Mon May 14 04:50:59 2007

Asterisk developer's documentation


rtp.h File Reference

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

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

Include dependency graph for rtp.h:

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

Go to the source code of this file.

Data Structures

struct  ast_rtp_protocol
struct  ast_rtp_quality

Defines

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

Typedefs

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

Enumerations

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

Functions

int ast_rtcp_fd (struct ast_rtp *rtp)
ast_frameast_rtcp_read (struct ast_rtp *rtp)
int ast_rtcp_send_h261fur (void *data)
 Send an H.261 fast update request. Some devices need this rather than the XML message in SIP.
size_t ast_rtp_alloc_size (void)
 Get the amount of space required to hold an RTP session.
int ast_rtp_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
 Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk.
int ast_rtp_codec_getformat (int pt)
ast_codec_prefast_rtp_codec_getpref (struct ast_rtp *rtp)
int ast_rtp_codec_setpref (struct ast_rtp *rtp, struct ast_codec_pref *prefs)
void ast_rtp_destroy (struct ast_rtp *rtp)
int ast_rtp_early_bridge (struct ast_channel *dest, struct ast_channel *src)
 If possible, create an early bridge directly between the devices without having to send a re-invite later.
int ast_rtp_fd (struct ast_rtp *rtp)
ast_rtpast_rtp_get_bridged (struct ast_rtp *rtp)
void ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats)
 Return the union of all of the codecs that were set by rtp_set...() calls They're returned as two distinct sets: AST_FORMATs, and AST_RTPs.
int ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them)
char * ast_rtp_get_quality (struct ast_rtp *rtp, struct ast_rtp_quality *qual)
 Return RTCP quality string.
int ast_rtp_get_rtpholdtimeout (struct ast_rtp *rtp)
 Get rtp hold timeout.
int ast_rtp_get_rtpkeepalive (struct ast_rtp *rtp)
 Get RTP keepalive interval.
int ast_rtp_get_rtptimeout (struct ast_rtp *rtp)
 Get rtp timeout.
void ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us)
int ast_rtp_getnat (struct ast_rtp *rtp)
void ast_rtp_init (void)
 Initialize the RTP system in Asterisk.
int ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code)
 Looks up an RTP code out of our *static* outbound list.
char * ast_rtp_lookup_mime_multiple (char *buf, size_t size, const int capability, const int isAstFormat, enum ast_rtp_options options)
 Build a string of MIME subtype names from a capability list.
const char * ast_rtp_lookup_mime_subtype (int isAstFormat, int code, enum ast_rtp_options options)
 Mapping an Asterisk code into a MIME subtype (string):.
rtpPayloadType ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt)
 Mapping between RTP payload format codes and Asterisk codes:.
int ast_rtp_make_compatible (struct ast_channel *dest, struct ast_channel *src, int media)
ast_rtpast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
 Initializate a RTP session.
void ast_rtp_new_init (struct ast_rtp *rtp)
 Initialize a new RTP structure.
ast_rtpast_rtp_new_with_bindaddr (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in)
 Initializate a RTP session using an in_addr structure.
int ast_rtp_proto_register (struct ast_rtp_protocol *proto)
 Register interface to channel driver.
void ast_rtp_proto_unregister (struct ast_rtp_protocol *proto)
 Unregister interface to channel driver.
void ast_rtp_pt_clear (struct ast_rtp *rtp)
 Setting RTP payload types from lines in a SDP description:.
void ast_rtp_pt_copy (struct ast_rtp *dest, struct ast_rtp *src)
 Copy payload types between RTP structures.
void ast_rtp_pt_default (struct ast_rtp *rtp)
 Set payload types to defaults.
ast_frameast_rtp_read (struct ast_rtp *rtp)
int ast_rtp_reload (void)
void ast_rtp_reset (struct ast_rtp *rtp)
int ast_rtp_sendcng (struct ast_rtp *rtp, int level)
 generate comfort noice (CNG)
int ast_rtp_senddigit_begin (struct ast_rtp *rtp, char digit)
 Send begin frames for DTMF.
int ast_rtp_senddigit_end (struct ast_rtp *rtp, char digit)
void ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback)
void ast_rtp_set_data (struct ast_rtp *rtp, void *data)
void ast_rtp_set_m_type (struct ast_rtp *rtp, int pt)
 Make a note of a RTP payload type that was seen in a SDP "m=" line. By default, use the well-known value for this type (although it may still be set to a different value by a subsequent "a=rtpmap:" line).
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
void ast_rtp_set_rtpmap_type (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype, enum ast_rtp_options options)
 Make a note of a RTP payload type (with MIME type) that was seen in an SDP "a=rtpmap:" line.
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)
int ast_rtp_write (struct ast_rtp *rtp, struct ast_frame *f)


Detailed Description

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

RTP is defined in RFC 3550.

Definition in file rtp.h.


Define Documentation

#define AST_RTP_CISCO_DTMF   (1 << 2)

DTMF (Cisco Proprietary)

Definition at line 47 of file rtp.h.

Referenced by ast_rtp_read().

#define AST_RTP_CN   (1 << 1)

'Comfort Noise' (RFC3389)

Definition at line 45 of file rtp.h.

Referenced by ast_rtp_read(), and ast_rtp_sendcng().

#define AST_RTP_DTMF   (1 << 0)

DTMF (RFC2833)

Definition at line 43 of file rtp.h.

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

#define AST_RTP_MAX   AST_RTP_CISCO_DTMF

Maximum RTP-specific code

Definition at line 49 of file rtp.h.

Referenced by add_sdp(), and ast_rtp_lookup_mime_multiple().

#define FLAG_3389_WARNING   (1 << 0)

Definition at line 93 of file rtp.h.

#define MAX_RTP_PT   256

Definition at line 51 of file rtp.h.

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


Typedef Documentation

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

Definition at line 95 of file rtp.h.


Enumeration Type Documentation

enum ast_rtp_get_result

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

Definition at line 57 of file rtp.h.

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

enum ast_rtp_options

Enumerator:
AST_RTP_OPT_G726_NONSTANDARD 

Definition at line 53 of file rtp.h.

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


Function Documentation

int ast_rtcp_fd ( struct ast_rtp rtp  ) 

Definition at line 517 of file rtp.c.

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

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

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

struct ast_frame* ast_rtcp_read ( struct ast_rtp rtp  ) 

Definition at line 822 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, 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().

