diff options
Diffstat (limited to '1.2-netsec/rtp.c')
| -rw-r--r-- | 1.2-netsec/rtp.c | 1896 |
1 files changed, 0 insertions, 1896 deletions
diff --git a/1.2-netsec/rtp.c b/1.2-netsec/rtp.c deleted file mode 100644 index 2df27e287..000000000 --- a/1.2-netsec/rtp.c +++ /dev/null @@ -1,1896 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 1999 - 2005, Digium, Inc. - * - * Mark Spencer <markster@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. - * - * RTP is deffined in RFC 3550. - */ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <sys/time.h> -#include <signal.h> -#include <errno.h> -#include <unistd.h> -#include <netinet/in.h> -#include <sys/time.h> -#include <sys/socket.h> -#include <arpa/inet.h> -#include <fcntl.h> - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include "asterisk/rtp.h" -#include "asterisk/frame.h" -#include "asterisk/logger.h" -#include "asterisk/options.h" -#include "asterisk/channel.h" -#include "asterisk/acl.h" -#include "asterisk/channel.h" -#include "asterisk/config.h" -#include "asterisk/lock.h" -#include "asterisk/utils.h" -#include "asterisk/cli.h" -#include "asterisk/unaligned.h" -#include "asterisk/utils.h" - -#define MAX_TIMESTAMP_SKEW 640 - -#define RTP_MTU 1200 - -#define DEFAULT_DTMF_TIMEOUT 3000 /* samples */ - -static int dtmftimeout = DEFAULT_DTMF_TIMEOUT; - -static int rtpstart = 0; -static int rtpend = 0; -static int rtpdebug = 0; /* Are we debugging? */ -static struct sockaddr_in rtpdebugaddr; /* Debug packets to/from this host */ -#ifdef SO_NO_CHECK -static int nochecksums = 0; -#endif - -/* The value of each payload format mapping: */ -struct rtpPayloadType { - int isAstFormat; /* whether the following code is an AST_FORMAT */ - int code; -}; - -#define MAX_RTP_PT 256 - -#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) - -struct ast_rtp { - int s; - char resp; - struct ast_frame f; - unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET]; - /*! Synchronization source, RFC 3550, page 10. */ - unsigned int ssrc; - unsigned int lastts; - unsigned int lastdigitts; - unsigned int lastrxts; - unsigned int lastividtimestamp; - unsigned int lastovidtimestamp; - unsigned int lasteventseqn; - unsigned int lasteventendseqn; - int lasttxformat; - int lastrxformat; - int dtmfcount; - unsigned int dtmfduration; - int nat; - unsigned int flags; - /*! Socket representation of the local endpoint. */ - struct sockaddr_in us; - /*! Socket representation of the remote endpoint. */ - struct sockaddr_in them; - struct timeval rxcore; - struct timeval txcore; - struct timeval dtmfmute; - struct ast_smoother *smoother; - int *ioid; - /*! Sequence number, RFC 3550, page 13. */ - unsigned short seqno; - unsigned short rxseqno; - struct sched_context *sched; - struct io_context *io; - void *data; - ast_rtp_callback callback; - struct rtpPayloadType current_RTP_PT[MAX_RTP_PT]; - /*! a cache for the result of rtp_lookup_code(): */ - int rtp_lookup_code_cache_isAstFormat; - int rtp_lookup_code_cache_code; - int rtp_lookup_code_cache_result; - int rtp_offered_from_local; - -#ifdef MIDCOM - struct sockaddr_in them_midcom_nat; -#endif - struct ast_rtcp *rtcp; -}; - -/*! - * \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 { - /*! Socket */ - int s; - /*! Socket representation of the local endpoint. */ - struct sockaddr_in us; - /*! Socket representation of the remote endpoint. */ - struct sockaddr_in them; -}; - -static struct ast_rtp_protocol *protos = NULL; - -int ast_rtp_fd(struct ast_rtp *rtp) -{ - return rtp->s; -} - -int ast_rtcp_fd(struct ast_rtp *rtp) -{ - if (rtp->rtcp) - return rtp->rtcp->s; - return -1; -} - -void ast_rtp_set_data(struct ast_rtp *rtp, void *data) -{ - rtp->data = data; -} - -void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback) -{ - rtp->callback = callback; -} - -void ast_rtp_setnat(struct ast_rtp *rtp, int nat) -{ - rtp->nat = nat; -} - -static