[svn-commits] file: branch file/rtp_engine-mark2 r183439 - /team/file/rtp_engine-mark2/res/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Mar 19 15:47:13 CDT 2009
Author: file
Date: Thu Mar 19 15:47:10 2009
New Revision: 183439
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183439
Log:
Add in further work. This is equal to rtp.c in trunk except it is in a module.
Added:
team/file/rtp_engine-mark2/res/res_rtp_asterisk.c (with props)
Added: team/file/rtp_engine-mark2/res/res_rtp_asterisk.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/res/res_rtp_asterisk.c?view=auto&rev=183439
==============================================================================
--- team/file/rtp_engine-mark2/res/res_rtp_asterisk.c (added)
+++ team/file/rtp_engine-mark2/res/res_rtp_asterisk.c Thu Mar 19 15:47:10 2009
@@ -1,0 +1,2497 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, 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 Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
+ *
+ * \author Mark Spencer <markster at digium.com>
+ *
+ * \note RTP is defined in RFC 3550.
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 138083 $")
+
+#include <sys/time.h>
+#include <signal.h>
+#include <fcntl.h>
+#include <math.h>
+
+#include "asterisk/stun.h"
+#include "asterisk/pbx.h"
+#include "asterisk/frame.h"
+#include "asterisk/channel.h"
+#include "asterisk/acl.h"
+#include "asterisk/config.h"
+#include "asterisk/lock.h"
+#include "asterisk/utils.h"
+#include "asterisk/netsock.h"
+#include "asterisk/cli.h"
+#include "asterisk/manager.h"
+#include "asterisk/unaligned.h"
+#include "asterisk/module.h"
+#include "asterisk/rtp_engine.h"
+
+#define MAX_TIMESTAMP_SKEW 640
+
+#define RTP_SEQ_MOD (1<<16) /*!< A sequence number can't be more than 16 bits */
+#define RTCP_DEFAULT_INTERVALMS 5000 /*!< Default milli-seconds between RTCP reports we send */
+#define RTCP_MIN_INTERVALMS 500 /*!< Min milli-seconds between RTCP reports we send */
+#define RTCP_MAX_INTERVALMS 60000 /*!< Max milli-seconds between RTCP reports we send */
+
+#define RTCP_PT_FUR 192
+#define RTCP_PT_SR 200
+#define RTCP_PT_RR 201
+#define RTCP_PT_SDES 202
+#define RTCP_PT_BYE 203
+#define RTCP_PT_APP 204
+
+#define RTP_MTU 1200
+
+#define DEFAULT_DTMF_TIMEOUT 3000 /*!< samples */
+
+#define ZFONE_PROFILE_ID 0x505a
+
+static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
+
+static int rtpstart = 5000; /*!< First port for RTP sessions (set in rtp.conf) */
+static int rtpend = 31000; /*!< Last port for RTP sessions (set in rtp.conf) */
+static int rtpdebug; /*!< Are we debugging? */
+static int rtcpdebug; /*!< Are we debugging RTCP? */
+static int rtcpstats; /*!< Are we debugging RTCP? */
+static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
+static struct sockaddr_in rtpdebugaddr; /*!< Debug packets to/from this host */
+static struct sockaddr_in rtcpdebugaddr; /*!< Debug RTCP packets to/from this host */
+#ifdef SO_NO_CHECK
+static int nochecksums;
+#endif
+static int strictrtp;
+
+enum strict_rtp_state {
+ STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
+ STRICT_RTP_LEARN, /*! Accept next packet as source */
+ STRICT_RTP_CLOSED, /*! Drop all RTP packets not coming from source that was learned */
+};
+
+#define FLAG_3389_WARNING (1 << 0)
+#define FLAG_NAT_ACTIVE (3 << 1)
+#define FLAG_NAT_INACTIVE (0 << 1)
+#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
+#define FLAG_NEED_MARKER_BIT (1 << 3)
+#define FLAG_DTMF_COMPENSATE (1 << 4)
+
+/*! \brief RTP session description */
+struct ast_rtp {
+ int s;
+ struct ast_frame f;
+ unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
+ unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
+ unsigned int themssrc; /*!< Their SSRC */
+ unsigned int rxssrc;
+ unsigned int lastts;
+ unsigned int lastrxts;
+ unsigned int lastividtimestamp;
+ unsigned int lastovidtimestamp;
+ unsigned int lastitexttimestamp;
+ unsigned int lastotexttimestamp;
+ unsigned int lasteventseqn;
+ int lastrxseqno; /*!< Last received sequence number */
+ unsigned short seedrxseqno; /*!< What sequence number did they start with?*/
+ unsigned int seedrxts; /*!< What RTP timestamp did they start with? */
+ unsigned int rxcount; /*!< How many packets have we received? */
+ unsigned int rxoctetcount; /*!< How many octets have we received? should be rxcount *160*/
+ unsigned int txcount; /*!< How many packets have we sent? */
