Fri Sep 25 19:28:44 2009

Asterisk developer's documentation


rtp.h File Reference

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

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

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)
struct 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)
struct 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)
struct 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):.
struct 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)
struct 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.
struct 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.
struct 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)

#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


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

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 };

Enumerator:
AST_RTP_OPT_G726_NONSTANDARD 

Definition at line 53 of file rtp.h.

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


Function Documentation

int ast_rtcp_fd ( struct ast_rtp rtp  ) 

Definition at line 517 of file rtp.c.

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

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

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

struct ast_frame* ast_rtcp_read ( struct ast_rtp rtp  )  [read]

Definition at line 826 of file rtp.c.

References ast_rtcp::accumulated_transit, AST_CONTROL_VIDUPDATE, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), CRASH, ast_frame::datalen, 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().

00827 {
00828    socklen_t len;
00829    int position, i, packetwords;
00830    int res;
00831    struct sockaddr_in sin;
00832    unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
00833    unsigned int *rtcpheader;
00834    int pt;
00835    struct timeval now;
00836    unsigned int length;
00837    int rc;
00838    double rttsec;
00839    uint64_t rtt = 0;
00840    unsigned int dlsr;
00841    unsigned int lsr;
00842    unsigned int msw;
00843    unsigned int lsw;
00844    unsigned int comp;
00845    struct ast_frame *f = &ast_null_frame;
00846    
00847    if (!rtp || !rtp->rtcp)
00848       return &ast_null_frame;
00849 
00850    len = sizeof(sin);
00851    
00852    res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
00853                0, (struct sockaddr *)&sin, &len);
00854    rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
00855    
00856    if (res < 0) {
00857       if (errno == EBADF)
00858          CRASH;
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 2333 of file rtp.c.

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

02334 {
02335    struct ast_rtp *rtp = data;
02336    int res;
02337 
02338    rtp->rtcp->sendfur = 1;
02339    res = ast_rtcp_write(data);
02340    
02341    return res;
02342 }

size_t ast_rtp_alloc_size ( void   ) 

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

Returns:
number of bytes required

Definition at line 397 of file rtp.c.

Referenced by process_sdp().

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

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

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

Definition at line 3253 of file rtp.c.

References AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_codec_pref_getsize(), ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_set_flag, ast_test_flag, ast_verbose(), bridge_native_loop(), bridge_p2p_loop(), ast_format_list::cur_ms, FLAG_DTMF_COMPENSATE, FLAG_HAS_DTMF, FLAG_P2P_NEED_DTMF, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, LOG_DEBUG, 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.

03254 {
03255    struct ast_rtp *p0 = NULL, *p1 = NULL;    /* Audio RTP Channels */
03256    struct ast_rtp *vp0 = NULL, *vp1 = NULL;  /* Video RTP channels */
03257    struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
03258    enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
03259    enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
03260    enum ast_bridge_result res = AST_BRIDGE_FAILED;
03261    int codec0 = 0, codec1 = 0;
03262    void *pvt0 = NULL, *pvt1 = NULL;
03263 
03264    /* Lock channels */
03265    ast_channel_lock(c0);
03266    while(ast_channel_trylock(c1)) {
03267       ast_channel_unlock(c0);
03268       usleep(1);
03269       ast_channel_lock(c0);
03270    }
03271 
03272    /* Find channel driver interfaces */
03273    if (!(pr0 = get_proto(c0))) {
03274       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
03275       ast_channel_unlock(c0);
03276       ast_channel_unlock(c1);
03277       return AST_BRIDGE_FAILED;
03278    }
03279    if (!(pr1 = get_proto(c1))) {
03280       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
03281       ast_channel_unlock(c0);
03282       ast_channel_unlock(c1);
03283       return AST_BRIDGE_FAILED;
03284    }
03285 
03286    /* Get channel specific interface structures */
03287    pvt0 = c0->tech_pvt;
03288    pvt1 = c1->tech_pvt;
03289 
03290    /* Get audio and video interface (if native bridge is possible) */
03291    audio_p0_res = pr0->get_rtp_info(c0, &p0);
03292    video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
03293    audio_p1_res = pr1->get_rtp_info(c1, &p1);
03294    video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
03295 
03296    /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
03297    if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
03298       audio_p0_res = AST_RTP_GET_FAILED;
03299    if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
03300       audio_p1_res = AST_RTP_GET_FAILED;
03301 
03302    /* Check if a bridge is possible (partial/native) */
03303    if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
03304       /* Somebody doesn't want to play... */
03305       ast_channel_unlock(c0);
03306       ast_channel_unlock(c1);
03307       return AST_BRIDGE_FAILED_NOWARN;
03308    }
03309 
03310    /* If we need to feed DTMF frames into the core then only do a partial native bridge */
03311    if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
03312       ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
03313       audio_p0_res = AST_RTP_TRY_PARTIAL;
03314    }
03315 
03316    if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
03317       ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
03318       audio_p1_res = AST_RTP_TRY_PARTIAL;
03319    }
03320 
03321    /* If both sides are not using the same method of DTMF transmission 
03322     * (ie: one is RFC2833, other is INFO... then we can not do direct media. 
03323     * --------------------------------------------------
03324     * | DTMF Mode |  HAS_DTMF  |  Accepts Begin Frames |
03325     * |-----------|------------|-----------------------|
03326     * | Inband    | False      | True                  |
03327     * | RFC2833   | True       | True                  |
03328     * | SIP INFO  | False      | False                 |
03329     * --------------------------------------------------
03330     * However, if DTMF from both channels is being monitored by the core, then
03331     * we can still do packet-to-packet bridging, because passing through the 
03332     * core will handle DTMF mode translation.
03333     */
03334    if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
03335        (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
03336       if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
03337          ast_channel_unlock(c0);
03338          ast_channel_unlock(c1);
03339          return AST_BRIDGE_FAILED_NOWARN;
03340       }
03341       audio_p0_res = AST_RTP_TRY_PARTIAL;
03342       audio_p1_res = AST_RTP_TRY_PARTIAL;
03343    }
03344 
03345    /* If the core will need to compensate and the P2P bridge will need to feed up DTMF frames then we can not reliably do so yet, so do not P2P bridge */
03346    if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF) && ast_test_flag(p0, FLAG_DTMF_COMPENSATE)) ||
03347        (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF) && ast_test_flag(p1, FLAG_DTMF_COMPENSATE))) {
03348       ast_channel_unlock(c0);
03349       ast_channel_unlock(c1);
03350       return AST_BRIDGE_FAILED_NOWARN;
03351    }
03352 
03353    /* Get codecs from both sides */
03354    codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
03355    codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
03356    if (codec0 && codec1 && !(codec0 & codec1)) {
03357       /* Hey, we can't do native bridging if both parties speak different codecs */
03358       if (option_debug)
03359          ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
03360       ast_channel_unlock(c0);
03361       ast_channel_unlock(c1);
03362       return AST_BRIDGE_FAILED_NOWARN;
03363    }
03364 
03365    /* If either side can only do a partial bridge, then don't try for a true native bridge */
03366    if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
03367       struct ast_format_list fmt0, fmt1;
03368 
03369       /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
03370       if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
03371          if (option_debug)
03372             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n");
03373          ast_channel_unlock(c0);
03374          ast_channel_unlock(c1);
03375          return AST_BRIDGE_FAILED_NOWARN;
03376       }
03377       /* They must also be using the same packetization */
03378       fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
03379       fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
03380       if (fmt0.cur_ms != fmt1.cur_ms) {
03381          if (option_debug)
03382             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n");
03383          ast_channel_unlock(c0);
03384          ast_channel_unlock(c1);
03385          return AST_BRIDGE_FAILED_NOWARN;
03386       }
03387 
03388       if (option_verbose > 2)
03389          ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
03390       res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
03391    } else {
03392       if (option_verbose > 2) 
03393          ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
03394       res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
03395    }
03396 
03397    return res;
03398 }