struct ast_frame *send_dtmf(struct ast_rtp *rtp) -{ - static struct ast_frame null_frame = { AST_FRAME_NULL, }; - char iabuf[INET_ADDRSTRLEN]; - - if (ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) { - if (option_debug) - ast_log(LOG_DEBUG, "Ignore potential DTMF echo from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr)); - rtp->resp = 0; - rtp->dtmfduration = 0; - return &null_frame; - } - if (option_debug) - ast_log(LOG_DEBUG, "Sending dtmf: %d (%c), at %s\n", rtp->resp, rtp->resp, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr)); - if (rtp->resp == 'X') { - rtp->f.frametype = AST_FRAME_CONTROL; - rtp->f.subclass = AST_CONTROL_FLASH; - } else { - rtp->f.frametype = AST_FRAME_DTMF; - rtp->f.subclass = rtp->resp; - } - rtp->f.datalen = 0; - rtp->f.samples = 0; - rtp->f.mallocd = 0; - rtp->f.src = "RTP"; - rtp->resp = 0; - rtp->dtmfduration = 0; - return &rtp->f; - -} - -static inline int rtp_debug_test_addr(struct sockaddr_in *addr) -{ - if (rtpdebug == 0) - 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 struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *data, int len) -{ - unsigned int event; - char resp = 0; - struct ast_frame *f = NULL; - event = ntohl(*((unsigned int *)(data))); - event &= 0x001F; -#if 0 - printf("Cisco Digit: %08x (len = %d)\n", event, len); -#endif - if (event < 10) { - resp = '0' + event; - } else if (event < 11) { - resp = '*'; - } else if (event < 12) { - resp = '#'; - } else if (event < 16) { - resp = 'A' + (event - 12); - } else if (event < 17) { - resp = 'X'; - } - if (rtp->resp && (rtp->resp != resp)) { - f = send_dtmf(rtp); - } - rtp->resp = resp; - rtp->dtmfcount = dtmftimeout; - return f; -} - -/*! - * \brief Process RTP DTMF and events according to RFC 2833. - * - * RFC 2833 is "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals". - * - * \param rtp - * \param data - * \param len - * \param seqno - * \returns - */ -static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len, unsigned int seqno) -{ - unsigned int event; - unsigned int event_end; - unsigned int duration; - char resp = 0; - struct ast_frame *f = NULL; - event = ntohl(*((unsigned int *)(data))); - event >>= 24; - event_end = ntohl(*((unsigned int *)(data))); - event_end <<= 8; - event_end >>= 24; - duration = ntohl(*((unsigned int *)(data))); - duration &= 0xFFFF; - if (rtpdebug) - ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len); - if (event < 10) { - resp = '0' + event; - } else if (event < 11) { - resp = '*'; - } else if (event < 12) { - resp = '#'; - } else if (event < 16) { - resp = 'A' + (event - 12); - } else if (event < 17) { /* Event 16: Hook flash */ - resp = 'X'; - } - if (rtp->resp && (rtp->resp != resp)) { - f = send_dtmf(rtp); - } else if(event_end & 0x80) { - if (rtp->resp) { - if(rtp->lasteventendseqn != seqno) { - f = send_dtmf(rtp); - rtp->lasteventendseqn = seqno; - } - rtp->resp = 0; - } - resp = 0; - duration = 0; - } else if (rtp->resp && rtp->dtmfduration && (duration < rtp->dtmfduration)) { - f = send_dtmf(rtp); - } - if (!(event_end & 0x80)) - rtp->resp = resp; - rtp->dtmfcount = dtmftimeout; - rtp->dtmfduration = duration; - return f; -} - -/*! - * \brief Process Comfort Noise RTP. - * - * This is incomplete at the moment. - * -*/ -static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *data, int len) -{ - struct ast_frame *f = NULL; - /* Convert comfort noise into audio with various codecs. Unfortunately this doesn't - totally help us out becuase we don't have an engine to keep it going and we are not - guaranteed to have it every 20ms or anything */ - if (rtpdebug) - ast_log(LOG_DEBUG, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len); - - if (!