Thu Apr 28 2011 17:16:20

Asterisk developer's documentation


rtp.h File Reference

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

#include "asterisk/network.h"
#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
#include "asterisk/linkedlists.h"
#include "asterisk/io.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
 This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem. More...
struct  ast_rtp_quality
 RTCP quality report storage. More...
struct  rtpPayloadType
 The value of each payload format mapping: More...

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
#define RED_MAX_GENERATION   5

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) }
enum  ast_rtp_qos_vars {
  AST_RTP_TXCOUNT, AST_RTP_RXCOUNT, AST_RTP_TXJITTER, AST_RTP_RXJITTER,
  AST_RTP_RXPLOSS, AST_RTP_TXPLOSS, AST_RTP_RTT
}
 

Variables used in ast_rtcp_get function.

More...
enum  ast_rtp_quality_type { RTPQOS_SUMMARY = 0, RTPQOS_JITTER, RTPQOS_LOSS, RTPQOS_RTT }

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)
 The RTP bridge.
void ast_rtp_change_source (struct ast_rtp *rtp)
 Indicate that we need to set the marker bit and change the ssrc.
int ast_rtp_codec_getformat (int pt)
 get format from predefined dynamic payload format
struct ast_codec_prefast_rtp_codec_getpref (struct ast_rtp *rtp)
 Get codec preference.
void ast_rtp_codec_setpref (struct ast_rtp *rtp, struct ast_codec_pref *prefs)
 Set codec preference.
void ast_rtp_destroy (struct ast_rtp *rtp)
int ast_rtp_early_bridge (struct ast_channel *c0, struct ast_channel *c1)
 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)
int ast_rtp_get_qos (struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen)
 Get QOS stats on a RTP channel.
unsigned int ast_rtp_get_qosvalue (struct ast_rtp *rtp, enum ast_rtp_qos_vars value)
 Return RTP and RTCP QoS values.
char * ast_rtp_get_quality (struct ast_rtp *rtp, struct ast_rtp_quality *qual, enum ast_rtp_quality_type qtype)
 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:
unsigned int ast_rtp_lookup_sample_rate (int isAstFormat, int code)
 Get the sample rate associated with known RTP payload types.
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.
void ast_rtp_new_source (struct ast_rtp *rtp)
 Indicate that we need to set the marker bit.
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 an RTP channel client.
void ast_rtp_proto_unregister (struct ast_rtp_protocol *proto)
 Unregister an RTP channel client.
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)
int ast_rtp_senddigit_end_with_duration (struct ast_rtp *rtp, char digit, unsigned int duration)
 Send end packets for DTMF.
void ast_rtp_set_alt_peer (struct ast_rtp *rtp, struct sockaddr_in *alt)
 set potential alternate source for RTP media
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)
 Set payload type to a known MIME media type for a codec.
int ast_rtp_set_rtpmap_type_rate (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype, enum ast_rtp_options options, unsigned int sample_rate)
 Set payload type to a known MIME media type for a codec with a specific sample rate.
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_set_vars (struct ast_channel *chan, struct ast_rtp *rtp)
 Set RTPAUDIOQOS(...) variables on a channel when it is being hung up.
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)
int ast_rtp_setqos (struct ast_rtp *rtp, int tos, int cos, char *desc)
void ast_rtp_setstun (struct ast_rtp *rtp, int stun_enable)
 Enable STUN capability.
void ast_rtp_stop (struct ast_rtp *rtp)
void ast_rtp_stun_request (struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username)
 Send STUN request for an RTP socket Deprecated, this is just a wrapper for ast_rtp_stun_request()
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)
int ast_stun_request (int s, struct sockaddr_in *dst, const char *username, struct sockaddr_in *answer)
 Generic STUN request send a generic stun request to the server specified.
void red_buffer_t140 (struct ast_rtp *rtp, struct ast_frame *f)
 Buffer t.140 data.
int rtp_red_init (struct ast_rtp *rtp, int ti, int *pt, int num_gen)
 Initalize t.140 redudancy.

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_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 57 of file rtp.h.

#define RED_MAX_GENERATION   5

T.140 Redundancy Maxium number of generations

Definition at line 55 of file rtp.h.

Referenced by process_sdp_a_text().


Typedef Documentation

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

RTP callback structure

Definition at line 130 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 63 of file rtp.h.

                        {
   /*! Failed to find the RTP structure */
   AST_RTP_GET_FAILED = 0,
   /*! RTP structure exists but true native bridge can not occur so try partial */
   AST_RTP_TRY_PARTIAL,
   /*! RTP structure exists and native bridge can occur */
   AST_RTP_TRY_NATIVE,
};
Enumerator:
AST_RTP_OPT_G726_NONSTANDARD 

Definition at line 59 of file rtp.h.

                     {
   AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
};

Variables used in ast_rtcp_get function.

Enumerator:
AST_RTP_TXCOUNT 
AST_RTP_RXCOUNT 
AST_RTP_TXJITTER 
AST_RTP_RXJITTER 
AST_RTP_RXPLOSS 
AST_RTP_TXPLOSS 
AST_RTP_RTT 

Definition at line 73 of file rtp.h.

Enumerator:
RTPQOS_SUMMARY 
RTPQOS_JITTER 
RTPQOS_LOSS 
RTPQOS_RTT 

Definition at line 109 of file rtp.h.


Function Documentation

int ast_rtcp_fd ( struct ast_rtp rtp)

Definition at line 722 of file rtp.c.

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

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

{
   if (rtp->rtcp)
      return rtp->rtcp->s;
   return -1;
}
struct ast_frame* ast_rtcp_read ( struct ast_rtp rtp) [read]

Definition at line 1182 of file rtp.c.

References ast_rtcp::accumulated_transit, ast_rtcp::altthem, ast_assert, AST_CONTROL_VIDUPDATE, ast_debug, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), ast_null_frame, ast_verbose(), ast_frame::datalen, errno, EVENT_FLAG_REPORTING, ast_rtp::f, f, ast_frame::frametype, len(), LOG_DEBUG, LOG_WARNING, ast_frame::mallocd, manager_event, ast_rtcp::maxrtt, ast_rtcp::minrtt, ast_rtp::nat, normdev_compute(), ast_rtcp::normdevrtt, option_debug, ast_rtcp::reported_jitter, ast_rtcp::reported_jitter_count, ast_rtcp::reported_lost, ast_rtcp::reported_maxjitter, ast_rtcp::reported_maxlost, ast_rtcp::reported_minjitter, ast_rtcp::reported_minlost, ast_rtcp::reported_normdev_jitter, ast_rtcp::reported_normdev_lost, ast_rtcp::reported_stdev_jitter, ast_rtcp::reported_stdev_lost, ast_rtp::rtcp, rtcp_debug_test_addr(), ast_rtcp::rtcp_info, RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, ast_rtcp::rtt, ast_rtcp::rtt_count, ast_rtcp::rxlsr, ast_rtcp::s, ast_frame::samples, ast_rtcp::soc, ast_rtcp::spc, ast_frame::src, stddev_compute(), ast_rtcp::stdevrtt, ast_frame::subclass, ast_rtcp::them, ast_rtcp::themrxlsr, and timeval2ntp().

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

{
   socklen_t len;
   int position, i, packetwords;
   int res;
   struct sockaddr_in sock_in;
   unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
   unsigned int *rtcpheader;
   int pt;
   struct timeval now;
   unsigned int length;
   int rc;
   double rttsec;
   uint64_t rtt = 0;
   unsigned int dlsr;
   unsigned int lsr;
   unsigned int msw;
   unsigned int lsw;
   unsigned int comp;
   struct ast_frame *f = &ast_null_frame;
   
   double reported_jitter;
   double reported_normdev_jitter_current;
   double normdevrtt_current;
   double reported_lost;
   double reported_normdev_lost_current;

   if (!rtp || !rtp->rtcp)
      return &ast_null_frame;

   len = sizeof(sock_in);
   
   res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
               0, (struct sockaddr *)&sock_in, &len);
   rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
   
   if (res < 0) {
      ast_assert(errno != EBADF);
      if (errno != EAGAIN) {
         ast_log(LOG_WARNING, "RTCP Read error: %s.  Hanging up.\n", strerror(errno));
         return NULL;
      }
      return &ast_null_frame;
   }

   packetwords = res / 4;
   
   if (rtp->nat) {
      /* Send to whoever sent to us */
      if (((rtp->rtcp->them.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
          (rtp->rtcp->them.sin_port != sock_in.sin_port)) && 
          ((rtp->rtcp->altthem.sin_addr.s_addr != sock_in.sin_addr.s_addr) || 
          (rtp->rtcp->altthem.sin_port != sock_in.sin_port))) {
         memcpy(&rtp->rtcp->them, &sock_in, sizeof(rtp->rtcp->them));
         if (option_debug || rtpdebug)
            ast_debug(0, "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));
      }
   }

   ast_debug(1, "Got RTCP report of %d bytes\n", res);

   /* Process a compound packet */
   position = 0;
   while (position < packetwords) {
      i = position;
      length = ntohl(rtcpheader[i]);
      pt = (length & 0xff0000) >> 16;
      rc = (length & 0x1f000000) >> 24;
      length &= 0xffff;
 
      if ((i + length) > packetwords) {
         if (option_debug || rtpdebug)
            ast_log(LOG_DEBUG, "RTCP Read too short\n");
         return &ast_null_frame;
      }
      
      if (rtcp_debug_test_addr(&sock_in)) {
         ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port));
         ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
         ast_verbose("Reception reports: %d\n", rc);
         ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
      }
 
      i += 2; /* Advance past header and ssrc */
      
      switch (pt) {
      case RTCP_PT_SR:
         gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
         rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
         rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
         rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
 
         if (rtcp_debug_test_addr(&sock_in)) {
            ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
            ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
            ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
         }
         i += 5;
         if (rc < 1)
            break;
         /* Intentional fall through */
      case RTCP_PT_RR:
         /* Don't handle multiple reception reports (rc > 1) yet */
         /* Calculate RTT per RFC */
         gettimeofday(&now, NULL);
         timeval2ntp(now, &msw, &lsw);
         if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */
            comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
            lsr = ntohl(rtcpheader[i + 4]);
            dlsr = ntohl(rtcpheader[i + 5]);
            rtt = comp - lsr - dlsr;