00823 {
00824    socklen_t len;
00825    int position, i, packetwords;
00826    int res;
00827    struct sockaddr_in sin;
00828    unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
00829    unsigned int *rtcpheader;
00830    int pt;
00831    struct timeval now;
00832    unsigned int length;
00833    int rc;
00834    double rtt = 0;
00835    double a;
00836    double dlsr;
00837    double lsr;
00838    unsigned int msw;
00839    unsigned int lsw;
00840    unsigned int comp;
00841    struct ast_frame *f = &ast_null_frame;
00842    
00843    if (!rtp || !rtp->rtcp)
00844       return &ast_null_frame;
00845 
00846    len = sizeof(sin);
00847    
00848    res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
00849                0, (struct sockaddr *)&sin, &len);
00850    rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
00851    
00852    if (res < 0) {
00853       if (errno == EBADF)
00854          CRASH;
00855       if (errno != EAGAIN) {
00856          ast_log(LOG_WARNING, "RTCP Read error: %s.  Hanging up.\n", strerror(errno));
00857          return NULL;
00858       }
00859       return &ast_null_frame;
00860    }
00861 
00862    packetwords = res / 4;
00863    
00864    if (rtp->nat) {
00865       /* Send to whoever sent to us */
00866       if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
00867           (rtp->rtcp->them.sin_port != sin.sin_port)) {
00868          memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
00869          if (option_debug || rtpdebug)
00870             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));
00871       }
00872    }
00873 
00874    if (option_debug)
00875       ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
00876 
00877    /* Process a compound packet */
00878    position = 0;
00879    while (position < packetwords) {
00880       i = position;
00881       length = ntohl(rtcpheader[i]);
00882       pt = (length & 0xff0000) >> 16;
00883       rc = (length & 0x1f000000) >> 24;
00884       length &= 0xffff;
00885     
00886       if ((i + length) > packetwords) {
00887          ast_log(LOG_WARNING, "RTCP Read too short\n");
00888          return &ast_null_frame;
00889       }
00890       
00891       if (rtcp_debug_test_addr(&sin)) {
00892          ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
00893          ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
00894          ast_verbose("Reception reports: %d\n", rc);
00895          ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
00896       }
00897     
00898       i += 2; /* Advance past header and ssrc */
00899       
00900       switch (pt) {
00901       case RTCP_PT_SR:
00902          gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
00903          rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
00904          rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
00905          rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
00906     
00907          if (rtcp_debug_test_addr(&sin)) {
00908             ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
00909             ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
00910             ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
00911          }
00912          i += 5;
00913          if (rc < 1)
00914             break;
00915          /* Intentional fall through */
00916       case RTCP_PT_RR:
00917          /* This is the place to calculate RTT */
00918          /* Don't handle multiple reception reports (rc > 1) yet */
00919          gettimeofday(&now, NULL);
00920          timeval2ntp(now, &msw, &lsw);
00921          /* Use the one we sent them in our SR instead, rtcp->txlsr could have been rewritten if the dlsr is large */
00922          if (ntohl(rtcpheader[i + 4])) { /* We must have the LSR */
00923             comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
00924             a = (double)((comp & 0xffff0000) >> 16) + (double)((double)(comp & 0xffff)/1000000.);
00925             lsr = (double)((ntohl(rtcpheader[i + 4]) & 0xffff0000) >> 16) + (double)((double)(ntohl(rtcpheader[i + 4]) & 0xffff) / 1000000.);
00926             dlsr = (double)(ntohl(rtcpheader[i + 5])/65536.);
00927             rtt = a - dlsr - lsr;
00928             if (rtt >= 0) {
00929                rtp->rtcp->accumulated_transit += rtt;
00930                rtp->rtcp->rtt = rtt;
00931                if (rtp->rtcp->maxrtt < rtt)
00932                   rtp->rtcp->maxrtt = rtt;
00933                if (rtp->rtcp->minrtt > rtt)
00934                   rtp->rtcp->minrtt = rtt;
00935             }
00936          }
00937          rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
00938          rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
00939          if (rtcp_debug_test_addr(&sin)) {
00940             ast_verbose("  Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
00941             ast_verbose("  Packets lost so far: %d\n", rtp->rtcp->reported_lost);
00942             ast_verbose("  Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
00943             ast_verbose("  Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
00944             ast_verbose("  Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
00945             ast_verbose("  Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
00946             ast_verbose("  DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
00947             if (rtt)
00948                ast_verbose("  RTT: %f(sec)\n", rtt);
00949          }
00950          break;
00951       case RTCP_PT_FUR:
00952          if (rtcp_debug_test_addr(&sin))
00953             ast_verbose("Received an RTCP Fast Update Request\n");
00954          rtp->f.frametype = AST_FRAME_CONTROL;
00955          rtp->f.subclass = AST_CONTROL_VIDUPDATE;
00956          rtp->f.datalen = 0;
00957          rtp->f.samples = 0;
00958          rtp->f.mallocd = 0;
00959          rtp->f.src = "RTP";
00960          f = &rtp->f;
00961          break;
00962       case RTCP_PT_SDES:
00963          if (rtcp_debug_test_addr(&sin))
00964             ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00965          break;
00966       case RTCP_PT_BYE:
00967          if (rtcp_debug_test_addr(&sin))
00968             ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00969          break;
00970       default:
00971          if (option_debug)
00972             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));
00973          break;
00974       }
00975       position += (length + 1);
00976    }
00977          
00978    return f;
00979 }

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 2291 of file rtp.c.

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

02292 {
02293    struct ast_rtp *rtp = data;
02294    int res;
02295 
02296    rtp->rtcp->sendfur = 1;
02297    res = ast_rtcp_write(data);
02298    
02299    return res;
02300 }

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 3185 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_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(), 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_WARNING, option_debug, option_verbose, ast_channel::rawreadformat, ast_channel::rawwriteformat, ast_channel_tech::send_digit_begin, ast_channel::tech, ast_channel::tech_pvt, and VERBOSE_PREFIX_3.