(ast_test_flag(rtp, FLAG_3389_WARNING))) { - char iabuf[INET_ADDRSTRLEN]; - - ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: %s\n", - ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr)); - ast_set_flag(rtp, FLAG_3389_WARNING); - } - - /* Must have at least one byte */ - if (!len) - return NULL; - if (len < 24) { - rtp->f.data = rtp->rawdata + AST_FRIENDLY_OFFSET; - rtp->f.datalen = len - 1; - rtp->f.offset = AST_FRIENDLY_OFFSET; - memcpy(rtp->f.data, data + 1, len - 1); - } else { - rtp->f.data = NULL; - rtp->f.offset = 0; - rtp->f.datalen = 0; - } - rtp->f.frametype = AST_FRAME_CNG; - rtp->f.subclass = data[0] & 0x7f; - rtp->f.datalen = len - 1; - rtp->f.samples = 0; - rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0; - f = &rtp->f; - return f; -} - -static int rtpread(int *id, int fd, short events, void *cbdata) -{ - struct ast_rtp *rtp = cbdata; - struct ast_frame *f; - f = ast_rtp_read(rtp); - if (f) { - if (rtp->callback) - rtp->callback(rtp, f, rtp->data); - } - return 1; -} - -struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp) -{ - static struct ast_frame null_frame = { AST_FRAME_NULL, }; - socklen_t len; - int hdrlen = 8; - int res; - struct sockaddr_in sin; - unsigned int rtcpdata[1024]; - char iabuf[INET_ADDRSTRLEN]; - - if (!rtp || !rtp->rtcp) - return &null_frame; - - len = sizeof(sin); - - res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata), - 0, (struct sockaddr *)&sin, &len); - - if (res < 0) { - if (errno != EAGAIN) - ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno)); - if (errno == EBADF) - CRASH; - return &null_frame; - } - - if (res < hdrlen) { - ast_log(LOG_WARNING, "RTP Read too short\n"); - return &null_frame; - } - - if (rtp->nat) { - /* Send to whoever sent to us */ - if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || - (rtp->rtcp->them.sin_port != sin.sin_port)) { - memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); - if (option_debug || rtpdebug) - ast_log(LOG_DEBUG, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); - } - } - if (option_debug) - ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res); - return &null_frame; -} - -static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark) -{ - struct timeval ts = ast_samp2tv( timestamp, 8000); - if (ast_tvzero(rtp->rxcore) || mark) { - rtp->rxcore = ast_tvsub(ast_tvnow(), ts); - /* Round to 20ms for nice, pretty timestamps */ - rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 20000; - } - *tv = ast_tvadd(rtp->rxcore, ts); -} - -struct ast_frame *ast_rtp_read(struct ast_rtp *rtp) -{ - int res; - struct sockaddr_in sin; - socklen_t len; - unsigned int seqno; - int version; - int payloadtype; - int hdrlen = 12; - int padding; - int mark; - int ext; - int x; - char iabuf[INET_ADDRSTRLEN]; - unsigned int timestamp; - unsigned int *rtpheader; - static struct ast_frame *f, null_frame = { AST_FRAME_NULL, }; - struct rtpPayloadType rtpPT; - - len = sizeof(sin); - - /* Cache where the header will go */ - res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, - 0, (struct sockaddr *)&sin, &len); - - - rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); - if (res < 0) { - if (errno != EAGAIN) - ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno)); - if (errno == EBADF) - CRASH; - return &null_frame; - } - if (res < hdrlen) { - ast_log(LOG_WARNING, "RTP Read too short\n"); - return &null_frame; - } - - /* Ignore if the other side hasn't been given an address - yet. */ - if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) - return &null_frame; - - if (rtp->nat) { - /* Send to whoever sent to us */ - if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || - (rtp->them.sin_port != sin.sin_port)) { - memcpy(&rtp->them, &sin, sizeof(rtp->them)); - rtp->rxseqno = 0; - ast_set_flag(rtp, FLAG_NAT_ACTIVE); - if (option_debug || rtpdebug) - ast_log(LOG_DEBUG, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port)); - } - } - - /* Get fields */ - seqno = ntohl(rtpheader[0]); - - /* Check RTP version */ - version = (seqno & 0xC0000000) >> 30; - if (version != 