+ unsigned int txoctetcount; /*!< How many octets have we sent? (txcount*160)*/
+ unsigned int cycles; /*!< Shifted count of sequence number cycles */
+ double rxjitter; /*!< Interarrival jitter at the moment */
+ double rxtransit; /*!< Relative transit time for previous packet */
+ int lasttxformat;
+ int lastrxformat;
+
+ int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
+ int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
+ int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
+
+ /* DTMF Reception Variables */
+ char resp;
+ unsigned int lastevent;
+ int dtmfcount;
+ unsigned int dtmfsamples;
+ /* DTMF Transmission Variables */
+ unsigned int lastdigitts;
+ char sending_digit; /*!< boolean - are we sending digits */
+ char send_digit; /*!< digit we are sending */
+ int send_payload;
+ int send_duration;
+ unsigned int flags;
+ struct timeval rxcore;
+ struct timeval txcore;
+ double drxcore; /*!< The double representation of the first received packet */
+ struct timeval lastrx; /*!< timeval when we last received a packet */
+ struct timeval dtmfmute;
+ struct ast_smoother *smoother;
+ int *ioid;
+ unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
+ unsigned short rxseqno;
+ struct sched_context *sched;
+ struct io_context *io;
+ void *data;
+ struct ast_rtcp *rtcp;
+ struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
+
+ enum strict_rtp_state strict_rtp_state; /*!< Current state that strict RTP protection is in */
+ struct sockaddr_in strict_rtp_address; /*!< Remote address information for strict RTP purposes */
+
+ struct rtp_red *red;
+};
+
+/*!
+ * \brief Structure defining an RTCP session.
+ *
+ * The concept "RTCP session" is not defined in RFC 3550, but since
+ * this structure is analogous to ast_rtp, which tracks a RTP session,
+ * it is logical to think of this as a RTCP session.
+ *
+ * RTCP packet is defined on page 9 of RFC 3550.
+ *
+ */
+struct ast_rtcp {
+ int rtcp_info;
+ int s; /*!< Socket */
+ struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
+ struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
+ unsigned int soc; /*!< What they told us */
+ unsigned int spc; /*!< What they told us */
+ unsigned int themrxlsr; /*!< The middle 32 bits of the NTP timestamp in the last received SR*/
+ struct timeval rxlsr; /*!< Time when we got their last SR */
+ struct timeval txlsr; /*!< Time when we sent or last SR*/
+ unsigned int expected_prior; /*!< no. packets in previous interval */
+ unsigned int received_prior; /*!< no. packets received in previous interval */
+ int schedid; /*!< Schedid returned from ast_sched_add() to schedule RTCP-transmissions*/
+ unsigned int rr_count; /*!< number of RRs we've sent, not including report blocks in SR's */
+ unsigned int sr_count; /*!< number of SRs we've sent */
+ unsigned int lastsrtxcount; /*!< Transmit packet count when last SR sent */
+ double accumulated_transit; /*!< accumulated a-dlsr-lsr */
+ double rtt; /*!< Last reported rtt */
+ unsigned int reported_jitter; /*!< The contents of their last jitter entry in the RR */
+ unsigned int reported_lost; /*!< Reported lost packets in their RR */
+
+ double reported_maxjitter;
+ double reported_minjitter;
+ double reported_normdev_jitter;
+ double reported_stdev_jitter;
+ unsigned int reported_jitter_count;
+
+ double reported_maxlost;
+ double reported_minlost;
+ double reported_normdev_lost;
+ double reported_stdev_lost;
+
+ double rxlost;
+ double maxrxlost;
+ double minrxlost;
+ double normdev_rxlost;
+ double stdev_rxlost;
+ unsigned int rxlost_count;
+
+ double maxrxjitter;
+ double minrxjitter;
+ double normdev_rxjitter;
+ double stdev_rxjitter;
+ unsigned int rxjitter_count;
+ double maxrtt;
+ double minrtt;
+ double normdevrtt;
+ double stdevrtt;
+ unsigned int rtt_count;
+};
+
+struct rtp_red {
+ struct ast_frame t140; /*!< Primary data */
+ struct ast_frame t140red; /*!< Redundant t140*/
+ unsigned char pt[AST_RED_MAX_GENERATION]; /*!< Payload types for redundancy data */
+ unsigned char ts[AST_RED_MAX_GENERATION]; /*!< Time stamps */
+ unsigned char len[AST_RED_MAX_GENERATION]; /*!< length of each generation */
+ int num_gen; /*!< Number of generations */
+ int schedid; /*!< Timer id */
+ int ti; /*!< How long to buffer data before send */
+ unsigned char t140red_data[64000];
+ unsigned char buf_data[64000]; /*!< buffered primary data */
+ int hdrlen;
+ long int prev_ts;
+};
+
+/* Forward Declarations */