03186 {
03187    struct ast_rtp *p0 = NULL, *p1 = NULL;    /* Audio RTP Channels */
03188    struct ast_rtp *vp0 = NULL, *vp1 = NULL;  /* Video RTP channels */
03189    struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
03190    enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
03191    enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
03192    enum ast_bridge_result res = AST_BRIDGE_FAILED;
03193    int codec0 = 0, codec1 = 0;
03194    void *pvt0 = NULL, *pvt1 = NULL;
03195 
03196    /* Lock channels */
03197    ast_channel_lock(c0);
03198    while(ast_channel_trylock(c1)) {
03199       ast_channel_unlock(c0);
03200       usleep(1);
03201       ast_channel_lock(c0);
03202    }
03203 
03204    /* Find channel driver interfaces */
03205    if (!(pr0 = get_proto(c0))) {
03206       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
03207       ast_channel_unlock(c0);
03208       ast_channel_unlock(c1);
03209       return AST_BRIDGE_FAILED;
03210    }
03211    if (!(pr1 = get_proto(c1))) {
03212       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
03213       ast_channel_unlock(c0);
03214       ast_channel_unlock(c1);
03215       return AST_BRIDGE_FAILED;
03216    }
03217 
03218    /* Get channel specific interface structures */
03219    pvt0 = c0->tech_pvt;
03220    pvt1 = c1->tech_pvt;
03221 
03222    /* Get audio and video interface (if native bridge is possible) */
03223    audio_p0_res = pr0->get_rtp_info(c0, &p0);
03224    video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
03225    audio_p1_res = pr1->get_rtp_info(c1, &p1);
03226    video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
03227 
03228    /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
03229    if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
03230       audio_p0_res = AST_RTP_GET_FAILED;
03231    if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
03232       audio_p1_res = AST_RTP_GET_FAILED;
03233 
03234    /* Check if a bridge is possible (partial/native) */
03235    if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
03236       /* Somebody doesn't want to play... */
03237       ast_channel_unlock(c0);
03238       ast_channel_unlock(c1);
03239       return AST_BRIDGE_FAILED_NOWARN;
03240    }
03241 
03242    /* If we need to feed DTMF frames into the core then only do a partial native bridge */
03243    if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
03244       ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
03245       audio_p0_res = AST_RTP_TRY_PARTIAL;
03246    }
03247 
03248    if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
03249       ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
03250       audio_p1_res = AST_RTP_TRY_PARTIAL;
03251    }
03252 
03253    /* If both sides are not using the same method of DTMF transmission 
03254     * (ie: one is RFC2833, other is INFO... then we can not do direct media. 
03255     * --------------------------------------------------
03256     * | DTMF Mode |  HAS_DTMF  |  Accepts Begin Frames |
03257     * |-----------|------------|-----------------------|
03258     * | Inband    | False      | True                  |
03259     * | RFC2833   | True       | True                  |
03260     * | SIP INFO  | False      | False                 |
03261     * --------------------------------------------------
03262     * However, if DTMF from both channels is being monitored by the core, then
03263     * we can still do packet-to-packet bridging, because passing through the 
03264     * core will handle DTMF mode translation.
03265     */
03266    if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
03267        (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
03268       if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
03269          ast_channel_unlock(c0);
03270          ast_channel_unlock(c1);
03271          return AST_BRIDGE_FAILED_NOWARN;
03272       }
03273       audio_p0_res = AST_RTP_TRY_PARTIAL;
03274       audio_p1_res = AST_RTP_TRY_PARTIAL;
03275    }
03276 
03277    /* 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 */
03278    if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF) && ast_test_flag(p0, FLAG_DTMF_COMPENSATE)) ||
03279        (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF) && ast_test_flag(p1, FLAG_DTMF_COMPENSATE))) {
03280       ast_channel_unlock(c0);
03281       ast_channel_unlock(c1);
03282       return AST_BRIDGE_FAILED_NOWARN;
03283    }
03284 
03285    /* Get codecs from both sides */
03286    codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
03287    codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
03288    if (codec0 && codec1 && !(codec0 & codec1)) {
03289       /* Hey, we can't do native bridging if both parties speak different codecs */
03290       if (option_debug)
03291          ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
03292       ast_channel_unlock(c0);
03293       ast_channel_unlock(c1);
03294       return AST_BRIDGE_FAILED_NOWARN;
03295    }
03296 
03297    /* If either side can only do a partial bridge, then don't try for a true native bridge */
03298    if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
03299       /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
03300       if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
03301          if (option_debug)
03302             ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n");
03303          ast_channel_unlock(c0);
03304          ast_channel_unlock(c1);
03305          return AST_BRIDGE_FAILED_NOWARN;
03306       }
03307       if (option_verbose > 2)
03308          ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
03309       res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
03310    } else {
03311       if (option_verbose > 2) 
03312          ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
03313       res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
03314    }
03315 
03316    return res;
03317 }

int ast_rtp_codec_getformat ( int  pt  ) 

Definition at line 2671 of file rtp.c.

References rtpPayloadType::code, and static_RTP_PT.

Referenced by process_sdp().

02672 {
02673    if (pt < 0 || pt > MAX_RTP_PT)
02674       return 0; /* bogus payload type */
02675 
02676    if (static_RTP_PT[pt].isAstFormat)
02677       return static_RTP_PT[pt].code;
02678    else
02679       return 0;
02680 }

struct ast_codec_pref* ast_rtp_codec_getpref ( struct ast_rtp rtp  ) 

Definition at line 2666 of file rtp.c.

References ast_rtp::pref.

Referenced by add_codec_to_sdp(), and process_sdp().

02667 {
02668    return &rtp->pref;
02669 }

int ast_rtp_codec_setpref ( struct ast_rtp rtp,
struct ast_codec_pref prefs 
)

Definition at line 2653 of file rtp.c.

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

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

02654 {
02655    int x;
02656    for (x = 0; x < 32; x++) {  /* Ugly way */
02657       rtp->pref.order[x] = prefs->order[x];
02658       rtp->pref.framing[x] = prefs->framing[x];
02659    }
02660    if (rtp->smoother)
02661       ast_smoother_free(rtp->smoother);
02662    rtp->smoother = NULL;
02663    return 0;
02664 }

void ast_rtp_destroy ( struct ast_rtp rtp  ) 

Definition at line 2073 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().

02074 {
02075    if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
02076       /*Print some info on the call here */
02077       ast_verbose("  RTP-stats\n");
02078       ast_verbose("* Our Receiver:\n");
02079       ast_verbose("  SSRC:     %u\n", rtp->themssrc);
02080       ast_verbose("  Received packets: %u\n", rtp->rxcount);
02081       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
02082       ast_verbose("  Jitter:      %.4f\n", rtp->rxjitter);
02083       ast_verbose("  Transit:     %.4f\n", rtp->rxtransit);
02084       ast_verbose("  RR-count:    %u\n", rtp->rtcp->rr_count);
02085       ast_verbose("* Our Sender:\n");
02086       ast_verbose("  SSRC:     %u\n", rtp->ssrc);
02087       ast_verbose("  Sent packets:   %u\n", rtp->txcount);
02088       ast_verbose("  Lost packets:   %u\n", rtp->rtcp->reported_lost);
02089       ast_verbose("  Jitter:      %u\n", rtp->rtcp->reported_jitter);
02090       ast_verbose("  SR-count:    %u\n", rtp->rtcp->sr_count);
02091       ast_verbose("  RTT:      %f\n", rtp->rtcp->rtt);
02092    }
02093 
02094    if (rtp->smoother)
02095       ast_smoother_free(rtp->smoother);
02096    if (rtp->ioid)
02097       ast_io_remove(rtp->io, rtp->ioid);
02098    if (rtp->s > -1)
02099       close(rtp->s);
02100    if (rtp->rtcp) {
02101       if (rtp->rtcp->schedid > 0)
02102          ast_sched_del(rtp->sched, rtp->rtcp->schedid);
02103       close(rtp->rtcp->s);
02104       free(rtp->rtcp);
02105       rtp->rtcp=NULL;
02106    }
02107 
02108    ast_mutex_destroy(&rtp->bridge_lock);
02109 
02110    free(rtp);
02111 }

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 1459 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().

01460 {
01461    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01462    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01463    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01464    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01465    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
01466    int srccodec, destcodec, nat_active = 0;
01467 
01468    /* Lock channels */
01469    ast_channel_lock(dest);
01470    if (src) {
01471       while(ast_channel_trylock(src)) {
01472          ast_channel_unlock(dest);
01473          usleep(1);
01474          ast_channel_lock(dest);
01475       }
01476    }
01477 
01478    /* Find channel driver interfaces */
01479    destpr = get_proto(dest);
01480    if (src)
01481       srcpr = get_proto(src);
01482    if (!destpr) {
01483       if (option_debug)
01484          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01485       ast_channel_unlock(dest);
01486       if (src)
01487          ast_channel_unlock(src);
01488       return 0;
01489    }
01490    if (!srcpr) {
01491       if (option_debug)
01492          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
01493       ast_channel_unlock(dest);
01494       if (src)
01495          ast_channel_unlock(src);
01496       return 0;
01497    }
01498 
01499    /* Get audio and video interface (if native bridge is possible) */
01500    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01501    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01502    if (srcpr) {
01503       audio_src_res = srcpr->get_rtp_info(src, &srcp);
01504       video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01505    }
01506 
01507    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01508    if (audio_dest_res != AST_RTP_TRY_NATIVE) {
01509       /* Somebody doesn't want to play... */
01510       ast_channel_unlock(dest);
01511       if (src)
01512          ast_channel_unlock(src);
01513       return 0;
01514    }
01515    if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
01516       srccodec = srcpr->get_codec(src);
01517    else
01518       srccodec = 0;
01519    if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
01520       destcodec = destpr->get_codec(dest);
01521    else
01522       destcodec = 0;
01523    /* Ensure we have at least one matching codec */
01524    if (!(srccodec & destcodec)) {
01525       ast_channel_unlock(dest);
01526       if (src)
01527          ast_channel_unlock(src);
01528       return 0;
01529    }
01530    /* Consider empty media as non-existant */
01531    if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
01532       srcp = NULL;
01533    /* If the client has NAT stuff turned on then just safe NAT is active */
01534    if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01535       nat_active = 1;
01536    /* Bridge media early */
01537    if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, nat_active))
01538       ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>");
01539    ast_channel_unlock(dest);
01540    if (src)
01541       ast_channel_unlock(src);
01542    if (option_debug)
01543       ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
01544    return 1;
01545 }