int ast_rtp_codec_getformat ( int  pt  ) 

Definition at line 2716 of file rtp.c.

References rtpPayloadType::code, and MAX_RTP_PT.

Referenced by process_sdp().

02717 {
02718    if (pt < 0 || pt > MAX_RTP_PT)
02719       return 0; /* bogus payload type */
02720 
02721    if (static_RTP_PT[pt].isAstFormat)
02722       return static_RTP_PT[pt].code;
02723    else
02724       return 0;
02725 }

struct ast_codec_pref* ast_rtp_codec_getpref ( struct ast_rtp rtp  )  [read]

Definition at line 2711 of file rtp.c.

References ast_rtp::pref.

Referenced by add_codec_to_sdp(), and process_sdp().

02712 {
02713    return &rtp->pref;
02714 }

int ast_rtp_codec_setpref ( struct ast_rtp rtp,
struct ast_codec_pref prefs 
)

Definition at line 2698 of file rtp.c.

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

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

02699 {
02700    int x;
02701    for (x = 0; x < 32; x++) {  /* Ugly way */
02702       rtp->pref.order[x] = prefs->order[x];
02703       rtp->pref.framing[x] = prefs->framing[x];
02704    }
02705    if (rtp->smoother)
02706       ast_smoother_free(rtp->smoother);
02707    rtp->smoother = NULL;
02708    return 0;
02709 }

void ast_rtp_destroy ( struct ast_rtp rtp  ) 

Definition at line 2115 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().

02116 {
02117    if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
02118       /*Print some info on the call here */
02119       ast_verbose("  RTP-stats\n");
02120       ast_verbose("* Our Receiver:\n");
02121       ast_verbose("  SSRC:     %u\n", rtp->themssrc);
02122       ast_verbose("  Received packets: %u\n", rtp->rxcount);
02123       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
02124       ast_verbose("  Jitter:      %.4f\n", rtp->rxjitter);
02125       ast_verbose("  Transit:     %.4f\n", rtp->rxtransit);
02126       ast_verbose("  RR-count:    %u\n", rtp->rtcp->rr_count);
02127       ast_verbose("* Our Sender:\n");
02128       ast_verbose("  SSRC:     %u\n", rtp->ssrc);
02129       ast_verbose("  Sent packets:   %u\n", rtp->txcount);
02130       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->reported_lost);
02131       ast_verbose("  Jitter:      %u\n", rtp->rtcp->reported_jitter / (unsigned int)65536.0);
02132       ast_verbose("  SR-count:    %u\n", rtp->rtcp->sr_count);
02133       ast_verbose("  RTT:      %f\n", rtp->rtcp->rtt);
02134    }
02135 
02136    if (rtp->smoother)
02137       ast_smoother_free(rtp->smoother);
02138    if (rtp->ioid)
02139       ast_io_remove(rtp->io, rtp->ioid);
02140    if (rtp->s > -1)
02141       close(rtp->s);
02142    if (rtp->rtcp) {
02143       if (rtp->rtcp->schedid > 0)
02144          ast_sched_del(rtp->sched, rtp->rtcp->schedid);
02145       close(rtp->rtcp->s);
02146       free(rtp->rtcp);
02147       rtp->rtcp=NULL;
02148    }
02149 
02150    ast_mutex_destroy(&rtp->bridge_lock);
02151 
02152    free(rtp);
02153 }

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 1471 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().

01472 {
01473    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01474    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01475    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01476    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01477    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
01478    int srccodec, destcodec, nat_active = 0;
01479 
01480    /* Lock channels */
01481    ast_channel_lock(dest);
01482    if (src) {
01483       while(ast_channel_trylock(src)) {
01484          ast_channel_unlock(dest);
01485          usleep(1);
01486          ast_channel_lock(dest);
01487       }
01488    }
01489 
01490    /* Find channel driver interfaces */
01491    destpr = get_proto(dest);
01492    if (src)
01493       srcpr = get_proto(src);
01494    if (!destpr) {
01495       if (option_debug)
01496          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01497       ast_channel_unlock(dest);
01498       if (src)
01499          ast_channel_unlock(src);
01500       return 0;
01501    }
01502    if (!srcpr) {
01503       if (option_debug)
01504          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
01505       ast_channel_unlock(dest);
01506       if (src)
01507          ast_channel_unlock(src);
01508       return 0;
01509    }
01510 
01511    /* Get audio and video interface (if native bridge is possible) */
01512    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01513    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01514    if (srcpr) {
01515       audio_src_res = srcpr->get_rtp_info(src, &srcp);
01516       video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01517    }
01518 
01519    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01520    if (audio_dest_res != AST_RTP_TRY_NATIVE) {
01521       /* Somebody doesn't want to play... */
01522       ast_channel_unlock(dest);
01523       if (src)
01524          ast_channel_unlock(src);
01525       return 0;
01526    }
01527    if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
01528       srccodec = srcpr->get_codec(src);
01529    else
01530       srccodec = 0;
01531    if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
01532       destcodec = destpr->get_codec(dest);
01533    else
01534       destcodec = 0;
01535    /* Ensure we have at least one matching codec */
01536    if (!(srccodec & destcodec)) {
01537       ast_channel_unlock(dest);
01538       if (src)
01539          ast_channel_unlock(src);
01540       return 0;
01541    }
01542    /* Consider empty media as non-existant */
01543    if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
01544       srcp = NULL;
01545    /* If the client has NAT stuff turned on then just safe NAT is active */
01546    if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01547       nat_active = 1;
01548    /* Bridge media early */
01549    if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active))
01550       ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>");
01551    ast_channel_unlock(dest);
01552    if (src)
01553       ast_channel_unlock(src);
01554    if (option_debug)
01555       ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
01556    return 1;
01557 }

int ast_rtp_fd ( struct ast_rtp rtp  ) 

Definition at line 512 of file rtp.c.

