rtp.h File Reference

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

#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
#include <netinet/in.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
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
Typedefs
typedef int(*	ast_rtp_callback )(struct ast_rtp *rtp, struct ast_frame *f, void *data)
Functions
int	ast_rtcp_fd (struct ast_rtp *rtp)
struct ast_frame *	ast_rtcp_read (struct ast_rtp *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)
void	ast_rtp_destroy (struct ast_rtp *rtp)
int	ast_rtp_fd (struct ast_rtp *rtp)
void	ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats)
int	ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them)
void	ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us)
void	ast_rtp_init (void)
int	ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code)
char *	ast_rtp_lookup_mime_multiple (char *buf, int size, const int capability, const int isAstFormat)
char *	ast_rtp_lookup_mime_subtype (int isAstFormat, int code)
struct rtpPayloadType	ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt)
struct ast_rtp *	ast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
	Initializate a RTP session.
struct ast_rtp *	ast_rtp_new_with_bindaddr (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in)
	Initializate a RTP session using an in_addr structure.
void	ast_rtp_offered_from_local (struct ast_rtp *rtp, int local)
int	ast_rtp_proto_register (struct ast_rtp_protocol *proto)
void	ast_rtp_proto_unregister (struct ast_rtp_protocol *proto)
void	ast_rtp_pt_clear (struct ast_rtp *rtp)
void	ast_rtp_pt_default (struct ast_rtp *rtp)
struct ast_frame *	ast_rtp_read (struct ast_rtp *rtp)
void	ast_rtp_reload (void)
void	ast_rtp_reset (struct ast_rtp *rtp)
int	ast_rtp_sendcng (struct ast_rtp *rtp, int level)
int	ast_rtp_senddigit (struct ast_rtp *rtp, char digit)
void	ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback)
void	ast_rtp_set_data (struct ast_rtp *rtp, void *data)
void	ast_rtp_set_m_type (struct ast_rtp *rtp, int pt)
void	ast_rtp_set_peer (struct ast_rtp *rtp, struct sockaddr_in *them)
void	ast_rtp_set_rtpmap_type (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype)
void	ast_rtp_setnat (struct ast_rtp *rtp, int nat)
int	ast_rtp_settos (struct ast_rtp *rtp, int tos)
void	ast_rtp_stop (struct ast_rtp *rtp)
int	ast_rtp_write (struct ast_rtp *rtp, struct ast_frame *f)

---

Detailed Description

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

RTP is defined in RFC 3550.

Definition in file rtp.h.

---

Define Documentation

#define AST_RTP_CISCO_DTMF   (1 << 2)

DTMF (Cisco Proprietary)

Definition at line 46 of file rtp.h.

Referenced by ast_rtp_read().

#define AST_RTP_CN   (1 << 1)

'Comfort Noise' (RFC3389)

Definition at line 44 of file rtp.h.

Referenced by ast_rtp_read(), and ast_rtp_sendcng().

#define AST_RTP_DTMF   (1 << 0)

DTMF (RFC2833)

Definition at line 42 of file rtp.h.

Referenced by add_noncodec_to_sdp(), add_sdp(), ast_rtp_read(), ast_rtp_senddigit(), check_user_full(), create_addr(), create_addr_from_peer(), oh323_alloc(), oh323_request(), process_sdp(), and sip_alloc().

#define AST_RTP_MAX   AST_RTP_CISCO_DTMF

Maximum RTP-specific code

Definition at line 48 of file rtp.h.

Referenced by add_sdp(), and ast_rtp_lookup_mime_multiple().

---

Typedef Documentation

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

Definition at line 70 of file rtp.h.

---

Function Documentation

int ast_rtcp_fd

(

struct ast_rtp *

rtp

)

Definition at line 158 of file rtp.c.

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

Referenced by sip_new().

{
   if (rtp->rtcp)
      return rtp->rtcp->s;
   return -1;
}

struct ast_frame* ast_rtcp_read

(

struct ast_rtp *

rtp

)

[read]

Definition at line 370 of file rtp.c.

References AST_FRAME_NULL, ast_inet_ntoa(), ast_log(), CRASH, LOG_DEBUG, LOG_WARNING, ast_rtp::nat, option_debug, ast_rtp::rtcp, ast_rtcp::s, and ast_rtcp::them.

Referenced by sip_rtp_read().

{
   static struct ast_frame null_frame = { AST_FRAME_NULL, };
   socklen_t len;
   int hdrlen = 8;
   int res;
   struct sockaddr_in sin;
   unsigned int rtcpdata[1024];
   char iabuf[INET_ADDRSTRLEN];
   
   if (!rtp || !rtp->rtcp)
      return &null_frame;

   len = sizeof(sin);
   
   res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata),
               0, (struct sockaddr *)&sin, &len);
   
   if (res < 0) {
      if (errno == EBADF)
         CRASH;
      if (errno != EAGAIN) {
         ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up now.\n", strerror(errno));
         return NULL;
      }
      return &null_frame;
   }

   if (res < hdrlen) {
      ast_log(LOG_WARNING, "RTP Read too short\n");
      return &null_frame;
   }

   if (rtp->nat) {
      /* Send to whoever sent to us */
      if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
          (rtp->rtcp->them.sin_port != sin.sin_port)) {
         memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
         if (option_debug || rtpdebug)
            ast_log(LOG_DEBUG, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
      }
   }
   if (option_debug)
      ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
   return &null_frame;
}

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

Definition at line 1539 of file rtp.c.