            /* Convert end to end delay to usec (keeping the calculation in 64bit space)
               sess->ee_delay = (eedelay * 1000) / 65536; */
            if (rtt < 4294) {
                rtt = (rtt * 1000000) >> 16;
            } else {
                rtt = (rtt * 1000) >> 16;
                rtt *= 1000;
            }
            rtt = rtt / 1000.;
            rttsec = rtt / 1000.;
            rtp->rtcp->rtt = rttsec;

            if (comp - dlsr >= lsr) {
               rtp->rtcp->accumulated_transit += rttsec;

               if (rtp->rtcp->rtt_count == 0) 
                  rtp->rtcp->minrtt = rttsec;

               if (rtp->rtcp->maxrtt<rttsec)
                  rtp->rtcp->maxrtt = rttsec;

               if (rtp->rtcp->minrtt>rttsec)
                  rtp->rtcp->minrtt = rttsec;

               normdevrtt_current = normdev_compute(rtp->rtcp->normdevrtt, rttsec, rtp->rtcp->rtt_count);

               rtp->rtcp->stdevrtt = stddev_compute(rtp->rtcp->stdevrtt, rttsec, rtp->rtcp->normdevrtt, normdevrtt_current, rtp->rtcp->rtt_count);

               rtp->rtcp->normdevrtt = normdevrtt_current;

               rtp->rtcp->rtt_count++;
            } else if (rtcp_debug_test_addr(&sock_in)) {
               ast_verbose("Internal RTCP NTP clock skew detected: "
                        "lsr=%u, now=%u, dlsr=%u (%d:%03dms), "
                        "diff=%d\n",
                        lsr, comp, dlsr, dlsr / 65536,
                        (dlsr % 65536) * 1000 / 65536,
                        dlsr - (comp - lsr));
            }
         }

         rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
         reported_jitter = (double) rtp->rtcp->reported_jitter;

         if (rtp->rtcp->reported_jitter_count == 0) 
            rtp->rtcp->reported_minjitter = reported_jitter;

         if (reported_jitter < rtp->rtcp->reported_minjitter) 
            rtp->rtcp->reported_minjitter = reported_jitter;

         if (reported_jitter > rtp->rtcp->reported_maxjitter) 
            rtp->rtcp->reported_maxjitter = reported_jitter;

         reported_normdev_jitter_current = normdev_compute(rtp->rtcp->reported_normdev_jitter, reported_jitter, rtp->rtcp->reported_jitter_count);

         rtp->rtcp->reported_stdev_jitter = stddev_compute(rtp->rtcp->reported_stdev_jitter, reported_jitter, rtp->rtcp->reported_normdev_jitter, reported_normdev_jitter_current, rtp->rtcp->reported_jitter_count);

         rtp->rtcp->reported_normdev_jitter = reported_normdev_jitter_current;

         rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;

         reported_lost = (double) rtp->rtcp->reported_lost;

         /* using same counter as for jitter */
         if (rtp->rtcp->reported_jitter_count == 0)
            rtp->rtcp->reported_minlost = reported_lost;

         if (reported_lost < rtp->rtcp->reported_minlost)
            rtp->rtcp->reported_minlost = reported_lost;

         if (reported_lost > rtp->rtcp->reported_maxlost) 
            rtp->rtcp->reported_maxlost = reported_lost;

         reported_normdev_lost_current = normdev_compute(rtp->rtcp->reported_normdev_lost, reported_lost, rtp->rtcp->reported_jitter_count);

         rtp->rtcp->reported_stdev_lost = stddev_compute(rtp->rtcp->reported_stdev_lost, reported_lost, rtp->rtcp->reported_normdev_lost, reported_normdev_lost_current, rtp->rtcp->reported_jitter_count);

         rtp->rtcp->reported_normdev_lost = reported_normdev_lost_current;

         rtp->rtcp->reported_jitter_count++;

         if (rtcp_debug_test_addr(&sock_in)) {
            ast_verbose("  Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
            ast_verbose("  Packets lost so far: %d\n", rtp->rtcp->reported_lost);
            ast_verbose("  Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
            ast_verbose("  Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
            ast_verbose("  Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
            ast_verbose("  Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
            ast_verbose("  DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
            if (rtt)
               ast_verbose("  RTT: %lu(sec)\n", (unsigned long) rtt);
         }

         if (rtt) {
            manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From: %s:%d\r\n"
                            "PT: %d(%s)\r\n"
                            "ReceptionReports: %d\r\n"
                            "SenderSSRC: %u\r\n"
                            "FractionLost: %ld\r\n"
                            "PacketsLost: %d\r\n"
                            "HighestSequence: %ld\r\n"
                            "SequenceNumberCycles: %ld\r\n"
                            "IAJitter: %u\r\n"
                            "LastSR: %lu.%010lu\r\n"
                            "DLSR: %4.4f(sec)\r\n"
                            "RTT: %llu(sec)\r\n",
                            ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port),
                            pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
                            rc,
                            rtcpheader[i + 1],
                            (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
                            rtp->rtcp->reported_lost,
                            (long) (ntohl(rtcpheader[i + 2]) & 0xffff),
                            (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
                            rtp->rtcp->reported_jitter,
                            (unsigned long) ntohl(rtcpheader[i + 4]) >> 16, ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
                            ntohl(rtcpheader[i + 5])/65536.0,
                            (unsigned long long)rtt);
         } else {
            manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From: %s:%d\r\n"
                            "PT: %d(%s)\r\n"
                            "ReceptionReports: %d\r\n"
                            "SenderSSRC: %u\r\n"
                            "FractionLost: %ld\r\n"
                            "PacketsLost: %d\r\n"
                            "HighestSequence: %ld\r\n"
                            "SequenceNumberCycles: %ld\r\n"
                            "IAJitter: %u\r\n"
                            "LastSR: %lu.%010lu\r\n"
                            "DLSR: %4.4f(sec)\r\n",
                            ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port),
                            pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
                            rc,
                            rtcpheader[i + 1],
                            (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
                            rtp->rtcp->reported_lost,
                            (long) (ntohl(rtcpheader[i + 2]) & 0xffff),
                            (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
                            rtp->rtcp->reported_jitter,
                            (unsigned long) ntohl(rtcpheader[i + 4]) >> 16,
                            ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
                            ntohl(rtcpheader[i + 5])/65536.0);
         }
         break;
      case RTCP_PT_FUR:
         if (rtcp_debug_test_addr(&sock_in))
            ast_verbose("Received an RTCP Fast Update Request\n");
         rtp->f.frametype = AST_FRAME_CONTROL;
         rtp->f.subclass = AST_CONTROL_VIDUPDATE;
         rtp->f.datalen = 0;
         rtp->f.samples = 0;
         rtp->f.mallocd = 0;
         rtp->f.src = "RTP";
         f = &rtp->f;
         break;
      case RTCP_PT_SDES:
         if (rtcp_debug_test_addr(&sock_in))
            ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
         break;
      case RTCP_PT_BYE:
         if (rtcp_debug_test_addr(&sock_in))
            ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
         break;
      default:
         ast_debug(1, "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));
         break;
      }
      position += (length + 1);
   }
   rtp->rtcp->rtcp_info = 1;  
   return f;
}
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 3357 of file rtp.c.

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

{
   struct ast_rtp *rtp = data;
   int res;

   rtp->rtcp->sendfur = 1;
   res = ast_rtcp_write(data);
   
   return res;
}
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 496 of file rtp.c.

Referenced by process_sdp().

{
   return sizeof(struct ast_rtp);
}
int ast_rtp_bridge ( struct ast_channel c0,
struct ast_channel c1,
int  flags,
struct ast_frame **  fo,
struct ast_channel **  rc,
int  timeoutms 
)

The RTP bridge.

Definition at line 4456 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_check_hangup(), ast_codec_pref_getsize(), ast_debug, ast_log(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_set_flag, ast_test_flag, ast_verb, bridge_native_loop(), bridge_p2p_loop(), ast_format_list::cur_ms, FLAG_HAS_DTMF, FLAG_P2P_NEED_DTMF, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_trtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, ast_rtp::pref, ast_channel::rawreadformat, ast_channel::rawwriteformat, ast_channel_tech::send_digit_begin, ast_channel::tech, and ast_channel::tech_pvt.

{
   struct ast_rtp *p0 = NULL, *p1 = NULL;    /* Audio RTP Channels */
   struct ast_rtp *vp0 = NULL, *vp1 = NULL;  /* Video RTP channels */
   struct ast_rtp *tp0 = NULL, *tp1 = NULL;  /* Text RTP channels */
   struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
   enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED, text_p0_res = AST_RTP_GET_FAILED;
   enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED, text_p1_res = AST_RTP_GET_FAILED;
   enum ast_bridge_result res = AST_BRIDGE_FAILED;
   int codec0 = 0, codec1 = 0;
   void *pvt0 = NULL, *pvt1 = NULL;

   /* Lock channels */
   ast_channel_lock(c0);
   while (ast_channel_trylock(c1)) {
      ast_channel_unlock(c0);
      usleep(1);
      ast_channel_lock(c0);
   }

   /* Ensure neither channel got hungup during lock avoidance */
   if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
      ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED;
   }
      
   /* Find channel driver interfaces */
   if (!(pr0 = get_proto(c0))) {
      ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED;
   }
   if (!(pr1 = get_proto(c1))) {
      ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED;
   }

   /* Get channel specific interface structures */
   pvt0 = c0->tech_pvt;
   pvt1 = c1->tech_pvt;

   /* Get audio and video interface (if native bridge is possible) */
   audio_p0_res = pr0->get_rtp_info(c0, &p0);
   video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
   text_p0_res = pr0->get_trtp_info ? pr0->get_trtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
   audio_p1_res = pr1->get_rtp_info(c1, &p1);
   video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
   text_p1_res = pr1->get_trtp_info ? pr1->get_trtp_info(c1, &vp1) : AST_RTP_GET_FAILED;

   /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
   if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
      audio_p0_res = AST_RTP_GET_FAILED;
   if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
      audio_p1_res = AST_RTP_GET_FAILED;

   /* Check if a bridge is possible (partial/native) */
   if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
      /* Somebody doesn't want to play... */
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED_NOWARN;
   }

   /* If we need to feed DTMF frames into the core then only do a partial native bridge */
   if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
      ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
      audio_p0_res = AST_RTP_TRY_PARTIAL;
   }

   if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
      ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
      audio_p1_res = AST_RTP_TRY_PARTIAL;
   }

   /* If both sides are not using the same method of DTMF transmission 
    * (ie: one is RFC2833, other is INFO... then we can not do direct media. 
    * --------------------------------------------------
    * | DTMF Mode |  HAS_DTMF  |  Accepts Begin Frames |
    * |-----------|------------|-----------------------|
    * | Inband    | False      | True                  |
    * | RFC2833   | True       | True                  |
    * | SIP INFO  | False      | False                 |
    * --------------------------------------------------
    * However, if DTMF from both channels is being monitored by the core, then
    * we can still do packet-to-packet bridging, because passing through the 
    * core will handle DTMF mode translation.
    */
   if ((ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
      (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
      if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
         ast_channel_unlock(c0);
         ast_channel_unlock(c1);
         return AST_BRIDGE_FAILED_NOWARN;
      }
      audio_p0_res = AST_RTP_TRY_PARTIAL;
      audio_p1_res = AST_RTP_TRY_PARTIAL;
   }

   /* If we need to feed frames into the core don't do a P2P bridge */
   if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) ||
       (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) {
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED_NOWARN;
   }

   /* Get codecs from both sides */
   codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
   codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
   if (codec0 && codec1 && !(codec0 & codec1)) {
      /* Hey, we can't do native bridging if both parties speak different codecs */
      ast_debug(3, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED_NOWARN;
   }

   /* If either side can only do a partial bridge, then don't try for a true native bridge */
   if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
      struct ast_format_list fmt0, fmt1;

      /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
      if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
         ast_debug(1, "Cannot packet2packet bridge - raw formats are incompatible\n");
         ast_channel_unlock(c0);
         ast_channel_unlock(c1);
         return AST_BRIDGE_FAILED_NOWARN;
      }
      /* They must also be using the same packetization */
      fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
      fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
      if (fmt0.cur_ms != fmt1.cur_ms) {
         ast_debug(1, "Cannot packet2packet bridge - packetization settings prevent it\n");
         ast_channel_unlock(c0);
         ast_channel_unlock(c1);
         return AST_BRIDGE_FAILED_NOWARN;
      }

      ast_verb(3, "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
      res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
   } else {
      ast_verb(3, "Native bridging %s and %s\n", c0->name, c1->name);
      res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, tp0, tp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
   }

   return res;
}
void ast_rtp_change_source ( struct ast_rtp rtp)

Indicate that we need to set the marker bit and change the ssrc.