2) - return &null_frame; - - payloadtype = (seqno & 0x7f0000) >> 16; - padding = seqno & (1 << 29); - mark = seqno & (1 << 23); - ext = seqno & (1 << 28); - seqno &= 0xffff; - timestamp = ntohl(rtpheader[1]); - - if (padding) { - /* Remove padding bytes */ - res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; - } - - if (ext) { - /* RTP Extension present */ - hdrlen += 4; - hdrlen += (ntohl(rtpheader[3]) & 0xffff) << 2; - } - - if (res < hdrlen) { - ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); - return &null_frame; - } - - if(rtp_debug_test_addr(&sin)) - ast_verbose("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len %d)\n" - , ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); - - rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); - if (!rtpPT.isAstFormat) { - /* This is special in-band data that's not one of our codecs */ - if (rtpPT.code == AST_RTP_DTMF) { - /* It's special -- rfc2833 process it */ - if(rtp_debug_test_addr(&sin)) { - unsigned char *data; - unsigned int event; - unsigned int event_end; - unsigned int duration; - data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen; - event = ntohl(*((unsigned int *)(data))); - event >>= 24; - event_end = ntohl(*((unsigned int *)(data))); - event_end <<= 8; - event_end >>= 24; - duration = ntohl(*((unsigned int *)(data))); - duration &= 0xFFFF; - ast_verbose("Got rfc2833 RTP packet from %s:%d (type %d, seq %d, ts %d, len %d, mark %d, event %08x, end %d, duration %d) \n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration); - } - if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) { - f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno); - rtp->lasteventseqn = seqno; - } else - f = NULL; - if (f) - return f; - else - return &null_frame; - } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { - /* It's really special -- process it the Cisco way */ - if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) { - f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); - rtp->lasteventseqn = seqno; - } else - f = NULL; - if (f) - return f; - else - return &null_frame; - } else if (rtpPT.code == AST_RTP_CN) { - /* Comfort Noise */ - f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); - if (f) - return f; - else - return &null_frame; - } else { - ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype); - return &null_frame; - } - } - rtp->f.subclass = rtpPT.code; - if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) - rtp->f.frametype = AST_FRAME_VOICE; - else - rtp->f.frametype = AST_FRAME_VIDEO; - rtp->lastrxformat = rtp->f.subclass; - - if (!rtp->lastrxts) - rtp->lastrxts = timestamp; - - if (rtp->rxseqno) { - for (x=rtp->rxseqno + 1; x < seqno; x++) { - /* Queue empty frames */ - rtp->f.mallocd = 0; - rtp->f.datalen = 0; - rtp->f.data = NULL; - rtp->f.offset = 0; - rtp->f.samples = 0; - rtp->f.src = "RTPMissedFrame"; - } - } - rtp->rxseqno = seqno; - - if (rtp->dtmfcount) { -#if 0 - printf("dtmfcount was %d\n", rtp->dtmfcount); -#endif - rtp->dtmfcount -= (timestamp - rtp->lastrxts); - if (rtp->dtmfcount < 0) - rtp->dtmfcount = 0; -#if 0 - if (dtmftimeout != rtp->dtmfcount) - printf("dtmfcount is %d\n", rtp->dtmfcount); -#endif - } - rtp->lastrxts = timestamp; - - /* Send any pending DTMF */ - if (rtp->resp && !rtp->dtmfcount) { - if (option_debug) - ast_log(LOG_DEBUG, "Sending pending DTMF\n"); - return send_dtmf(rtp); - } - rtp->f.mallocd = 0; - rtp->f.datalen = res - hdrlen; - rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; - rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; - if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { - rtp->f.samples = ast_codec_get_samples(&rtp->f); - if (rtp->f.subclass == AST_FORMAT_SLINEAR) - ast_frame_byteswap_be(&rtp->f); - calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark); - } else { - /* Video -- samples is # of samples vs. 90000 */ - if (!rtp->lastividtimestamp) - rtp->lastividtimestamp = timestamp; - rtp->f.samples = timestamp - rtp->lastividtimestamp; - rtp->lastividtimestamp = timestamp; - rtp->f.delivery.tv_sec = 0; - rtp->f.delivery.tv_usec = 0; - if (mark) - rtp->f.subclass |= 0x1; - - } - rtp->f.src = "RTP"; - return &rtp->f; -} - -/* The following array defines the MIME Media type (and subtype) for each - of our codecs, or RTP-specific data type. */ -static struct { - struct rtpPayloadType payloadType; - char* type; - char* subtype; -} mimeTypes[] = { - {{1, AST_FORMAT_G723_1}, "audio", "G723"}, - {{1, AST_FORMAT_GSM}, "audio", "GSM"}, - {{1, AST_FORMAT_ULAW}, "audio", "PCMU"}, - {{1, AST_FORMAT_ALAW}, "audio", "PCMA"}, - {{1, AST_FORMAT_G726}, "audio", "G726-32"}, - {{1, AST_FORMAT_ADPCM}, "audio", "DVI4"}, - {{1, AST_FORMAT_SLINEAR}, "audio", "L16"}, - {{1, AST_FORMAT_LPC10}, "audio", "LPC"}, - {{1, AST_FORMAT_G729A}, "audio", "G729"}, - {{1, AST_FORMAT_SPEEX}, "audio", "speex"}, - {{1, AST_FORMAT_ILBC}, "audio", "iLBC"}, - {{0, AST_RTP_DTMF}, "audio", "telephone-event"}, - {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"}, - {{0, AST_RTP_CN}, "audio", "CN"}, - {{1, AST_FORMAT_JPEG}, "video", "JPEG"}, - {{1, AST_FORMAT_PNG}, "video", "PNG"}, - {{1, AST_FORMAT_H261}, "video", "H261"}, - {{1, AST_FORMAT_H263}, "video", "H263"}, - {{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998"}, -}; - -/* Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s: - also, our own choices for dynamic payload types. This is our master - table for transmission */ -static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = { - [0] = {1, AST_FORMAT_ULAW}, -#ifdef USE_DEPRECATED_G726 - [2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */ -#endif - [3] = {1, AST_FORMAT_GSM}, - [4] = {1, AST_FORMAT_G723_1}, - [5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */ - [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */ - [7] = {1, AST_FORMAT_LPC10}, - [8] = {1, AST_FORMAT_ALAW}, - [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */ - [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */ - [13] = {0, AST_RTP_CN}, - [16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */ - [17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */ - [18] = {1, AST_FORMAT_G729A}, - [19] = {0, AST_RTP_CN}, /* Also used for CN */ - [26] = {1, AST_FORMAT_JPEG}, - [31] = {1, AST_FORMAT_H261}, - [34] = {1, AST_FORMAT_H263}, - [103] = {1, AST_FORMAT_H263_PLUS}, - [97] = {1, AST_FORMAT_ILBC}, - [101] = {0, AST_RTP_DTMF}, - [110] = {1, AST_FORMAT_SPEEX}, - [111] = {1, AST_FORMAT_G726}, - [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */ -}; - -void ast_rtp_pt_clear(struct ast_rtp* rtp) -{ - int i; - if (!rtp) - return; - - for (i = 0; i < MAX_RTP_PT; ++i) { - rtp->current_RTP_PT[i].isAstFormat = 0; - rtp->current_RTP_PT[i].code = 0; - } - - rtp->rtp_lookup_code_cache_isAstFormat = 0; - rtp->rtp_lookup_code_cache_code = 0; - rtp->rtp_lookup_code_cache_result = 0; -} - -void ast_rtp_pt_default(struct ast_rtp* rtp) -{ - int i; - - /* Initialize to default payload types */ - for (i = 0; i < MAX_RTP_PT; ++i) { - rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat; - rtp->current_RTP_PT[i].code = static_RTP_PT[i].code; - } - - rtp->rtp_lookup_code_cache_isAstFormat = 0; - rtp->rtp_lookup_code_cache_code = 0; - rtp->rtp_lookup_code_cache_result = 0; -} - -/* Make a note of a RTP paymoad type that was seen in a SDP "m=" line. */ -/* By default, use the well-known value for this type (although it may */ -/* still be set to a different value by a subsequent "a=rtpmap:" line): */ -void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) { - if (pt < 0 || pt > MAX_RTP_PT) - return; /* bogus payload type */ - - if (static_RTP_PT[pt].code != 0) { - rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; - } -} - -/* Make a note of a RTP payload type (with MIME type) that was seen in */ -/* a SDP "a=rtpmap:" line. */ -void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt, - char* mimeType, char* mimeSubtype) { - int i; - - if (pt < 0 || pt > MAX_RTP_PT) - return; /* bogus payload type */ - - for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) { - if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && - strcasecmp(mimeType, mimeTypes[i].type) == 0) { - rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; - return; - } - } -} - -/* Return the union of all of the codecs that were set by rtp_set...