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Last updated: Sat Feb 3 05:01:52 2007

Asterisk developer's documentation :: Codename Pineapple

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sip3_sdprtp.c File Reference

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Detailed Description

Various SIP SDP and RTP handling functions Version 3 of chan_sip.

Author:

Mark Spencer <markster@digium.com>

Olle E. Johansson <oej@edvina.net> (all the chan_sip3 changes)

See Also:

• CREDITS

Definition in file sip3_sdprtp.c.

#include "asterisk.h"
#include <stdio.h>
#include <ctype.h>
#include <string.h>
#include <unistd.h>
#include <sys/socket.h>
#include <sys/ioctl.h>
#include <net/if.h>
#include <errno.h>
#include <stdlib.h>
#include <fcntl.h>
#include <netdb.h>
#include <signal.h>
#include <sys/signal.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <arpa/inet.h>
#include <netinet/ip.h>
#include <regex.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/compiler.h"
#include "sip3.h"
#include "sip3funcs.h"

Include dependency graph for sip3_sdprtp.c:

Go to the source code of this file.

Functions
static void	add_codec_to_sdp (const struct sip_dialog *p, int codec, int sample_rate, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, int debug, int *min_packet_size)
	Add codec offer to SDP offer/answer body in INVITE or 200 OK.
static void	add_noncodec_to_sdp (const struct sip_dialog *p, int format, int sample_rate, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, int debug)
	Add RFC 2833 DTMF offer to SDP.
int	add_sdp (struct sip_request *resp, struct sip_dialog *p)
	Add Session Description Protocol message.
int	find_sdp (struct sip_request *req)
	Determine whether a SIP message contains an SDP in its body.
char *	get_body (struct sip_request *req, char *name)
	Get the message body part identified by name=.
static char *	get_body_by_line (const char *line, const char *name, int nameLen)
	Reads one line of SIP message body.
static void	get_our_media_address (struct sip_dialog *p, int needvideo, struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *dest, struct sockaddr_in *vdest)
	Set all IP media addresses for this call.
static const char *	get_sdp (struct sip_request *req, const char *name)
	Get a line from an SDP message body.
static const char *	get_sdp_iterate (int *start, struct sip_request *req, const char *name)
	Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.
int	process_sdp (struct sip_dialog *p, struct sip_request *req)
	Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call.
void	register_rtp_and_udptl (void)
	Register RTP and UDPTL to the subsystems.
static int	sip_get_codec (struct ast_channel *chan)
	Return SIP UA's codec (part of the RTP interface).
enum ast_rtp_get_result	sip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite audio (part of RTP interface).
static struct ast_udptl *	sip_get_udptl_peer (struct ast_channel *chan)
	Get UDPTL peer address (part of UDPTL interface).
enum ast_rtp_get_result	sip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite video (part of RTP interface).
GNURK struct ast_frame *	sip_read (struct ast_channel *ast)
	Read SIP RTP from channel.
static struct ast_frame *	sip_rtp_read (struct ast_channel *ast, struct sip_dialog *p, int *faxdetect)
	Read RTP from network.
int	sip_set_rtp_peer (struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
	Set the RTP peer for this call.
static int	sip_set_udptl_peer (struct ast_channel *chan, struct ast_udptl *udptl)
	Determine UDPTL peer address for re-invite (part of UDPTL interface).
void	stop_media_flows (struct sip_dialog *dialog)
	Immediately stop RTP, VRTP and UDPTL as applicable.
void	unregister_rtp_and_udptl (void)
	UNRegister RTP and UDPTL to the subsystems.
Variables
static struct ast_rtp_protocol	sip_rtp
	Interface structure with callbacks used to connect to RTP module.
static struct ast_udptl_protocol	sip_udptl
	Interface structure with callbacks used to connect to UDPTL module.

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Function Documentation

static void add_codec_to_sdp (  const struct sip_dialog *  p, int  codec, int  sample_rate, char **  m_buf, size_t *  m_size, char **  a_buf, size_t *  a_size, int  debug, int *  min_packet_size )  [static]

Add codec offer to SDP offer/answer body in INVITE or 200 OK. Definition at line 957 of file sip3_sdprtp.c. References ast_build_string(), ast_codec_pref_getsize(), AST_FORMAT_G729A, AST_FORMAT_ILBC, ast_getformatname(), ast_rtp_codec_getpref(), ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), AST_RTP_OPT_G726_NONSTANDARD, ast_test_flag, ast_verbose(), sip_dialog::flags, fmt, sip_dialog::rtp, and SIP_G726_NONSTANDARD. { int rtp_code; struct ast_format_list fmt; if (debug) ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec)); if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1) return; if (p->rtp) { struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp); fmt = ast_codec_pref_getsize(pref, codec); } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */ return; ast_build_string(m_buf, m_size, " %d", rtp_code); ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, ast_rtp_lookup_mime_subtype(1, codec, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0), sample_rate); if (codec == AST_FORMAT_G729A) { /* Indicate that we don't support VAD (G.729 annex B) */ ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code); } else if (codec == AST_FORMAT_ILBC) { /* Add information about us using only 20/30 ms packetization */ ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms); } if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size)) *min_packet_size = fmt.cur_ms; /* Our first codec packetization processed cannot be zero */ if ((*min_packet_size)==0 && fmt.cur_ms) *min_packet_size = fmt.cur_ms; }

static void add_noncodec_to_sdp (  const struct sip_dialog *  p, int  format, int  sample_rate, char **  m_buf, size_t *  m_size, char **  a_buf, size_t *  a_size, int  debug )  [static]

Add RFC 2833 DTMF offer to SDP. Definition at line 936 of file sip3_sdprtp.c. References ast_build_string(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), ast_verbose(), and sip_dialog::rtp. { int rtp_code; if (debug) ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format, 0)); if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1) return; ast_build_string(m_buf, m_size, " %d", rtp_code); ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, ast_rtp_lookup_mime_subtype(0, format, 0), sample_rate); if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */ ast_build_string(a_buf, a_size, "a=fmtp:%d 0-16\r\n", rtp_code); }

int add_sdp (  struct sip_request *  resp, struct sip_dialog *  p )