Definition at line 2692 of file rtp.c.

References ast_debug, ast_random(), ast_rtp::set_marker_bit, and ast_rtp::ssrc.

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

{
   if (rtp) {
      unsigned int ssrc = ast_random();

      rtp->set_marker_bit = 1;
      ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
      rtp->ssrc = ssrc;
   }
}
int ast_rtp_codec_getformat ( int  pt)

get format from predefined dynamic payload format

Definition at line 3837 of file rtp.c.

References rtpPayloadType::code, and MAX_RTP_PT.

Referenced by process_sdp_a_audio().

{
   if (pt < 0 || pt >= MAX_RTP_PT)
      return 0; /* bogus payload type */

   if (static_RTP_PT[pt].isAstFormat)
      return static_RTP_PT[pt].code;
   else
      return 0;
}
struct ast_codec_pref* ast_rtp_codec_getpref ( struct ast_rtp rtp) [read]

Get codec preference.

Definition at line 3832 of file rtp.c.

References ast_rtp::pref.

Referenced by add_codec_to_sdp(), and process_sdp_a_audio().

{
   return &rtp->pref;
}
void ast_rtp_codec_setpref ( struct ast_rtp rtp,
struct ast_codec_pref prefs 
)

Set codec preference.

Definition at line 3786 of file rtp.c.

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

Referenced by __oh323_rtp_create(), check_peer_ok(), create_addr_from_peer(), gtalk_new(), jingle_new(), process_sdp_a_audio(), register_verify(), set_peer_capabilities(), sip_alloc(), start_rtp(), and transmit_response_with_sdp().

{
   struct ast_format_list current_format_old, current_format_new;

   /* if no packets have been sent through this session yet, then
    *  changing preferences does not require any extra work
    */
   if (rtp->lasttxformat == 0) {
      rtp->pref = *prefs;
      return;
   }

   current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);

   rtp->pref = *prefs;

   current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);

   /* if the framing desired for the current format has changed, we may have to create
    * or adjust the smoother for this session
    */
   if ((current_format_new.inc_ms != 0) &&
       (current_format_new.cur_ms != current_format_old.cur_ms)) {
      int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms;

      if (rtp->smoother) {
         ast_smoother_reconfigure(rtp->smoother, new_size);
         if (option_debug) {
            ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size);
         }
      } else {
         if (!(rtp->smoother = ast_smoother_new(new_size))) {
            ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
            return;
         }
         if (current_format_new.flags) {
            ast_smoother_set_flags(rtp->smoother, current_format_new.flags);
         }
         if (option_debug) {
            ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
         }
      }
   }

}
void ast_rtp_destroy ( struct ast_rtp rtp)

Destroy RTP session

Definition at line 3105 of file rtp.c.

References ast_free, ast_io_remove(), ast_mutex_destroy(), AST_SCHED_DEL, ast_smoother_free(), ast_verbose(), EVENT_FLAG_REPORTING, ast_rtcp::expected_prior, ast_rtp::io, ast_rtp::ioid, manager_event, 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_peer_ok(), cleanup_connection(), create_addr_from_peer(), destroy_endpoint(), gtalk_free_pvt(), jingle_free_pvt(), mgcp_hangup(), oh323_alloc(), skinny_hangup(), start_rtp(), unalloc_sub(), and unistim_hangup().

{
   if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
      /*Print some info on the call here */
      ast_verbose("  RTP-stats\n");
      ast_verbose("* Our Receiver:\n");
      ast_verbose("  SSRC:     %u\n", rtp->themssrc);
      ast_verbose("  Received packets: %u\n", rtp->rxcount);
      ast_verbose("  Lost packets:   %u\n", rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0);
      ast_verbose("  Jitter:      %.4f\n", rtp->rxjitter);
      ast_verbose("  Transit:     %.4f\n", rtp->rxtransit);
      ast_verbose("  RR-count:    %u\n", rtp->rtcp ? rtp->rtcp->rr_count : 0);
      ast_verbose("* Our Sender:\n");
      ast_verbose("  SSRC:     %u\n", rtp->ssrc);
      ast_verbose("  Sent packets:   %u\n", rtp->txcount);
      ast_verbose("  Lost packets:   %u\n", rtp->rtcp ? rtp->rtcp->reported_lost : 0);
      ast_verbose("  Jitter:      %u\n", rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int)65536.0) : 0);
      ast_verbose("  SR-count:    %u\n", rtp->rtcp ? rtp->rtcp->sr_count : 0);
      ast_verbose("  RTT:      %f\n", rtp->rtcp ? rtp->rtcp->rtt : 0);
   }

   manager_event(EVENT_FLAG_REPORTING, "RTPReceiverStat", "SSRC: %u\r\n"
                   "ReceivedPackets: %u\r\n"
                   "LostPackets: %u\r\n"
                   "Jitter: %.4f\r\n"
                   "Transit: %.4f\r\n"
                   "RRCount: %u\r\n",
                   rtp->themssrc,
                   rtp->rxcount,
                   rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0,
                   rtp->rxjitter,
                   rtp->rxtransit,
                   rtp->rtcp ? rtp->rtcp->rr_count : 0);
   manager_event(EVENT_FLAG_REPORTING, "RTPSenderStat", "SSRC: %u\r\n"
                   "SentPackets: %u\r\n"
                   "LostPackets: %u\r\n"
                   "Jitter: %u\r\n"
                   "SRCount: %u\r\n"
                   "RTT: %f\r\n",
                   rtp->ssrc,
                   rtp->txcount,
                   rtp->rtcp ? rtp->rtcp->reported_lost : 0,
                   rtp->rtcp ? rtp->rtcp->reported_jitter : 0,
                   rtp->rtcp ? rtp->rtcp->sr_count : 0,
                   rtp->rtcp ? rtp->rtcp->rtt : 0);
   if (rtp->smoother)
      ast_smoother_free(rtp->smoother);
   if (rtp->ioid)
      ast_io_remove(rtp->io, rtp->ioid);
   if (rtp->s > -1)
      close(rtp->s);
   if (rtp->rtcp) {
      AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
      close(rtp->rtcp->s);
      ast_free(rtp->rtcp);
      rtp->rtcp=NULL;
   }
#ifdef P2P_INTENSE
   ast_mutex_destroy(&rtp->bridge_lock);
#endif
   ast_free(rtp);
}
int ast_rtp_early_bridge ( struct ast_channel c0,
struct ast_channel c1 
)

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

Definition at line 2114 of file rtp.c.

References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_debug, 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_trtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, and ast_rtp_protocol::set_rtp_peer.

{
   struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
   struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
   struct ast_rtp *tdestp = NULL, *tsrcp = NULL;      /* Text RTP channels */
   struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
   enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
   enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED;
   int srccodec, destcodec, nat_active = 0;

   /* Lock channels */
   ast_channel_lock(c0);
   if (c1) {
      while (ast_channel_trylock(c1)) {
         ast_channel_unlock(c0);
         usleep(1);
         ast_channel_lock(c0);
      }
   }

   /* Find channel driver interfaces */
   destpr = get_proto(c0);
   if (c1)
      srcpr = get_proto(c1);
   if (!destpr) {
      ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c0->name);
      ast_channel_unlock(c0);
      if (c1)
         ast_channel_unlock(c1);
      return -1;
   }
   if (!srcpr) {
      ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c1 ? c1->name : "<unspecified>");
      ast_channel_unlock(c0);
      if (c1)
         ast_channel_unlock(c1);
      return -1;
   }

   /* Get audio, video  and text interface (if native bridge is possible) */
   audio_dest_res = destpr->get_rtp_info(c0, &destp);
   video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(c0, &vdestp) : AST_RTP_GET_FAILED;
   text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(c0, &tdestp) : AST_RTP_GET_FAILED;
   if (srcpr) {
      audio_src_res = srcpr->get_rtp_info(c1, &srcp);
      video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(c1, &vsrcp) : AST_RTP_GET_FAILED;
      text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED;
   }

   /* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
   if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {
      /* Somebody doesn't want to play... */
      ast_channel_unlock(c0);
      if (c1)
         ast_channel_unlock(c1);
      return -1;
   }
   if (audio_src_res == AST_RTP_TRY_NATIVE && (video_src_res == AST_RTP_GET_FAILED || video_src_res == AST_RTP_TRY_NATIVE) && srcpr->get_codec)
      srccodec = srcpr->get_codec(c1);
   else
      srccodec = 0;
   if (audio_dest_res == AST_RTP_TRY_NATIVE && (video_dest_res == AST_RTP_GET_FAILED || video_dest_res == AST_RTP_TRY_NATIVE) && destpr->get_codec)
      destcodec = destpr->get_codec(c0);
   else
      destcodec = 0;
   /* Ensure we have at least one matching codec */
   if (srcp && !(srccodec & destcodec)) {
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return 0;
   }
   /* Consider empty media as non-existent */
   if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
      srcp = NULL;
   if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
      nat_active = 1;
   /* Bridge media early */
   if (destpr->set_rtp_peer(c0, srcp, vsrcp, tsrcp, srccodec, nat_active))
      ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
   ast_channel_unlock(c0);
   if (c1)
      ast_channel_unlock(c1);
   ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
   return 0;
}
int ast_rtp_fd ( struct ast_rtp rtp)
struct ast_rtp* ast_rtp_get_bridged ( struct ast_rtp rtp) [read]

Definition at line 2746 of file rtp.c.

References ast_rtp::bridged, rtp_bridge_lock(), and rtp_bridge_unlock().

Referenced by __sip_destroy(), ast_rtp_read(), and dialog_needdestroy().

{
   struct ast_rtp *bridged = NULL;

   rtp_bridge_lock(rtp);
   bridged = rtp->bridged;
   rtp_bridge_unlock(rtp);

   return bridged;
}
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 2362 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().

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

{
   int pt;
   
   rtp_bridge_lock(rtp);
   
   *astFormats = *nonAstFormats = 0;
   for (pt = 0; pt < MAX_RTP_PT; ++pt) {
      if (rtp->current_RTP_PT[pt].isAstFormat) {
         *astFormats |= rtp->current_RTP_PT[pt].code;
      } else {
         *nonAstFormats |= rtp->current_RTP_PT[pt].code;
      }
   }

   rtp_bridge_unlock(rtp);
}
int ast_rtp_get_peer ( struct ast_rtp rtp,
struct sockaddr_in *  them 
)

Definition at line 2728 of file rtp.c.

References ast_rtp::them.

Referenced by acf_channel_read(), add_sdp(), bridge_native_loop(), check_rtp_timeout(), gtalk_update_stun(), oh323_set_rtp_peer(), process_sdp(), sip_set_rtp_peer(), skinny_set_rtp_peer(), and transmit_modify_with_sdp().

{
   if ((them->sin_family != AF_INET) ||
      (them->sin_port != rtp->them.sin_port) ||
      (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
      them->sin_family = AF_INET;
      them->sin_port = rtp->them.sin_port;
      them->sin_addr = rtp->them.sin_addr;
      return 1;
   }
   return 0;
}
int ast_rtp_get_qos ( struct ast_rtp rtp,
const char *  qos,
char *  buf,
unsigned int  buflen 
)

Get QOS stats on a RTP channel.