int ast_rtp_fd ( struct ast_rtp rtp  ) 

Definition at line 512 of file rtp.c.

References ast_rtp::s.

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

00513 {
00514    return rtp->s;
00515 }

struct ast_rtp* ast_rtp_get_bridged ( struct ast_rtp rtp  ) 

Definition at line 1995 of file rtp.c.

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

Referenced by ast_rtp_read(), and do_monitor().

01996 {
01997    struct ast_rtp *bridged = NULL;
01998 
01999    ast_mutex_lock(&rtp->bridge_lock);
02000    bridged = rtp->bridged;
02001    ast_mutex_unlock(&rtp->bridge_lock);
02002 
02003    return bridged;
02004 }

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 1665 of file rtp.c.

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

Referenced by process_sdp().

01667 {
01668    int pt;
01669    
01670    ast_mutex_lock(&rtp->bridge_lock);
01671    
01672    *astFormats = *nonAstFormats = 0;
01673    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01674       if (rtp->current_RTP_PT[pt].isAstFormat) {
01675          *astFormats |= rtp->current_RTP_PT[pt].code;
01676       } else {
01677          *nonAstFormats |= rtp->current_RTP_PT[pt].code;
01678       }
01679    }
01680    
01681    ast_mutex_unlock(&rtp->bridge_lock);
01682    
01683    return;
01684 }

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

Definition at line 1977 of file rtp.c.

References ast_rtp::them.

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

01978 {
01979    if ((them->sin_family != AF_INET) ||
01980       (them->sin_port != rtp->them.sin_port) ||
01981       (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
01982       them->sin_family = AF_INET;
01983       them->sin_port = rtp->them.sin_port;
01984       them->sin_addr = rtp->them.sin_addr;
01985       return 1;
01986    }
01987    return 0;
01988 }

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

Return RTCP quality string.

Definition at line 2043 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().

02044 {
02045    /*
02046    *ssrc          our ssrc
02047    *themssrc      their ssrc
02048    *lp            lost packets
02049    *rxjitter      our calculated jitter(rx)
02050    *rxcount       no. received packets
02051    *txjitter      reported jitter of the other end
02052    *txcount       transmitted packets
02053    *rlp           remote lost packets
02054    *rtt           round trip time
02055    */
02056 
02057    if (qual) {
02058       qual->local_ssrc = rtp->ssrc;
02059       qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
02060       qual->local_jitter = rtp->rxjitter;
02061       qual->local_count = rtp->rxcount;
02062       qual->remote_ssrc = rtp->themssrc;
02063       qual->remote_lostpackets = rtp->rtcp->reported_lost;
02064       qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
02065       qual->remote_count = rtp->txcount;
02066       qual->rtt = rtp->rtcp->rtt;
02067    }
02068    snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", rtp->ssrc, rtp->themssrc, rtp->rtcp->expected_prior - rtp->rtcp->received_prior, rtp->rxjitter, rtp->rxcount, (double)rtp->rtcp->reported_jitter/65536., rtp->txcount, rtp->rtcp->reported_lost, rtp->rtcp->rtt);
02069    
02070    return rtp->rtcp->quality;
02071 }

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 1990 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().

01991 {
01992    *us = rtp->us;
01993 }

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 3702 of file rtp.c.

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

Referenced by main().

03703 {
03704    ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
03705    ast_rtp_reload();
03706 }

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 1708 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().

01709 {
01710    int pt = 0;
01711 
01712    ast_mutex_lock(&rtp->bridge_lock);
01713 
01714    if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
01715       code == rtp->rtp_lookup_code_cache_code) {
01716       /* Use our cached mapping, to avoid the overhead of the loop below */
01717       pt = rtp->rtp_lookup_code_cache_result;
01718       ast_mutex_unlock(&rtp->bridge_lock);
01719       return pt;
01720    }
01721 
01722    /* Check the dynamic list first */
01723    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01724       if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
01725          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01726          rtp->rtp_lookup_code_cache_code = code;
01727          rtp->rtp_lookup_code_cache_result = pt;
01728          ast_mutex_unlock(&rtp->bridge_lock);
01729          return pt;
01730       }
01731    }
01732 
01733    /* Then the static list */
01734    for (pt = 0; pt < MAX_RTP_PT; ++pt) {
01735       if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
01736          rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
01737          rtp->rtp_lookup_code_cache_code = code;
01738          rtp->rtp_lookup_code_cache_result = pt;
01739          ast_mutex_unlock(&rtp->bridge_lock);
01740          return pt;
01741       }
01742    }
01743 
01744    ast_mutex_unlock(&rtp->bridge_lock);
01745 
01746    return -1;
01747 }

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 1768 of file rtp.c.

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

Referenced by process_sdp().

01770 {
01771    int format;
01772    unsigned len;
01773    char *end = buf;
01774    char *start = buf;
01775 
01776    if (!buf || !size)
01777       return NULL;
01778 
01779    snprintf(end, size, "0x%x (", capability);
01780 
01781    len = strlen(end);
01782    end += len;
01783    size -= len;
01784    start = end;
01785 
01786    for (format = 1; format < AST_RTP_MAX; format <<= 1) {
01787       if (capability & format) {
01788          const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
01789 
01790          snprintf(end, size, "%s|", name);
01791          len = strlen(end);
01792          end += len;
01793          size -= len;
01794       }
01795    }
01796 
01797    if (start == end)
01798       snprintf(start, size, "nothing)"); 
01799    else if (size > 1)
01800       *(end -1) = ')';
01801    
01802    return buf;
01803 }

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 1749 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().

01751 {
01752    unsigned int i;
01753 
01754    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01755       if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
01756          if (isAstFormat &&
01757              (code == AST_FORMAT_G726_AAL2) &&
01758              (options & AST_RTP_OPT_G726_NONSTANDARD))
01759             return "G726-32";
01760          else
01761             return mimeTypes[i].subtype;
01762       }
01763    }
01764 
01765    return "";
01766 }

struct rtpPayloadType ast_rtp_lookup_pt ( struct ast_rtp rtp,
int  pt 
)

Mapping between RTP payload format codes and Asterisk codes:.

Definition at line 1686 of file rtp.c.