References ast_channel::_softhangup, AST_BRIDGE_COMPLETE, AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, AST_BRIDGE_IGNORE_SIGS, AST_BRIDGE_RETRY, ast_check_hangup(), AST_CONTROL_HOLD, AST_CONTROL_UNHOLD, AST_CONTROL_VIDUPDATE, AST_FRAME_CONTROL, AST_FRAME_DTMF, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frfree(), ast_indicate(), ast_inet_ntoa(), ast_log(), ast_mutex_lock(), ast_mutex_trylock(), ast_mutex_unlock(), ast_read(), ast_rtp_get_peer(), ast_test_flag, ast_waitfor_n(), ast_write(), FLAG_NAT_ACTIVE, ast_frame::frametype, ast_rtp_protocol::get_codec, get_proto(), ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, inaddrcmp(), ast_channel::lock, LOG_DEBUG, LOG_WARNING, ast_channel::masq, ast_channel::masqr, ast_channel::name, option_debug, ast_rtp_protocol::set_rtp_peer, ast_frame::subclass, and ast_channel::tech_pvt.

{
   struct ast_frame *f;
   struct ast_channel *who, *cs[3];
   struct ast_rtp *p0, *p1;      /* Audio RTP Channels */
   struct ast_rtp *vp0, *vp1;    /* Video RTP channels */
   struct ast_rtp_protocol *pr0, *pr1;
   struct sockaddr_in ac0, ac1;
   struct sockaddr_in vac0, vac1;
   struct sockaddr_in t0, t1;
   struct sockaddr_in vt0, vt1;
   char iabuf[INET_ADDRSTRLEN];
   
   void *pvt0, *pvt1;
   int codec0,codec1, oldcodec0, oldcodec1;
   
   memset(&vt0, 0, sizeof(vt0));
   memset(&vt1, 0, sizeof(vt1));
   memset(&vac0, 0, sizeof(vac0));
   memset(&vac1, 0, sizeof(vac1));

   /* if need DTMF, cant native bridge */
   if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
      return AST_BRIDGE_FAILED_NOWARN;

   /* Lock channels */
   ast_mutex_lock(&c0->lock);
   while(ast_mutex_trylock(&c1->lock)) {
      ast_mutex_unlock(&c0->lock);
      usleep(1);
      ast_mutex_lock(&c0->lock);
   }

   /* Find channel driver interfaces */
   pr0 = get_proto(c0);
   pr1 = get_proto(c1);
   if (!pr0) {
      ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
      ast_mutex_unlock(&c0->lock);
      ast_mutex_unlock(&c1->lock);
      return AST_BRIDGE_FAILED;
   }
   if (!pr1) {
      ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
      ast_mutex_unlock(&c0->lock);
      ast_mutex_unlock(&c1->lock);
      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) */
   p0 = pr0->get_rtp_info(c0);
   if (pr0->get_vrtp_info)
      vp0 = pr0->get_vrtp_info(c0);
   else
      vp0 = NULL;
   p1 = pr1->get_rtp_info(c1);
   if (pr1->get_vrtp_info)
      vp1 = pr1->get_vrtp_info(c1);
   else
      vp1 = NULL;

   /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
   if (!p0 || !p1) {
      /* Somebody doesn't want to play... */
      ast_mutex_unlock(&c0->lock);
      ast_mutex_unlock(&c1->lock);
      return AST_BRIDGE_FAILED_NOWARN;
   }
   /* Get codecs from both sides */
   if (pr0->get_codec)
      codec0 = pr0->get_codec(c0);
   else
      codec0 = 0;
   if (pr1->get_codec)
      codec1 = pr1->get_codec(c1);
   else
      codec1 = 0;
   if (pr0->get_codec && pr1->get_codec) {
      /* Hey, we can't do reinvite if both parties speak different codecs */
      if (!(codec0 & codec1)) {
         if (option_debug)
            ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
         ast_mutex_unlock(&c0->lock);
         ast_mutex_unlock(&c1->lock);
         return AST_BRIDGE_FAILED_NOWARN;
      }
   }