References ast_rtp::s.

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

00513 {
00514    return rtp->s;
00515 }

struct ast_rtp* ast_rtp_get_bridged ( struct ast_rtp rtp  )  [read]

Definition at line 2023 of file rtp.c.

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

Referenced by ast_rtp_read().

02024 {
02025    struct ast_rtp *bridged = NULL;
02026 
02027    ast_mutex_lock(&rtp->bridge_lock);
02028    bridged = rtp->bridged;
02029    ast_mutex_unlock(&rtp->bridge_lock);
02030 
02031    return bridged;
02032 }

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 1693 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().

01695 {
01696    int pt;
01697    
01698    ast_mutex_lock(&rtp->bridge_lock);
01699    
01700    *astFormats = *nonAstFormats = 0;
01701    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01702       if (rtp->current_RTP_PT[pt].isAstFormat) {
01703          *astFormats |= rtp->current_RTP_PT[pt].code;
01704       } else {
01705          *nonAstFormats |= rtp->current_RTP_PT[pt].code;
01706       }
01707    }
01708    
01709    ast_mutex_unlock(&rtp->bridge_lock);
01710    
01711    return;
01712 }

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

Definition at line 2005 of file rtp.c.

References ast_rtp::them.

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

02006 {
02007    if ((them->sin_family != AF_INET) ||
02008       (them->sin_port != rtp->them.sin_port) ||
02009       (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
02010       them->sin_family = AF_INET;
02011       them->sin_port = rtp->them.sin_port;
02012       them->sin_addr = rtp->them.sin_addr;
02013       return 1;
02014    }
02015    return 0;
02016 }

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

Return RTCP quality string.

Definition at line 2071 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().

02072 {
02073    /*
02074    *ssrc          our ssrc
02075    *themssrc      their ssrc
02076    *lp            lost packets
02077    *rxjitter      our calculated jitter(rx)
02078    *rxcount       no. received packets
02079    *txjitter      reported jitter of the other end
02080    *txcount       transmitted packets
02081    *rlp           remote lost packets
02082    *rtt           round trip time
02083    */
02084 
02085    if (qual && rtp) {
02086       qual->local_ssrc = rtp->ssrc;
02087       qual->local_jitter = rtp->rxjitter;
02088       qual->local_count = rtp->rxcount;
02089       qual->remote_ssrc = rtp->themssrc;
02090       qual->remote_count = rtp->txcount;
02091       if (rtp->rtcp) {
02092          qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
02093          qual->remote_lostpackets = rtp->rtcp->reported_lost;
02094          qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
02095          qual->rtt = rtp->rtcp->rtt;
02096       }
02097    }
02098    if (rtp->rtcp) {
02099       snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality),
02100          "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
02101          rtp->ssrc,
02102          rtp->themssrc,
02103          rtp->rtcp->expected_prior - rtp->rtcp->received_prior,
02104          rtp->rxjitter,
02105          rtp->rxcount,
02106          (double)rtp->rtcp->reported_jitter / 65536.0,
02107          rtp->txcount,
02108          rtp->rtcp->reported_lost,
02109          rtp->rtcp->rtt);
02110       return rtp->rtcp->quality;
02111    } else
02112       return "<Unknown> - RTP/RTCP has already been destroyed";
02113 }

int ast_rtp_get_rtpholdtimeout ( struct ast_rtp rtp  ) 

Get rtp hold timeout.

Definition at line 567 of file rtp.c.

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

Referenced by do_monitor().

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

int ast_rtp_get_rtpkeepalive ( struct ast_rtp rtp  ) 

Get RTP keepalive interval.

Definition at line 575 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by do_monitor().

00576 {
00577    return rtp->rtpkeepalive;
00578 }

int ast_rtp_get_rtptimeout ( struct ast_rtp rtp  ) 

Get rtp timeout.

Definition at line 559 of file rtp.c.

References ast_rtp::rtptimeout.

Referenced by do_monitor().

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

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

Definition at line 2018 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().

02019 {
02020    *us = rtp->us;
02021 }

int ast_rtp_getnat ( struct ast_rtp rtp  ) 

Definition at line 595 of file rtp.c.

References ast_test_flag, and FLAG_NAT_ACTIVE.

Referenced by sip_get_rtp_peer().

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

void ast_rtp_init ( void   ) 

Initialize the RTP system in Asterisk.

Definition at line 3783 of file rtp.c.

References ast_cli_register_multiple(), and ast_rtp_reload().

Referenced by main().

03784 {
03785    ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
03786    ast_rtp_reload();
03787 }

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 1736 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().

01737 {
01738    int pt = 0;
01739 
01740    ast_mutex_lock(&rtp->bridge_lock);
01741 
01742    if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
01743       code == rtp->rtp_lookup_code_cache_code) {
01744       /* Use our cached mapping, to avoid the overhead of the loop below */
01745       pt = rtp->rtp_lookup_code_cache_result;
01746       ast_mutex_unlock(&rtp->bridge_lock);
01747       return pt;
01748    }
01749 
01750    /* Check the dynamic list first */
01751    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01752       if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
01753          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01754          rtp->rtp_lookup_code_cache_code = code;
01755          rtp->rtp_lookup_code_cache_result = pt;
01756          ast_mutex_unlock(&rtp->bridge_lock);
01757          return pt;
01758       }
01759    }
01760 
01761    /* Then the static list */
01762    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01763       if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
01764          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01765          rtp->rtp_lookup_code_cache_code = code;
01766          rtp->rtp_lookup_code_cache_result = pt;
01767          ast_mutex_unlock(&rtp->bridge_lock);
01768          return pt;
01769       }
01770    }
01771 
01772    ast_mutex_unlock(&rtp->bridge_lock);
01773 
01774    return -1;
01775 }

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 1796 of file rtp.c.

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

Referenced by process_sdp().