References ast_mutex_lock(), ast_mutex_unlock(), MAX_RTP_PT, result, and static_RTP_PT.

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

01687 {
01688    struct rtpPayloadType result;
01689 
01690    result.isAstFormat = result.code = 0;
01691 
01692    if (pt < 0 || pt > MAX_RTP_PT) 
01693       return result; /* bogus payload type */
01694 
01695    /* Start with negotiated codecs */
01696    ast_mutex_lock(&rtp->bridge_lock);
01697    result = rtp->current_RTP_PT[pt];
01698    ast_mutex_unlock(&rtp->bridge_lock);
01699 
01700    /* If it doesn't exist, check our static RTP type list, just in case */
01701    if (!result.code) 
01702       result = static_RTP_PT[pt];
01703 
01704    return result;
01705 }

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

Definition at line 1547 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().

01548 {
01549    struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
01550    struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
01551    struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
01552    enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
01553    enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED; 
01554    int srccodec, destcodec;
01555 
01556    /* Lock channels */
01557    ast_channel_lock(dest);
01558    while(ast_channel_trylock(src)) {
01559       ast_channel_unlock(dest);
01560       usleep(1);
01561       ast_channel_lock(dest);
01562    }
01563 
01564    /* Find channel driver interfaces */
01565    if (!(destpr = get_proto(dest))) {
01566       if (option_debug)
01567          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
01568       ast_channel_unlock(dest);
01569       ast_channel_unlock(src);
01570       return 0;
01571    }
01572    if (!(srcpr = get_proto(src))) {
01573       if (option_debug)
01574          ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
01575       ast_channel_unlock(dest);
01576       ast_channel_unlock(src);
01577       return 0;
01578    }
01579 
01580    /* Get audio and video interface (if native bridge is possible) */
01581    audio_dest_res = destpr->get_rtp_info(dest, &destp);
01582    video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
01583    audio_src_res = srcpr->get_rtp_info(src, &srcp);
01584    video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
01585 
01586    /* Ensure we have at least one matching codec */
01587    if (srcpr->get_codec)
01588       srccodec = srcpr->get_codec(src);
01589    else
01590       srccodec = 0;
01591    if (destpr->get_codec)
01592       destcodec = destpr->get_codec(dest);
01593    else
01594       destcodec = 0;
01595 
01596    /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
01597    if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
01598       /* Somebody doesn't want to play... */
01599       ast_channel_unlock(dest);
01600       ast_channel_unlock(src);
01601       return 0;
01602    }
01603    ast_rtp_pt_copy(destp, srcp);
01604    if (vdestp && vsrcp)
01605       ast_rtp_pt_copy(vdestp, vsrcp);
01606    if (media) {
01607       /* Bridge early */
01608       if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
01609          ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
01610    }
01611    ast_channel_unlock(dest);
01612    ast_channel_unlock(src);
01613    if (option_debug)
01614       ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
01615    return 1;
01616 }

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

Initializate a RTP session.

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

Definition at line 1949 of file rtp.c.

References ast_rtp_new_with_bindaddr(), io, and sched.

01950 {
01951    struct in_addr ia;
01952 
01953    memset(&ia, 0, sizeof(ia));
01954    return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
01955 }

void ast_rtp_new_init ( struct ast_rtp rtp  ) 

Initialize a new RTP structure.

Definition at line 1849 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().

01850 {
01851    ast_mutex_init(&rtp->bridge_lock);
01852 
01853    rtp->them.sin_family = AF_INET;
01854    rtp->us.sin_family = AF_INET;
01855    rtp->ssrc = ast_random();
01856    rtp->seqno = ast_random() & 0xffff;
01857    ast_set_flag(rtp, FLAG_HAS_DTMF);
01858 
01859    return;
01860 }

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

Initializate a RTP session using an in_addr structure.

This fuction gets called by ast_rtp_new().

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

Definition at line 1862 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, FLAG_CALLBACK_MODE, free, io, LOG_ERROR, rtp_socket(), rtpread(), and sched.

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

01863 {
01864    struct ast_rtp *rtp;
01865    int x;
01866    int first;
01867    int startplace;
01868    
01869    if (!(rtp = ast_calloc(1, sizeof(*rtp))))
01870       return NULL;
01871 
01872    ast_rtp_new_init(rtp);
01873 
01874    rtp->s = rtp_socket();
01875    if (rtp->s < 0) {
01876       free(rtp);
01877       ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
01878       return NULL;
01879    }
01880    if (sched && rtcpenable) {
01881       rtp->sched = sched;
01882       rtp->rtcp = ast_rtcp_new();
01883    }
01884    
01885    /* Select a random port number in the range of possible RTP */
01886    x = (ast_random() % (rtpend-rtpstart)) + rtpstart;
01887    x = x & ~1;
01888    /* Save it for future references. */
01889    startplace = x;
01890    /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
01891    for (;;) {
01892       /* Must be an even port number by RTP spec */
01893       rtp->us.sin_port = htons(x);
01894       rtp->us.sin_addr = addr;
01895       /* If there's rtcp, initialize it as well. */
01896       if (rtp->rtcp) {
01897          rtp->rtcp->us.sin_port = htons(x + 1);
01898          rtp->rtcp->us.sin_addr = addr;
01899       }
01900       /* Try to bind it/them. */
01901       if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
01902          (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
01903          break;
01904       if (!first) {
01905          /* Primary bind succeeded! Gotta recreate it */
01906          close(rtp->s);
01907          rtp->s = rtp_socket();
01908       }
01909       if (errno != EADDRINUSE) {
01910          /* We got an error that wasn't expected, abort! */
01911          ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
01912          close(rtp->s);
01913          if (rtp->rtcp) {
01914             close(rtp->rtcp->s);
01915             free(rtp->rtcp);
01916          }
01917          free(rtp);
01918          return NULL;
01919       }
01920       /* The port was used, increment it (by two). */
01921       x += 2;
01922       /* Did we go over the limit ? */
01923       if (x > rtpend)
01924          /* then, start from the begingig. */
01925          x = (rtpstart + 1) & ~1;
01926       /* Check if we reached the place were we started. */
01927       if (x == startplace) {
01928          /* If so, there's no ports available. */
01929          ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
01930          close(rtp->s);
01931          if (rtp->rtcp) {
01932             close(rtp->rtcp->s);
01933             free(rtp->rtcp);
01934          }
01935          free(rtp);
01936          return NULL;
01937       }
01938    }
01939    rtp->sched = sched;
01940    rtp->io = io;
01941    if (callbackmode) {
01942       rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
01943       ast_set_flag(rtp, FLAG_CALLBACK_MODE);
01944    }
01945    ast_rtp_pt_default(rtp);
01946    return rtp;
01947 }

int ast_rtp_proto_register ( struct ast_rtp_protocol proto  ) 

Register interface to channel driver.

Definition at line 2770 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().

02771 {
02772    struct ast_rtp_protocol *cur;
02773 
02774    AST_LIST_LOCK(&protos);
02775    AST_LIST_TRAVERSE(&protos, cur, list) {   
02776       if (!strcmp(cur->type, proto->type)) {
02777          ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
02778          AST_LIST_UNLOCK(&protos);
02779          return -1;
02780       }
02781    }
02782    AST_LIST_INSERT_HEAD(&protos, proto, list);
02783    AST_LIST_UNLOCK(&protos);
02784    
02785    return 0;
02786 }

void ast_rtp_proto_unregister ( struct ast_rtp_protocol proto  ) 

Unregister interface to channel driver.