Since:
1.6.1

Definition at line 2867 of file rtp.c.

References __ast_rtp_get_qos().

Referenced by acf_channel_read().

{
   double value;
   int found;

   value = __ast_rtp_get_qos(rtp, qos, &found);

   if (!found)
      return -1;

   snprintf(buf, buflen, "%.0lf", value);

   return 0;
}
unsigned int ast_rtp_get_qosvalue ( struct ast_rtp rtp,
enum ast_rtp_qos_vars  value 
)

Return RTP and RTCP QoS values.

Since:
1.6.1

Get QoS values from RTP and RTCP data (used in "sip show channelstats")

Definition at line 2801 of file rtp.c.

References ast_log(), AST_RTP_RTT, AST_RTP_RXCOUNT, AST_RTP_RXJITTER, AST_RTP_RXPLOSS, AST_RTP_TXCOUNT, AST_RTP_TXJITTER, AST_RTP_TXPLOSS, ast_rtcp::expected_prior, LOG_DEBUG, option_debug, ast_rtcp::received_prior, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtp::rtcp, ast_rtcp::rtt, ast_rtp::rxcount, ast_rtp::rxjitter, and ast_rtp::txcount.

Referenced by show_chanstats_cb().

{
   if (rtp == NULL) {
      if (option_debug > 1)
         ast_log(LOG_DEBUG, "NO RTP Structure? Kidding me? \n");
      return 0;
   }
   if (option_debug > 1 && rtp->rtcp == NULL) {
      ast_log(LOG_DEBUG, "NO RTCP structure. Maybe in RTP p2p bridging mode? \n");
   }

   switch (value) {
   case AST_RTP_TXCOUNT:
      return (unsigned int) rtp->txcount;
   case AST_RTP_RXCOUNT:
      return (unsigned int) rtp->rxcount;
   case AST_RTP_TXJITTER:
      return (unsigned int) (rtp->rxjitter * 1000.0);
   case AST_RTP_RXJITTER:
      return (unsigned int) (rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int) 65536.0) : 0);
   case AST_RTP_RXPLOSS:
      return rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0;
   case AST_RTP_TXPLOSS:
      return rtp->rtcp ? rtp->rtcp->reported_lost : 0;
   case AST_RTP_RTT:
      return (unsigned int) (rtp->rtcp ? (rtp->rtcp->rtt * 100) : 0);
   }
   return 0;   /* To make the compiler happy */
}
char* ast_rtp_get_quality ( struct ast_rtp rtp,
struct ast_rtp_quality qual,
enum ast_rtp_quality_type  qtype 
)

Return RTCP quality string.

Parameters:
rtpAn rtp structure to get qos information about.
qualAn (optional) rtp quality structure that will be filled with the quality information described in the ast_rtp_quality structure. This structure is not dependent on any qtype, so a call for any type of information would yield the same results because ast_rtp_quality is not a data type specific to any qos type.
qtypeThe quality type you'd like, default should be RTPQOS_SUMMARY which returns basic information about the call. The return from RTPQOS_SUMMARY is basically ast_rtp_quality in a string. The other types are RTPQOS_JITTER, RTPQOS_LOSS and RTPQOS_RTT which will return more specific statistics.
Version:
1.6.1 added qtype parameter

Definition at line 3074 of file rtp.c.

References __ast_rtp_get_quality(), __ast_rtp_get_quality_jitter(), __ast_rtp_get_quality_loss(), __ast_rtp_get_quality_rtt(), 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::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, RTPQOS_JITTER, RTPQOS_LOSS, RTPQOS_RTT, RTPQOS_SUMMARY, 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(), ast_rtp_set_vars(), handle_request_bye(), and sip_hangup().

{
   if (qual && rtp) {
      qual->local_ssrc   = rtp->ssrc;
      qual->local_jitter = rtp->rxjitter;
      qual->local_count  = rtp->rxcount;
      qual->remote_ssrc  = rtp->themssrc;
      qual->remote_count = rtp->txcount;

      if (rtp->rtcp) {
         qual->local_lostpackets  = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
         qual->remote_lostpackets = rtp->rtcp->reported_lost;
         qual->remote_jitter      = rtp->rtcp->reported_jitter / 65536.0;
         qual->rtt                = rtp->rtcp->rtt;
      }
   }

   switch (qtype) {
   case RTPQOS_SUMMARY:
      return __ast_rtp_get_quality(rtp);
   case RTPQOS_JITTER:
      return __ast_rtp_get_quality_jitter(rtp);
   case RTPQOS_LOSS:
      return __ast_rtp_get_quality_loss(rtp);
   case RTPQOS_RTT:
      return __ast_rtp_get_quality_rtt(rtp);
   }

   return NULL;
}
int ast_rtp_get_rtpholdtimeout ( struct ast_rtp rtp)

Get rtp hold timeout.

Definition at line 777 of file rtp.c.

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

Referenced by check_rtp_timeout().

{
   if (rtp->rtptimeout < 0)   /* We're not checking, but remembering the setting (during T.38 transmission) */
      return 0;
   return rtp->rtpholdtimeout;
}
int ast_rtp_get_rtpkeepalive ( struct ast_rtp rtp)

Get RTP keepalive interval.

Definition at line 785 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by check_rtp_timeout().

{
   return rtp->rtpkeepalive;
}
int ast_rtp_get_rtptimeout ( struct ast_rtp rtp)

Get rtp timeout.

Definition at line 769 of file rtp.c.

References ast_rtp::rtptimeout.

Referenced by check_rtp_timeout().

{
   if (rtp->rtptimeout < 0)   /* We're not checking, but remembering the setting (during T.38 transmission) */
      return 0;
   return rtp->rtptimeout;
}
void ast_rtp_get_us ( struct ast_rtp rtp,
struct sockaddr_in *  us 
)
int ast_rtp_getnat ( struct ast_rtp rtp)

Definition at line 805 of file rtp.c.

References ast_test_flag, and FLAG_NAT_ACTIVE.

Referenced by sip_get_rtp_peer().

{
   return ast_test_flag(rtp, FLAG_NAT_ACTIVE);
}
void ast_rtp_init ( void  )

Initialize the RTP system in Asterisk.

Definition at line 4877 of file rtp.c.

References __ast_rtp_reload(), and ast_cli_register_multiple().

Referenced by main().

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 2403 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), rtp_bridge_unlock(), 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(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_rtp_sendcng(), ast_rtp_senddigit_begin(), ast_rtp_write(), bridge_p2p_rtp_write(), and start_rtp().

{
   int pt = 0;

   rtp_bridge_lock(rtp);

   if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
      code == rtp->rtp_lookup_code_cache_code) {
      /* Use our cached mapping, to avoid the overhead of the loop below */
      pt = rtp->rtp_lookup_code_cache_result;
      rtp_bridge_unlock(rtp);
      return pt;
   }

   /* Check the dynamic list first */
   for (pt = 0; pt < MAX_RTP_PT; ++pt) {
      if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
         rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
         rtp->rtp_lookup_code_cache_code = code;
         rtp->rtp_lookup_code_cache_result = pt;
         rtp_bridge_unlock(rtp);
         return pt;
      }
   }

   /* Then the static list */
   for (pt = 0; pt < MAX_RTP_PT; ++pt) {
      if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
         rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
         rtp->rtp_lookup_code_cache_code = code;
         rtp->rtp_lookup_code_cache_result = pt;
         rtp_bridge_unlock(rtp);
         return pt;
      }
   }

   rtp_bridge_unlock(rtp);

   return -1;
}
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 2476 of file rtp.c.

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

Referenced by process_sdp().

{
   int format;
   unsigned len;
   char *end = buf;
   char *start = buf;

   if (!buf || !size)
      return NULL;

   snprintf(end, size, "0x%x (", capability);

   len = strlen(end);
   end += len;
   size -= len;
   start = end;

   for (format = 1; format < AST_RTP_MAX; format <<= 1) {
      if (capability & format) {
         const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);

         snprintf(end, size, "%s|", name);
         len = strlen(end);
         end += len;
         size -= len;
      }
   }

   if (start == end)
      ast_copy_string(start, "nothing)", size); 
   else if (size > 1)
      *(end -1) = ')';
   
   return buf;
}
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 2444 of file rtp.c.

References ARRAY_LEN, AST_FORMAT_G726_AAL2, AST_RTP_OPT_G726_NONSTANDARD, rtpPayloadType::isAstFormat, mimeTypes, mimeType::payloadType, and mimeType::subtype.

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

{
   unsigned int i;

   for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
      if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
         if (isAstFormat &&
             (code == AST_FORMAT_G726_AAL2) &&
             (options & AST_RTP_OPT_G726_NONSTANDARD))
            return "G726-32";
         else
            return mimeTypes[i].subtype;
      }
   }

   return "";
}
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 2381 of file rtp.c.

References rtpPayloadType::code, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().

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

{
   struct rtpPayloadType result;

   result.isAstFormat = result.code = 0;

   if (pt < 0 || pt >= MAX_RTP_PT) 
      return result; /* bogus payload type */

   /* Start with negotiated codecs */
   rtp_bridge_lock(rtp);
   result = rtp->current_RTP_PT[pt];
   rtp_bridge_unlock(rtp);

   /* If it doesn't exist, check our static RTP type list, just in case */
   if (!result.code) 
      result = static_RTP_PT[pt];

   return result;
}
unsigned int ast_rtp_lookup_sample_rate ( int  isAstFormat,
int  code 
)

Get the sample rate associated with known RTP payload types.

Parameters:
isAstFormatTrue if the value in the 'code' parameter is an AST_FORMAT value
codeFormat code, either from AST_FORMAT list or from AST_RTP list
Returns:
the sample rate if the format was found, zero if it was not found

Definition at line 2463 of file rtp.c.

References ARRAY_LEN, rtpPayloadType::isAstFormat, mimeTypes, mimeType::payloadType, and mimeType::sample_rate.

Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_tcodec_to_sdp(), and add_vcodec_to_sdp().

{
   unsigned int i;

   for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
      if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
         return mimeTypes[i].sample_rate;
      }
   }

   return 0;
}
int ast_rtp_make_compatible ( struct ast_channel dest,
struct ast_channel src,
int  media 
)

Definition at line 2200 of file rtp.c.

References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_debug, 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_trtp_info, ast_rtp_protocol::get_vrtp_info, LOG_WARNING, ast_channel::name, and ast_rtp_protocol::set_rtp_peer.

Referenced by dial_exec_full(), and do_forward().