   /* Ok, we should be able to redirect the media. Start with one channel */
   if (pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE))) 
      ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
   else {
      /* Store RTP peer */
      ast_rtp_get_peer(p1, &ac1);
      if (vp1)
         ast_rtp_get_peer(vp1, &vac1);
   }
   /* Then test the other channel */
   if (pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
      ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
   else {
      /* Store RTP peer */
      ast_rtp_get_peer(p0, &ac0);
      if (vp0)
         ast_rtp_get_peer(vp0, &vac0);
   }
   ast_mutex_unlock(&c0->lock);
   ast_mutex_unlock(&c1->lock);
   /* External RTP Bridge up, now loop and see if something happes that force us to take the
      media back to Asterisk */
   cs[0] = c0;
   cs[1] = c1;
   cs[2] = NULL;
   oldcodec0 = codec0;
   oldcodec1 = codec1;
   for (;;) {
      /* Check if something changed... */
      if ((c0->tech_pvt != pvt0)  ||
         (c1->tech_pvt != pvt1) ||
         (c0->masq || c0->masqr || c1->masq || c1->masqr)) {
            ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
            if (c0->tech_pvt == pvt0) {
               if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0)) 
                  ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
            }
            if (c1->tech_pvt == pvt1) {
               if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0)) 
                  ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
            }
            return AST_BRIDGE_RETRY;
      }
      /* Now check if they have changed address */
      ast_rtp_get_peer(p1, &t1);
      ast_rtp_get_peer(p0, &t0);
      if (pr0->get_codec)
         codec0 = pr0->get_codec(c0);
      if (pr1->get_codec)
         codec1 = pr1->get_codec(c1);
      if (vp1)
         ast_rtp_get_peer(vp1, &vt1);
      if (vp0)
         ast_rtp_get_peer(vp0, &vt0);
      if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) {
         if (option_debug > 1) {
            ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", 
               c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t1.sin_addr), ntohs(t1.sin_port), codec1);
            ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n", 
               c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vt1.sin_addr), ntohs(vt1.sin_port), codec1);
            ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
               c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
            ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
               c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
         }
         if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE))) 
            ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
         memcpy(&ac1, &t1, sizeof(ac1));
         memcpy(&vac1, &vt1, sizeof(vac1));
         oldcodec1 = codec1;
      }
      if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) {
         if (option_debug) {
            ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", 
               c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t0.sin_addr), ntohs(t0.sin_port), codec0);
            ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
               c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
         }
         if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
            ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
         memcpy(&ac0, &t0, sizeof(ac0));
         memcpy(&vac0, &vt0, sizeof(vac0));
         oldcodec0 = codec0;
      }
      who = ast_waitfor_n(cs, 2, &timeoutms);
      if (!who) {
         if (!timeoutms) {
            if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
            if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
            return AST_BRIDGE_RETRY;
         }
         if (option_debug)
            ast_log(LOG_DEBUG, "Ooh, empty read...\n");
         /* check for hangup / whentohangup */
         if (ast_check_hangup(c0) || ast_check_hangup(c1))
            break;
         continue;
      }
      f = ast_read(who);
      if (!f || ((f->frametype == AST_FRAME_DTMF) &&
               (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 
                ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
         *fo = f;
         *rc = who;
         if (option_debug)
            ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
         if ((c0->tech_pvt == pvt0) && (!c0->_softhangup)) {
            if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0)) 
               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
         }
         if ((c1->tech_pvt == pvt1) && (!c1->_softhangup)) {
            if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0)) 
               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
         }
         return AST_BRIDGE_COMPLETE;
      } else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
         if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
             (f->subclass == AST_CONTROL_VIDUPDATE)) {
            ast_indicate(who == c0 ? c1 : c0, f->subclass);
            ast_frfree(f);
         } else {
            *fo = f;
            *rc = who;
            ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name);
            return AST_BRIDGE_COMPLETE;
         }
      } else {
         if ((f->frametype == AST_FRAME_DTMF) || 
            (f->frametype == AST_FRAME_VOICE) || 
            (f->frametype == AST_FRAME_VIDEO)) {
            /* Forward voice or DTMF frames if they happen upon us */
            if (who == c0) {
               ast_write(c1, f);
            } else if (who == c1) {
               ast_write(c0, f);
            }
         }
         ast_frfree(f);
      }
      /* Swap priority not that it's a big deal at this point */
      cs[2] = cs[0];
      cs[0] = cs[1];
      cs[1] = cs[2];
      
   }

   if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
      ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
   if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
      ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);

   return AST_BRIDGE_FAILED;
}

void ast_rtp_destroy

(

struct ast_rtp *

rtp

)

Definition at line 1105 of file rtp.c.

References ast_io_remove(), ast_smoother_free(), free, ast_rtp::io, ast_rtp::ioid, ast_rtp::rtcp, ast_rtcp::s, ast_rtp::s, and ast_rtp::smoother.

Referenced by __oh323_destroy(), __sip_destroy(), cleanup_connection(), destroy_endpoint(), mgcp_hangup(), skinny_hangup(), start_rtp(), and unalloc_sub().

{
   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) {
      close(rtp->rtcp->s);
      free(rtp->rtcp);
   }
   free(rtp);
}

int ast_rtp_fd

(

struct ast_rtp *

rtp

)

Definition at line 153 of file rtp.c.

References ast_rtp::s.

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

{
   return rtp->s;
}

void ast_rtp_get_current_formats	(	struct ast_rtp *	rtp,
		int *	astFormats,
		int *	nonAstFormats
	)

Definition at line 782 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, and MAX_RTP_PT.