Definition at line 2762 of file rtp.c.

References AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, and protos.

Referenced by load_module(), and unload_module().

02763 {
02764    AST_LIST_LOCK(&protos);
02765    AST_LIST_REMOVE(&protos, proto, list);
02766    AST_LIST_UNLOCK(&protos);
02767 }

void ast_rtp_pt_clear ( struct ast_rtp rtp  ) 

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

Definition at line 1383 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().

01384 {
01385    int i;
01386 
01387    if (!rtp)
01388       return;
01389 
01390    ast_mutex_lock(&rtp->bridge_lock);
01391 
01392    for (i = 0; i < MAX_RTP_PT; ++i) {
01393       rtp->current_RTP_PT[i].isAstFormat = 0;
01394       rtp->current_RTP_PT[i].code = 0;
01395    }
01396 
01397    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01398    rtp->rtp_lookup_code_cache_code = 0;
01399    rtp->rtp_lookup_code_cache_result = 0;
01400 
01401    ast_mutex_unlock(&rtp->bridge_lock);
01402 }

void ast_rtp_pt_copy ( struct ast_rtp dest,
struct ast_rtp src 
)

Copy payload types between RTP structures.

Definition at line 1423 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().

01424 {
01425    unsigned int i;
01426 
01427    ast_mutex_lock(&dest->bridge_lock);
01428    ast_mutex_lock(&src->bridge_lock);
01429 
01430    for (i=0; i < MAX_RTP_PT; ++i) {
01431       dest->current_RTP_PT[i].isAstFormat = 
01432          src->current_RTP_PT[i].isAstFormat;
01433       dest->current_RTP_PT[i].code = 
01434          src->current_RTP_PT[i].code; 
01435    }
01436    dest->rtp_lookup_code_cache_isAstFormat = 0;
01437    dest->rtp_lookup_code_cache_code = 0;
01438    dest->rtp_lookup_code_cache_result = 0;
01439 
01440    ast_mutex_unlock(&src->bridge_lock);
01441    ast_mutex_unlock(&dest->bridge_lock);
01442 }

void ast_rtp_pt_default ( struct ast_rtp rtp  ) 

Set payload types to defaults.

Definition at line 1404 of file rtp.c.

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

Referenced by ast_rtp_new_with_bindaddr().

01405 {
01406    int i;
01407 
01408    ast_mutex_lock(&rtp->bridge_lock);
01409 
01410    /* Initialize to default payload types */
01411    for (i = 0; i < MAX_RTP_PT; ++i) {
01412       rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
01413       rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
01414    }
01415 
01416    rtp->rtp_lookup_code_cache_isAstFormat = 0;
01417    rtp->rtp_lookup_code_cache_code = 0;
01418    rtp->rtp_lookup_code_cache_result = 0;
01419 
01420    ast_mutex_unlock(&rtp->bridge_lock);
01421 }

struct ast_frame* ast_rtp_read ( struct ast_rtp rtp  ) 

Definition at line 1081 of file rtp.c.

References ast_backtrace(), 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, 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_ERROR, 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().

01082 {
01083    int res;
01084    struct sockaddr_in sin;
01085    socklen_t len;
01086    unsigned int seqno;
01087    int version;
01088    int payloadtype;
01089    int hdrlen = 12;
01090    int padding;
01091    int mark;
01092    int ext;
01093    unsigned int ssrc;
01094    unsigned int timestamp;
01095    unsigned int *rtpheader;
01096    struct rtpPayloadType rtpPT;
01097    struct ast_rtp *bridged = NULL;
01098    
01099    if( !rtp ) {
01100        ast_log(LOG_ERROR, "ast_rtp_read(): called with rtp == NULL\n");
01101        ast_backtrace();
01102        return &ast_null_frame;
01103    }
01104 
01105    /* If time is up, kill it */
01106    if (rtp->sending_digit)
01107       ast_rtp_senddigit_continuation(rtp);
01108 
01109    len = sizeof(sin);
01110    
01111    /* Cache where the header will go */
01112    res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
01113                0, (struct sockaddr *)&sin, &len);
01114 
01115    rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
01116    if (res < 0) {
01117       if (errno == EBADF)
01118          CRASH;
01119       if (errno != EAGAIN) {
01120          ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up.\n", strerror(errno));
01121          return NULL;
01122       }
01123       return &ast_null_frame;
01124    }
01125    
01126    if (res < hdrlen) {
01127       ast_log(LOG_WARNING, "RTP Read too short\n");
01128       return &ast_null_frame;
01129    }
01130 
01131    /* Get fields */
01132    seqno = ntohl(rtpheader[0]);
01133 
01134    /* Check RTP version */
01135    version = (seqno & 0xC0000000) >> 30;
01136    if (!version) {
01137       if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) &&
01138          (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
01139          memcpy(&rtp->them, &sin, sizeof(rtp->them));
01140       }
01141       return &ast_null_frame;
01142    }
01143 
01144 #if 0 /* Allow to receive RTP stream with closed transmission path */
01145    /* If we don't have the other side's address, then ignore this */
01146    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
01147       return &ast_null_frame;
01148 #endif
01149 
01150    /* Send to whoever send to us if NAT is turned on */
01151    if (rtp->nat) {
01152       if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
01153           (rtp->them.sin_port != sin.sin_port)) {
01154          rtp->them = sin;
01155          if (rtp->rtcp) {
01156             memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
01157             rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
01158          }
01159          rtp->rxseqno = 0;
01160          ast_set_flag(rtp, FLAG_NAT_ACTIVE);
01161          if (option_debug || rtpdebug)
01162             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));
01163       }
01164    }
01165 
01166    /* If we are bridged to another RTP stream, send direct */
01167    if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
01168       return &ast_null_frame;
01169 
01170    if (version != 2)
01171       return &ast_null_frame;
01172 
01173    payloadtype = (seqno & 0x7f0000) >> 16;
01174    padding = seqno & (1 << 29);
01175    mark = seqno & (1 << 23);
01176    ext = seqno & (1 << 28);
01177    seqno &= 0xffff;
01178    timestamp = ntohl(rtpheader[1]);
01179    ssrc = ntohl(rtpheader[2]);
01180    
01181    if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
01182       if (option_debug || rtpdebug)
01183          ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
01184       mark = 1;
01185    }
01186 
01187    rtp->rxssrc = ssrc;
01188    
01189    if (padding) {
01190       /* Remove padding bytes */
01191       res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
01192    }
01193    
01194    if (ext) {
01195       /* RTP Extension present */
01196       hdrlen += 4;
01197       hdrlen += (ntohl(rtpheader[3]) & 0xffff) << 2;
01198       if (option_debug) {
01199          int profile;
01200          profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
01201          if (profile == 0x505a)
01202             ast_log(LOG_DEBUG, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
01203          else
01204             ast_log(LOG_DEBUG, "Found unknown RTP Extensions %x\n", profile);
01205       }
01206    }
01207 
01208    if (res < hdrlen) {
01209       ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
01210       return &ast_null_frame;
01211    }
01212 
01213    rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
01214 
01215    if (rtp->rxcount==1) {
01216       /* This is the first RTP packet successfully received from source */
01217       rtp->seedrxseqno = seqno;
01218    }
01219 
01220    /* Do not schedule RR if RTCP isn't run */
01221    if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
01222       /* Schedule transmission of Receiver Report */
01223       rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
01224    }
01225    if ( (int)rtp->lastrxseqno - (int)seqno  > 100) /* if so it would indicate that the sender cycled; allow for misordering */
01226       rtp->cycles += RTP_SEQ_MOD;
01227 
01228    rtp->lastrxseqno = seqno;
01229    
01230    if (rtp->themssrc==0)
01231       rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
01232    
01233    if (rtp_debug_test_addr(&sin))
01234       ast_verbose("Got  RTP packet from    %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
01235          ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
01236 
01237    rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
01238    if (!rtpPT.isAstFormat) {
01239       struct ast_frame *f = NULL;
01240 
01241       /* This is special in-band data that's not one of our codecs */
01242       if (rtpPT.code == AST_RTP_DTMF) {
01243          /* It's special -- rfc2833 process it */
01244          if (rtp_debug_test_addr(&sin)) {
01245             unsigned char *data;
01246             unsigned int event;
01247             unsigned int event_end;
01248             unsigned int duration;
01249             data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
01250             event = ntohl(*((unsigned int *)(data)));
01251             event >>= 24;
01252             event_end = ntohl(*((unsigned int *)(data)));
01253             event_end <<= 8;
01254             event_end >>= 24;
01255             duration = ntohl(*((unsigned int *)(data)));
01256             duration &= 0xFFFF;
01257             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);
01258          }
01259          f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
01260       } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
01261          /* It's really special -- process it the Cisco way */
01262          if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
01263             f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01264             rtp->lastevent = seqno;
01265          }
01266       } else if (rtpPT.code == AST_RTP_CN) {
01267          /* Comfort Noise */
01268          f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
01269       } else {
01270          ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
01271       }
01272       return f ? f : &ast_null_frame;
01273    }
01274    rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
01275    rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
01276 
01277    if (!rtp->lastrxts)
01278       rtp->lastrxts = timestamp;
01279 
01280    rtp->rxseqno = seqno;
01281 
01282    /* Record received timestamp as last received now */
01283    rtp->lastrxts = timestamp;
01284 
01285    rtp->f.mallocd = 0;
01286    rtp->f.datalen = res - hdrlen;
01287    rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
01288    rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
01289    if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
01290       rtp->f.samples = ast_codec_get_samples(&rtp->f);
01291       if (rtp->f.subclass == AST_FORMAT_SLINEAR) 
01292          ast_frame_byteswap_be(&rtp->f);
01293       calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
01294       /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
01295       rtp->f.has_timing_info = 1;
01296       rtp->f.ts = timestamp / 8;
01297       rtp->f.len = rtp->f.samples / 8;
01298       rtp->f.seqno = seqno;
01299    } else {
01300       /* Video -- samples is # of samples vs. 90000 */
01301       if (!rtp->lastividtimestamp)
01302          rtp->lastividtimestamp = timestamp;
01303       rtp->f.samples = timestamp - rtp->lastividtimestamp;
01304       rtp->lastividtimestamp = timestamp;
01305       rtp->f.delivery.tv_sec = 0;
01306       rtp->f.delivery.tv_usec = 0;
01307       if (mark)
01308          rtp->f.subclass |= 0x1;
01309       
01310    }
01311    rtp->f.src = "RTP";
01312    return &rtp->f;
01313 }