Add Session Description Protocol message. Todo: XXX We need to select one codec, not many, since there's no transcoding Definition at line 1029 of file sip3_sdprtp.c. References add_codec_to_sdp(), ast_build_string(), ast_codec_pref_index(), AST_FORMAT_AUDIO_MASK, AST_FORMAT_VIDEO_MASK, ast_getformatname_multiple(), ast_inet_ntoa(), ast_log(), ast_test_flag, ast_verbose(), capability, FALSE, sip_dialog::flags, get_our_media_address(), sip_dialog::jointcapability, len, LOG_WARNING, sip_dialog::maxcallbitrate, option_debug, sip_dialog::ourip, sip_dialog::prefcodec, sip_dialog::prefs, sip_dialog::rtp, sip_dialog::sessionid, sip_dialog::sessionversion, sip_debug_test_pvt(), SIP_NOVIDEO, SIP_PAGE2_CALL_ONHOLD_INACTIVE, SIP_PAGE2_CALL_ONHOLD_ONEDIR, SIP_PAGE2_T38SUPPORT_RTP, sip_dialog::t38, t38properties::t38support, TRUE, and sip_dialog::vrtp. { int len = 0; int alreadysent = 0; struct sockaddr_in sin; struct sockaddr_in vsin; struct sockaddr_in dest; struct sockaddr_in vdest = { 0, }; /* SDP fields */ char *version = "v=0\r\n"; /* Protocol version */ char *subject = "s=session\r\n"; /* Subject of the session */ char owner[256]; /* Session owner/creator */ char connection[256]; /* Connection data */ char *stime = "t=0 0\r\n"; /* Time the session is active */ char bandwidth[256] = ""; /* Max bitrate */ char *hold; char m_audio[256]; /* Media declaration line for audio */ char m_video[256]; /* Media declaration line for video */ char a_audio[1024]; /* Attributes for audio */ char a_video[1024]; /* Attributes for video */ char *m_audio_next = m_audio; char *m_video_next = m_video; size_t m_audio_left = sizeof(m_audio); size_t m_video_left = sizeof(m_video); char *a_audio_next = a_audio; char *a_video_next = a_video; size_t a_audio_left = sizeof(a_audio); size_t a_video_left = sizeof(a_video); int x; int capability; int needvideo = FALSE; int debug = sip_debug_test_pvt(p); int min_audio_packet_size = 0; int min_video_packet_size = 0; m_video[0] = '\0'; /* Reset the video media string if it's not needed */ if (!p->rtp) { ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); return -1; } /* Set RTP Session ID and version */ if (!p->sessionid) { p->sessionid = getpid(); p->sessionversion = p->sessionid; } else p->sessionversion++; /* Ok, let's start working with codec selection here */ capability = p->jointcapability; if (option_debug > 1) { char codecbuf[BUFSIZ]; ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False"); ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec)); } #ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS if ((ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP))) { ast_build_string(&m_audio_next, &m_audio_left, " %d", 191); ast_build_string(&a_audio_next, &a_audio_left, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000); } #endif /* Check if we need video in this call */ if((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) { if (p->vrtp) { needvideo = TRUE; if (option_debug > 1) ast_log(LOG_DEBUG, "This call needs video offers! \n"); } else if (option_debug > 1) ast_log(LOG_DEBUG, "This call needs video offers, but there's no video support enabled ! \n"); } /* Get our media IP addresses for RTP */ get_our_media_address(p, needvideo, &sin, &vsin, &dest, &vdest); /* Ok, we need video. Let's add what we need for video and set codecs. Video is handled differently than audio since we can not transcode. */ if (needvideo) { ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port)); /* Build max bitrate string */ if (p->maxcallbitrate) snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate); if (debug) ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(vsin.sin_port)); /*! \todo XXX We need to select one codec, not many, since there's no transcoding */ } if (debug) ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(sin.sin_port)); /* Start building generic SDP headers */ /* We break with the "recommendation" and send our IP, in order that our peer doesn't have to ast_gethostbyname() us */ snprintf(owner, sizeof(owner), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr)); snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr)); ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port)); if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR)) hold = "a=recvonly\r\n"; else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE)) hold = "a=inactive\r\n"; else hold = "a=sendrecv\r\n"; /* Now, start adding audio codecs. These are added in this order: - First what was requested by the calling channel - Then preferences in order from sip.conf device config for this peer/user - Then other codecs in capabilities, including video */ /* Prefer the audio codec we were requested to use, first, no matter what Note that p->prefcodec can include video codecs, so mask them out */ if (capability & p->prefcodec) { add_codec_to_sdp(p, p->prefcodec & AST_FORMAT_AUDIO_MASK, 8000, &m_audio_next, &m_audio_left, &a_audio_next, &a_audio_left, debug, &min_audio_packet_size); alreadysent |= p->prefcodec & AST_FORMAT_AUDIO_MASK; } /* Start by sending our preferred audio codecs */ for (x = 0; x < 32; x++) { int pref_codec; if (!(pref_codec = ast_codec_pref_index(&p->prefs, x))) break; if (!(capability & pref_codec)) continue; if (alreadysent & pref_codec) continue; add_codec_to_sdp(p, pref_codec, 8000, &m_audio_next, &m_audio_left, &a_audio_next, &a_audio_left, debug, &min_audio_packet_size); alreadysent |= pref_codec; } /* Now send any other common audio and video codecs, and non-codec formats: */ for (x = 1; x <= (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) { if (!(capability & x)) /* Codec not requested */ continue; if (alreadysent & x) /* Already added to SDP */ continue; if (x <= AST_FORMAT_MAX_AUDIO) add_codec_to_sdp(p, x, 8000, &m_audio_next, &m_audio_left, &a_audio_next, &a_audio_left, debug, &min_audio_packet_size); else add_codec_to_sdp(p, x, 90000, &m_video_next, &m_video_left, &a_video_next, &a_video_left, debug, &min_video_packet_size); } /* Now add DTMF RFC2833 telephony-event as a codec */ for (x = 1; x <= AST_RTP_MAX; x <<= 1) { if (!(p->jointnoncodeccapability & x)) continue; add_noncodec_to_sdp(p, x, 8000, &m_audio_next, &m_audio_left, &a_audio_next, &a_audio_left, debug); } if (option_debug > 2) ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n"); if(!p->owner || !ast_internal_timing_enabled(p->owner)) ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n"); if (min_audio_packet_size) ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size); if (min_video_packet_size) ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size); if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0)) ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); ast_build_string(&m_audio_next, &m_audio_left, "\r\n"); if (needvideo) ast_build_string(&m_video_next, &m_video_left, "\r\n"); len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + strlen(m_audio) + strlen(a_audio) + strlen(hold); if (needvideo) /* only if video response is appropriate */ len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold); add_header(resp, "Content-Type", "application/sdp"); add_header_contentLength(resp, len); add_line(resp, version); add_line(resp, owner); add_line(resp, subject); add_line(resp, connection); if (needvideo) /* only if video response is appropriate */ add_line(resp, bandwidth); add_line(resp, stime); add_line(resp, m_audio); add_line(resp, a_audio); add_line(resp, hold); if (needvideo) { /* only if video response is appropriate */ add_line(resp, m_video); add_line(resp, a_video); add_line(resp, hold); /* Repeat hold for the video stream */ } /* Update lastrtprx when we send our SDP */ p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ if (option_debug > 2) { char buf[BUFSIZ]; ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability)); } return 0; }