+static int ast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data);
+static int ast_rtp_destroy(struct ast_rtp_instance *instance);
+static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
+static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
+static void ast_rtp_new_source(struct ast_rtp_instance *instance);
+static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
+static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
+static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
+static int ast_rtp_fd(struct ast_rtp_instance *instance, int rtcp);
+static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct sockaddr_in *sin);
+static int rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations);
+static int rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame);
+static int ast_rtp_local_bridge(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1);
+static int ast_rtp_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat);
+static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
+
+/* RTP Engine Declaration */
+static struct ast_rtp_engine asterisk_rtp_engine = {
+ .name = "asterisk",
+ .new = ast_rtp_new,
+ .destroy = ast_rtp_destroy,
+ .dtmf_begin = ast_rtp_dtmf_begin,
+ .dtmf_end = ast_rtp_dtmf_end,
+ .new_source = ast_rtp_new_source,
+ .write = ast_rtp_write,
+ .read = ast_rtp_read,
+ .prop_set = ast_rtp_prop_set,
+ .fd = ast_rtp_fd,
+ .remote_address_set = ast_rtp_remote_address_set,
+ .red_init = rtp_red_init,
+ .red_buffer = rtp_red_buffer,
+ .local_bridge = ast_rtp_local_bridge,
+ .get_stat = ast_rtp_get_stat,
+ .dtmf_compatible = ast_rtp_dtmf_compatible,
+};
+
+static inline int rtp_debug_test_addr(struct sockaddr_in *addr)
+{
+ if (!rtpdebug) {
+ return 0;
+ }
+
+ if (rtpdebugaddr.sin_addr.s_addr) {
+ if (((ntohs(rtpdebugaddr.sin_port) != 0)
+ && (rtpdebugaddr.sin_port != addr->sin_port))
+ || (rtpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
+ return 0;
+ }
+
+ return 1;
+}
+
+static inline int rtcp_debug_test_addr(struct sockaddr_in *addr)
+{
+ if (!rtcpdebug) {
+ return 0;
+ }
+
+ if (rtcpdebugaddr.sin_addr.s_addr) {
+ if (((ntohs(rtcpdebugaddr.sin_port) != 0)
+ && (rtcpdebugaddr.sin_port != addr->sin_port))
+ || (rtcpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
+ return 0;
+ }
+
+ return 1;
+}
+
+static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
+{
+ unsigned int interval;
+ /*! \todo XXX Do a more reasonable calculation on this one
+ * Look in RFC 3550 Section A.7 for an example*/
+ interval = rtcpinterval;
+ return interval;
+}
+
+/*! \brief Calculate normal deviation */
+static double normdev_compute(double normdev, double sample, unsigned int sample_count)
+{
+ normdev = normdev * sample_count + sample;
+ sample_count++;
+
+ return normdev / sample_count;
+}
+
+static double stddev_compute(double stddev, double sample, double normdev, double normdev_curent, unsigned int sample_count)
+{
+/*
+ for the formula check http://www.cs.umd.edu/~austinjp/constSD.pdf
+ return sqrt( (sample_count*pow(stddev,2) + sample_count*pow((sample-normdev)/(sample_count+1),2) + pow(sample-normdev_curent,2)) / (sample_count+1));
+ we can compute the sigma^2 and that way we would have to do the sqrt only 1 time at the end and would save another pow 2 compute
+ optimized formula
+*/
+#define SQUARE(x) ((x) * (x))
+
+ stddev = sample_count * stddev;
+ sample_count++;
+
+ return stddev +
+ ( sample_count * SQUARE( (sample - normdev) / sample_count ) ) +
+ ( SQUARE(sample - normdev_curent) / sample_count );
+
+#undef SQUARE
+}
+
+static int create_new_socket(const char *type)
+{
+ int sock = socket(AF_INET, SOCK_DGRAM, 0);
+
+ if (sock < 0) {
+ if (!type) {
+ type = "RTP/RTCP";
+ }
+ ast_log(LOG_WARNING, "Unable to allocate %s socket: %s\n", type, strerror(errno));
+ } else {
+ long flags = fcntl(sock, F_GETFL);
+ fcntl(sock, F_SETFL, flags | O_NONBLOCK);
+#ifdef SO_NO_CHECK
+ if (nochecksums) {
+ setsockopt(sock, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
+ }
+#endif
+ }
+
+ return sock;
+}
+
+static int ast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data)
+{
+ struct ast_rtp *rtp = NULL;
+ int x, startplace;
+
+ /* Create a new RTP structure to hold all of our data */
+ if (!(rtp = ast_calloc(1, sizeof(*rtp)))) {
+ return -1;
+ }
+
+ /* Set default parameters on the newly created RTP structure */
+ rtp->ssrc = ast_random();
+ rtp->seqno = ast_random() & 0xffff;
+ rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
+
+ /* Create a new socket for us to listen on and use */
+ if ((rtp->s = create_new_socket("RTP")) < 0) {
+ ast_debug(1, "Failed to create a new socket for RTP instance '%p'\n", instance);
+ ast_free(rtp);
+ return -1;
+ }
+
+ /* Now actually find a free RTP port to use */
+ x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart;
+ x = x & ~1;
+ startplace = x;
+
+ for (;;) {
+ instance->local_address.sin_port = htons(x);
+ /* Try to bind, this will tell us whether the port is available or not */
+ if (!bind(rtp->s, (struct sockaddr*)&instance->local_address, sizeof(instance->local_address))) {
+ ast_debug(1, "Allocated port %d for RTP instance '%p'\n", x, instance);
+ break;
+ }
+
+ x += 2;
+ if (x > rtpend) {
+ x = (rtpstart + 1) & ~1;
+ }
+
+ /* See if we ran out of ports or if the bind actually failed because of something other than the address being in use */
+ if (x == startplace || errno != EADDRINUSE) {
+ ast_log(LOG_ERROR, "Oh dear... we couldn't allocate a port for RTP instance '%p'\n", instance);
+ return -1;
+ }
+ }
+
+ /* Record any information we may need */
+ rtp->sched = sched;
+
+ /* Associate the RTP structure with the RTP instance and be done */
+ instance->data = rtp;
+
+ return 0;
+}
+
+static int ast_rtp_destroy(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = instance->data;
+
+ /* Destroy the smoother that was smoothing out audio if present */
+ if (rtp->smoother) {
+ ast_smoother_free(rtp->smoother);
+ }
+
+ /* Close our own socket so we no longer get packets */
+ if (rtp->s > -1) {
+ close(rtp->s);
+ }
+
+ /* Destroy RTCP if it was being used */
+ if (rtp->rtcp) {
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ close(rtp->rtcp->s);
+ ast_free(rtp->rtcp);
+ }
+
+ /* Destroy RED if it was being used */
+ if (rtp->red) {
+ AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
+ ast_free(rtp->red);
+ }
+
+ /* Finally destroy ourselves */
+ ast_free(rtp);
+
+ return 0;
+}
+
+static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
+{
+ struct ast_rtp *rtp = instance->data;
+ int hdrlen = 12, res = 0, i = 0, payload = 101;
+ char data[256];
+ unsigned int *rtpheader = (unsigned int*)data;
+
+ /* If we have no remote address information bail out now */
+ if (!instance->remote_address.sin_addr.s_addr || !instance->remote_address.sin_port) {
+ return -1;
+ }
+
+ /* Convert given digit into what we want to transmit */
+ 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;
+ }
+
+ /* Grab the payload that they expect the RFC2833 packet to be received in */
+ payload = ast_rtp_codecs_payload_code(&instance->codecs, 0, AST_RTP_DTMF);
+
+ rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
+ rtp->send_duration = 160;
+ rtp->lastdigitts = rtp->lastts + rtp->send_duration;
+
+ /* Create the actual packet that we will be sending */
+ rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
+ rtpheader[1] = htonl(rtp->lastdigitts);
+ rtpheader[2] = htonl(rtp->ssrc);
+
+ /* Actually send the packet */
+ for (i = 0; i < 2; i++) {
+ rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
+ res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &instance->remote_address, sizeof(instance->remote_address));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
+ ast_inet_ntoa(instance->remote_address.sin_addr), ntohs(instance->remote_address.sin_port), strerror(errno));
+ }
+ if (rtp_debug_test_addr(&instance->remote_address)) {
+ ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(instance->remote_address.sin_addr),
+ ntohs(instance->remote_address.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ }
+ rtp->seqno++;
+ rtp->send_duration += 160;
+ rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
+ }
+
+ /* Record that we are in the process of sending a digit and information needed to continue doing so */
+ rtp->sending_digit = 1;
+ rtp->send_digit = digit;
+ rtp->send_payload = payload;
+
+ return 0;
+}
+
+static int ast_rtp_dtmf_continuation(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = instance->data;
+ int hdrlen = 12, res = 0;
+ char data[256];
+ unsigned int *rtpheader = (unsigned int*)data;
+
+ /* Make sure we know where the other side is so we can send them the packet */
+ if (!instance->remote_address.sin_addr.s_addr || !instance->remote_address.sin_port) {
+ return -1;
+ }
+
+ /* Actually create the packet we will be sending */
+ rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