Referenced by process_sdp().

                                                   {
   int pt;

   *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;
      }
   }
}

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

Definition at line 1057 of file rtp.c.

References ast_rtp::them.

Referenced by add_sdp(), ast_rtp_bridge(), do_monitor(), oh323_set_rtp_peer(), sip_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;
}

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

Definition at line 1070 of file rtp.c.

References ast_rtp::us.

Referenced by add_sdp(), external_rtp_create(), handle_message(), and oh323_set_rtp_peer().

{
   memcpy(us, &rtp->us, sizeof(rtp->us));
}

void ast_rtp_init

(

void

)

Definition at line 1913 of file rtp.c.

References ast_cli_register(), and ast_rtp_reload().

Referenced by main().

{
   ast_cli_register(&cli_debug);
   ast_cli_register(&cli_debug_ip);
   ast_cli_register(&cli_no_debug);
   ast_rtp_reload();
}

int ast_rtp_lookup_code	(	struct ast_rtp *	rtp,
		int	isAstFormat,
		int	code
	)

Definition at line 822 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), ast_rtp_sendcng(), ast_rtp_senddigit(), and ast_rtp_write().

                                                                                    {

   int pt;

   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 */
      return rtp->rtp_lookup_code_cache_result;
   }

   /* 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;
         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;
         return pt;
      }
   }
   return -1;
}

char* ast_rtp_lookup_mime_multiple	(	char *	buf,
		int	size,
		const int	capability,
		const int	isAstFormat
	)

Definition at line 867 of file rtp.c.

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

   if (start == end)
      snprintf(start, size, "nothing)"); 
   else if (size > 1)
      *(end -1) = ')';
   
   return buf;
}

char* ast_rtp_lookup_mime_subtype	(	int	isAstFormat,
		int	code
	)

Definition at line 855 of file rtp.c.

References rtpPayloadType::code, mimeTypes, and payloadType.

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

                                                                         {

   int i;

   for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
   if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat) {
            return mimeTypes[i].subtype;
      }
   }
   return "";
}

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

Definition at line 803 of file rtp.c.

References rtpPayloadType::code, rtpPayloadType::isAstFormat, and MAX_RTP_PT.

Referenced by ast_rtp_read(), 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 the negotiated codecs */
   if (!rtp->rtp_offered_from_local)
      result = rtp->current_RTP_PT[pt];

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

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 1029 of file rtp.c.

References ast_rtp_new_with_bindaddr().

Referenced by start_rtp().

{
   struct in_addr ia;

   memset(&ia, 0, sizeof(ia));
   return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
}

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 940 of file rtp.c.

References ast_io_add(), AST_IO_IN, ast_log(), ast_rtcp_new(), ast_rtp_pt_default(), free, ast_rtp::io, ast_rtp::ioid, LOG_ERROR, malloc, ast_rtp::rtcp, rtp_socket(), rtpread(), ast_rtcp::s, ast_rtp::s, ast_rtp::sched, ast_rtp::seqno, ast_rtp::ssrc, ast_rtp::them, ast_rtcp::us, and ast_rtp::us.

Referenced by ast_rtp_new(), oh323_alloc(), sip_alloc(), and start_rtp().

{
   struct ast_rtp *rtp;
   int x;
   int first;
   int startplace;
   rtp = malloc(sizeof(struct ast_rtp));
   if (!rtp)
      return NULL;
   memset(rtp, 0, sizeof(struct ast_rtp));
   rtp->them.sin_family = AF_INET;
   rtp->us.sin_family = AF_INET;
   rtp->s = rtp_socket();
   rtp->ssrc = rand();
   rtp->seqno = rand() & 0xffff;
   if (rtp->s < 0) {
      free(rtp);
      ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
      return NULL;
   }
   if (sched && rtcpenable) {
      rtp->sched = sched;
      rtp->rtcp = ast_rtcp_new();
   }
   
   /* Select a random port number in the range of possible RTP */
   x = (rand() % (rtpend-rtpstart)) + rtpstart;
   x = x & ~1;
   /* Save it for future references. */
   startplace = x;
   /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
   for (;;) {
      /* Must be an even port number by RTP spec */
      rtp->us.sin_port = htons(x);
      rtp->us.sin_addr = addr;
      /* If there's rtcp, initialize it as well. */
      if (rtp->rtcp) {
         rtp->rtcp->us.sin_addr = addr;
         rtp->rtcp->us.sin_port = htons(x + 1);
      }
      /* Try to bind it/them. */
      if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
         (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
         break;
      if (!first) {
         /* Primary bind succeeded! Gotta recreate it */
         close(rtp->s);
         rtp->s = rtp_socket();
      }
      if (errno != EADDRINUSE) {
         /* We got an error that wasn't expected, abort! */
         ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
         close(rtp->s);
         if (rtp->rtcp) {
            close(rtp->rtcp->s);
            free(rtp->rtcp);
         }
         free(rtp);
         return NULL;
      }
      /* The port was used, increment it (by two). */
      x += 2;
      /* Did we go over the limit ? */
      if (x > rtpend)
         /* then, start from the begingig. */
         x = (rtpstart + 1) & ~1;
      /* Check if we reached the place were we started. */
      if (x == startplace) {
         /* If so, there's no ports available. */
         ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
         close(rtp->s);
         if (rtp->rtcp) {
            close(rtp->rtcp->s);
            free(rtp->rtcp);
         }
         free(rtp);
         return NULL;
      }
   }
   if (io && sched && callbackmode) {
      /* Operate this one in a callback mode */
      rtp->sched = sched;
      rtp->io = io;
      rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
   }
   ast_rtp_pt_default(rtp);
   return rtp;
}