{
   struct ast_rtp *destp = NULL, *srcp = NULL;     /* Audio RTP Channels */
   struct ast_rtp *vdestp = NULL, *vsrcp = NULL;      /* Video RTP channels */
   struct ast_rtp *tdestp = NULL, *tsrcp = NULL;      /* Text RTP channels */
   struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
   enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
   enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED; 
   int srccodec, destcodec;

   /* Lock channels */
   ast_channel_lock(dest);
   while (ast_channel_trylock(src)) {
      ast_channel_unlock(dest);
      usleep(1);
      ast_channel_lock(dest);
   }

   /* Find channel driver interfaces */
   if (!(destpr = get_proto(dest))) {
      ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", dest->name);
      ast_channel_unlock(dest);
      ast_channel_unlock(src);
      return 0;
   }
   if (!(srcpr = get_proto(src))) {
      ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", src->name);
      ast_channel_unlock(dest);
      ast_channel_unlock(src);
      return 0;
   }

   /* Get audio and video interface (if native bridge is possible) */
   audio_dest_res = destpr->get_rtp_info(dest, &destp);
   video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
   text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(dest, &tdestp) : AST_RTP_GET_FAILED;
   audio_src_res = srcpr->get_rtp_info(src, &srcp);
   video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
   text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(src, &tsrcp) : AST_RTP_GET_FAILED;

   /* Ensure we have at least one matching codec */
   if (srcpr->get_codec)
      srccodec = srcpr->get_codec(src);
   else
      srccodec = 0;
   if (destpr->get_codec)
      destcodec = destpr->get_codec(dest);
   else
      destcodec = 0;

   /* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
   if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) {
      /* Somebody doesn't want to play... */
      ast_channel_unlock(dest);
      ast_channel_unlock(src);
      return 0;
   }
   ast_rtp_pt_copy(destp, srcp);
   if (vdestp && vsrcp)
      ast_rtp_pt_copy(vdestp, vsrcp);
   if (tdestp && tsrcp)
      ast_rtp_pt_copy(tdestp, tsrcp);
   if (media) {
      /* Bridge early */
      if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
         ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
   }
   ast_channel_unlock(dest);
   ast_channel_unlock(src);
   ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
   return 1;
}
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 2671 of file rtp.c.

References ast_rtp_new_with_bindaddr().

{
   struct in_addr ia;

   memset(&ia, 0, sizeof(ia));
   return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
}
void ast_rtp_new_init ( struct ast_rtp rtp)

Initialize a new RTP structure.

reload rtp configuration

Definition at line 2562 of file rtp.c.

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

Referenced by ast_rtp_new_with_bindaddr(), and process_sdp().

{
#ifdef P2P_INTENSE
   ast_mutex_init(&rtp->bridge_lock);
#endif

   rtp->them.sin_family = AF_INET;
   rtp->us.sin_family = AF_INET;
   rtp->ssrc = ast_random();
   rtp->seqno = ast_random() & 0xffff;
   ast_set_flag(rtp, FLAG_HAS_DTMF);
   rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
}
void ast_rtp_new_source ( struct ast_rtp rtp)

Indicate that we need to set the marker bit.

Definition at line 2684 of file rtp.c.

References ast_debug, and ast_rtp::set_marker_bit.

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

{
   if (rtp) {
      rtp->set_marker_bit = 1;
      ast_debug(3, "Setting the marker bit due to a source update\n");
   }
}
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 2576 of file rtp.c.

References ast_calloc, ast_free, 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, io, ast_rtp::io, ast_rtp::ioid, LOG_ERROR, ast_rtp::rtcp, rtp_socket(), rtpread(), rtpstart, ast_rtcp::s, ast_rtp::s, sched, ast_rtp::sched, ast_rtcp::us, and ast_rtp::us.

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

{
   struct ast_rtp *rtp;
   int x;
   int startplace;
   
   if (!(rtp = ast_calloc(1, sizeof(*rtp))))
      return NULL;

   ast_rtp_new_init(rtp);

   rtp->s = rtp_socket("RTP");
   if (rtp->s < 0)
      goto fail;
   if (sched && rtcpenable) {
      rtp->sched = sched;
      rtp->rtcp = ast_rtcp_new();
   }
   
   /*
    * Try to bind the RTP port, x, and possibly the RTCP port, x+1 as well.
    * Start from a random (even, by RTP spec) port number, and
    * iterate until success or no ports are available.
    * Note that the requirement of RTP port being even, or RTCP being the
    * next one, cannot be enforced in presence of a NAT box because the
    * mapping is not under our control.
    */
   x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart;
   x = x & ~1;    /* make it an even number */
   startplace = x;      /* remember the starting point */
   /* this is constant across the loop */
   rtp->us.sin_addr = addr;
   if (rtp->rtcp)
      rtp->rtcp->us.sin_addr = addr;
   for (;;) {
      rtp->us.sin_port = htons(x);
      if (!bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) {
         /* bind succeeded, if no rtcp then we are done */
         if (!rtp->rtcp)
            break;
         /* have rtcp, try to bind it */
         rtp->rtcp->us.sin_port = htons(x + 1);
         if (!bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))
            break;   /* success again, we are really done */
         /*
          * RTCP bind failed, so close and recreate the
          * already bound RTP socket for the next round.
          */
         close(rtp->s);
         rtp->s = rtp_socket("RTP");
         if (rtp->s < 0)
            goto fail;
      }
      /*
       * If we get here, there was an error in one of the bind()
       * calls, so make sure it is nothing unexpected.
       */
      if (errno != EADDRINUSE) {
         /* We got an error that wasn't expected, abort! */
         ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
         goto fail;
      }
      /*
       * One of the ports is in use. For the next iteration,
       * increment by two and handle wraparound.
       * If we reach the starting point, then declare failure.
       */
      x += 2;
      if (x > rtpend)
         x = (rtpstart + 1) & ~1;
      if (x == startplace) {
         ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
         goto fail;
      }
   }
   rtp->sched = sched;
   rtp->io = io;
   if (callbackmode) {
      rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
      ast_set_flag(rtp, FLAG_CALLBACK_MODE);
   }
   ast_rtp_pt_default(rtp);
   return rtp;

fail:
   if (rtp->s >= 0)
      close(rtp->s);
   if (rtp->rtcp) {
      close(rtp->rtcp->s);
      ast_free(rtp->rtcp);
   }
   ast_free(rtp);
   return NULL;
}
int ast_rtp_proto_register ( struct ast_rtp_protocol proto)

Register an RTP channel client.

Definition at line 3954 of file rtp.c.

References ast_log(), AST_RWLIST_INSERT_HEAD, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_rtp_protocol::list, LOG_WARNING, and ast_rtp_protocol::type.

Referenced by load_module().

{
   struct ast_rtp_protocol *cur;

   AST_RWLIST_WRLOCK(&protos);
   AST_RWLIST_TRAVERSE(&protos, cur, list) { 
      if (!strcmp(cur->type, proto->type)) {
         ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
         AST_RWLIST_UNLOCK(&protos);
         return -1;
      }
   }
   AST_RWLIST_INSERT_HEAD(&protos, proto, list);
   AST_RWLIST_UNLOCK(&protos);
   
   return 0;
}
void ast_rtp_proto_unregister ( struct ast_rtp_protocol proto)

Unregister an RTP channel client.

Definition at line 3946 of file rtp.c.

References AST_RWLIST_REMOVE, AST_RWLIST_UNLOCK, and AST_RWLIST_WRLOCK.

Referenced by load_module(), and unload_module().

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

Definition at line 1576 of file rtp.c.

References ast_rtp::altthem, ast_assert, ast_codec_get_samples(), AST_CONTROL_SRCCHANGE, ast_debug, AST_FORMAT_AUDIO_MASK, ast_format_rate(), AST_FORMAT_SLINEAR, AST_FORMAT_T140, AST_FORMAT_T140RED, AST_FORMAT_VIDEO_MASK, ast_frame_byteswap_be, AST_FRAME_CONTROL, AST_FRAME_DTMF_END, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_frisolate(), ast_inet_ntoa(), AST_LIST_EMPTY, AST_LIST_FIRST, AST_LIST_HEAD_INIT_NOLOCK, AST_LIST_INSERT_TAIL, 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_samp2tv(), ast_sched_add(), ast_set_flag, ast_tv(), ast_tvdiff_ms(), ast_verbose(), bridge_p2p_rtp_write(), ast_rtp::bridged, calc_rxstamp(), rtpPayloadType::code, create_dtmf_frame(), ast_rtp::cycles, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp::dtmf_duration, ast_rtp::dtmf_timeout, errno, ext, ast_rtp::f, f, FLAG_NAT_ACTIVE, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lastevent, ast_rtp::lastitexttimestamp, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxseqno, ast_rtp::lastrxts, ast_frame::len, len(), LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_frame::ptr, ast_rtp::rawdata, ast_rtp::resp, ast_rtp::rtcp, rtp_debug_test_addr(), rtp_get_rate(), 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, ast_rtp::strict_rtp_address, STRICT_RTP_CLOSED, STRICT_RTP_LEARN, ast_rtp::strict_rtp_state, STUN_ACCEPT, stun_handle_packet(), ast_frame::subclass, ast_rtcp::them, ast_rtp::them, ast_rtp::themssrc, ast_frame::ts, and version.

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

{
   int res;
   struct sockaddr_in sock_in;
   socklen_t len;
   unsigned int seqno;
   int version;
   int payloadtype;
   int hdrlen = 12;
   int padding;
   int mark;
   int ext;
   int cc;
   unsigned int ssrc;
   unsigned int timestamp;
   unsigned int *rtpheader;
   struct rtpPayloadType rtpPT;
   struct ast_rtp *bridged = NULL;
   int prev_seqno;
   struct frame_list frames;
   
   /* If time is up, kill it */
   if (rtp->sending_digit)
      ast_rtp_senddigit_continuation(rtp);

   len = sizeof(sock_in);
   
   /* Cache where the header will go */
   res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
               0, (struct sockaddr *)&sock_in, &len);

   /* If strict RTP protection is enabled see if we need to learn this address or if the packet should be dropped */
   if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
      /* Copy over address that this packet was received on */
      memcpy(&rtp->strict_rtp_address, &sock_in, sizeof(rtp->strict_rtp_address));
      /* Now move over to actually protecting the RTP port */
      rtp->strict_rtp_state = STRICT_RTP_CLOSED;
      ast_debug(1, "Learned remote address is %s:%d for strict RTP purposes, now protecting the port.\n", ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port));
   } else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
      /* If the address we previously learned doesn't match the address this packet came in on simply drop it */
      if ((rtp->strict_rtp_address.sin_addr.s_addr != sock_in.sin_addr.s_addr) || (rtp->strict_rtp_address.sin_port != sock_in.sin_port)) {
         ast_debug(1, "Received RTP packet from %s:%d, dropping due to strict RTP protection. Expected it to be from %s:%d\n", ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port), ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port));
         return &ast_null_frame;
      }
   }

   rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
   if (res < 0) {
      ast_assert(errno != EBADF);
      if (errno != EAGAIN) {
         ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up.\n", strerror(errno));
         return NULL;
      }
      return &ast_null_frame;
   }
   
   if (res < hdrlen) {
      ast_log(LOG_WARNING, "RTP Read too short\n");
      return &ast_null_frame;
   }

   /* Get fields */
   seqno = ntohl(rtpheader[0]);

   /* Check RTP version */
   version = (seqno & 0xC0000000) >> 30;
   if (!version) {
      /* If the two high bits are 0, this might be a
       * STUN message, so process it. stun_handle_packet()
       * answers to requests, and it returns STUN_ACCEPT
       * if the request is valid.
       */
      if ((stun_handle_packet(rtp->s, &sock_in, rtp->rawdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == STUN_ACCEPT) &&
         (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
         memcpy(&rtp->them, &sock_in, sizeof(rtp->them));
      }
      return &ast_null_frame;
   }

#if 0 /* Allow to receive RTP stream with closed transmission path */
   /* If we don't have the other side's address, then ignore this */
   if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
      return &ast_null_frame;
#endif