int ast_rtp_reload ( void   ) 

Definition at line 3637 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().

03638 {
03639    struct ast_config *cfg;
03640    const char *s;
03641 
03642    rtpstart = 5000;
03643    rtpend = 31000;
03644    dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03645    cfg = ast_config_load("rtp.conf");
03646    if (cfg) {
03647       if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
03648          rtpstart = atoi(s);
03649          if (rtpstart < 1024)
03650             rtpstart = 1024;
03651          if (rtpstart > 65535)
03652             rtpstart = 65535;
03653       }
03654       if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
03655          rtpend = atoi(s);
03656          if (rtpend < 1024)
03657             rtpend = 1024;
03658          if (rtpend > 65535)
03659             rtpend = 65535;
03660       }
03661       if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
03662          rtcpinterval = atoi(s);
03663          if (rtcpinterval == 0)
03664             rtcpinterval = 0; /* Just so we're clear... it's zero */
03665          if (rtcpinterval < RTCP_MIN_INTERVALMS)
03666             rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
03667          if (rtcpinterval > RTCP_MAX_INTERVALMS)
03668             rtcpinterval = RTCP_MAX_INTERVALMS;
03669       }
03670       if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
03671 #ifdef SO_NO_CHECK
03672          if (ast_false(s))
03673             nochecksums = 1;
03674          else
03675             nochecksums = 0;
03676 #else
03677          if (ast_false(s))
03678             ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
03679 #endif
03680       }
03681       if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
03682          dtmftimeout = atoi(s);
03683          if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
03684             ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
03685                dtmftimeout, DEFAULT_DTMF_TIMEOUT);
03686             dtmftimeout = DEFAULT_DTMF_TIMEOUT;
03687          };
03688       }
03689       ast_config_destroy(cfg);
03690    }
03691    if (rtpstart >= rtpend) {
03692       ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
03693       rtpstart = 5000;
03694       rtpend = 31000;
03695    }
03696    if (option_verbose > 1)
03697       ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
03698    return 0;
03699 }

void ast_rtp_reset ( struct ast_rtp rtp  ) 

Definition at line 2023 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.

02024 {
02025    memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
02026    memset(&rtp->txcore, 0, sizeof(rtp->txcore));
02027    memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
02028    rtp->lastts = 0;
02029    rtp->lastdigitts = 0;
02030    rtp->lastrxts = 0;
02031    rtp->lastividtimestamp = 0;
02032    rtp->lastovidtimestamp = 0;
02033    rtp->lasteventseqn = 0;
02034    rtp->lastevent = 0;
02035    rtp->lasttxformat = 0;
02036    rtp->lastrxformat = 0;
02037    rtp->dtmfcount = 0;
02038    rtp->dtmfsamples = 0;
02039    rtp->seqno = 0;
02040    rtp->rxseqno = 0;
02041 }

int ast_rtp_sendcng ( struct ast_rtp rtp,
int  level 
)

generate comfort noice (CNG)

Definition at line 2530 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, 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().

02531 {
02532    unsigned int *rtpheader;
02533    int hdrlen = 12;
02534    int res;
02535    int payload;
02536    char data[256];
02537    level = 127 - (level & 0x7f);
02538    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
02539 
02540    /* If we have no peer, return immediately */ 
02541    if (!rtp->them.sin_addr.s_addr)
02542       return 0;
02543 
02544    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02545 
02546    /* Get a pointer to the header */
02547    rtpheader = (unsigned int *)data;
02548    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
02549    rtpheader[1] = htonl(rtp->lastts);
02550    rtpheader[2] = htonl(rtp->ssrc); 
02551    data[12] = level;
02552    if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
02553       res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
02554       if (res <0) 
02555          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));
02556       if (rtp_debug_test_addr(&rtp->them))
02557          ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
02558                , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);         
02559          
02560    }
02561    return 0;
02562 }

int ast_rtp_senddigit_begin ( struct ast_rtp rtp,
char  digit 
)

Send begin frames for DTMF.