+ rtpheader[1] = htonl(rtp->lastdigitts);
+ rtpheader[2] = htonl(rtp->ssrc);
+ rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
+ rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
+
+ /* Boom, send it on out */
+ res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &instance->remote_address, sizeof(instance->remote_address));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
+ ast_inet_ntoa(instance->remote_address.sin_addr),
+ ntohs(instance->remote_address.sin_port), strerror(errno));
+ }
+
+ if (rtp_debug_test_addr(&instance->remote_address)) {
+ ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(instance->remote_address.sin_addr),
+ ntohs(instance->remote_address.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ }
+
+ /* And now we increment some values for the next time we swing by */
+ rtp->seqno++;
+ rtp->send_duration += 160;
+
+ return 0;
+}
+
+static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
+{
+ struct ast_rtp *rtp = instance->data;
+ int hdrlen = 12, res = 0, i = 0;
+ char data[256];
+ unsigned int *rtpheader = (unsigned int*)data;
+
+ /* Make sure we know where the remote side is so we can send them the packet we construct */
+ if (!instance->remote_address.sin_addr.s_addr || !instance->remote_address.sin_port) {
+ return -1;
+ }
+
+ /* Convert the given digit to the one we are going to send */
+ 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;
+ }
+
+ rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
+
+ /* Construct the packet we are going to send */
+ rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
+ rtpheader[1] = htonl(rtp->lastdigitts);
+ rtpheader[2] = htonl(rtp->ssrc);
+ rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
+ rtpheader[3] |= htonl((1 << 23));
+ rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
+
+ /* Send it 3 times, that's the magical number */
+ for (i = 0; i < 3; i++) {
+ res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &instance->remote_address, sizeof(instance->remote_address));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
+ ast_inet_ntoa(instance->remote_address.sin_addr),
+ ntohs(instance->remote_address.sin_port), strerror(errno));
+ }
+ if (rtp_debug_test_addr(&instance->remote_address)) {
+ ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(instance->remote_address.sin_addr),
+ ntohs(instance->remote_address.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ }
+ }
+
+ /* Oh and we can't forget to turn off the stuff that says we are sending DTMF */
+ rtp->lastts += rtp->send_duration;
+ rtp->sending_digit = 0;
+ rtp->send_digit = 0;
+
+ return 0;
+}
+
+static void ast_rtp_new_source(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = instance->data;
+
+ /* We simply set this bit so that the next packet sent will have the marker bit turned on */
+ ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+
+ return;
+}
+
+static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
+{
+ struct timeval t;
+ long ms;
+
+ if (ast_tvzero(rtp->txcore)) {
+ rtp->txcore = ast_tvnow();
+ rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
+ }
+
+ t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
+ if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
+ ms = 0;
+ }
+ rtp->txcore = t;
+
+ return (unsigned int) ms;
+}
+
+static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
+{
+ unsigned int sec, usec, frac;
+ sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
+ usec = tv.tv_usec;
+ frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
+ *msw = sec;
+ *lsw = frac;
+}
+
+/*! \brief Send RTCP recipient's report */
+static int ast_rtcp_write_rr(const void *data)
+{
+ struct ast_rtp *rtp = (struct ast_rtp *)data;
+ int res;
+ int len = 32;
+ unsigned int lost;
+ unsigned int extended;
+ unsigned int expected;
+ unsigned int expected_interval;
+ unsigned int received_interval;
+ int lost_interval;
+ struct timeval now;
+ unsigned int *rtcpheader;
+ char bdata[1024];
+ struct timeval dlsr;
+ int fraction;
+
+ double rxlost_current;
+
+ if (!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
+ return 0;
+
+ if (!rtp->rtcp->them.sin_addr.s_addr) {
+ ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted\n");
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ extended = rtp->cycles + rtp->lastrxseqno;
+ expected = extended - rtp->seedrxseqno + 1;
+ lost = expected - rtp->rxcount;