01798 {
01799    int format;
01800    unsigned len;
01801    char *end = buf;
01802    char *start = buf;
01803 
01804    if (!buf || !size)
01805       return NULL;
01806 
01807    snprintf(end, size, "0x%x (", capability);
01808 
01809    len = strlen(end);
01810    end += len;
01811    size -= len;
01812    start = end;
01813 
01814    for (format = 1; format < AST_RTP_MAX; format <<= 1) {
01815       if (capability & format) {
01816          const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
01817 
01818          snprintf(end, size, "%s|", name);
01819          len = strlen(end);
01820          end += len;
01821          size -= len;
01822       }
01823    }
01824 
01825    if (start == end)
01826       snprintf(start, size, "nothing)"); 
01827    else if (size > 1)
01828       *(end -1) = ')';
01829    
01830    return buf;
01831 }

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 1777 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().

01779 {
01780    unsigned int i;
01781 
01782    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01783       if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
01784          if (isAstFormat &&
01785              (code == AST_FORMAT_G726_AAL2) &&
01786              (options & AST_RTP_OPT_G726_NONSTANDARD))
01787             return "G726-32";
01788          else
01789             return mimeTypes[i].subtype;
01790       }
01791    }
01792 
01793    return "";
01794 }

struct rtpPayloadType ast_rtp_lookup_pt ( struct ast_rtp rtp,
int  pt 
) [read]

Mapping between RTP payload format codes and Asterisk codes:.

Definition at line 1714 of file rtp.c.

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

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

01715 {
01716    struct rtpPayloadType result;
01717 
01718    result.isAstFormat = result.code = 0;
01719 
01720    if (pt < 0 || pt > MAX_RTP_PT) 
01721       return result; /* bogus payload type */
01722 
01723    /* Start with negotiated codecs */
01724    ast_mutex_lock(&rtp->bridge_lock);
01725    result = rtp->current_RTP_PT[pt];
01726    ast_mutex_unlock(&rtp->bridge_lock);
01727 
01728    /* If it doesn't exist, check our static RTP type list, just in case */
01729    if (!result.code) 
01730       result = static_RTP_PT[pt];
01731 
01732    return result;
01733 }

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

Definition at line 1559 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().

01560 {
01561    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01562    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01563    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01564    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01565    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 
01566    int srccodec, destcodec;
01567 
01568    /* Lock channels */
01569    ast_channel_lock(dest);
01570    while(ast_channel_trylock(src)) {
01571       ast_channel_unlock(dest);
01572       usleep(1);
01573       ast_channel_lock(dest);
01574    }
01575 
01576    /* Find channel driver interfaces */
01577    if (!(destpr = get_proto(dest))) {
01578       if (option_debug)
01579          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01580       ast_channel_unlock(dest);
01581       ast_channel_unlock(src);
01582       return 0;
01583    }
01584    if (!(srcpr = get_proto(src))) {
01585       if (option_debug)
01586          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
01587       ast_channel_unlock(dest);
01588       ast_channel_unlock(src);
01589       return 0;
01590    }
01591 
01592    /* Get audio and video interface (if native bridge is possible) */
01593    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01594    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01595    audio_src_res = srcpr->get_rtp_info(src, &srcp);
01596    video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01597 
01598    /* Ensure we have at least one matching codec */
01599    if (srcpr->get_codec)
01600       srccodec = srcpr->get_codec(src);
01601    else
01602       srccodec = 0;
01603    if (destpr->get_codec)
01604       destcodec = destpr->get_codec(dest);
01605    else
01606       destcodec = 0;
01607 
01608    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01609    if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
01610       /* Somebody doesn't want to play... */
01611       ast_channel_unlock(dest);
01612       ast_channel_unlock(src);
01613       return 0;
01614    }
01615    ast_rtp_pt_copy(destp, srcp);
01616    if (vdestp && vsrcp)
01617       ast_rtp_pt_copy(vdestp, vsrcp);
01618    if (media) {
01619       /* Bridge early */
01620       if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01621          ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
01622    }
01623    ast_channel_unlock(dest);
01624    ast_channel_unlock(src);
01625    if (option_debug)
01626       ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
01627    return 1;
01628 }

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

Initializate a RTP session.

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

Definition at line 1977 of file rtp.c.

References ast_rtp_new_with_bindaddr().

01978 {
01979    struct in_addr ia;
01980 
01981    memset(&ia, 0, sizeof(ia));
01982    return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
01983 }

void ast_rtp_new_init ( struct ast_rtp rtp  ) 

Initialize a new RTP structure.

Definition at line 1877 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().

01878 {
01879    ast_mutex_init(&rtp->bridge_lock);
01880 
01881    rtp->them.sin_family = AF_INET;
01882    rtp->us.sin_family = AF_INET;
01883    rtp->ssrc = ast_random();
01884    rtp->seqno = ast_random() & 0xffff;
01885    ast_set_flag(rtp, FLAG_HAS_DTMF);
01886 
01887    return;
01888 }

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

Initializate a RTP session using an in_addr structure.

This fuction gets called by ast_rtp_new().

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

Definition at line 1890 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, ast_rtp::io, ast_rtp::ioid, LOG_ERROR, ast_rtp::rtcp, rtp_socket(), rtpread(), ast_rtcp::s, ast_rtp::s, ast_rtp::sched, ast_rtcp::us, and ast_rtp::us.