Definition at line 2133 of file rtp.c.

References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_tvadd(), ast_verbose(), ast_rtp::dtmfmute, 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().

02134 {
02135    unsigned int *rtpheader;
02136    int hdrlen = 12, res = 0, i = 0, payload = 0;
02137    char data[256];
02138 
02139    if ((digit <= '9') && (digit >= '0'))
02140       digit -= '0';
02141    else if (digit == '*')
02142       digit = 10;
02143    else if (digit == '#')
02144       digit = 11;
02145    else if ((digit >= 'A') && (digit <= 'D'))
02146       digit = digit - 'A' + 12;
02147    else if ((digit >= 'a') && (digit <= 'd'))
02148       digit = digit - 'a' + 12;
02149    else {
02150       ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
02151       return 0;
02152    }
02153 
02154    /* If we have no peer, return immediately */ 
02155    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
02156       return 0;
02157 
02158    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
02159 
02160    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
02161    rtp->send_duration = 160;
02162    
02163    /* Get a pointer to the header */
02164    rtpheader = (unsigned int *)data;
02165    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
02166    rtpheader[1] = htonl(rtp->lastdigitts);
02167    rtpheader[2] = htonl(rtp->ssrc); 
02168 
02169    for (i = 0; i < 2; i++) {
02170       rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
02171       res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
02172       if (res < 0) 
02173          ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
02174             ast_inet_ntoa(rtp->them.sin_addr),
02175             ntohs(rtp->them.sin_port), strerror(errno));
02176       if (rtp_debug_test_addr(&rtp->them))
02177          ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
02178                 ast_inet_ntoa(rtp->them.sin_addr),
02179                 ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
02180       /* Increment sequence number */
02181       rtp->seqno++;
02182       /* Increment duration */
02183       rtp->send_duration += 160;
02184       /* Clear marker bit and set seqno */
02185       rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
02186    }
02187 
02188    /* Since we received a begin, we can safely store the digit and disable any compensation */
02189    rtp->sending_digit = 1;
02190    rtp->send_digit = digit;
02191    rtp->send_payload = payload;
02192 
02193    return 0;
02194 }

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 
)

Make a note of a RTP payload type that was seen in a SDP "m=" line. By default, use the well-known value for this type (although it may still be set to a different value by a subsequent "a=rtpmap:" line).

Definition at line 1622 of file rtp.c.

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

Referenced by gtalk_newcall(), and process_sdp().

01623 {
01624    if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0) 
01625       return; /* bogus payload type */
01626 
01627    ast_mutex_lock(&rtp->bridge_lock);
01628    rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
01629    ast_mutex_unlock(&rtp->bridge_lock);
01630 } 

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

Definition at line 1966 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().

01967 {
01968    rtp->them.sin_port = them->sin_port;
01969    rtp->them.sin_addr = them->sin_addr;
01970    if (rtp->rtcp) {
01971       rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
01972       rtp->rtcp->them.sin_addr = them->sin_addr;
01973    }
01974    rtp->rxseqno = 0;
01975 }

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 }

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

Make a note of a RTP payload type (with MIME type) that was seen in an SDP "a=rtpmap:" line.

Definition at line 1635 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().

01638 {
01639    unsigned int i;
01640 
01641    if (pt < 0 || pt > MAX_RTP_PT) 
01642       return; /* bogus payload type */
01643    
01644    ast_mutex_lock(&rtp->bridge_lock);
01645 
01646    for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
01647       if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
01648           strcasecmp(mimeType, mimeTypes[i].type) == 0) {
01649          rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
01650          if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
01651              mimeTypes[i].payloadType.isAstFormat &&
01652              (options & AST_RTP_OPT_G726_NONSTANDARD))
01653             rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
01654          break;
01655       }
01656    }
01657 
01658    ast_mutex_unlock(&rtp->bridge_lock);
01659 
01660    return;
01661 } 

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 1957 of file rtp.c.

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

Referenced by __oh323_rtp_create(), and sip_alloc().

01958 {
01959    int res;
01960 
01961    if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
01962       ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
01963    return res;
01964 }

void ast_rtp_stop ( struct ast_rtp rtp  ) 

Definition at line 2006 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().

02007 {
02008    if (rtp->rtcp && rtp->rtcp->schedid > 0) {
02009       ast_sched_del(rtp->sched, rtp->rtcp->schedid);
02010       rtp->rtcp->schedid = -1;
02011    }
02012 
02013    memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
02014    memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
02015    if (rtp->rtcp) {
02016       memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
02017       memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
02018    }
02019    
02020    ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
02021 }

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

Definition at line 402 of file rtp.c.

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

Referenced by gtalk_update_stun().

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

int ast_rtp_write ( struct ast_rtp rtp,
struct ast_frame f 
)

Definition at line 2682 of file rtp.c.

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

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

02683 {
02684    struct ast_frame *f;
02685    int codec;
02686    int hdrlen = 12;
02687    int subclass;
02688    
02689 
02690    /* If we have no peer, return immediately */ 
02691    if (!rtp->them.sin_addr.s_addr)
02692       return 0;
02693 
02694    /* If there is no data length, return immediately */
02695    if (!_f->datalen) 
02696       return 0;
02697    
02698    /* Make sure we have enough space for RTP header */
02699    if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
02700       ast_log(LOG_WARNING, "RTP can only send voice and video\n");
02701       return -1;
02702    }
02703 
02704    subclass = _f->subclass;
02705    if (_f->frametype == AST_FRAME_VIDEO)
02706       subclass &= ~0x1;
02707 
02708    codec = ast_rtp_lookup_code(rtp, 1, subclass);
02709    if (codec < 0) {
02710       ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
02711       return -1;
02712    }
02713 
02714    if (rtp->lasttxformat != subclass) {
02715       /* New format, reset the smoother */
02716       if (option_debug)
02717          ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
02718       rtp->lasttxformat = subclass;
02719       if (rtp->smoother)
02720          ast_smoother_free(rtp->smoother);
02721       rtp->smoother = NULL;
02722    }
02723 
02724    if (!rtp->smoother) {
02725       struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
02726       if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
02727          if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
02728             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));
02729             return -1;
02730          }
02731          if (fmt.flags)
02732             ast_smoother_set_flags(rtp->smoother, fmt.flags);
02733          if (option_debug)
02734             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));
02735       }
02736    }
02737    if (rtp->smoother) {
02738       if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
02739          ast_smoother_feed_be(rtp->smoother, _f);
02740       } else {
02741          ast_smoother_feed(rtp->smoother, _f);
02742       }
02743 
02744       while((f = ast_smoother_read(rtp->smoother)))
02745          ast_rtp_raw_write(rtp, f, codec);
02746    } else {
02747            /* Don't buffer outgoing frames; send them one-per-packet: */
02748       if (_f->offset < hdrlen) {
02749          f = ast_frdup(_f);
02750       } else {
02751          f = _f;
02752       }
02753       ast_rtp_raw_write(rtp, f, codec);
02754       if (f != _f)
02755          ast_frfree(f);
02756    }
02757       
02758    return 0;
02759 }


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