() calls */ -/* They're returned as two distinct sets: AST_FORMATs, and AST_RTPs */ -void ast_rtp_get_current_formats(struct ast_rtp* rtp, - int* astFormats, int* nonAstFormats) { - int pt; - - *astFormats = *nonAstFormats = 0; - for (pt = 0; pt < MAX_RTP_PT; ++pt) { - if (rtp->current_RTP_PT[pt].isAstFormat) { - *astFormats |= rtp->current_RTP_PT[pt].code; - } else { - *nonAstFormats |= rtp->current_RTP_PT[pt].code; - } - } -} - -void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) { - if (rtp) - rtp->rtp_offered_from_local = local; - else - ast_log(LOG_WARNING, "rtp structure is null\n"); -} - -struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt) -{ - struct rtpPayloadType result; - - result.isAstFormat = result.code = 0; - if (pt < 0 || pt > MAX_RTP_PT) - return result; /* bogus payload type */ - - /* Start with the negotiated codecs */ - if (!rtp->rtp_offered_from_local) - result = rtp->current_RTP_PT[pt]; - - /* If it doesn't exist, check our static RTP type list, just in case */ - if (!result.code) - result = static_RTP_PT[pt]; - return result; -} - -/* Looks up an RTP code out of our *static* outbound list */ -int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int code) { - - int pt; - - if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat && - code == rtp->rtp_lookup_code_cache_code) { - - /* Use our cached mapping, to avoid the overhead of the loop below */ - return rtp->rtp_lookup_code_cache_result; - } - - /* Check the dynamic list first */ - for (pt = 0; pt < MAX_RTP_PT; ++pt) { - if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) { - rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; - rtp->rtp_lookup_code_cache_code = code; - rtp->rtp_lookup_code_cache_result = pt; - return pt; - } - } - - /* Then the static list */ - for (pt = 0; pt < MAX_RTP_PT; ++pt) { - if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) { - rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat; - rtp->rtp_lookup_code_cache_code = code; - rtp->rtp_lookup_code_cache_result = pt; - return pt; - } - } - return -1; -} - -char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code) { - - int i; - - for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) { - if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat) { - return mimeTypes[i].subtype; - } - } - return ""; -} - -char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, const int isAstFormat) -{ - int format; - unsigned len; - char *end = buf; - char *start = buf; - - if (!buf || !size) - return NULL; - - snprintf(end, size, "0x%x (", capability); - - len = strlen(end); - end += len; - size -= len; - start = end; - - for (format = 1; format < AST_RTP_MAX; format <<= 1) { - if (capability & format) { - const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format); - snprintf(end, size, "%s|", name); - len = strlen(end); - end += len; - size -= len; - } - } - - if (start == end) - snprintf(start, size, "nothing)"); - else if (size > 1) - *(end -1) = ')'; - - return buf; -} - -static int rtp_socket(void) -{ - int s; - long flags; - s = socket(AF_INET, SOCK_DGRAM, 0); - if (s > -1) { - flags = fcntl(s, F_GETFL); - fcntl(s, F_SETFL, flags | O_NONBLOCK); -#ifdef SO_NO_CHECK - if (nochecksums) - setsockopt(s, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums)); -#endif - } - return s; -} - -/*! - * \brief