+ expected_interval = expected - rtp->rtcp->expected_prior;
+ rtp->rtcp->expected_prior = expected;
+ received_interval = rtp->rxcount - rtp->rtcp->received_prior;
+ rtp->rtcp->received_prior = rtp->rxcount;
+ lost_interval = expected_interval - received_interval;
+
+ if (lost_interval <= 0)
+ rtp->rtcp->rxlost = 0;
+ else rtp->rtcp->rxlost = rtp->rtcp->rxlost;
+ if (rtp->rtcp->rxlost_count == 0)
+ rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
+ if (lost_interval < rtp->rtcp->minrxlost)
+ rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
+ if (lost_interval > rtp->rtcp->maxrxlost)
+ rtp->rtcp->maxrxlost = rtp->rtcp->rxlost;
+
+ rxlost_current = normdev_compute(rtp->rtcp->normdev_rxlost, rtp->rtcp->rxlost, rtp->rtcp->rxlost_count);
+ rtp->rtcp->stdev_rxlost = stddev_compute(rtp->rtcp->stdev_rxlost, rtp->rtcp->rxlost, rtp->rtcp->normdev_rxlost, rxlost_current, rtp->rtcp->rxlost_count);
+ rtp->rtcp->normdev_rxlost = rxlost_current;
+ rtp->rtcp->rxlost_count++;
+
+ if (expected_interval == 0 || lost_interval <= 0)
+ fraction = 0;
+ else
+ fraction = (lost_interval << 8) / expected_interval;
+ gettimeofday(&now, NULL);
+ timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
+ rtcpheader = (unsigned int *)bdata;
+ rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_RR << 16) | ((len/4)-1));
+ rtcpheader[1] = htonl(rtp->ssrc);
+ rtcpheader[2] = htonl(rtp->themssrc);
+ rtcpheader[3] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
+ rtcpheader[4] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
+ rtcpheader[5] = htonl((unsigned int)(rtp->rxjitter * 65536.));
+ rtcpheader[6] = htonl(rtp->rtcp->themrxlsr);
+ rtcpheader[7] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
+
+ /*! \note Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos
+ it can change mid call, and SDES can't) */
+ rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
+ rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
+ len += 12;
+
+ res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
+
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted: %s\n",strerror(errno));
+ /* Remove the scheduler */
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ rtp->rtcp->rr_count++;
+ if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
+ ast_verbose("\n* Sending RTCP RR to %s:%d\n"
+ " Our SSRC: %u\nTheir SSRC: %u\niFraction lost: %d\nCumulative loss: %u\n"
+ " IA jitter: %.4f\n"
+ " Their last SR: %u\n"
+ " DLSR: %4.4f (sec)\n\n",
+ ast_inet_ntoa(rtp->rtcp->them.sin_addr),
+ ntohs(rtp->rtcp->them.sin_port),
+ rtp->ssrc, rtp->themssrc, fraction, lost,
+ rtp->rxjitter,
+ rtp->rtcp->themrxlsr,
+ (double)(ntohl(rtcpheader[7])/65536.0));
+ }
+
+ return res;
+}
+
+/*! \brief Send RTCP sender's report */
+static int ast_rtcp_write_sr(const void *data)
+{
+ struct ast_rtp *rtp = (struct ast_rtp *)data;
+ int res;
+ int len = 0;
+ struct timeval now;
+ unsigned int now_lsw;
+ unsigned int now_msw;
+ unsigned int *rtcpheader;
+ unsigned int lost;
+ unsigned int extended;
+ unsigned int expected;
+ unsigned int expected_interval;
+ unsigned int received_interval;
+ int lost_interval;
+ int fraction;
+ struct timeval dlsr;
+ char bdata[512];
+
+ /* Commented condition is always not NULL if rtp->rtcp is not NULL */
+ if (!rtp || !rtp->rtcp/* || (&rtp->rtcp->them.sin_addr == 0)*/)
+ return 0;
+
+ if (!rtp->rtcp->them.sin_addr.s_addr) { /* This'll stop rtcp for this rtp session */
+ ast_verbose("RTCP SR transmission error, rtcp halted\n");
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ gettimeofday(&now, NULL);
+ timeval2ntp(now, &now_msw, &now_lsw); /* fill thses ones in from utils.c*/
+ rtcpheader = (unsigned int *)bdata;
+ rtcpheader[1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[2] = htonl(now_msw); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970*/
+ rtcpheader[3] = htonl(now_lsw); /* now, LSW */
+ rtcpheader[4] = htonl(rtp->lastts); /* FIXME shouldn't be that, it should be now */
+ rtcpheader[5] = htonl(rtp->txcount); /* No. packets sent */
+ rtcpheader[6] = htonl(rtp->txoctetcount); /* No. bytes sent */
+ len += 28;
+
+ extended = rtp->cycles + rtp->lastrxseqno;
+ expected = extended - rtp->seedrxseqno + 1;
+ if (rtp->rxcount > expected)
+ expected += rtp->rxcount - expected;
+ lost = expected - rtp->rxcount;
+ expected_interval = expected - rtp->rtcp->expected_prior;
+ rtp->rtcp->expected_prior = expected;
+ received_interval = rtp->rxcount - rtp->rtcp->received_prior;