int find_sdp (  struct sip_request *  req  )

Determine whether a SIP message contains an SDP in its body. Parameters: req  the SIP request to process Returns: TRUE if SDP found, FALSE if not found Also updates req->sdp_start and req->sdp_end to indicate where the SDP lives in the message body. Definition at line 880 of file sip3_sdprtp.c. References ast_strdupa, ast_strlen_zero(), FALSE, get_header(), sip_request::line, sip_request::lines, sip_request::sdp_end, sip_request::sdp_start, strcasestr(), and TRUE. { const char *content_type; const char *search; char *boundary; unsigned int x; content_type = get_header(req, "Content-Type"); /* if the body contains only SDP, this is easy */ if (!strcasecmp(content_type, "application/sdp")) { req->sdp_start = 0; req->sdp_end = req->lines; return TRUE; } /* if it's not multipart/mixed, there cannot be an SDP */ if (strncasecmp(content_type, "multipart/mixed", 15)) return FALSE; /* if there is no boundary marker, it's invalid */ if (!(search = strcasestr(content_type, ";boundary="))) return FALSE; search += 10; if (ast_strlen_zero(search)) return FALSE; /* make a duplicate of the string, with two extra characters at the beginning */ boundary = ast_strdupa(search - 2); boundary[0] = boundary[1] = '-'; /* search for the boundary marker, but stop when there are not enough lines left for it, the Content-Type header and at least one line of body */ for (x = 0; x < (req->lines - 2); x++) { if (!strncasecmp(req->line[x], boundary, strlen(boundary)) && !strcasecmp(req->line[x + 1], "Content-Type: application/sdp")) { x += 2; req->sdp_start = x; /* search for the end of the body part */ for ( ; x < req->lines; x++) { if (!strncasecmp(req->line[x], boundary, strlen(boundary))) break; } req->sdp_end = x; return TRUE; } } return FALSE; }

GNURK char * get_body (  struct sip_request *  req, char *  name )

Get the message body part identified by name=. Definition at line 166 of file sip3_sdprtp.c. References get_body_by_line(), len, sip_request::line, and sip_request::lines. { int x; int len = strlen(name); char *r; for (x = 0; x < req->lines; x++) { r = get_body_by_line(req->line[x], name, len); if (r[0] != '\0') return r; } return ""; }

static char* get_body_by_line (  const char *  line, const char *  name, int  nameLen )  [static]

Reads one line of SIP message body. Definition at line 132 of file sip3_sdprtp.c. { if (strncasecmp(line, name, nameLen) == 0 && line[nameLen] == '=') return ast_skip_blanks(line + nameLen + 1); return ""; }

static void get_our_media_address (  struct sip_dialog *  p, int  needvideo, struct sockaddr_in *  sin, struct sockaddr_in *  vsin, struct sockaddr_in *  dest, struct sockaddr_in *  vdest )  [static]

Set all IP media addresses for this call. Note: called from add_sdp() Definition at line 999 of file sip3_sdprtp.c. References ast_rtp_get_us(), sip_dialog::ourip, sip_dialog::redirip, sip_dialog::rtp, sip_dialog::vredirip, and sip_dialog::vrtp. { /* First, get our address */ ast_rtp_get_us(p->rtp, sin); if (p->vrtp) ast_rtp_get_us(p->vrtp, vsin); /* Now, try to figure out where we want them to send data */ /* Is this a re-invite to move the media out, then use the original offer from caller */ if (p->redirip.sin_addr.s_addr) { /* If we have a redirection IP, use it */ dest->sin_port = p->redirip.sin_port; dest->sin_addr = p->redirip.sin_addr; } else { dest->sin_addr = p->ourip; dest->sin_port = sin->sin_port; } if (needvideo) { /* Determine video destination */ if (p->vredirip.sin_addr.s_addr) { vdest->sin_addr = p->vredirip.sin_addr; vdest->sin_port = p->vredirip.sin_port; } else { vdest->sin_addr = p->ourip; vdest->sin_port = vsin->sin_port; } } }

static const char* get_sdp (  struct sip_request *  req, const char *  name )  [static]

Get a line from an SDP message body. Definition at line 158 of file sip3_sdprtp.c. References get_sdp_iterate(). { int dummy = 0; return get_sdp_iterate(&dummy, req, name); }

static const char* get_sdp_iterate (  int *  start, struct sip_request *  req, const char *  name )  [static]

Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number. Definition at line 144 of file sip3_sdprtp.c. References get_body_by_line(), len, and sip_request::line. { int len = strlen(name); while (*start < req->sdp_end) { const char *r = get_body_by_line(req->line[(*start)++], name, len); if (r[0] != '\0') return r; } return ""; }

int process_sdp (  struct sip_dialog *  p, struct sip_request *  req )

Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call. The SDP contains a session description one or multiple media descriptions that starts with m= < RTP Audio port number < RTP Video port number < media socket address < Video socket address < RTP Audio host IP < RTP video host IP Definition at line 189 of file sip3_sdprtp.c. References ast_clear_flag, ast_gethostbyname(), ast_log(), ast_rtp_alloc_size(), ast_rtp_new_init(), ast_rtp_pt_clear(), ast_rtp_set_m_type(), ast_set_flag, ast_strlen_zero(), ast_verbose(), debug(), FALSE, sip_dialog::flags, get_sdp(), get_sdp_iterate(), hp, sip_dialog::lastrtprx, sip_dialog::lastrtptx, len, LOG_ERROR, LOG_WARNING, sip_dialog::rtp, s, sip_request::sdp_start, sip_debug_test_pvt(), SIP_NOVIDEO, and TRUE. { const char *m; /* SDP media offer */ const char *c; const char *a; char host[258]; int len = -1; int portno = -1; /*!< RTP Audio port number */ int vportno = -1; /*!< RTP Video port number */ int udptlportno = -1; int peert38capability = 0; char s[256]; int old = 0; /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */ int peercapability = 0, peernoncodeccapability = 0; int vpeercapability = 0, vpeernoncodeccapability = 0; struct sockaddr_in sin; /*!< media socket address */ struct sockaddr_in vsin; /*!< Video socket address */ const char *codecs; struct hostent *hp; /*!< RTP Audio host IP */ struct hostent *vhp = NULL; /*!< RTP video host IP */ struct ast_hostent audiohp; struct ast_hostent videohp; int codec; int destiterator = 0; int iterator; int sendonly = 0; int numberofports; struct ast_channel *bridgepeer = NULL; struct ast_rtp *newaudiortp, *newvideortp; /* Buffers for codec handling */ int newjointcapability; /* Negotiated capability */ int newpeercapability; int newnoncodeccapability; int numberofmediastreams = 0; int debug = sip_debug_test_pvt(p); int found_rtpmap_codecs[32]; int last_rtpmap_codec=0; if (!p->rtp) { ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n"); return -1; } /* Initialize the temporary RTP structures we use to evaluate the offer from the peer */ newaudiortp = alloca(ast_rtp_alloc_size()); memset(newaudiortp, 0, ast_rtp_alloc_size()); ast_rtp_new_init(newaudiortp); ast_rtp_pt_clear(newaudiortp); newvideortp = alloca(ast_rtp_alloc_size()); memset(newvideortp, 0, ast_rtp_alloc_size()); ast_rtp_new_init(newvideortp); ast_rtp_pt_clear(newvideortp); /* Update our last rtprx when we receive an SDP, too */ p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ /* Try to find first media stream */ m = get_sdp(req, "m"); destiterator = req->sdp_start; c = get_sdp_iterate(&destiterator, req, "c"); if (ast_strlen_zero(m) || ast_strlen_zero(c)) { ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c); return -1; } /* Check for IPv4 address (not IPv6 yet) */ if (sscanf(c, "IN IP4 %256s", host) != 1) { ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c); return -1; } /* XXX This could block for a long time, and block the main thread! XXX */ hp = ast_gethostbyname(host, &audiohp); if (!hp) { ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c); return -1; } vhp = hp; /* Copy to video address as default too */ iterator = req->sdp_start; ast_set_flag(&p->flags[0], SIP_NOVIDEO); /* Find media streams in this SDP offer */ while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') { int x; int audio = FALSE; numberofports = 1; if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) || (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) { audio = TRUE; numberofmediastreams++; /* Found audio stream in this media definition */ portno = x; /* Scan through the RTP payload types specified in a "m=" line: */ for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { if (sscanf(codecs, "%d%n", &codec, &len) != 1) { ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); return -1; } if (debug) ast_verbose("Found RTP audio format %d\n", codec); ast_rtp_set_m_type(newaudiortp, codec); } } else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) || (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) { /* If it is not audio - is it video ? */ ast_clear_flag(&p->flags[0], SIP_NOVIDEO); numberofmediastreams++; vportno = x; /* Scan through the RTP payload types specified in a "m=" line: */ for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { if (sscanf(codecs, "%d%n", &codec, &len) != 1) { ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); return -1; } if (debug) ast_verbose("Found RTP video format %d\n", codec); ast_rtp_set_m_type(newvideortp, codec); } } else if (p->udptl && ((sscanf(m, "image %d udptl t38%n", &x, &len) == 1))) { if (debug) ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid); udptlportno = x; numberofmediastreams++; if (p->owner && p->lastinvite) { p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */ if (option_debug > 1) ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" ); } else { p->t38.state = T38_PEER_DIRECT; /* T38 Offered directly from peer in first invite */ if (option_debug > 1) ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } } else ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m); if (numberofports > 1) ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports); /* Check for Media-description-level-address for audio */ c = get_sdp_iterate(&destiterator, req, "c"); if (!ast_strlen_zero(c)) { if (sscanf(c, "IN IP4 %256s", host) != 1) { ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c); } else { /* XXX This could block for a long time, and block the main thread! XXX */ if (audio) { if ( !(hp = ast_gethostbyname(host, &audiohp))) ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c); } else if (!