Initialize a new RTCP session. - * - * \returns The newly initialized RTCP session. - */ -static struct ast_rtcp *ast_rtcp_new(void) -{ - struct ast_rtcp *rtcp; - rtcp = malloc(sizeof(struct ast_rtcp)); - if (!rtcp) - return NULL; - memset(rtcp, 0, sizeof(struct ast_rtcp)); - rtcp->s = rtp_socket(); - rtcp->us.sin_family = AF_INET; - if (rtcp->s < 0) { - free(rtcp); - ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno)); - return NULL; - } - return rtcp; -} - -struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr addr) -{ - struct ast_rtp *rtp; - int x; - int first; - int startplace; - rtp = malloc(sizeof(struct ast_rtp)); - if (!rtp) - return NULL; - memset(rtp, 0, sizeof(struct ast_rtp)); - rtp->them.sin_family = AF_INET; - rtp->us.sin_family = AF_INET; - rtp->s = rtp_socket(); - rtp->ssrc = rand(); - rtp->seqno = rand() & 0xffff; - if (rtp->s < 0) { - free(rtp); - ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno)); - return NULL; - } - if (sched && rtcpenable) { - rtp->sched = sched; - rtp->rtcp = ast_rtcp_new(); - } - - /* Select a random port number in the range of possible RTP */ - x = (rand() % (rtpend-rtpstart)) + rtpstart; - x = x & ~1; - /* Save it for future references. */ - startplace = x; - /* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */ - for (;;) { - /* Must be an even port number by RTP spec */ - rtp->us.sin_port = htons(x); - rtp->us.sin_addr = addr; - /* If there's rtcp, initialize it as well. */ - if (rtp->rtcp) - rtp->rtcp->us.sin_port = htons(x + 1); - /* Try to bind it/them. */ - if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) && - (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))) - break; - if (!first) { - /* Primary bind succeeded! Gotta recreate it */ - close(rtp->s); - rtp->s = rtp_socket(); - } - if (errno != EADDRINUSE) { - /* We got an error that wasn't expected, abort! */ - ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); - close(rtp->s); - if (rtp->rtcp) { - close(rtp->rtcp->s); - free(rtp->rtcp); - } - free(rtp); - return NULL; - } - /* The port was used, increment it (by two). */ - x += 2; - /* Did we go over the limit ? */ - if (x > rtpend) - /* then, start from the begingig. */ - x = (rtpstart + 1) & ~1; - /* Check if we reached the place were we started. */ - if (x == startplace) { - /* If so, there's no ports available. */ - ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); - close(rtp->s); - if (rtp->rtcp) { - close(rtp->rtcp->s); - free(rtp->rtcp); - } - free(rtp); - return NULL; - } - } - if (io && sched && callbackmode) { - /* Operate this one in a callback mode */ - rtp->sched = sched; - rtp->io = io; - rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); - } - ast_rtp_pt_default(rtp); - return rtp; -} - -struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode) -{ - struct in_addr ia; - - memset(&ia, 0, sizeof(ia)); - return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia); -} - -int ast_rtp_settos(struct ast_rtp *rtp, int tos) -{ - int res; - - if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) - ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); - return res; -} - -void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them) -{ - rtp->them.sin_port = them->sin_port; - rtp->them.sin_addr = them->sin_addr; - if (rtp->rtcp) { - rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1); - rtp->rtcp->them.sin_addr = them->sin_addr; - } - rtp->rxseqno = 0; -} - -void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them) -{ - them->sin_family = AF_INET; - them->sin_port = rtp->them.sin_port; - them->sin_addr = rtp->them.sin_addr; -} - -void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us) -{ - memcpy(us, &rtp->us, sizeof(rtp->us)); -} - -#ifdef MIDCOM /* RANCH */ -void ast_rtp_nat_us(struct ast_rtp *rtp, struct sockaddr_in *our_nat) -{ - memcpy(&rtp->them_midcom_nat, our_nat, sizeof(rtp->them_midcom_nat)); -} - -void