[svn-commits] oej: branch oej/codename-pineapple r45259 -
/team/oej/codename-pineapple/chan...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Mon Oct 16 12:28:04 MST 2006
Author: oej
Date: Mon Oct 16 14:28:03 2006
New Revision: 45259
URL: http://svn.digium.com/view/asterisk?rev=45259&view=rev
Log:
Missing file. Need to go through all those .h files that are included
Added:
team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c (with props)
Added: team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c?rev=45259&view=auto
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c (added)
+++ team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c Mon Oct 16 14:28:03 2006
@@ -1,0 +1,820 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2006, Digium, Inc.
+ *
+ * Mark Spencer <markster at digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Various SIP SDP and RTP handling functions
+ * Version 3 of chan_sip
+ *
+ * \author Mark Spencer <markster at digium.com>
+ * \author Olle E. Johansson <oej at edvina.net> (all the chan_sip3 changes)
+ *
+ * See Also:
+ * \arg \ref AstCREDITS
+ *
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#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/acl.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"
+
+/*! \brief Reads one line of SIP message body */
+static char *get_body_by_line(const char *line, const char *name, int nameLen)
+{
+ if (strncasecmp(line, name, nameLen) == 0 && line[nameLen] == '=')
+ return ast_skip_blanks(line + nameLen + 1);
+
+ return "";
+}
+
+/*! \brief Lookup 'name' in the SDP starting
+ * at the 'start' line. Returns the matching line, and 'start'
+ * is updated with the next line number.
+ */
+static const char *get_sdp_iterate(int *start, struct sip_request *req, const char *name)
+{
+ 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 "";
+}
+
+/*! \brief Get a line from an SDP message body */
+static const char *get_sdp(struct sip_request *req, const char *name)
+{
+ int dummy = 0;
+
+ return get_sdp_iterate(&dummy, req, name);
+}
+
+/*! \brief Get a specific line from the message body */
+static char *get_body(struct sip_request *req, char *name)
+{
+ 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 "";
+}
+
+/*! \brief Process SIP SDP offer, select formats and activate RTP channels
+ If offer is rejected, we will not change any properties of the call
+*/
+static int process_sdp(struct sip_pvt *p, struct sip_request *req)
+{
+ 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_pt_clear(newaudiortp);
+
+ newvideortp = alloca(ast_rtp_alloc_size());
+ memset(newvideortp, 0, ast_rtp_alloc_size());
+ 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));
+
+ if (p->rtp) {
+ if (portno > 0) {
+ sin.sin_port = htons(portno);
+ 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 {
+ ast_rtp_stop(p->rtp);
+ if (debug)
+ ast_verbose("Peer doesn't provide audio\n");
+ }
+ }
+ /* Setup video port number */
+ if (vportno != -1)
+ vsin.sin_port = htons(vportno);
+
+ /* 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");
+ }
+ }
+
+ /* 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;
+ } else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
+ if (debug)
+ ast_verbose("Got unsupported a:ptime 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 = global.dtmf_capability & 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, global.dtmf_capability, 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->noncodeccapability = 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)) {
+ 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;
+}
+
+
+/*! \brief Set the RTP peer for this call */
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
+{
+ struct sip_pvt *p;
+ int changed = 0;
+
+ p = chan->tech_pvt;
+ if (!p)
+ return -1;
+ ast_mutex_lock(&p->lock);
+ if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) {
+ /* If we're destroyed, don't bother */
+ ast_mutex_unlock(&p->lock);
+ 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)) {
+ ast_mutex_unlock(&p->lock);
+ 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 (global.recordhistory)
+ 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);
+ } 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);
+ ast_mutex_unlock(&p->lock);
+ return 0;
+}
+
+/*! \brief Returns null if we can't reinvite audio (part of RTP interface) */
+static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+{
+ struct sip_pvt *p = NULL;
+ enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
+
+ if (!(p = chan->tech_pvt))
+ return AST_RTP_GET_FAILED;
+
+ ast_mutex_lock(&p->lock);
+ if (!(p->rtp)) {
+ ast_mutex_unlock(&p->lock);
+ return AST_RTP_GET_FAILED;
+ }
+
+ *rtp = p->rtp;
+
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ res = AST_RTP_TRY_NATIVE;
+
+ ast_mutex_unlock(&p->lock);
+
+ return res;
+}
+
+/*! \brief Returns null if we can't reinvite video (part of RTP interface) */
+static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+{
+ struct sip_pvt *p = NULL;
+ enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
+
+ if (!(p = chan->tech_pvt))
+ return AST_RTP_GET_FAILED;
+
+ ast_mutex_lock(&p->lock);
+ if (!(p->vrtp)) {
+ ast_mutex_unlock(&p->lock);
+ return AST_RTP_GET_FAILED;
+ }
+
+ *rtp = p->vrtp;
+
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ res = AST_RTP_TRY_NATIVE;
+
+ ast_mutex_unlock(&p->lock);
+
+ return res;
+}
+
Propchange: team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
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svn:eol-style = native
Propchange: team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
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