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

01891 {
01892    struct ast_rtp *rtp;
01893    int x;
01894    int first;
01895    int startplace;
01896    
01897    if (!(rtp = ast_calloc(1, sizeof(*rtp))))
01898       return NULL;
01899 
01900    ast_rtp_new_init(rtp);
01901 
01902    rtp->s = rtp_socket();
01903    if (rtp->s < 0) {
01904       free(rtp);
01905       ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
01906       return NULL;
01907    }
01908    if (sched && rtcpenable) {
01909       rtp->sched = sched;
01910       rtp->rtcp = ast_rtcp_new();
01911    }
01912    
01913    /* Select a random port number in the range of possible RTP */
01914    x = (ast_random() % (rtpend-rtpstart)) + rtpstart;
01915    x = x & ~1;
01916    /* Save it for future references. */
01917    startplace = x;
01918    /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
01919    for (;;) {
01920       /* Must be an even port number by RTP spec */
01921       rtp->us.sin_port = htons(x);
01922       rtp->us.sin_addr = addr;
01923       /* If there's rtcp, initialize it as well. */
01924       if (rtp->rtcp) {
01925          rtp->rtcp->us.sin_port = htons(x + 1);
01926          rtp->rtcp->us.sin_addr = addr;
01927       }
01928       /* Try to bind it/them. */
01929       if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
01930          (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
01931          break;
01932       if (!first) {
01933          /* Primary bind succeeded! Gotta recreate it */
01934          close(rtp->s);
01935          rtp->s = rtp_socket();
01936       }
01937       if (errno != EADDRINUSE) {
01938          /* We got an error that wasn't expected, abort! */
01939          ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
01940          close(rtp->s);
01941          if (rtp->rtcp) {
01942             close(rtp->rtcp->s);
01943             free(rtp->rtcp);
01944          }
01945          free(rtp);
01946          return NULL;
01947       }
01948       /* The port was used, increment it (by two). */
01949       x += 2;
01950       /* Did we go over the limit ? */
01951       if (x > rtpend)
01952          /* then, start from the begingig. */
01953          x = (rtpstart + 1) & ~1;
01954       /* Check if we reached the place were we started. */
01955       if (x == startplace) {
01956          /* If so, there's no ports available. */
01957          ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
01958          close(rtp->s);
01959          if (rtp->rtcp) {
01960             close(rtp->rtcp->s);
01961             free(rtp->rtcp);
01962          }
01963          free(rtp);
01964          return NULL;
01965       }
01966    }
01967    rtp->sched = sched;
01968    rtp->io = io;
01969    if (callbackmode) {
01970       rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
01971       ast_set_flag(rtp, FLAG_CALLBACK_MODE);
01972    }
01973    ast_rtp_pt_default(rtp);
01974    return rtp;
01975 }

int ast_rtp_proto_register ( struct ast_rtp_protocol proto  ) 

Register interface to channel driver.

Definition at line 2816 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().

02817 {
02818    struct ast_rtp_protocol *cur;
02819 
02820    AST_LIST_LOCK(&protos);
02821    AST_LIST_TRAVERSE(&protos, cur, list) {   
02822       if (!strcmp(cur->type, proto->type)) {
02823          ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
02824          AST_LIST_UNLOCK(&protos);
02825          return -1;
02826       }
02827    }
02828    AST_LIST_INSERT_HEAD(&protos, proto, list);
02829    AST_LIST_UNLOCK(&protos);
02830    
02831    return 0;
02832 }

void ast_rtp_proto_unregister ( struct ast_rtp_protocol proto  ) 

Unregister interface to channel driver.

Definition at line 2808 of file rtp.c.

References AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, and protos.

Referenced by load_module(), and unload_module().

02809 {
02810    AST_LIST_LOCK(&protos);
02811    AST_LIST_REMOVE(&protos, proto, list);
02812    AST_LIST_UNLOCK(&protos);
02813 }

void ast_rtp_pt_clear ( struct ast_rtp rtp  ) 

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

Definition at line 1395 of file rtp.c.

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

Referenced by process_sdp().

01396 {
01397    int i;
01398 
01399    if (!rtp)
01400       return;
01401 
01402    ast_mutex_lock(&rtp->bridge_lock);
01403 
01404    for (i = 0; i < MAX_RTP_PT; ++i) {
01405       rtp->current_RTP_PT[i].isAstFormat = 0;
01406       rtp->current_RTP_PT[i].code = 0;
01407    }
01408 
01409    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01410    rtp->rtp_lookup_code_cache_code = 0;
01411    rtp->rtp_lookup_code_cache_result = 0;
01412 
01413    ast_mutex_unlock(&rtp->bridge_lock);
01414 }

void ast_rtp_pt_copy ( struct ast_rtp dest,
struct ast_rtp src 
)

Copy payload types between RTP structures.

Definition at line 1435 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().

01436 {
01437    unsigned int i;
01438 
01439    ast_mutex_lock(&dest->bridge_lock);
01440    ast_mutex_lock(&src->bridge_lock);
01441 
01442    for (i=0; i < MAX_RTP_PT; ++i) {
01443       dest->current_RTP_PT[i].isAstFormat = 
01444          src->current_RTP_PT[i].isAstFormat;
01445       dest->current_RTP_PT[i].code = 
01446          src->current_RTP_PT[i].code; 
01447    }
01448    dest->rtp_lookup_code_cache_isAstFormat = 0;
01449    dest->rtp_lookup_code_cache_code = 0;
01450    dest->rtp_lookup_code_cache_result = 0;
01451 
01452    ast_mutex_unlock(&src->bridge_lock);
01453    ast_mutex_unlock(&dest->bridge_lock);
01454 }

void ast_rtp_pt_default ( struct ast_rtp rtp  ) 

Set payload types to defaults.

Definition at line 1416 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_new_with_bindaddr().

01417 {
01418    int i;
01419 
01420    ast_mutex_lock(&rtp->bridge_lock);
01421 
01422    /* Initialize to default payload types */
01423    for (i = 0; i < MAX_RTP_PT; ++i) {
01424       rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
01425       rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
01426    }
01427 
01428    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01429    rtp->rtp_lookup_code_cache_code = 0;
01430    rtp->rtp_lookup_code_cache_result = 0;
01431 
01432    ast_mutex_unlock(&rtp->bridge_lock);
01433 }

struct ast_frame* ast_rtp_read ( struct ast_rtp rtp  )  [read]

Definition at line 1101 of file rtp.c.

References ast_codec_get_samples(), AST_FORMAT_MAX_AUDIO, AST_FORMAT_SLINEAR, ast_frame_byteswap_be, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_get_bridged(), ast_rtp_lookup_pt(), ast_rtp_senddigit_continuation(), ast_sched_add(), ast_set_flag, ast_verbose(), bridge_p2p_rtp_write(), ast_rtp::bridged, calc_rxstamp(), rtpPayloadType::code, CRASH, ast_rtp::cycles, ast_frame::data, ast_frame::datalen, ast_frame::delivery, errno, event, ext, ast_rtp::f, f, FLAG_NAT_ACTIVE, ast_frame::frametype, ast_frame::has_timing_info, rtpPayloadType::isAstFormat, ast_rtp::lastevent, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len, LOG_DEBUG, LOG_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().