(vhp = ast_gethostbyname(host, &videohp))) ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c); } } } if (portno == -1 && vportno == -1 && udptlportno == -1) /* No acceptable offer found in SDP - we have no ports */ /* Do not change RTP or VRTP if this is a re-invite */ return -2; if (numberofmediastreams > 2) /* We have too many fax, audio and/or video media streams, fail this offer */ return -3; /* RTP addresses and ports for audio and video */ sin.sin_family = AF_INET; vsin.sin_family = AF_INET; memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr)); if (vhp) memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr)); /* Setup UDPTL port number */ if (p->udptl) { if (udptlportno > 0) { sin.sin_port = htons(udptlportno); ast_udptl_set_peer(p->udptl, &sin); if (debug) ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); } else { ast_udptl_stop(p->udptl); if (debug) ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n"); } } if (p->rtp) { if (portno > 0) { sin.sin_port = htons(portno); if (!sin.sin_port) ast_log(LOG_WARNING, "No SIN structure here, will crash\n"); ast_rtp_set_peer(p->rtp, &sin); if (debug) ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); } else { if (udptlportno > 0) { if (debug) ast_log(LOG_DEBUG, "Got T.38 re-invite without audio. Keeping RTP active during T.38 session. Call-Id %s\n", p->callid); } else ast_rtp_stop(p->rtp); if (debug) ast_verbose("Peer doesn't provide audio: Call-Id: %s\n", p->callid); } } /* Setup video port number */ if (vportno != -1) vsin.sin_port = htons(vportno); /* Next, scan through each "a=rtpmap:" line, noting each * specified RTP payload type (with corresponding MIME subtype): */ /* XXX This needs to be done per media stream, since it's media stream specific */ iterator = req->sdp_start; while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */ if (option_debug > 1) { int breakout = FALSE; /* If we're debugging, check for unsupported sdp options */ if (!strncasecmp(a, "rtcp:", (size_t) 5)) { if (debug) ast_verbose("Got unsupported a:rtcp in SDP offer \n"); breakout = TRUE; } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) { /* Format parameters: Not supported */ /* Note: This is used for codec parameters, like bitrate for G722 and video formats for H263 and H264 See RFC2327 for an example */ if (debug) ast_verbose("Got unsupported a:fmtp in SDP offer \n"); breakout = TRUE; } else if (!strncasecmp(a, "framerate:", (size_t) 10)) { /* Video stuff: Not supported */ if (debug) ast_verbose("Got unsupported a:framerate in SDP offer \n"); breakout = TRUE; } else if (!strncasecmp(a, "maxprate:", (size_t) 9)) { /* Video stuff: Not supported */ if (debug) ast_verbose("Got unsupported a:maxprate in SDP offer \n"); breakout = TRUE; } else if (!strncasecmp(a, "crypto:", (size_t) 7)) { /* SRTP stuff, not yet supported */ if (debug) ast_verbose("Got unsupported a:crypto in SDP offer \n"); breakout = TRUE; } if (breakout) /* We have a match, skip to next header */ continue; } if (!strcasecmp(a, "sendonly")) { sendonly = 1; continue; } else if (!strcasecmp(a, "inactive")) { sendonly = 2; continue; } else if (!strcasecmp(a, "sendrecv")) { sendonly = 0; continue; } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) { char *tmp = strrchr(a, ':'); long int framing = 0; if (tmp) { tmp++; framing = strtol(tmp, NULL, 10); if (framing == LONG_MIN || framing == LONG_MAX) { framing = 0; if (option_debug) ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a); } } if (framing && last_rtpmap_codec) { if (p->autoframing || global.autoframing) { struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp); int codec_n; int format = 0; for (codec_n = 0; codec_n < last_rtpmap_codec; codec_n++) { format = ast_rtp_codec_getformat(found_rtpmap_codecs[codec_n]); if (!format) /* non-codec or not found */ continue; if (option_debug) ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing); ast_codec_pref_setsize(pref, format, framing); } ast_rtp_codec_setpref(p->rtp, pref); } } memset(&found_rtpmap_codecs, 0, sizeof(found_rtpmap_codecs)); last_rtpmap_codec = 0; continue; } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) { /* We have a rtpmap to handle */ if (debug) ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec); found_rtpmap_codecs[last_rtpmap_codec] = codec; last_rtpmap_codec++; /* Note: should really look at the 'freq' and '#chans' params too */ ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0); if (p->vrtp) ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0); } } if (udptlportno != -1) { int found = 0, x; old = 0; /* Scan trough the a= lines for T38 attributes and set apropriate fileds */ iterator = req->sdp_start; while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { if ((sscanf(a, "T38FaxMaxBuffer:%d", &x) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "MaxBufferSize:%d\n",x); } else if ((sscanf(a, "T38MaxBitRate:%d", &x) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG,"T38MaxBitRate: %d\n",x); switch (x) { case 14400: peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; break; case 12000: peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; break; case 9600: peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; break; case 7200: peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; break; case 4800: peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400; break; case 2400: peert38capability |= T38FAX_RATE_2400; break; } } else if ((sscanf(a, "T38FaxVersion:%d", &x) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "FaxVersion: %d\n",x); if (x == 0) peert38capability |= T38FAX_VERSION_0; else if (x == 1) peert38capability |= T38FAX_VERSION_1; } else if ((sscanf(a, "T38FaxMaxDatagram:%d", &x) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "FaxMaxDatagram: %d\n",x); ast_udptl_set_far_max_datagram(p->udptl, x); ast_udptl_set_local_max_datagram(p->udptl, x); } else if ((sscanf(a, "T38FaxFillBitRemoval:%d", &x) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "FillBitRemoval: %d\n",x); if (x == 1) peert38capability |= T38FAX_FILL_BIT_REMOVAL; } else if ((sscanf(a, "T38FaxTranscodingMMR:%d", &x) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "Transcoding MMR: %d\n",x); if (x == 1) peert38capability |= T38FAX_TRANSCODING_MMR; } if ((sscanf(a, "T38FaxTranscodingJBIG:%d", &x) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "Transcoding JBIG: %d\n",x); if (x == 1) peert38capability |= T38FAX_TRANSCODING_JBIG; } else if ((sscanf(a, "T38FaxRateManagement:%s", s) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "RateMangement: %s\n", s); if (!strcasecmp(s, "localTCF")) peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF; else if (!strcasecmp(s, "transferredTCF")) peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF; } else if ((sscanf(a, "T38FaxUdpEC:%s", s) == 1)) { found = 1; if (option_debug > 2) ast_log(LOG_DEBUG, "UDP EC: %s\n", s); if (!strcasecmp(s, "t38UDPRedundancy")) { peert38capability |= T38FAX_UDP_EC_REDUNDANCY; ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY); } else if (!strcasecmp(s, "t38UDPFEC")) { peert38capability |= T38FAX_UDP_EC_FEC; ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC); } else { peert38capability |= T38FAX_UDP_EC_NONE; ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE); } } } if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */ p->t38.peercapability = peert38capability; p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */ peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400); p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */ } if (debug) ast_log(LOG_DEBUG, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.peercapability, p->t38.jointcapability); } else { p->t38.state = T38_DISABLED; if (option_debug > 2) ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } /* Now gather all of the codecs that we are asked for: */ ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability); ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability); newjointcapability = p->capability & (peercapability | vpeercapability); newpeercapability = (peercapability | vpeercapability); newnoncodeccapability = p->noncodeccapability & peernoncodeccapability; if (debug) { /* shame on whoever coded this.... */ char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ]; ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n", ast_getformatname_multiple(s1, BUFSIZ, p->capability), ast_getformatname_multiple(s2, BUFSIZ, newpeercapability), ast_getformatname_multiple(s3, BUFSIZ, vpeercapability), ast_getformatname_multiple(s4, BUFSIZ, newjointcapability)); ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n", ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0), ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0), ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0)); } if (!newjointcapability) { /* If T.38 was not negotiated either, totally bail out... */ if (!p->t38.jointcapability) { ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); /* Do NOT Change current setting */ return -1; } else { if (option_debug > 2) ast_log(LOG_DEBUG, "Have T.38 but no audio codecs, accepting offer anyway\n"); return 0; } } /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since they are acceptable */ p->jointcapability = newjointcapability; /* Our joint codec profile for this call */ p->peercapability = newpeercapability; /* The other sides capability in latest offer */ p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */ ast_rtp_pt_copy(p->rtp, newaudiortp); if (p->vrtp) ast_rtp_pt_copy(p->vrtp, newvideortp); if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) { ast_clear_flag(&p->flags[0], SIP_DTMF); if (newnoncodeccapability & AST_RTP_DTMF) { /* XXX Would it be reasonable to drop the DSP at this point? XXX */ ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833); } else { ast_set_flag(&p->flags[0], SIP_DTMF_INBAND); } } /* Setup audio port number */ if (p->rtp && sin.sin_port) { ast_rtp_set_peer(p->rtp, &sin); if (debug) ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); } /* Setup video port number */ if (p->vrtp && vsin.sin_port) { ast_rtp_set_peer(p->vrtp, &vsin); if (debug) ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port)); } /* Ok, we're going with this offer */ if (option_debug > 1) { char buf[BUFSIZ]; ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability)); } if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */ return 0; if (option_debug > 3) ast_log(LOG_DEBUG, "We have an owner, now see if we need to change this call\n"); if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) { if (debug) { char s1[BUFSIZ], s2[BUFSIZ]; ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability), ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats)); } p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability); ast_set_read_format(p->owner, p->owner->readformat); ast_set_write_format(p->owner, p->owner->writeformat); } /* Turn on/off music on hold if we are holding/unholding */ if ((bridgepeer = ast_bridged_channel(p->owner))) { if (sin.sin_addr.s_addr && !sendonly) { ast_queue_control(p->owner, AST_CONTROL_UNHOLD); /* Activate a re-invite */ ast_queue_frame(p->owner, &ast_null_frame); } else if (!sin.sin_addr.s_addr || sendonly) { ast_queue_control_data(p->owner, AST_CONTROL_HOLD, S_OR(p->mohsuggest, NULL), !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0); if (sendonly) ast_rtp_stop(p->rtp); /* RTCP needs to go ahead, even if we're on hold!!! */ /* Activate a re-invite */ ast_queue_frame(p->owner, &ast_null_frame); } } /* Manager Hold and Unhold events must be generated, if necessary */ if (sin.sin_addr.s_addr && !sendonly) { if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) { append_history(p, "Unhold", "%s", req->data); if (global.callevents) manager_event(EVENT_FLAG_CALL, "Unhold", "Channel: %s\r\n" "Uniqueid: %s\r\n", p->owner->name, p->owner->uniqueid); sip_peer_hold(p, 0); } ast_clear_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */ } else if (!sin.sin_addr.s_addr || sendonly ) { /* No address for RTP, we're on hold */ append_history(p, "Hold", "%s", req->data); if (global.callevents && !ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) { manager_event(EVENT_FLAG_CALL, "Hold", "Channel: %s\r\n" "Uniqueid: %s\r\n", p->owner->name, p->owner->uniqueid); } if (sendonly == 1) /* One directional hold (sendonly/recvonly) */ ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR); else if (sendonly == 2) /* Inactive stream */ ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE); sip_peer_hold(p, 1); } return 0; }