void ast_rtp_offered_from_local	(	struct ast_rtp *	rtp,
		int	local
	)

Definition at line 796 of file rtp.c.

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

Referenced by transmit_connect_with_sdp(), transmit_invite(), transmit_modify_with_sdp(), transmit_reinvite_with_sdp(), and transmit_response_with_sdp().

                                                                {
   if (rtp)
      rtp->rtp_offered_from_local = local;
   else
      ast_log(LOG_WARNING, "rtp structure is null\n");
}

int ast_rtp_proto_register

(

struct ast_rtp_protocol *

proto

)

Definition at line 1505 of file rtp.c.

References ast_log(), LOG_WARNING, ast_rtp_protocol::next, and ast_rtp_protocol::type.

Referenced by load_module(), and unload_module().

{
   struct ast_rtp_protocol *cur;
   cur = protos;
   while(cur) {
      if (cur->type == proto->type) {
         ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
         return -1;
      }
      cur = cur->next;
   }
   proto->next = protos;
   protos = proto;
   return 0;
}

void ast_rtp_proto_unregister

(

struct ast_rtp_protocol *

proto

)

Definition at line 1485 of file rtp.c.

References ast_rtp_protocol::next.

Referenced by unload_module().

{
   struct ast_rtp_protocol *cur, *prev;

   cur = protos;
   prev = NULL;
   while(cur) {
      if (cur == proto) {
         if (prev)
            prev->next = proto->next;
         else
            protos = proto->next;
         return;
      }
      prev = cur;
      cur = cur->next;
   }
}

void ast_rtp_pt_clear

(

struct ast_rtp *

rtp

)

Definition at line 719 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by process_sdp().

{
   int i;
   if (!rtp)
      return;

   for (i = 0; i < MAX_RTP_PT; ++i) {
      rtp->current_RTP_PT[i].isAstFormat = 0;
      rtp->current_RTP_PT[i].code = 0;
   }

   rtp->rtp_lookup_code_cache_isAstFormat = 0;
   rtp->rtp_lookup_code_cache_code = 0;
   rtp->rtp_lookup_code_cache_result = 0;
}

void ast_rtp_pt_default

(

struct ast_rtp *

rtp

)

Definition at line 735 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, MAX_RTP_PT, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by ast_rtp_new_with_bindaddr().

{
   int i;

   /* Initialize to default payload types */
   for (i = 0; i < MAX_RTP_PT; ++i) {
      rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
      rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
   }

   rtp->rtp_lookup_code_cache_isAstFormat = 0;
   rtp->rtp_lookup_code_cache_code = 0;
   rtp->rtp_lookup_code_cache_result = 0;
}

struct ast_frame* ast_rtp_read

(

struct ast_rtp *

rtp

)

[read]

Definition at line 428 of file rtp.c.

References ast_codec_get_samples(), AST_FORMAT_MAX_AUDIO, AST_FORMAT_SLINEAR, ast_frame_byteswap_be, AST_FRAME_NULL, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_inet_ntoa(), ast_log(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_lookup_pt(), ast_set_flag, ast_verbose(), calc_rxstamp(), rtpPayloadType::code, CRASH, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp::dtmfcount, ast_rtp::f, FLAG_NAT_ACTIVE, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lasteventseqn, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxts, LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, option_debug, process_cisco_dtmf(), process_rfc2833(), process_rfc3389(), ast_rtp::rawdata, ast_rtp::resp, rtp_debug_test_addr(), ast_rtp::rxseqno, ast_rtp::rxssrc, ast_rtp::s, ast_frame::samples, send_dtmf(), ast_frame::src, ast_frame::subclass, and ast_rtp::them.