01102 {
01103    int res;
01104    struct sockaddr_in sin;
01105    socklen_t len;
01106    unsigned int seqno;
01107    int version;
01108    int payloadtype;
01109    int hdrlen = 12;
01110    int padding;
01111    int mark;
01112    int ext;
01113    int cc;
01114    unsigned int ssrc;
01115    unsigned int timestamp;
01116    unsigned int *rtpheader;
01117    struct rtpPayloadType rtpPT;
01118    struct ast_rtp *bridged = NULL;
01119    
01120    /* If time is up, kill it */
01121    if (rtp->sending_digit)
01122       ast_rtp_senddigit_continuation(rtp);
01123 
01124    len = sizeof(sin);
01125    
01126    /* Cache where the header will go */
01127    res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
01128                0, (struct sockaddr *)&sin, &len);
01129 
01130    rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
01131    if (res < 0) {
01132       if (errno == EBADF)
01133          CRASH;
01134       if (errno != EAGAIN) {
01135          ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up.\n", strerror(errno));
01136          return NULL;
01137       }
01138       return &ast_null_frame;
01139    }
01140    
01141    if (res < hdrlen) {
01142       ast_log(LOG_WARNING, "RTP Read too short\n");
01143       return &ast_null_frame;
01144    }
01145 
01146    /* Get fields */
01147    seqno = ntohl(rtpheader[0]);
01148 
01149    /* Check RTP version */
01150    version = (seqno & 0xC0000000) >> 30;
01151    if (!version) {
01152       if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) &&
01153          (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
01154          memcpy(&rtp->them, &sin, sizeof(rtp->them));
01155       }
01156       return &ast_null_frame;
01157    }
01158 
01159 #if 0 /* Allow to receive RTP stream with closed transmission path */
01160    /* If we don't have the other side's address, then ignore this */
01161    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
01162       return &ast_null_frame;
01163 #endif
01164 
01165    /* Send to whoever send to us if NAT is turned on */
01166    if (rtp->nat) {
01167       if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
01168           (rtp->them.sin_port != sin.sin_port)) {
01169          rtp->them = sin;
01170          if (rtp->rtcp) {
01171             memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
01172             rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
01173          }
01174          rtp->rxseqno = 0;
01175          ast_set_flag(rtp, FLAG_NAT_ACTIVE);
01176          if (option_debug || rtpdebug)
01177             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));
01178       }
01179    }
01180 
01181    /* If we are bridged to another RTP stream, send direct */
01182    if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
01183       return &ast_null_frame;
01184 
01185    if (version != 2)
01186       return &ast_null_frame;
01187 
01188    payloadtype = (seqno & 0x7f0000) >> 16;
01189    padding = seqno & (1 << 29);
01190    mark = seqno & (1 << 23);
01191    ext = seqno & (1 << 28);
01192    cc = (seqno & 0xF000000) >> 24;
01193    seqno &= 0xffff;
01194    timestamp = ntohl(rtpheader[1]);
01195    ssrc = ntohl(rtpheader[2]);
01196    
01197    if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
01198       if (option_debug || rtpdebug)
01199          ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
01200       mark = 1;
01201    }
01202 
01203    rtp->rxssrc = ssrc;
01204    
01205    if (padding) {
01206       /* Remove padding bytes */
01207       res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
01208    }
01209    
01210    if (cc) {
01211       /* CSRC fields present */
01212       hdrlen += cc*4;
01213    }
01214 
01215    if (ext) {
01216       /* RTP Extension present */
01217       hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
01218       hdrlen += 4;
01219    }
01220 
01221    if (res < hdrlen) {
01222       ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
01223       return &ast_null_frame;
01224    }
01225 
01226    rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
01227 
01228    if (rtp->rxcount==1) {
01229       /* This is the first RTP packet successfully received from source */
01230       rtp->seedrxseqno = seqno;
01231    }
01232 
01233    /* Do not schedule RR if RTCP isn't run */
01234    if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
01235       /* Schedule transmission of Receiver Report */
01236       rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
01237    }
01238    if ( (int)rtp->lastrxseqno - (int)seqno  > 100) /* if so it would indicate that the sender cycled; allow for misordering */
01239       rtp->cycles += RTP_SEQ_MOD;
01240 
01241    rtp->lastrxseqno = seqno;
01242    
01243    if (rtp->themssrc==0)
01244       rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
01245    
01246    if (rtp_debug_test_addr(&sin))
01247       ast_verbose("Got  RTP packet from    %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
01248          ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
01249 
01250    rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
01251    if (!rtpPT.isAstFormat) {
01252       struct ast_frame *f = NULL;
01253 
01254       /* This is special in-band data that's not one of our codecs */
01255       if (rtpPT.code == AST_RTP_DTMF) {
01256          /* It's special -- rfc2833 process it */
01257          if (rtp_debug_test_addr(&sin)) {
01258             unsigned char *data;
01259             unsigned int event;
01260             unsigned int event_end;
01261             unsigned int duration;
01262             data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
01263             event = ntohl(*((unsigned int *)(data)));
01264             event >>= 24;
01265             event_end = ntohl(*((unsigned int *)(data)));
01266             event_end <<= 8;
01267             event_end >>= 24;
01268             duration = ntohl(*((unsigned int *)(data)));
01269             duration &= 0xFFFF;
01270             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);
01271          }
01272          f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
01273       } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
01274          /* It's really special -- process it the Cisco way */
01275          if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
01276             f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01277             rtp->lastevent = seqno;
01278          }
01279       } else if (rtpPT.code == AST_RTP_CN) {
01280          /* Comfort Noise */
01281          f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01282       } else {
01283          ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
01284       }
01285       return f ? f : &ast_null_frame;
01286    }
01287    rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
01288    rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
01289 
01290    if (!rtp->lastrxts)
01291       rtp->lastrxts = timestamp;
01292 
01293    rtp->rxseqno = seqno;
01294 
01295    /* Record received timestamp as last received now */
01296    rtp->lastrxts = timestamp;
01297 
01298    rtp->f.mallocd = 0;
01299    rtp->f.datalen = res - hdrlen;
01300    rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
01301    rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
01302    rtp->f.seqno = seqno;
01303    if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
01304       rtp->f.samples = ast_codec_get_samples(&rtp->f);
01305       if (rtp->f.subclass == AST_FORMAT_SLINEAR) 
01306          ast_frame_byteswap_be(&rtp->f);
01307       calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
01308       /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
01309       rtp->f.has_timing_info = 1;
01310       rtp->f.ts = timestamp / 8;
01311       rtp->f.len = rtp->f.samples / 8;
01312    } else {
01313       /* Video -- samples is # of samples vs. 90000 */
01314       if (!rtp->lastividtimestamp)
01315          rtp->lastividtimestamp = timestamp;
01316       rtp->f.samples = timestamp - rtp->lastividtimestamp;
01317       rtp->lastividtimestamp = timestamp;
01318       rtp->f.delivery.tv_sec = 0;
01319       rtp->f.delivery.tv_usec = 0;
01320       if (mark)
01321          rtp->f.subclass |= 0x1;
01322       
01323    }
01324    rtp->f.src = "RTP";
01325    return &rtp->f;
01326 }

int ast_rtp_reload ( void   ) 

Definition at line 3718 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().

