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

Definition at line 403 of file rtp.c.

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

Referenced by gtalk_update_stun().

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

void ast_rtp_unset_m_type ( struct ast_rtp rtp,
int  pt 
)

clear payload type

Definition at line 1651 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().

01652 {
01653    if (pt < 0 || pt > MAX_RTP_PT)
01654       return; /* bogus payload type */
01655 
01656    ast_mutex_lock(&rtp->bridge_lock);
01657    rtp->current_RTP_PT[pt].isAstFormat = 0;
01658    rtp->current_RTP_PT[pt].code = 0;
01659    ast_mutex_unlock(&rtp->bridge_lock);
01660 }

int ast_rtp_write ( struct ast_rtp rtp,
struct ast_frame f 
)

Definition at line 2734 of file rtp.c.

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

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

02735 {
02736    struct ast_frame *f;
02737    int codec;
02738    int hdrlen = 12;
02739    int subclass;
02740    
02741 
02742    /* If we have no peer, return immediately */ 
02743    if (!rtp->them.sin_addr.s_addr)
02744       return 0;
02745 
02746    /* If there is no data length, return immediately */
02747    if (!_f->datalen) 
02748       return 0;
02749    
02750    /* Make sure we have enough space for RTP header */
02751    if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
02752       ast_log(LOG_WARNING, "RTP can only send voice and video\n");
02753       return -1;
02754    }
02755 
02756    subclass = _f->subclass;
02757    if (_f->frametype == AST_FRAME_VIDEO)
02758       subclass &= ~0x1;
02759 
02760    codec = ast_rtp_lookup_code(rtp, 1, subclass);
02761    if (codec < 0) {
02762       ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
02763       return -1;
02764    }
02765 
02766    if (rtp->lasttxformat != subclass) {
02767       /* New format, reset the smoother */
02768       if (option_debug)
02769          ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
02770       rtp->lasttxformat = subclass;
02771       if (rtp->smoother)
02772          ast_smoother_free(rtp->smoother);
02773       rtp->smoother = NULL;
02774    }
02775 
02776    if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) {
02777       struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
02778       if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
02779          if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
02780             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));
02781             return -1;
02782          }
02783          if (fmt.flags)
02784             ast_smoother_set_flags(rtp->smoother, fmt.flags);
02785          if (option_debug)
02786             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));
02787       }
02788    }
02789    if (rtp->smoother) {
02790       if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
02791          ast_smoother_feed_be(rtp->smoother, _f);
02792       } else {
02793          ast_smoother_feed(rtp->smoother, _f);
02794       }
02795 
02796       while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) {
02797          if (f->subclass == AST_FORMAT_G722) {
02798             /* G.722 is silllllllllllllly */
02799             f->samples /= 2;
02800          }
02801 
02802          ast_rtp_raw_write(rtp, f, codec);
02803       }
02804    } else {
02805       /* Don't buffer outgoing frames; send them one-per-packet: */
02806       if (_f->offset < hdrlen) {
02807          f = ast_frdup(_f);
02808       } else {
02809          f = _f;
02810       }
02811       if (f->data) {
02812          if (f->subclass == AST_FORMAT_G722) {
02813             /* G.722 is silllllllllllllly */
02814             f->samples /= 2;
02815          }
02816          ast_rtp_raw_write(rtp, f, codec);
02817       }
02818       if (f != _f)
02819          ast_frfree(f);
02820    }
02821       
02822    return 0;
02823 }


Generated on Sat Jul 26 06:14:29 2008 for Asterisk - the Open Source PBX by  doxygen 1.5.1