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

{
   int res;
   struct sockaddr_in sin;
   socklen_t len;
   unsigned int seqno;
   int version;
   int payloadtype;
   int hdrlen = 12;
   int padding;
   int mark;
   int ext;
   int cc;
   int x;
   char iabuf[INET_ADDRSTRLEN];
   unsigned int ssrc;
   unsigned int timestamp;
   unsigned int *rtpheader;
   struct ast_frame *f;
   static struct ast_frame null_frame = { AST_FRAME_NULL, };
   struct rtpPayloadType rtpPT;
   
   len = sizeof(sin);
   
   /* Cache where the header will go */
   res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
               0, (struct sockaddr *)&sin, &len);


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

   /* Ignore if the other side hasn't been given an address
      yet.  */
   if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
      return &null_frame;

   if (rtp->nat) {
      /* Send to whoever sent to us */
      if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
          (rtp->them.sin_port != sin.sin_port)) {
         memcpy(&rtp->them, &sin, sizeof(rtp->them));
         rtp->rxseqno = 0;
         ast_set_flag(rtp, FLAG_NAT_ACTIVE);
         if (option_debug || rtpdebug)
            ast_log(LOG_DEBUG, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
      }
   }

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

   /* Check RTP version */
   version = (seqno & 0xC0000000) >> 30;
   if (version != 2)
      return &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]);
   
   if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
      if (option_debug || rtpdebug)
         ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
      mark = 1;
   }

   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 (res < hdrlen) {
      ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
      return &null_frame;
   }

   if(rtp_debug_test_addr(&sin))
      ast_verbose("Got RTP packet from %s:%u (type %d, seq %u, ts %u, len %d)\n"
         , ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);

   rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
   if (!rtpPT.isAstFormat) {
      /* 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(&sin)) {
            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 rfc2833 RTP packet from %s:%d (type %d, seq %d, ts %d, len %d, mark %d, event %08x, end %d, duration %d) \n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
         }
         if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) {
            f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno);
            rtp->lasteventseqn = seqno;
         } else
            f = NULL;
         if (f)
            return f;
         else
            return &null_frame;
      } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
         /* It's really special -- process it the Cisco way */
         if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) {
            f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
            rtp->lasteventseqn = seqno;
         } else 
            f = NULL;
            if (f) 
            return f; 
         else 
            return &null_frame;
      } else if (rtpPT.code == AST_RTP_CN) {
         /* Comfort Noise */
         f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
         if (f) 
            return f; 
         else 
            return &null_frame;
      } else {
         ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
         return &null_frame;
      }
   }
   rtp->f.subclass = rtpPT.code;
   if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO)
      rtp->f.frametype = AST_FRAME_VOICE;
   else
      rtp->f.frametype = AST_FRAME_VIDEO;
   rtp->lastrxformat = rtp->f.subclass;

   if (!rtp->lastrxts)
      rtp->lastrxts = timestamp;

   if (rtp->rxseqno) {
      for (x=rtp->rxseqno + 1; x < seqno; x++) {
         /* Queue empty frames */
         rtp->f.mallocd = 0;
         rtp->f.datalen = 0;
         rtp->f.data = NULL;
         rtp->f.offset = 0;
         rtp->f.samples = 0;
         rtp->f.src = "RTPMissedFrame";
      }
   }
   rtp->rxseqno = seqno;

   if (rtp->dtmfcount) {
#if 0
      printf("dtmfcount was %d\n", rtp->dtmfcount);
#endif      
      rtp->dtmfcount -= (timestamp - rtp->lastrxts);
      if (rtp->dtmfcount < 0)
         rtp->dtmfcount = 0;
#if 0
      if (dtmftimeout != rtp->dtmfcount)
         printf("dtmfcount is %d\n", rtp->dtmfcount);
#endif
   }
   rtp->lastrxts = timestamp;

   /* Send any pending DTMF */
   if (rtp->resp && !rtp->dtmfcount) {
      if (option_debug)
         ast_log(LOG_DEBUG, "Sending pending DTMF\n");
      return send_dtmf(rtp);
   }
   rtp->f.mallocd = 0;
   rtp->f.datalen = res - hdrlen;
   rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
   rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
   if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
      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);
   } else {
      /* 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;
      if (mark)
         rtp->f.subclass |= 0x1;
      
   }
   rtp->f.src = "RTP";
   return &rtp->f;
}

void ast_rtp_reload

(

void

)

Definition at line 1857 of file rtp.c.

References ast_config_destroy(), ast_config_load(), ast_false(), ast_log(), ast_variable_retrieve(), ast_verbose(), cfg, DEFAULT_DTMF_TIMEOUT, LOG_WARNING, option_verbose, s, and VERBOSE_PREFIX_2.

Referenced by ast_module_reload(), ast_rtp_init(), and main().

{
   struct ast_config *cfg;
   char *s;

   rtpstart = 5000;
   rtpend = 31000;
   dtmftimeout = DEFAULT_DTMF_TIMEOUT;
   cfg = ast_config_load("rtp.conf");
   if (cfg) {
      if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
         rtpstart = atoi(s);
         if (rtpstart < 1024)
            rtpstart = 1024;
         if (rtpstart > 65535)
            rtpstart = 65535;
      }
      if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
         rtpend = atoi(s);
         if (rtpend < 1024)
            rtpend = 1024;
         if (rtpend > 65535)
            rtpend = 65535;
      }
      if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
#ifdef SO_NO_CHECK
         if (ast_false(s))
            nochecksums = 1;
         else
            nochecksums = 0;
#else
         if (ast_false(s))
            ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
#endif
      }
      if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
         dtmftimeout = atoi(s);
         if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
            ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
               dtmftimeout, DEFAULT_DTMF_TIMEOUT);
            dtmftimeout = DEFAULT_DTMF_TIMEOUT;
         };
      }
      ast_config_destroy(cfg);
   }
   if (rtpstart >= rtpend) {
      ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
      rtpstart = 5000;
      rtpend = 31000;
   }
   if (option_verbose > 1)
      ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
   
}

void ast_rtp_reset

(

struct ast_rtp *

rtp

)

Definition at line 1085 of file rtp.c.