+ rtp->rtcp->received_prior = rtp->rxcount;
+ lost_interval = expected_interval - received_interval;
+ if (expected_interval == 0 || lost_interval <= 0)
+ fraction = 0;
+ else
+ fraction = (lost_interval << 8) / expected_interval;
+ timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
+ rtcpheader[7] = htonl(rtp->themssrc);
+ rtcpheader[8] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
+ rtcpheader[9] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
+ rtcpheader[10] = htonl((unsigned int)(rtp->rxjitter * 65536.));
+ rtcpheader[11] = htonl(rtp->rtcp->themrxlsr);
+ rtcpheader[12] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
+ len += 24;
+
+ rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SR << 16) | ((len/4)-1));
+
+ /* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
+ /* it can change mid call, and SDES can't) */
+ rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
+ rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
+ len += 12;
+
+ res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTCP SR transmission error to %s:%d, rtcp halted %s\n",ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port), strerror(errno));
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ /* FIXME Don't need to get a new one */
+ gettimeofday(&rtp->rtcp->txlsr, NULL);
+ rtp->rtcp->sr_count++;
+
+ rtp->rtcp->lastsrtxcount = rtp->txcount;
+
+ if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
+ ast_verbose("* Sent RTCP SR to %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ ast_verbose(" Our SSRC: %u\n", rtp->ssrc);
+ ast_verbose(" Sent(NTP): %u.%010u\n", (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096);
+ ast_verbose(" Sent(RTP): %u\n", rtp->lastts);
+ ast_verbose(" Sent packets: %u\n", rtp->txcount);
+ ast_verbose(" Sent octets: %u\n", rtp->txoctetcount);
+ ast_verbose(" Report block:\n");
+ ast_verbose(" Fraction lost: %u\n", fraction);
+ ast_verbose(" Cumulative loss: %u\n", lost);
+ ast_verbose(" IA jitter: %.4f\n", rtp->rxjitter);
+ ast_verbose(" Their last SR: %u\n", rtp->rtcp->themrxlsr);
+ ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(ntohl(rtcpheader[12])/65536.0));
+ }
+ manager_event(EVENT_FLAG_REPORTING, "RTCPSent", "To %s:%d\r\n"
+ "OurSSRC: %u\r\n"
+ "SentNTP: %u.%010u\r\n"
+ "SentRTP: %u\r\n"
+ "SentPackets: %u\r\n"
+ "SentOctets: %u\r\n"
+ "ReportBlock:\r\n"
+ "FractionLost: %u\r\n"
+ "CumulativeLoss: %u\r\n"
+ "IAJitter: %.4f\r\n"
+ "TheirLastSR: %u\r\n"
+ "DLSR: %4.4f (sec)\r\n",
+ ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port),
+ rtp->ssrc,
+ (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096,
+ rtp->lastts,
+ rtp->txcount,
+ rtp->txoctetcount,
+ fraction,
+ lost,
+ rtp->rxjitter,
+ rtp->rtcp->themrxlsr,
+ (double)(ntohl(rtcpheader[12])/65536.0));
+ return res;
+}
+
+/*! \brief Write and RTCP packet to the far end
+ * \note Decide if we are going to send an SR (with Reception Block) or RR
+ * RR is sent if we have not sent any rtp packets in the previous interval */
+static int ast_rtcp_write(const void *data)
+{
+ struct ast_rtp *rtp = (struct ast_rtp *)data;
+ int res;
+
+ if (!rtp || !rtp->rtcp)
+ return 0;
+
+ if (rtp->txcount > rtp->rtcp->lastsrtxcount)
+ res = ast_rtcp_write_sr(data);
+ else
+ res = ast_rtcp_write_rr(data);
+
+ return res;
+}
+
+static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
+{
+ struct ast_rtp *rtp = instance->data;
+ int pred, mark = 0;
+ unsigned int ms = calc_txstamp(rtp, &frame->delivery);
+
+ if (rtp->sending_digit) {
+ return 0;
+ }
+
+ if (frame->frametype == AST_FRAME_VOICE) {
+ pred = rtp->lastts + frame->samples;
+
+ /* Re-calculate last TS */
+ rtp->lastts = rtp->lastts + ms * 8;
+ if (ast_tvzero(frame->delivery)) {
+ /* If this isn't an absolute delivery time, Check if it is close to our prediction,
+ and if so, go with our prediction */
+ if (abs(rtp->lastts - pred) < MAX_TIMESTAMP_SKEW) {
+ rtp->lastts = pred;
+ } else {
+ ast_debug(3, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
+ mark = 1;
+ }
+ }
+ } else if (frame->frametype == AST_FRAME_VIDEO) {
+ mark = frame->subclass & 0x1;
+ pred = rtp->lastovidtimestamp + frame->samples;
+ /* Re-calculate last TS */
+ rtp->lastts = rtp->lastts + ms * 90;
+ /* If it's close to our prediction, go for it */
+ if (ast_tvzero(frame->delivery)) {
+ if (abs(rtp->lastts - pred) < 7200) {
+ rtp->lastts = pred;
+ rtp->lastovidtimestamp += frame->samples;
+ } else {