03719 {
03720    struct ast_config *cfg;
03721    const char *s;
03722 
03723    rtpstart = 5000;
03724    rtpend = 31000;
03725    dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03726    cfg = ast_config_load("rtp.conf");
03727    if (cfg) {
03728       if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
03729          rtpstart = atoi(s);
03730          if (rtpstart < 1024)
03731             rtpstart = 1024;
03732          if (rtpstart > 65535)
03733             rtpstart = 65535;
03734       }
03735       if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
03736          rtpend = atoi(s);
03737          if (rtpend < 1024)
03738             rtpend = 1024;
03739          if (rtpend > 65535)
03740             rtpend = 65535;
03741       }
03742       if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
03743          rtcpinterval = atoi(s);
03744          if (rtcpinterval == 0)
03745             rtcpinterval = 0; /* Just so we're clear... it's zero */
03746          if (rtcpinterval < RTCP_MIN_INTERVALMS)
03747             rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
03748          if (rtcpinterval > RTCP_MAX_INTERVALMS)
03749             rtcpinterval = RTCP_MAX_INTERVALMS;
03750       }
03751       if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
03752 #ifdef SO_NO_CHECK
03753          if (ast_false(s))
03754             nochecksums = 1;
03755          else
03756             nochecksums = 0;
03757 #else
03758          if (ast_false(s))
03759             ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
03760 #endif
03761       }
03762       if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
03763          dtmftimeout = atoi(s);
03764          if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
03765             ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
03766                dtmftimeout, DEFAULT_DTMF_TIMEOUT);
03767             dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03768          };
03769       }
03770       ast_config_destroy(cfg);
03771    }
03772    if (rtpstart >= rtpend) {
03773       ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
03774       rtpstart = 5000;
03775       rtpend = 31000;
03776    }
03777    if (option_verbose > 1)
03778       ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
03779    return 0;
03780 }

void ast_rtp_reset ( struct ast_rtp rtp  ) 

Definition at line 2051 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.

02052 {
02053    memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
02054    memset(&rtp->txcore, 0, sizeof(rtp->txcore));
02055    memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
02056    rtp->lastts = 0;
02057    rtp->lastdigitts = 0;
02058    rtp->lastrxts = 0;
02059    rtp->lastividtimestamp = 0;
02060    rtp->lastovidtimestamp = 0;
02061    rtp->lasteventseqn = 0;
02062    rtp->lastevent = 0;
02063    rtp->lasttxformat = 0;
02064    rtp->lastrxformat = 0;
02065    rtp->dtmfcount = 0;
02066    rtp->dtmfsamples = 0;
02067    rtp->seqno = 0;
02068    rtp->rxseqno = 0;
02069 }

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 2175 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 oh323_digit_begin(), and sip_senddigit_begin().

02176 {
02177    unsigned int *rtpheader;
02178    int hdrlen = 12, res = 0, i = 0, payload = 0;
02179    char data[256];
02180 
02181    if ((digit <= '9') && (digit >= '0'))
02182       digit -= '0';
02183    else if (digit == '*')
02184       digit = 10;
02185    else if (digit == '#')
02186       digit = 11;
02187    else if ((digit >= 'A') && (digit <= 'D'))
02188       digit = digit - 'A' + 12;
02189    else if ((digit >= 'a') && (digit <= 'd'))
02190       digit = digit - 'a' + 12;
02191    else {
02192       ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
02193       return 0;
02194    }
02195 
02196    /* If we have no peer, return immediately */ 
02197    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
02198       return 0;
02199 
02200    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
02201 
02202    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02203    rtp->send_duration = 160;
02204    
02205    /* Get a pointer to the header */
02206    rtpheader = (unsigned int *)data;
02207    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
02208    rtpheader[1] = htonl(rtp->lastdigitts);
02209    rtpheader[2] = htonl(rtp->ssrc); 
02210 
02211    for (i = 0; i < 2; i++) {
02212       rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
02213       res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
02214       if (res < 0) 
02215          ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
02216             ast_inet_ntoa(rtp->them.sin_addr),
02217             ntohs(rtp->them.sin_port), strerror(errno));
02218       if (rtp_debug_test_addr(&rtp->them))
02219          ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
02220                 ast_inet_ntoa(rtp->them.sin_addr),
02221                 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
02222       /* Increment sequence number */
02223       rtp->seqno++;
02224       /* Increment duration */
02225       rtp->send_duration += 160;
02226       /* Clear marker bit and set seqno */
02227       rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
02228    }
02229 
02230    /* Since we received a begin, we can safely store the digit and disable any compensation */
02231    rtp->sending_digit = 1;
02232    rtp->send_digit = digit;
02233    rtp->send_payload = payload;
02234 
02235    return 0;
02236 }

int ast_rtp_senddigit_end ( struct ast_rtp rtp,
char  digit 
)

void ast_rtp_set_callback ( struct ast_rtp rtp,
ast_rtp_callback  callback 
)

Definition at line 585 of file rtp.c.

References ast_rtp::callback.

Referenced by start_rtp().

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

void ast_rtp_set_data ( struct ast_rtp rtp,
void *  data 
)

Definition at line 580 of file rtp.c.

References ast_rtp::data.

Referenced by start_rtp().

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

void ast_rtp_set_m_type ( struct ast_rtp rtp,
int  pt 
)

Activate payload type.

Definition at line 1634 of file rtp.c.

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

Referenced by gtalk_newcall(), and process_sdp().

01635 {
01636    if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 
01637       return; /* bogus payload type */
01638 
01639    ast_mutex_lock(&rtp->bridge_lock);
01640    rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
01641    ast_mutex_unlock(&rtp->bridge_lock);
01642 } 

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

Definition at line 1994 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().

01995 {
01996    rtp->them.sin_port = them->sin_port;
01997    rtp->them.sin_addr = them->sin_addr;
01998    if (rtp->rtcp) {
01999       rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
02000       rtp->rtcp->them.sin_addr = them->sin_addr;
02001    }
02002    rtp->rxseqno = 0;
02003 }

void ast_rtp_set_rtpholdtimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp hold timeout.

Definition at line 547 of file rtp.c.

References ast_rtp::rtpholdtimeout.

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

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

void ast_rtp_set_rtpkeepalive ( struct ast_rtp rtp,
int  period 
)

set RTP keepalive interval

Definition at line 553 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by create_addr_from_peer(), and sip_alloc().

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

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

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

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 1661 of file rtp.c.