   /* Send to whoever send to us if NAT is turned on */
   if (rtp->nat) {
      if (((rtp->them.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
          (rtp->them.sin_port != sock_in.sin_port)) && 
          ((rtp->altthem.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
          (rtp->altthem.sin_port != sock_in.sin_port))) {
         rtp->them = sock_in;
         if (rtp->rtcp) {
            int h = 0;
            memcpy(&rtp->rtcp->them, &sock_in, sizeof(rtp->rtcp->them));
            h = ntohs(rtp->them.sin_port);
            rtp->rtcp->them.sin_port = htons(h + 1);
         }
         rtp->rxseqno = 0;
         ast_set_flag(rtp, FLAG_NAT_ACTIVE);
         if (option_debug || rtpdebug)
            ast_debug(0, "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));
      }
   }

   /* If we are bridged to another RTP stream, send direct */
   if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
      return &ast_null_frame;

   if (version != 2)
      return &ast_null_frame;

   payloadtype = (seqno & 0x7f0000) >> 16;
   padding = seqno & (1 << 29);
   mark = seqno & (1 << 23);
   ext = seqno & (1 << 28);
   cc = (seqno & 0xF000000) >> 24;
   seqno &= 0xffff;
   timestamp = ntohl(rtpheader[1]);
   ssrc = ntohl(rtpheader[2]);
   
   AST_LIST_HEAD_INIT_NOLOCK(&frames);
   /* Force a marker bit and change SSRC if the SSRC changes */
   if (rtp->rxssrc && rtp->rxssrc != ssrc) {
      struct ast_frame *f, srcupdate = {
         AST_FRAME_CONTROL,
         .subclass = AST_CONTROL_SRCCHANGE,
      };

      if (!mark) {
         if (option_debug || rtpdebug) {
            ast_debug(0, "Forcing Marker bit, because SSRC has changed\n");
         }
         mark = 1;
      }
      f = ast_frisolate(&srcupdate);
      AST_LIST_INSERT_TAIL(&frames, f, frame_list);
   }

   rtp->rxssrc = ssrc;
   
   if (padding) {
      /* Remove padding bytes */
      res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
   }
   
   if (cc) {
      /* CSRC fields present */
      hdrlen += cc*4;
   }

   if (ext) {
      /* RTP Extension present */
      hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
      hdrlen += 4;
      if (option_debug) {
         int profile;
         profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
         if (profile == 0x505a)
            ast_debug(1, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
         else
            ast_debug(1, "Found unknown RTP Extensions %x\n", profile);
      }
   }

   if (res < hdrlen) {
      ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
      return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
   }

   rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */

   if (rtp->rxcount==1) {
      /* This is the first RTP packet successfully received from source */
      rtp->seedrxseqno = seqno;
   }

   /* Do not schedule RR if RTCP isn't run */
   if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
      /* Schedule transmission of Receiver Report */
      rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
   }
   if ((int)rtp->lastrxseqno - (int)seqno  > 100) /* if so it would indicate that the sender cycled; allow for misordering */
      rtp->cycles += RTP_SEQ_MOD;
   
   prev_seqno = rtp->lastrxseqno;

   rtp->lastrxseqno = seqno;
   
   if (!rtp->themssrc)
      rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
   
   if (rtp_debug_test_addr(&sock_in))
      ast_verbose("Got  RTP packet from    %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
         ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port), payloadtype, seqno, timestamp,res - hdrlen);

   rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
   if (!rtpPT.isAstFormat) {
      struct ast_frame *f = NULL;

      /* This is special in-band data that's not one of our codecs */
      if (rtpPT.code == AST_RTP_DTMF) {
         /* It's special -- rfc2833 process it */
         if (rtp_debug_test_addr(&sock_in)) {
            unsigned char *data;
            unsigned int event;
            unsigned int event_end;
            unsigned int duration;
            data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
            event = ntohl(*((unsigned int *)(data)));
            event >>= 24;
            event_end = ntohl(*((unsigned int *)(data)));
            event_end <<= 8;
            event_end >>= 24;
            duration = ntohl(*((unsigned int *)(data)));
            duration &= 0xFFFF;
            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(sock_in.sin_addr), ntohs(sock_in.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
         }
         /* process_rfc2833 may need to return multiple frames. We do this
          * by passing the pointer to the frame list to it so that the method
          * can append frames to the list as needed
          */
         process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &frames);
      } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
         /* It's really special -- process it the Cisco way */
         if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
            f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
            rtp->lastevent = seqno;
         }
      } else if (rtpPT.code == AST_RTP_CN) {
         /* Comfort Noise */
         f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
      } else {
         ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
      }
      if (f) {
         AST_LIST_INSERT_TAIL(&frames, f, frame_list);
      }
      /* Even if no frame was returned by one of the above methods,
       * we may have a frame to return in our frame list
       */
      if (!AST_LIST_EMPTY(&frames)) {
         return AST_LIST_FIRST(&frames);
      }
      return &ast_null_frame;
   }
   rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
   rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;

   rtp->rxseqno = seqno;

   if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) {
      rtp->dtmf_timeout = 0;

      if (rtp->resp) {
         struct ast_frame *f;
         f = create_dtmf_frame(rtp, AST_FRAME_DTMF_END);
         f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
         rtp->resp = 0;
         rtp->dtmf_timeout = rtp->dtmf_duration = 0;
         AST_LIST_INSERT_TAIL(&frames, f, frame_list);
         return AST_LIST_FIRST(&frames);
      }
   }

   /* Record received timestamp as last received now */
   rtp->lastrxts = timestamp;

   rtp->f.mallocd = 0;
   rtp->f.datalen = res - hdrlen;
   rtp->f.data.ptr = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
   rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
   rtp->f.seqno = seqno;

   if (rtp->f.subclass == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
        unsigned char *data = rtp->f.data.ptr;
        
        memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
        rtp->f.datalen +=3;
        *data++ = 0xEF;
        *data++ = 0xBF;
        *data = 0xBD;
   }
 
   if (rtp->f.subclass == AST_FORMAT_T140RED) {
      unsigned char *data = rtp->f.data.ptr;
      unsigned char *header_end;
      int num_generations;
      int header_length;
      int length;
      int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
      int x;

      rtp->f.subclass = AST_FORMAT_T140;
      header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
      if (header_end == NULL) {
         return &ast_null_frame;
      }
      header_end++;
      
      header_length = header_end - data;
      num_generations = header_length / 4;
      length = header_length;

      if (!diff) {
         for (x = 0; x < num_generations; x++)
            length += data[x * 4 + 3];
         
         if (!(rtp->f.datalen - length))
            return &ast_null_frame;
         
         rtp->f.data.ptr += length;
         rtp->f.datalen -= length;
      } else if (diff > num_generations && diff < 10) {
         length -= 3;
         rtp->f.data.ptr += length;
         rtp->f.datalen -= length;
         
         data = rtp->f.data.ptr;
         *data++ = 0xEF;
         *data++ = 0xBF;
         *data = 0xBD;
      } else   {
         for ( x = 0; x < num_generations - diff; x++) 
            length += data[x * 4 + 3];
         
         rtp->f.data.ptr += length;
         rtp->f.datalen -= length;
      }
   }

   if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) {
      rtp->f.samples = ast_codec_get_samples(&rtp->f);
      if (rtp->f.subclass == AST_FORMAT_SLINEAR) 
         ast_frame_byteswap_be(&rtp->f);
      calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
      /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
      ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
      rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass) / 1000);
      rtp->f.len = rtp->f.samples / ((ast_format_rate(rtp->f.subclass) / 1000));
   } else if (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
      /* Video -- samples is # of samples vs. 90000 */
      if (!rtp->lastividtimestamp)
         rtp->lastividtimestamp = timestamp;
      rtp->f.samples = timestamp - rtp->lastividtimestamp;
      rtp->lastividtimestamp = timestamp;
      rtp->f.delivery.tv_sec = 0;
      rtp->f.delivery.tv_usec = 0;
      /* Pass the RTP marker bit as bit 0 in the subclass field.
       * This is ok because subclass is actually a bitmask, and
       * the low bits represent audio formats, that are not
       * involved here since we deal with video.
       */
      if (mark)
         rtp->f.subclass |= 0x1;
   } else {
      /* TEXT -- samples is # of samples vs. 1000 */
      if (!rtp->lastitexttimestamp)
         rtp->lastitexttimestamp = timestamp;
      rtp->f.samples = timestamp - rtp->lastitexttimestamp;
      rtp->lastitexttimestamp = timestamp;
      rtp->f.delivery.tv_sec = 0;
      rtp->f.delivery.tv_usec = 0;
   }
   rtp->f.src = "RTP";

   AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
   return AST_LIST_FIRST(&frames);
}
int ast_rtp_reload ( void  )

Initialize RTP subsystem

Definition at line 4871 of file rtp.c.

References __ast_rtp_reload().

{
   return __ast_rtp_reload(1);
}
int ast_rtp_sendcng ( struct ast_rtp rtp,
int  level 
)

generate comfort noice (CNG)

Definition at line 3632 of file rtp.c.

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

Referenced by check_rtp_timeout().

{
   unsigned int *rtpheader;
   int hdrlen = 12;
   int res;
   int payload;
   char data[256];
   level = 127 - (level & 0x7f);
   payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);

   /* If we have no peer, return immediately */ 
   if (!rtp->them.sin_addr.s_addr)
      return 0;

   rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));

   /* Get a pointer to the header */
   rtpheader = (unsigned int *)data;
   rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
   rtpheader[1] = htonl(rtp->lastts);
   rtpheader[2] = htonl(rtp->ssrc); 
   data[12] = level;
   if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
      res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
      if (res <0) 
         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));
      if (rtp_debug_test_addr(&rtp->them))
         ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
               , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);         
         
   }
   return 0;
}
int ast_rtp_senddigit_begin ( struct ast_rtp rtp,
char  digit 
)

Send begin frames for DTMF.

Definition at line 3188 of file rtp.c.

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

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

{
   unsigned int *rtpheader;
   int hdrlen = 12, res = 0, i = 0, payload = 0;
   char data[256];

   if ((digit <= '9') && (digit >= '0'))
      digit -= '0';
   else if (digit == '*')
      digit = 10;
   else if (digit == '#')
      digit = 11;
   else if ((digit >= 'A') && (digit <= 'D'))
      digit = digit - 'A' + 12;
   else if ((digit >= 'a') && (digit <= 'd'))
      digit = digit - 'a' + 12;
   else {
      ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
      return 0;
   }

   /* If we have no peer, return immediately */ 
   if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
      return 0;

   payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);

   rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
   rtp->send_duration = 160;
   rtp->lastdigitts = rtp->lastts + rtp->send_duration;
   
   /* Get a pointer to the header */
   rtpheader = (unsigned int *)data;
   rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
   rtpheader[1] = htonl(rtp->lastdigitts);
   rtpheader[2] = htonl(rtp->ssrc); 

   for (i = 0; i < 2; i++) {
      rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
      res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
      if (res < 0) 
         ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
            ast_inet_ntoa(rtp->them.sin_addr),
            ntohs(rtp->them.sin_port), strerror(errno));
      if (rtp_debug_test_addr(&rtp->them))
         ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
                ast_inet_ntoa(rtp->them.sin_addr),
                ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
      /* Increment sequence number */
      rtp->seqno++;
      /* Increment duration */
      rtp->send_duration += 160;
      /* Clear marker bit and set seqno */
      rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
   }

   /* Since we received a begin, we can safely store the digit and disable any compensation */
   rtp->sending_digit = 1;
   rtp->send_digit = digit;
   rtp->send_payload = payload;

   return 0;
}
int ast_rtp_senddigit_end ( struct ast_rtp rtp,
char  digit 
)

Definition at line 3289 of file rtp.c.