void register_rtp_and_udptl (  void   )

Register RTP and UDPTL to the subsystems. Definition at line 111 of file sip3_sdprtp.c. References ast_rtp_proto_register(), ast_udptl_proto_register(), sip_rtp, and sip_udptl. Referenced by load_module(). { /* Tell the RTP subdriver that we're here */ ast_rtp_proto_register(&sip_rtp); /* Tell the UDPTL subdriver that we're here */ ast_udptl_proto_register(&sip_udptl); }

static int sip_get_codec (  struct ast_channel *  chan  )  [static]

Return SIP UA's codec (part of the RTP interface). Definition at line 1264 of file sip3_sdprtp.c. References sip_dialog::peercapability, and ast_channel::tech_pvt. { struct sip_dialog *p = chan->tech_pvt; return p->peercapability; }

enum ast_rtp_get_result sip_get_rtp_peer (  struct ast_channel *  chan, struct ast_rtp **  rtp )

Returns null if we can't reinvite audio (part of RTP interface). Definition at line 825 of file sip3_sdprtp.c. References AST_JB_FORCED, AST_RTP_GET_FAILED, ast_rtp_getnat(), AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_test_flag, dialog_lock(), FALSE, sip_dialog::flags, global, sip_globals::jbconf, sip_dialog::rtp, SIP_CAN_REINVITE, SIP_CAN_REINVITE_NAT, ast_channel::tech_pvt, and TRUE. { struct sip_dialog *p = NULL; enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL; if (!(p = chan->tech_pvt) || !(p->rtp)) return AST_RTP_GET_FAILED; dialog_lock(p, TRUE); *rtp = p->rtp; if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) res = AST_RTP_TRY_PARTIAL; else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) res = AST_RTP_TRY_NATIVE; else if (ast_test_flag(&global.jbconf, AST_JB_FORCED)) res = AST_RTP_GET_FAILED; dialog_lock(p, FALSE); return res; }

static struct ast_udptl * sip_get_udptl_peer (  struct ast_channel *  chan  )  [static]

Get UDPTL peer address (part of UDPTL interface). Definition at line 1370 of file sip3_sdprtp.c. References dialog_lock(), FALSE, ast_channel::tech_pvt, TRUE, and sip_dialog::udptl. { struct sip_dialog *p; struct ast_udptl *udptl = NULL; p = chan->tech_pvt; if (!p || !(p->udptl)) return NULL; dialog_lock(p, TRUE); /* T38 reinvites is something very different from media re-invites to optimize the media stream. They should not rely on the CAN_REINVITE setting if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) */ udptl = p->udptl; dialog_lock(p, FALSE); return udptl; }

enum ast_rtp_get_result sip_get_vrtp_peer (  struct ast_channel *  chan, struct ast_rtp **  rtp )

Returns null if we can't reinvite video (part of RTP interface). Definition at line 850 of file sip3_sdprtp.c. References ast_mutex_lock(), ast_mutex_unlock(), AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_test_flag, sip_dialog::flags, sip_dialog::lock, sip_dialog::rtp, SIP_CAN_REINVITE, ast_channel::tech_pvt, and sip_dialog::vrtp. { struct sip_dialog *p = NULL; enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL; if (!(p = chan->tech_pvt) || !(p->vrtp)) return AST_RTP_GET_FAILED; ast_mutex_lock(&p->lock); *rtp = p->vrtp; if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) res = AST_RTP_TRY_NATIVE; /* The jitterbuffer is not implemented for video */ ast_mutex_unlock(&p->lock); return res; }

GNURK struct ast_frame * sip_read (  struct ast_channel *  ast  )

Read SIP RTP from channel. Definition at line 1333 of file sip3_sdprtp.c. References ast_bridged_channel(), ast_log(), ast_set_flag, ast_test_flag, dialog_lock(), FALSE, sip_dialog::flags, sip_dialog::lastrtprx, option_debug, sip_dialog::pendinginvite, SIP_GOTREFER, SIP_NEEDREINVITE, SIP_PAGE2_T38SUPPORT_UDPTL, SIP_PENDINGBYE, sip_rtp_read(), t38properties::state, sip_dialog::t38, T38_DISABLED, T38_LOCAL_REINVITE, t38properties::t38support, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), and TRUE. { struct ast_frame *fr; struct sip_dialog *p = ast->tech_pvt; int faxdetected = FALSE; dialog_lock(p, TRUE); fr = sip_rtp_read(ast, p, &faxdetected); p->lastrtprx = time(NULL); /* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */ /* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */ if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) { if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (!p->pendinginvite) { if (option_debug > 2) ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name); p->t38.state = T38_LOCAL_REINVITE; transmit_reinvite_with_sdp(p, TRUE); if (option_debug > 1) ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name); } } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { if (option_debug > 2) ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name); ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); } else if (option_debug > 3) { ast_log(LOG_DEBUG, "Fax deteced but discarded because another transaction is already in progress.\n"); /* \note Should we turn on a flag and send re-invite for fax at a later stage? */ } } dialog_lock(p, FALSE); return fr; }

static struct ast_frame* sip_rtp_read (  struct ast_channel *  ast, struct sip_dialog *  p, int *  faxdetect )  [static]

Read RTP from network. Definition at line 1271 of file sip3_sdprtp.c. References ast_dsp_process(), AST_FORMAT_AUDIO_MASK, AST_FORMAT_VIDEO_MASK, AST_FRAME_DTMF, AST_FRAME_VOICE, ast_log(), ast_null_frame, ast_rtcp_read(), ast_rtp_read(), ast_set_read_format(), ast_set_write_format(), ast_test_flag, ast_udptl_read(), ast_channel::fdno, sip_dialog::flags, ast_frame::frametype, ast_channel::nativeformats, option_debug, sip_dialog::owner, ast_channel::readformat, sip_dialog::rtp, SIP_DTMF, SIP_DTMF_INBAND, SIP_DTMF_RFC2833, SIP_PAGE2_T38SUPPORT_UDPTL, ast_frame::subclass, sip_dialog::t38, t38properties::t38support, sip_dialog::udptl, sip_dialog::vad, sip_dialog::vrtp, and ast_channel::writeformat. { /* Retrieve audio/etc from channel. Assumes p->lock is already held. */ struct ast_frame *f; if (!p->rtp) { /* We have no RTP allocated for this channel */ return &ast_null_frame; } switch(ast->fdno) { case 0: f = ast_rtp_read(p->rtp); /* RTP Audio */ break; case 1: f = ast_rtcp_read(p->rtp); /* RTCP Control Channel */ break; case 2: f = ast_rtp_read(p->vrtp); /* RTP Video */ break; case 3: f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */ break; case 5: f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */ break; default: f = &ast_null_frame; } /* Don't forward RFC2833 if we're not supposed to */ if (f && (f->frametype == AST_FRAME_DTMF) && (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833)) return &ast_null_frame; if (p->owner) { /* We already hold the channel lock */ if (f->frametype == AST_FRAME_VOICE) { if (f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) { if (option_debug) ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass); p->owner->nativeformats = (p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass; ast_set_read_format(p->owner, p->owner->readformat); ast_set_write_format(p->owner, p->owner->writeformat); } if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) { f = ast_dsp_process(p->owner, p->vad, f); if (f && f->frametype == AST_FRAME_DTMF) { if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') { if (option_debug) ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name); *faxdetect = 1; } else if (option_debug) { ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass); } } } } } return f; }

int sip_set_rtp_peer (  struct ast_channel *  chan, struct ast_rtp *  rtp, struct ast_rtp *  vrtp, int  codecs, int  nat_active )