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

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

01664 {
01665    unsigned int i;
01666    int found = 0;
01667 
01668    if (pt < 0 || pt > MAX_RTP_PT) 
01669       return -1; /* bogus payload type */
01670    
01671    ast_mutex_lock(&rtp->bridge_lock);
01672 
01673    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01674       if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
01675           strcasecmp(mimeType, mimeTypes[i].type) == 0) {
01676          found = 1;
01677          rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
01678          if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
01679              mimeTypes[i].payloadType.isAstFormat &&
01680              (options & AST_RTP_OPT_G726_NONSTANDARD))
01681             rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
01682          break;
01683       }
01684    }
01685 
01686    ast_mutex_unlock(&rtp->bridge_lock);
01687 
01688    return (found ? 0 : -1);
01689 } 

void ast_rtp_set_rtptimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp timeout.

Definition at line 541 of file rtp.c.

References ast_rtp::rtptimeout.

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

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

void ast_rtp_set_rtptimers_onhold ( struct ast_rtp rtp  ) 

Definition at line 534 of file rtp.c.

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

Referenced by handle_response_invite().

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

void ast_rtp_setdtmf ( struct ast_rtp rtp,
int  dtmf 
)

Indicate whether this RTP session is carrying DTMF or not.

Definition at line 600 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_DTMF.

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

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

void ast_rtp_setdtmfcompensate ( struct ast_rtp rtp,
int  compensate 
)

Compensate for devices that send RFC2833 packets all at once.

Definition at line 605 of file rtp.c.

References ast_set2_flag, and FLAG_DTMF_COMPENSATE.

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

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

void ast_rtp_setnat ( struct ast_rtp rtp,
int  nat 
)

Definition at line 590 of file rtp.c.

References ast_rtp::nat.

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

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

void ast_rtp_setstun ( struct ast_rtp rtp,
int  stun_enable 
)

Enable STUN capability.

Definition at line 610 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_STUN.

Referenced by gtalk_new().

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

int ast_rtp_settos ( struct ast_rtp rtp,
int  tos 
)

Definition at line 1985 of file rtp.c.

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

Referenced by __oh323_rtp_create(), and sip_alloc().

01986 {
01987    int res;
01988 
01989    if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
01990       ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
01991    return res;
01992 }

void ast_rtp_stop ( struct ast_rtp rtp  ) 

Definition at line 2034 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().

02035 {
02036    if (rtp->rtcp && rtp->rtcp->schedid > 0) {
02037       ast_sched_del(rtp->sched, rtp->rtcp->schedid);
02038       rtp->rtcp->schedid = -1;
02039    }
02040 
02041    memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
02042    memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
02043    if (rtp->rtcp) {
02044       memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
02045       memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
02046    }
02047    
02048    ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
02049 }

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

Definition at line 402 of file rtp.c.

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

Referenced by gtalk_update_stun().

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

void ast_rtp_unset_m_type ( struct ast_rtp rtp,
int  pt 
)

clear payload type

Definition at line 1646 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().

01647 {
01648    if (pt < 0 || pt > MAX_RTP_PT)
01649       return; /* bogus payload type */
01650 
01651    ast_mutex_lock(&rtp->bridge_lock);
01652    rtp->current_RTP_PT[pt].isAstFormat = 0;
01653    rtp->current_RTP_PT[pt].code = 0;
01654    ast_mutex_unlock(&rtp->bridge_lock);
01655 }

int ast_rtp_write ( struct ast_rtp rtp,
struct ast_frame f 
)

Definition at line 2727 of file rtp.c.

References ast_codec_pref_getsize(), AST_FORMAT_G723_1, AST_FORMAT_SPEEX, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frdup(), ast_frfree, ast_getformatname(), ast_log(), ast_rtp_lookup_code(), ast_rtp_raw_write(), ast_smoother_feed, ast_smoother_feed_be, AST_SMOOTHER_FLAG_BE, ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_smoother_set_flags(), ast_smoother_test_flag(), ast_format_list::cur_ms, ast_frame::data, ast_frame::datalen, f, ast_format_list::flags, ast_format_list::fr_len, ast_frame::frametype, ast_format_list::inc_ms, ast_rtp::lasttxformat, LOG_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().

02728 {
02729    struct ast_frame *f;
02730    int codec;
02731    int hdrlen = 12;
02732    int subclass;
02733    
02734 
02735    /* If we have no peer, return immediately */ 
02736    if (!rtp->them.sin_addr.s_addr)
02737       return 0;
02738 
02739    /* If there is no data length, return immediately */
02740    if (!_f->datalen) 
02741       return 0;
02742    
02743    /* Make sure we have enough space for RTP header */
02744    if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
02745       ast_log(LOG_WARNING, "RTP can only send voice and video\n");
02746       return -1;
02747    }
02748 
02749    subclass = _f->subclass;
02750    if (_f->frametype == AST_FRAME_VIDEO)
02751       subclass &= ~0x1;
02752 
02753    codec = ast_rtp_lookup_code(rtp, 1, subclass);
02754    if (codec < 0) {
02755       ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
02756       return -1;
02757    }
02758 
02759    if (rtp->lasttxformat != subclass) {
02760       /* New format, reset the smoother */
02761       if (option_debug)
02762          ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
02763       rtp->lasttxformat = subclass;
02764       if (rtp->smoother)
02765          ast_smoother_free(rtp->smoother);
02766       rtp->smoother = NULL;
02767    }
02768 
02769    if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) {
02770       struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
02771       if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
02772          if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
02773             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));
02774             return -1;
02775          }
02776          if (fmt.flags)
02777             ast_smoother_set_flags(rtp->smoother, fmt.flags);
02778          if (option_debug)
02779             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));
02780       }
02781    }
02782    if (rtp->smoother) {
02783       if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
02784          ast_smoother_feed_be(rtp->smoother, _f);
02785       } else {
02786          ast_smoother_feed(rtp->smoother, _f);
02787       }
02788 
02789       while((f = ast_smoother_read(rtp->smoother)) && (f->data))
02790          ast_rtp_raw_write(rtp, f, codec);
02791    } else {
02792            /* Don't buffer outgoing frames; send them one-per-packet: */
02793       if (_f->offset < hdrlen) {
02794          f = ast_frdup(_f);
02795       } else {
02796          f = _f;
02797       }
02798       if (f->data)
02799          ast_rtp_raw_write(rtp, f, codec);
02800       if (f != _f)
02801          ast_frfree(f);
02802    }
02803       
02804    return 0;
02805 }


Generated on Fri Sep 25 19:28:44 2009 for Asterisk - the Open Source PBX by  doxygen 1.5.5