References ast_rtp_senddigit_end_with_duration().

Referenced by mgcp_senddigit_end(), and oh323_digit_end().

{
   return ast_rtp_senddigit_end_with_duration(rtp, digit, 0);
}
int ast_rtp_senddigit_end_with_duration ( struct ast_rtp rtp,
char  digit,
unsigned int  duration 
)

Send end packets for DTMF.

Definition at line 3295 of file rtp.c.

References ast_debug, ast_inet_ntoa(), ast_log(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), ast_rtp::dtmfmute, errno, ast_rtp::f, ast_rtp::lastdigitts, ast_rtp::lastts, LOG_ERROR, LOG_WARNING, rtp_debug_test_addr(), rtp_get_rate(), 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, ast_frame::subclass, and ast_rtp::them.

Referenced by ast_rtp_senddigit_end(), and sip_senddigit_end().

{
   unsigned int *rtpheader;
   int hdrlen = 12, res = 0, i = 0;
   char data[256];
   unsigned int measured_samples;
   
   /* If no address, then bail out */
   if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
      return 0;
   
   if ((digit <= '9') && (digit >= '0'))
      digit -= '0';
   else if (digit == '*')
      digit = 10;
   else if (digit == '#')
      digit = 11;
   else if ((digit >= 'A') && (digit <= 'D'))
      digit = digit - 'A' + 12;
   else if ((digit >= 'a') && (digit <= 'd'))
      digit = digit - 'a' + 12;
   else {
      ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
      return 0;
   }

   rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));

   if (duration > 0 && (measured_samples = duration * rtp_get_rate(rtp->f.subclass) / 1000) > rtp->send_duration) {
      ast_debug(2, "Adjusting final end duration from %u to %u\n", rtp->send_duration, measured_samples);
      rtp->send_duration = measured_samples;
   }

   rtpheader = (unsigned int *)data;
   rtpheader[1] = htonl(rtp->lastdigitts);
   rtpheader[2] = htonl(rtp->ssrc);
   rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
   /* Set end bit */
   rtpheader[3] |= htonl((1 << 23));

   /* Send 3 termination packets */
   for (i = 0; i < 3; i++) {
      rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
      res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
      rtp->seqno++;
      if (res < 0)
         ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
            ast_inet_ntoa(rtp->them.sin_addr),
            ntohs(rtp->them.sin_port), strerror(errno));
      if (rtp_debug_test_addr(&rtp->them))
         ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
                ast_inet_ntoa(rtp->them.sin_addr),
                ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
   }
   rtp->lastts += rtp->send_duration;
   rtp->sending_digit = 0;
   rtp->send_digit = 0;

   return res;
}
void ast_rtp_set_alt_peer ( struct ast_rtp rtp,
struct sockaddr_in *  alt 
)

set potential alternate source for RTP media

Since:
1.4.26 This function may be used to give the RTP stack a hint that there is a potential second source of media. One case where this is used is when the SIP stack receives a REINVITE to which it will be replying with a 491. In such a scenario, the IP and port information in the SDP of that REINVITE lets us know that we may receive media from that source/those sources even though the SIP transaction was unable to be completed successfully
Parameters:
rtpThe RTP structure we wish to set up an alternate host/port on
altThe address information for the alternate media source
Return values:
void

Definition at line 2718 of file rtp.c.

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

Referenced by handle_request_invite().

{
   rtp->altthem.sin_port = alt->sin_port;
   rtp->altthem.sin_addr = alt->sin_addr;
   if (rtp->rtcp) {
      rtp->rtcp->altthem.sin_port = htons(ntohs(alt->sin_port) + 1);
      rtp->rtcp->altthem.sin_addr = alt->sin_addr;
   }
}
void ast_rtp_set_callback ( struct ast_rtp rtp,
ast_rtp_callback  callback 
)

Definition at line 795 of file rtp.c.

References ast_rtp::callback.

Referenced by start_rtp().

{
   rtp->callback = callback;
}
void ast_rtp_set_data ( struct ast_rtp rtp,
void *  data 
)

Definition at line 790 of file rtp.c.

References ast_rtp::data.

Referenced by start_rtp().

{
   rtp->data = data;
}
void ast_rtp_set_m_type ( struct ast_rtp rtp,
int  pt 
)

Activate payload type.

Definition at line 2277 of file rtp.c.

References ast_rtp::current_RTP_PT, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().

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

{
   if (pt < 0 || pt >= MAX_RTP_PT || static_RTP_PT[pt].code == 0) 
      return; /* bogus payload type */

   rtp_bridge_lock(rtp);
   rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
   rtp_bridge_unlock(rtp);
} 
void ast_rtp_set_peer ( struct ast_rtp rtp,
struct sockaddr_in *  them 
)

Definition at line 2703 of file rtp.c.

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

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

{
   rtp->them.sin_port = them->sin_port;
   rtp->them.sin_addr = them->sin_addr;
   if (rtp->rtcp) {
      int h = ntohs(them->sin_port);
      rtp->rtcp->them.sin_port = htons(h + 1);
      rtp->rtcp->them.sin_addr = them->sin_addr;
   }
   rtp->rxseqno = 0;
   /* If strict RTP protection is enabled switch back to the learn state so we don't drop packets from above */
   if (strictrtp)
      rtp->strict_rtp_state = STRICT_RTP_LEARN;
}
void ast_rtp_set_rtpholdtimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp hold timeout.

Definition at line 757 of file rtp.c.

References ast_rtp::rtpholdtimeout.

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

{
   rtp->rtpholdtimeout = timeout;
}
void ast_rtp_set_rtpkeepalive ( struct ast_rtp rtp,
int  period 
)

set RTP keepalive interval

Definition at line 763 of file rtp.c.

References ast_rtp::rtpkeepalive.

Referenced by create_addr_from_peer(), and sip_alloc().

{
   rtp->rtpkeepalive = period;
}
int ast_rtp_set_rtpmap_type ( struct ast_rtp rtp,
int  pt,
char *  mimeType,
char *  mimeSubtype,
enum ast_rtp_options  options 
)

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

Parameters:
rtpRTP structure to modify
ptPayload type entry to modify
mimeTypetop-level MIME type of media stream (typically "audio", "video", "text", etc.)
mimeSubtypeMIME subtype of media stream (typically a codec name)
optionsZero or more flags from the ast_rtp_options enum

This function 'fills in' an entry in the list of possible formats for a media stream associated with an RTP structure.

Return values:
0on success
-1if the payload type is out of range
-2if the mimeType/mimeSubtype combination was not found

Definition at line 2353 of file rtp.c.

References ast_rtp_set_rtpmap_type_rate().

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

{
   return ast_rtp_set_rtpmap_type_rate(rtp, pt, mimeType, mimeSubtype, options, 0);
}
int ast_rtp_set_rtpmap_type_rate ( struct ast_rtp rtp,
int  pt,
char *  mimeType,
char *  mimeSubtype,
enum ast_rtp_options  options,
unsigned int  sample_rate 
)

Set payload type to a known MIME media type for a codec with a specific sample rate.

Parameters:
rtpRTP structure to modify
ptPayload type entry to modify
mimeTypetop-level MIME type of media stream (typically "audio", "video", "text", etc.)
mimeSubtypeMIME subtype of media stream (typically a codec name)
optionsZero or more flags from the ast_rtp_options enum
sample_rateThe sample rate of the media stream

This function 'fills in' an entry in the list of possible formats for a media stream associated with an RTP structure.

Return values:
0on success
-1if the payload type is out of range
-2if the mimeType/mimeSubtype combination was not found

Set payload type to a known MIME media type for a codec with a specific sample rate.

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

Definition at line 2304 of file rtp.c.

References ARRAY_LEN, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_RTP_OPT_G726_NONSTANDARD, rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, mimeTypes, mimeType::payloadType, rtp_bridge_lock(), rtp_bridge_unlock(), mimeType::sample_rate, mimeType::subtype, and mimeType::type.

Referenced by ast_rtp_set_rtpmap_type(), process_sdp_a_audio(), and process_sdp_a_video().

{
   unsigned int i;
   int found = 0;

   if (pt < 0 || pt >= MAX_RTP_PT)
      return -1; /* bogus payload type */

   rtp_bridge_lock(rtp);

   for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
      const struct mimeType *t = &mimeTypes[i];

      if (strcasecmp(mimeSubtype, t->subtype)) {
         continue;
      }

      if (strcasecmp(mimeType, t->type)) {
         continue;
      }

      /* if both sample rates have been supplied, and they don't match,
         then this not a match; if one has not been supplied, then the
         rates are not compared */
      if (sample_rate && t->sample_rate &&
          (sample_rate != t->sample_rate)) {
         continue;
      }

      found = 1;
      rtp->current_RTP_PT[pt] = t->payloadType;

      if ((t->payloadType.code == AST_FORMAT_G726) &&
          t->payloadType.isAstFormat &&
          (options & AST_RTP_OPT_G726_NONSTANDARD)) {
         rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
      }

      break;
   }

   rtp_bridge_unlock(rtp);

   return (found ? 0 : -2);
}
void ast_rtp_set_rtptimeout ( struct ast_rtp rtp,
int  timeout 
)

Set rtp timeout.

Definition at line 751 of file rtp.c.

References ast_rtp::rtptimeout.

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

{
   rtp->rtptimeout = timeout;
}
void ast_rtp_set_rtptimers_onhold ( struct ast_rtp rtp)

Definition at line 744 of file rtp.c.

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

Referenced by handle_response_invite().

{
   rtp->rtptimeout = (-1) * rtp->rtptimeout;
   rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
}
void ast_rtp_set_vars ( struct ast_channel chan,
struct ast_rtp rtp 
)

Set RTPAUDIOQOS(...) variables on a channel when it is being hung up.

Since:
1.6.1

Definition at line 2882 of file rtp.c.

References ast_bridged_channel(), ast_rtp_get_quality(), ast_channel::bridge, pbx_builtin_setvar_helper(), RTPQOS_JITTER, RTPQOS_LOSS, RTPQOS_RTT, and RTPQOS_SUMMARY.

Referenced by handle_request_bye(), and sip_hangup().

                                                                     {
   char *audioqos;
   char *audioqos_jitter;
   char *audioqos_loss;
   char *audioqos_rtt;
   struct ast_channel *bridge;

   if (!rtp || !chan)
      return;

   bridge = ast_bridged_channel(chan);

   audioqos        = ast_rtp_get_quality(rtp, NULL, RTPQOS_SUMMARY);
   audioqos_jitter = ast_rtp_get_quality(rtp, NULL, RTPQOS_JITTER);
   audioqos_loss   = ast_rtp_get_quality(rtp, NULL, RTPQOS_LOSS);
   audioqos_rtt    = ast_rtp_get_quality(rtp, NULL, RTPQOS_RTT);

   pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", audioqos);
   pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", audioqos_jitter);
   pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", audioqos_loss);
   pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", audioqos_rtt);

   if (!bridge)
      return;

   pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", audioqos);
   pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", audioqos_jitter);
   pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", audioqos_loss);
   pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", audioqos_rtt);
}
void ast_rtp_setdtmf ( struct ast_rtp rtp,
int  dtmf 
)

Indicate whether this RTP session is carrying DTMF or not.

Definition at line 810 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().

{
   ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
}
void ast_rtp_setdtmfcompensate ( struct ast_rtp rtp,
int  compensate 
)

Compensate for devices that send RFC2833 packets all at once.

