Sun Aug 6 15:15:39 2006

Asterisk developer's documentation

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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.

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)
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)
void	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)
rtpPayloadType	ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt)
ast_rtp *	ast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
	Initializate a RTP session.
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)
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)

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Detailed Description

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

RTP is defined in RFC 3550.

Definition in file rtp.h.

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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().

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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.

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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  )

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, rtpdebug, 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 != EAGAIN) ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno)); if (errno == EBADF) CRASH; 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 1518 of file rtp.c. References 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::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) 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]; } return AST_BRIDGE_FAILED; }

void ast_rtp_destroy (  struct ast_rtp *  rtp  )

Definition at line 1086 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 771 of file rtp.c. References rtpPayloadType::code, ast_rtp::current_RTP_PT, and rtpPayloadType::isAstFormat. 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; } } }

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

Definition at line 1044 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(). { them->sin_family = AF_INET; them->sin_port = rtp->them.sin_port; them->sin_addr = rtp->them.sin_addr; }

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

Definition at line 1051 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 1881 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 811 of file rtp.c. References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, 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 856 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 844 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 )

Definition at line 792 of file rtp.c. References rtpPayloadType::code, and rtpPayloadType::isAstFormat. 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 )

Initializate a RTP session. Parameters: sched  io  rtcpenable  callbackmode  Returns: A representation (structure) of an RTP session. Definition at line 1016 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 )

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 929 of file rtp.c. References ast_io_add(), AST_IO_IN, ast_log(), ast_rtcp_new(), ast_rtp_pt_default(), free, LOG_ERROR, malloc, rtp_socket(), rtpend, rtpread(), and rtpstart. 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_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 785 of file rtp.c. References ast_log(), and LOG_WARNING. 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 1484 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 1464 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 708 of file rtp.c. 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 724 of file rtp.c. References rtpPayloadType::code, and rtpPayloadType::isAstFormat. 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  )

Definition at line 426 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, dtmftimeout, 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(), rtpdebug, 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 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 != EAGAIN) ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno)); if (errno == EBADF) CRASH; 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); 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 (ext) { /* RTP Extension present */ hdrlen += 4; hdrlen += (ntohl(rtpheader[3]) & 0xffff) << 2; } 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:%d (type %d, seq %d, ts %d, 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 1825 of file rtp.c. References ast_config_destroy(), ast_config_load(), ast_false(), ast_log(), ast_variable_retrieve(), ast_verbose(), cfg, DEFAULT_DTMF_TIMEOUT, dtmftimeout, LOG_WARNING, option_verbose, rtpend, rtpstart, 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 1066 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 1202 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 1120 of file rtp.c. References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_tvadd(), ast_verbose(), LOG_ERROR, LOG_WARNING, and rtp_debug_test_addr(). 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 742 of file rtp.c. References rtpPayloadType::code. 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 1033 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 753 of file rtp.c. References ast_rtp::current_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 1024 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 1056 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 1318 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_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, LOG_DEBUG, LOG_WARNING, ast_frame::offset, option_debug, 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); } return 0; }

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