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

{
   memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
   memset(&rtp->txcore, 0, sizeof(rtp->txcore));
   memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
   rtp->lastts = 0;
   rtp->lastdigitts = 0;
   rtp->lastrxts = 0;
   rtp->lastividtimestamp = 0;
   rtp->lastovidtimestamp = 0;
   rtp->lasteventseqn = 0;
   rtp->lasteventendseqn = 0;
   rtp->lasttxformat = 0;
   rtp->lastrxformat = 0;
   rtp->dtmfcount = 0;
   rtp->dtmfduration = 0;
   rtp->seqno = 0;
   rtp->rxseqno = 0;
}

int ast_rtp_sendcng	(	struct ast_rtp *	rtp,
		int	level
	)

Definition at line 1221 of file rtp.c.

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

Referenced by do_monitor().

{
   unsigned int *rtpheader;
   int hdrlen = 12;
   int res;
   int payload;
   char data[256];
   char iabuf[INET_ADDRSTRLEN];
   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(iabuf, sizeof(iabuf), 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:%d (type %d, seq %d, ts %d, len %d)\n"
               , ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);        
         
   }
   return 0;
}

int ast_rtp_senddigit	(	struct ast_rtp *	rtp,
		char	digit
	)

Definition at line 1139 of file rtp.c.

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

Referenced by oh323_digit(), and sip_senddigit().

{
   unsigned int *rtpheader;
   int hdrlen = 12;
   int res;
   int x;
   int payload;
   char data[256];
   char iabuf[INET_ADDRSTRLEN];

   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 -1;
   }
   payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);

   /* 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->lastdigitts);
   rtpheader[2] = htonl(rtp->ssrc); 
   rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
   for (x = 0; x < 6; x++) {
      if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
         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:%d: %s\n",
               ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
               ntohs(rtp->them.sin_port), strerror(errno));
         if (rtp_debug_test_addr(&rtp->them))
            ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n",
                   ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
                   ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
      }
      /* Sequence number of last two end packets does not get incremented */
      if (x < 3)
         rtp->seqno++;
      /* Clear marker bit and set seqno */
      rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
      /* For the last three packets, set the duration and the end bit */
      if (x == 2) {
#if 0
         /* No, this is wrong...  Do not increment lastdigitts, that's not according
            to the RFC, as best we can determine */
         rtp->lastdigitts++; /* or else the SPA3000 will click instead of beeping... */
         rtpheader[1] = htonl(rtp->lastdigitts);
#endif         
         /* Make duration 800 (100ms) */
         rtpheader[3] |= htonl((800));
         /* Set the End bit */
         rtpheader[3] |= htonl((1 << 23));
      }
   }
   /* Increment the digit timestamp by 120ms, to ensure that digits
      sent sequentially with no intervening non-digit packets do not
      get sent with the same timestamp, and that sequential digits
      have some 'dead air' in between them
   */
   rtp->lastdigitts += 960;
   /* Increment the sequence number to reflect the last packet
      that was sent
   */
   rtp->seqno++;
   return 0;
}

void ast_rtp_set_callback	(	struct ast_rtp *	rtp,
		ast_rtp_callback	callback
	)

Definition at line 170 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 165 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
	)

Definition at line 753 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, and MAX_RTP_PT.

Referenced by process_sdp().

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

   if (static_RTP_PT[pt].code != 0) {
      rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
   }
} 

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

Definition at line 1046 of file rtp.c.

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

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

{
   rtp->them.sin_port = them->sin_port;
   rtp->them.sin_addr = them->sin_addr;
   if (rtp->rtcp) {
      rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
      rtp->rtcp->them.sin_addr = them->sin_addr;
   }
   rtp->rxseqno = 0;
}

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

Definition at line 764 of file rtp.c.

References ast_rtp::current_RTP_PT, MAX_RTP_PT, mimeTypes, subtype, and type.

Referenced by process_sdp(), and set_dtmf_payload().

                                             {
   int i;

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

   for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
      if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
           strcasecmp(mimeType, mimeTypes[i].type) == 0) {
         rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
      return;
      }
   }
} 

void ast_rtp_setnat	(	struct ast_rtp *	rtp,
		int	nat
	)

Definition at line 175 of file rtp.c.

References ast_rtp::nat.

Referenced by check_user_full(), create_addr(), create_addr_from_peer(), oh323_request(), oh323_rtp_read(), sip_alloc(), and start_rtp().