Definition at line 815 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().

{
   ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
}
void ast_rtp_setnat ( struct ast_rtp rtp,
int  nat 
)

Definition at line 800 of file rtp.c.

References nat, and ast_rtp::nat.

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

{
   rtp->nat = nat;
}
int ast_rtp_setqos ( struct ast_rtp rtp,
int  tos,
int  cos,
char *  desc 
)

Definition at line 2679 of file rtp.c.

References ast_netsock_set_qos(), and ast_rtp::s.

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

{
   return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
}
void ast_rtp_setstun ( struct ast_rtp rtp,
int  stun_enable 
)

Enable STUN capability.

Definition at line 820 of file rtp.c.

References ast_set2_flag, and FLAG_HAS_STUN.

Referenced by gtalk_new().

{
   ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
}
void ast_rtp_stop ( struct ast_rtp rtp)

Stop RTP session, do not destroy structure

Definition at line 2757 of file rtp.c.

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

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

{
   if (rtp->rtcp) {
      AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
   }
   if (rtp->red) {
      AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
      free(rtp->red);
      rtp->red = NULL;
   }

   memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
   memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
   if (rtp->rtcp) {
      memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
      memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
   }
   
   ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
}
void ast_rtp_stun_request ( struct ast_rtp rtp,
struct sockaddr_in *  suggestion,
const char *  username 
)

Send STUN request for an RTP socket Deprecated, this is just a wrapper for ast_rtp_stun_request()

Definition at line 699 of file rtp.c.

References ast_stun_request(), and ast_rtp::s.

Referenced by gtalk_update_stun(), and jingle_update_stun().

{
   ast_stun_request(rtp->s, suggestion, username, NULL);
}
void ast_rtp_unset_m_type ( struct ast_rtp rtp,
int  pt 
)

clear payload type

Definition at line 2289 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, rtp_bridge_lock(), and rtp_bridge_unlock().

Referenced by process_sdp_a_audio(), and process_sdp_a_video().

{
   if (pt < 0 || pt >= MAX_RTP_PT)
      return; /* bogus payload type */

   rtp_bridge_lock(rtp);
   rtp->current_RTP_PT[pt].isAstFormat = 0;
   rtp->current_RTP_PT[pt].code = 0;
   rtp_bridge_unlock(rtp);
}
int ast_rtp_write ( struct ast_rtp rtp,
struct ast_frame f 
)

Bug:
XXX this might never be free'd. Why do we do this?

Definition at line 3848 of file rtp.c.

References ast_codec_pref_getsize(), ast_debug, AST_FORMAT_G723_1, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SPEEX, AST_FRAME_TEXT, 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_WARNING, ast_frame::offset, ast_rtp::pref, ast_frame::ptr, ast_rtp::red, red_t140_to_red(), ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.

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

{
   struct ast_frame *f;
   int codec;
   int hdrlen = 12;
   int subclass;
   

   /* If we have no peer, return immediately */ 
   if (!rtp->them.sin_addr.s_addr)
      return 0;

   /* If there is no data length, return immediately */
   if (!_f->datalen && !rtp->red)
      return 0;
   
   /* Make sure we have enough space for RTP header */
   if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO) && (_f->frametype != AST_FRAME_TEXT)) {
      ast_log(LOG_WARNING, "RTP can only send voice, video and text\n");
      return -1;
   }

   if (rtp->red) {
      /* return 0; */
      /* no primary data or generations to send */
      if ((_f = red_t140_to_red(rtp->red)) == NULL) 
         return 0;
   }

   /* The bottom bit of a video subclass contains the marker bit */
   subclass = _f->subclass;
   if (_f->frametype == AST_FRAME_VIDEO)
      subclass &= ~0x1;

   codec = ast_rtp_lookup_code(rtp, 1, subclass);
   if (codec < 0) {
      ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
      return -1;
   }

   if (rtp->lasttxformat != subclass) {
      /* New format, reset the smoother */
      ast_debug(1, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
      rtp->lasttxformat = subclass;
      if (rtp->smoother)
         ast_smoother_free(rtp->smoother);
      rtp->smoother = NULL;
   }

   if (!rtp->smoother) {
      struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);

      switch (subclass) {
      case AST_FORMAT_SPEEX:
      case AST_FORMAT_G723_1:
      case AST_FORMAT_SIREN7:
      case AST_FORMAT_SIREN14:
         /* these are all frame-based codecs and cannot be safely run through
            a smoother */
         break;
      default:
         if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
            if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
               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));
               return -1;
            }
            if (fmt.flags)
               ast_smoother_set_flags(rtp->smoother, fmt.flags);
            ast_debug(1, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
         }
      }
   }
   if (rtp->smoother) {
      if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
         ast_smoother_feed_be(rtp->smoother, _f);
      } else {
         ast_smoother_feed(rtp->smoother, _f);
      }

      while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
         ast_rtp_raw_write(rtp, f, codec);
      }
   } else {
      /* Don't buffer outgoing frames; send them one-per-packet: */
      if (_f->offset < hdrlen) 
         f = ast_frdup(_f);   /*! \bug XXX this might never be free'd. Why do we do this? */
      else
         f = _f;
      if (f->data.ptr)
         ast_rtp_raw_write(rtp, f, codec);
      if (f != _f)
         ast_frfree(f);
   }
      
   return 0;
}
int ast_stun_request ( int  s,
struct sockaddr_in *  dst,
const char *  username,
struct sockaddr_in *  answer 
)

Generic STUN request send a generic stun request to the server specified.

Parameters:
sthe socket used to send the request
dstthe address of the STUN server
usernameif non null, add the username in the request
answerif non null, the function waits for a response and puts here the externally visible address.
Returns:
0 on success, other values on error. The interface it may change in the future.

Generic STUN request send a generic stun request to the server specified.

Parameters:
sthe socket used to send the request
dstthe address of the STUN server
usernameif non null, add the username in the request
answerif non null, the function waits for a response and puts here the externally visible address.
Returns:
0 on success, other values on error.

Definition at line 636 of file rtp.c.

References append_attr_string(), ast_log(), ast_poll, stun_attr::attr, stun_header::ies, LOG_WARNING, stun_header::msglen, stun_header::msgtype, s, STUN_BINDREQ, stun_get_mapped(), stun_handle_packet(), stun_req_id(), stun_send(), and STUN_USERNAME.

Referenced by ast_rtp_stun_request(), ast_sip_ouraddrfor(), and reload_config().

{
   struct stun_header *req;
   unsigned char reqdata[1024];
   int reqlen, reqleft;
   struct stun_attr *attr;
   int res = 0;
   int retry;
   
   req = (struct stun_header *)reqdata;
   stun_req_id(req);
   reqlen = 0;
   reqleft = sizeof(reqdata) - sizeof(struct stun_header);
   req->msgtype = 0;
   req->msglen = 0;
   attr = (struct stun_attr *)req->ies;
   if (username)
      append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
   req->msglen = htons(reqlen);
   req->msgtype = htons(STUN_BINDREQ);
   for (retry = 0; retry < 3; retry++) {  /* XXX make retries configurable */
      /* send request, possibly wait for reply */
      unsigned char reply_buf[1024];
      struct pollfd pfds = { .fd = s, .events = POLLIN, };
      struct sockaddr_in src;
      socklen_t srclen;

      res = stun_send(s, dst, req);
      if (res < 0) {
         ast_log(LOG_WARNING, "ast_stun_request send #%d failed error %d, retry\n",
            retry, res);
         continue;
      }
      if (answer == NULL)
         break;
      res = ast_poll(&pfds, 1, 3000);
      if (res <= 0)  /* timeout or error */
         continue;
      memset(&src, '\0', sizeof(src));
      srclen = sizeof(src);
      /* XXX pass -1 in the size, because stun_handle_packet might
       * write past the end of the buffer.
       */
      res = recvfrom(s, reply_buf, sizeof(reply_buf) - 1,
         0, (struct sockaddr *)&src, &srclen);
      if (res < 0) {
         ast_log(LOG_WARNING, "ast_stun_request recvfrom #%d failed error %d, retry\n",
            retry, res);
         continue;
      }
      memset(answer, '\0', sizeof(struct sockaddr_in));
      stun_handle_packet(s, &src, reply_buf, res,
         stun_get_mapped, answer);
      res = 0; /* signal regular exit */
      break;
   }
   return res;
}
void red_buffer_t140 ( struct ast_rtp rtp,
struct ast_frame f 
)

Buffer t.140 data.

Buffer t.140 data.

Parameters:
rtp
fframe

Definition at line 4981 of file rtp.c.

References rtp_red::buf_data, ast_frame::data, ast_frame::datalen, ast_frame::ptr, ast_rtp::red, rtp_red::t140, and ast_frame::ts.

Referenced by sip_write().

{
   if (f->datalen > -1) {
      struct rtp_red *red = rtp->red;
      memcpy(&red->buf_data[red->t140.datalen], f->data.ptr, f->datalen);
      red->t140.datalen += f->datalen;
      red->t140.ts = f->ts;
   }
}
int rtp_red_init ( struct ast_rtp rtp,
int  ti,
int *  red_data_pt,
int  num_gen 
)

Initalize t.140 redudancy.

Parameters:
titime between each t140red frame is sent
red_ptpayloadtype for RTP packet
ptpayloadtype numbers for each generation including primary data
num_gennumber of redundant generations, primary data excluded
Since:
1.6.1

Initalize t.140 redudancy.

Parameters:
rtp
tibuffer t140 for ti (msecs) before sending redundant frame
red_data_ptPayloadtypes for primary- and generation-data
num_gennumbers of generations (primary generation not encounted)

Definition at line 4942 of file rtp.c.

References ast_calloc, AST_FORMAT_T140RED, AST_FRAME_TEXT, ast_sched_add(), rtp_red::buf_data, ast_frame::data, ast_frame::datalen, ast_frame::frametype, rtp_red::hdrlen, rtp_red::num_gen, rtp_red::prev_ts, rtp_red::pt, ast_frame::ptr, ast_rtp::red, red_write(), ast_rtp::sched, rtp_red::schedid, ast_frame::subclass, rtp_red::t140, rtp_red::t140red, rtp_red::t140red_data, rtp_red::ti, and ast_frame::ts.

Referenced by process_sdp().

{
   struct rtp_red *r;
   int x;
   
   if (!(r = ast_calloc(1, sizeof(struct rtp_red))))
      return -1;

   r->t140.frametype = AST_FRAME_TEXT;
   r->t140.subclass = AST_FORMAT_T140RED;
   r->t140.data.ptr = &r->buf_data; 

   r->t140.ts = 0;
   r->t140red = r->t140;
   r->t140red.data.ptr = &r->t140red_data;
   r->t140red.datalen = 0;
   r->ti = ti;
   r->num_gen = num_gen;
   r->hdrlen = num_gen * 4 + 1;
   r->prev_ts = 0;

   for (x = 0; x < num_gen; x++) {
      r->pt[x] = red_data_pt[x];
      r->pt[x] |= 1 << 7; /* mark redundant generations pt */ 
      r->t140red_data[x*4] = r->pt[x];
   }
   r->t140red_data[x*4] = r->pt[x] = red_data_pt[x]; /* primary pt */
   r->schedid = ast_sched_add(rtp->sched, ti, red_write, rtp);
   rtp->red = r;

   r->t140.datalen = 0;
   
   return 0;
}