+ ast_debug(3, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, frame->samples);
+ rtp->lastovidtimestamp = rtp->lastts;
+ }
+ }
+ } else {
+ pred = rtp->lastotexttimestamp + frame->samples;
+ /* Re-calculate last TS */
+ rtp->lastts = rtp->lastts + ms * 90;
+ /* If it's close to our prediction, go for it */
+ if (ast_tvzero(frame->delivery)) {
+ if (abs(rtp->lastts - pred) < 7200) {
+ rtp->lastts = pred;
+ rtp->lastotexttimestamp += frame->samples;
+ } else {
+ ast_debug(3, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, frame->samples);
+ rtp->lastotexttimestamp = rtp->lastts;
+ }
+ }
+ }
+
+ /* If we have been explicitly told to set the marker bit then do so */
+ if (ast_test_flag(rtp, FLAG_NEED_MARKER_BIT)) {
+ mark = 1;
+ ast_clear_flag(rtp, FLAG_NEED_MARKER_BIT);
+ }
+
+ /* If the timestamp for non-digt packets has moved beyond the timestamp for digits, update the digit timestamp */
+ if (rtp->lastts > rtp->lastdigitts) {
+ rtp->lastdigitts = rtp->lastts;
+ }
+
+ if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
+ rtp->lastts = frame->ts * 8;
+ }
+
+ /* If we know the remote address construct a packet and send it out */
+ if (instance->remote_address.sin_port && instance->remote_address.sin_addr.s_addr) {
+ int hdrlen = 12, res;
+ unsigned char *rtpheader = (unsigned char *)(frame->data.ptr - hdrlen);
+
+ put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (rtp->seqno) | (mark << 23)));
+ put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
+ put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc));
+
+ if ((res = sendto(rtp->s, (void *)rtpheader, frame->datalen + hdrlen, 0, (struct sockaddr *)&instance->remote_address, sizeof(instance->remote_address))) < 0) {
+ if (!instance->properties[AST_RTP_PROPERTY_NAT] || (instance->properties[AST_RTP_PROPERTY_NAT] && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
+ ast_debug(1, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(instance->remote_address.sin_addr), ntohs(instance->remote_address.sin_port), strerror(errno));
+ } else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
+ /* Only give this error message once if we are not RTP debugging */
+ if (option_debug || rtpdebug)
+ ast_debug(0, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(instance->remote_address.sin_addr), ntohs(instance->remote_address.sin_port));
+ ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
+ }
+ } else {
+ rtp->txcount++;
+ rtp->txoctetcount += (res - hdrlen);
+
+ if (rtp->rtcp && rtp->rtcp->schedid < 1) {
+ ast_debug(1, "Starting RTCP transmission on RTP instance '%p'\n", instance);
+ rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
+ }
+ }
+
+ if (rtp_debug_test_addr(&instance->remote_address)) {
+ ast_verbose("Sent RTP packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(instance->remote_address.sin_addr), ntohs(instance->remote_address.sin_port), codec, rtp->seqno, rtp->lastts, res - hdrlen);
+ }
+ }
+
+ rtp->seqno++;
+
+ return 0;
+}
+
+static struct ast_frame *red_t140_to_red(struct rtp_red *red) {
+ unsigned char *data = red->t140red.data.ptr;
+ int len = 0;
+ int i;
+
+ /* replace most aged generation */
+ if (red->len[0]) {
+ for (i = 1; i < red->num_gen+1; i++)
+ len += red->len[i];
+
+ memmove(&data[red->hdrlen], &data[red->hdrlen+red->len[0]], len);
+ }
+
+ /* Store length of each generation and primary data length*/
+ for (i = 0; i < red->num_gen; i++)
+ red->len[i] = red->len[i+1];
+ red->len[i] = red->t140.datalen;
+
+ /* write each generation length in red header */
+ len = red->hdrlen;
+ for (i = 0; i < red->num_gen; i++)
+ len += data[i*4+3] = red->len[i];
+
+ /* add primary data to buffer */
+ memcpy(&data[len], red->t140.data.ptr, red->t140.datalen);
+ red->t140red.datalen = len + red->t140.datalen;
+
+ /* no primary data and no generations to send */
+ if (len == red->hdrlen && !red->t140.datalen)
+ return NULL;
+
+ /* reset t.140 buffer */
+ red->t140.datalen = 0;
+
+ return &red->t140red;
+}
+
+static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+ struct ast_rtp *rtp = instance->data;
+ int codec, subclass;
+
+ /* If we don't actually know the remote address don't even bother doing anything */
+ if (!instance->remote_address.sin_addr.s_addr) {
+ ast_debug(1, "No remote address on RTP instance '%p' so dropping frame\n", instance);
+ return -1;
+ }
+
[... 1439 lines stripped ...]
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