Set the RTP peer for this call. Definition at line 760 of file sip3_sdprtp.c. References ast_channel::_state, append_history, ast_inet_ntoa(), ast_log(), ast_rtp_get_peer(), ast_set_flag, AST_STATE_UP, ast_test_flag, dialog_lock(), FALSE, sip_dialog::flags, sip_dialog::lastrtprx, sip_dialog::lastrtptx, option_debug, sip_dialog::ourip, sip_dialog::pendinginvite, sip_dialog::redircodecs, sip_dialog::redirip, sip_dialog::rtp, SIP_ALREADYGONE, SIP_CAN_REINVITE_NAT, SIP_GOTREFER, SIP_NEEDREINVITE, SIP_NO_HISTORY, SIP_PENDINGBYE, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), TRUE, sip_dialog::vredirip, and sip_dialog::vrtp. { struct sip_dialog *p; int changed = 0; p = chan->tech_pvt; if (!p) return -1; dialog_lock(p, TRUE); if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) { /* If we're destroyed, don't bother */ dialog_lock(p, FALSE); return 0; } /* if this peer cannot handle reinvites of the media stream to devices that are known to be behind a NAT, then stop the process now */ if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) { dialog_lock(p, FALSE); return 0; } if (rtp) { changed |= ast_rtp_get_peer(rtp, &p->redirip); } else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) { memset(&p->redirip, 0, sizeof(p->redirip)); changed = 1; } if (vrtp) { changed |= ast_rtp_get_peer(vrtp, &p->vredirip); } else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) { memset(&p->vredirip, 0, sizeof(p->vredirip)); changed = 1; } if (codecs && (p->redircodecs != codecs)) { p->redircodecs = codecs; changed = 1; } if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (chan->_state != AST_STATE_UP) { /* We are in early state */ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal."); if (option_debug) ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); } else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */ if (option_debug > 2) { ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); } transmit_reinvite_with_sdp(p, FALSE); } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { if (option_debug > 2) { ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); } /* We have a pending Invite. Send re-invite when we're done with the invite */ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); } } /* Reset lastrtprx timer */ p->lastrtprx = p->lastrtptx = time(NULL); dialog_lock(p, FALSE); return 0; }

static int sip_set_udptl_peer (  struct ast_channel *  chan, struct ast_udptl *  udptl )  [static]

Determine UDPTL peer address for re-invite (part of UDPTL interface). Definition at line 1389 of file sip3_sdprtp.c. References ast_inet_ntoa(), ast_log(), ast_mutex_lock(), ast_mutex_unlock(), ast_set_flag, ast_test_flag, ast_udptl_get_peer(), sip_dialog::flags, sip_dialog::lastrtprx, sip_dialog::lastrtptx, sip_dialog::lock, option_debug, sip_dialog::ourip, sip_dialog::pendinginvite, SIP_GOTREFER, SIP_NEEDREINVITE, SIP_PENDINGBYE, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), TRUE, sip_dialog::udptl, and sip_dialog::udptlredirip. { struct sip_dialog *p; p = chan->tech_pvt; if (!p) return -1; ast_mutex_lock(&p->lock); if (udptl) ast_udptl_get_peer(udptl, &p->udptlredirip); else memset(&p->udptlredirip, 0, sizeof(p->udptlredirip)); if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (!p->pendinginvite) { if (option_debug > 2) ast_log(LOG_DEBUG, "Sending T38 reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); transmit_reinvite_with_sdp(p, TRUE); } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { if (option_debug > 2) { ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); } ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); } } /* Reset lastrtprx timer */ p->lastrtprx = p->lastrtptx = time(NULL); ast_mutex_unlock(&p->lock); return 0; }

void stop_media_flows (  struct sip_dialog *  dialog  )

Immediately stop RTP, VRTP and UDPTL as applicable. Definition at line 1421 of file sip3_sdprtp.c. References ast_log(), ast_rtp_stop(), ast_udptl_stop(), option_debug, sip_dialog::rtp, sipdebug, sip_dialog::udptl, and sip_dialog::vrtp. { if (option_debug > 4 && sipdebug) ast_log(LOG_DEBUG, "Stopping media flow for %s\n", dialog->callid); /* Immediately stop RTP, VRTP and UDPTL as applicable */ if (dialog->rtp) ast_rtp_stop(dialog->rtp); if (dialog->vrtp) ast_rtp_stop(dialog->vrtp); if (dialog->udptl) ast_udptl_stop(dialog->udptl); }

void unregister_rtp_and_udptl (  void   )

UNRegister RTP and UDPTL to the subsystems. Definition at line 121 of file sip3_sdprtp.c. References ast_rtp_proto_unregister(), ast_udptl_proto_unregister(), sip_rtp, and sip_udptl. Referenced by unload_module(). { /* Tell the RTP subdriver that we're gone */ ast_rtp_proto_unregister(&sip_rtp); /* Tell the UDPTL subdriver that we're gone */ ast_udptl_proto_unregister(&sip_udptl); }

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Variable Documentation

struct ast_rtp_protocol sip_rtp [static]

Interface structure with callbacks used to connect to RTP module. Definition at line 95 of file sip3_sdprtp.c.

struct ast_udptl_protocol sip_udptl [static]

Initial value: { type: "SIP", get_udptl_info: sip_get_udptl_peer, set_udptl_peer: sip_set_udptl_peer, } Interface structure with callbacks used to connect to UDPTL module. Definition at line 104 of file sip3_sdprtp.c.

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