{
   rtp->nat = nat;
}

int ast_rtp_settos	(	struct ast_rtp *	rtp,
		int	tos
	)

Definition at line 1037 of file rtp.c.

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

Referenced by oh323_alloc(), and sip_alloc().

{
   int res;

   if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
      ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
   return res;
}

void ast_rtp_stop

(

struct ast_rtp *

rtp

)

Definition at line 1075 of file rtp.c.

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

Referenced by handle_request_bye(), handle_request_cancel(), handle_response(), and process_sdp().

{
   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->them.sin_addr));
      memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->them.sin_port));
   }
}

int ast_rtp_write	(	struct ast_rtp *	rtp,
		struct ast_frame *	f
	)

Definition at line 1337 of file rtp.c.

References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_H261, AST_FORMAT_H263, AST_FORMAT_H263_PLUS, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SLINEAR, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, 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_G729, ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_smoother_set_flags(), ast_frame::datalen, ast_frame::frametype, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, option_debug, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.

Referenced by mgcp_write(), oh323_write(), sip_write(), and skinny_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) 
      return 0;
   
   /* Make sure we have enough space for RTP header */
   if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
      ast_log(LOG_WARNING, "RTP can only send voice\n");
      return -1;
   }

   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 */
      if (option_debug)
         ast_log(LOG_DEBUG, "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;
   }


   switch(subclass) {
   case AST_FORMAT_SLINEAR:
      if (!rtp->smoother) {
         rtp->smoother = ast_smoother_new(320);
      }
      if (!rtp->smoother) {
         ast_log(LOG_WARNING, "Unable to create smoother :(\n");
         return -1;
      }
      ast_smoother_feed_be(rtp->smoother, _f);
      
      while((f = ast_smoother_read(rtp->smoother)))
         ast_rtp_raw_write(rtp, f, codec);
      break;
   case AST_FORMAT_ULAW:
   case AST_FORMAT_ALAW:
      if (!rtp->smoother) {
         rtp->smoother = ast_smoother_new(160);
      }
      if (!rtp->smoother) {
         ast_log(LOG_WARNING, "Unable to create smoother :(\n");
         return -1;
      }
      ast_smoother_feed(rtp->smoother, _f);
      
      while((f = ast_smoother_read(rtp->smoother)))
         ast_rtp_raw_write(rtp, f, codec);
      break;
   case AST_FORMAT_ADPCM:
   case AST_FORMAT_G726:
      if (!rtp->smoother) {
         rtp->smoother = ast_smoother_new(80);
      }
      if (!rtp->smoother) {
         ast_log(LOG_WARNING, "Unable to create smoother :(\n");
         return -1;
      }
      ast_smoother_feed(rtp->smoother, _f);
      
      while((f = ast_smoother_read(rtp->smoother)))
         ast_rtp_raw_write(rtp, f, codec);
      break;
   case AST_FORMAT_G729A:
      if (!rtp->smoother) {
         rtp->smoother = ast_smoother_new(20);
         if (rtp->smoother)
            ast_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_G729);
      }
      if (!rtp->smoother) {
         ast_log(LOG_WARNING, "Unable to create g729 smoother :(\n");
         return -1;
      }
      ast_smoother_feed(rtp->smoother, _f);
      
      while((f = ast_smoother_read(rtp->smoother)))
         ast_rtp_raw_write(rtp, f, codec);
      break;
   case AST_FORMAT_GSM:
      if (!rtp->smoother) {
         rtp->smoother = ast_smoother_new(33);
      }
      if (!rtp->smoother) {
         ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n");
         return -1;
      }
      ast_smoother_feed(rtp->smoother, _f);
      while((f = ast_smoother_read(rtp->smoother)))
         ast_rtp_raw_write(rtp, f, codec);
      break;
   case AST_FORMAT_ILBC:
      if (!rtp->smoother) {
         rtp->smoother = ast_smoother_new(50);
      }
      if (!rtp->smoother) {
         ast_log(LOG_WARNING, "Unable to create ILBC smoother :(\n");
         return -1;
      }
      ast_smoother_feed(rtp->smoother, _f);
      while((f = ast_smoother_read(rtp->smoother)))
         ast_rtp_raw_write(rtp, f, codec);
      break;
   default: 
      ast_log(LOG_WARNING, "Not sure about sending format %s packets\n", ast_getformatname(subclass));
      /* fall through to... */
   case AST_FORMAT_H261:
   case AST_FORMAT_H263:
   case AST_FORMAT_H263_PLUS:
   case AST_FORMAT_G723_1:
   case AST_FORMAT_LPC10:
   case AST_FORMAT_SPEEX:
           /* Don't buffer outgoing frames; send them one-per-packet: */
      if (_f->offset < hdrlen) {
         f = ast_frdup(_f);
      } else {
         f = _f;
      }
      ast_rtp_raw_write(rtp, f, codec);
      if (f != _f)
         ast_frfree(f);
   }
      
   return 0;
}