ast_rtp_get_their_nat(struct ast_rtp *rtp, struct sockaddr_in *their_nat) -{ - their_nat->sin_family = AF_INET; - their_nat->sin_port = rtp->them_midcom_nat.sin_port; - their_nat->sin_addr = rtp->them_midcom_nat.sin_addr; -} -#endif - -void ast_rtp_stop(struct ast_rtp *rtp) -{ - memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); - memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port)); - if (rtp->rtcp) { - memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->them.sin_addr)); - memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->them.sin_port)); - } -} - -void ast_rtp_reset(struct ast_rtp *rtp) -{ - memset(&rtp->rxcore, 0, sizeof(rtp->rxcore)); - memset(&rtp->txcore, 0, sizeof(rtp->txcore)); - memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute)); - rtp->lastts = 0; - rtp->lastdigitts = 0; - rtp->lastrxts = 0; - rtp->lastividtimestamp = 0; - rtp->lastovidtimestamp = 0; - rtp->lasteventseqn = 0; - rtp->lasteventendseqn = 0; - rtp->lasttxformat = 0; - rtp->lastrxformat = 0; - rtp->dtmfcount = 0; - rtp->dtmfduration = 0; - rtp->seqno = 0; - rtp->rxseqno = 0; -} - -void ast_rtp_destroy(struct ast_rtp *rtp) -{ - if (rtp->smoother) - ast_smoother_free(rtp->smoother); - if (rtp->ioid) - ast_io_remove(rtp->io, rtp->ioid); - if (rtp->s > -1) - close(rtp->s); - if (rtp->rtcp) { - close(rtp->rtcp->s); - free(rtp->rtcp); - } - free(rtp); -} - -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(); - /* Round to 20ms for nice, pretty timestamps */ - rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000; - } - /* Use previous txcore if available */ - t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow(); - ms = ast_tvdiff_ms(t, rtp->txcore); - if (ms < 0) - ms = 0; - /* Use what we just got for next time */ - rtp->txcore = t; - return (unsigned int) ms; -} - -int ast_rtp_senddigit(struct ast_rtp *rtp, char digit) -{ - unsigned int *rtpheader; - int hdrlen = 12; - int res; - int x; - int payload; - char data[256]; - char iabuf[INET_ADDRSTRLEN]; - - if ((digit <= '9') && (digit >= '0')) - digit -= '0'; - else if (digit == '*') - digit = 10; - else if (digit == '#') - digit = 11; - else if ((digit >= 'A') && (digit <= 'D')) - digit = digit - 'A' + 12; - else if ((digit >= 'a') && (digit <= 'd')) - digit = digit - 'a' + 12; - else { - ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit); - return -1; - } - payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); - - /* If we have no peer, return immediately */ - if (!rtp->them.sin_addr.s_addr) - return 0; - - rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); - - /* Get a pointer to the header */ - rtpheader = (unsigned int *)data; - rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno)); - rtpheader[1] = htonl(rtp->lastdigitts); - rtpheader[2] = htonl(rtp->ssrc); - rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0)); - for (x = 0; x < 6; x++) { - if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { - res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); - if (res < 0) - ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n", - ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), - ntohs(rtp->them.sin_port), strerror(errno)); - if (rtp_debug_test_addr(&rtp->them)) - ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n", - ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), - ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); - } - /* Sequence number of last two end packets does not get incremented */ - if (x < 3) - rtp->seqno++; - /* Clear marker bit and set seqno */ - rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); - /* For the last three packets, set the duration and the end bit */ - if (x == 2) { -#if 0 - /* No, this is wrong... Do not increment lastdigitts, that's not according - to the RFC, as best we can determine */ - rtp->lastdigitts++; /* or else the SPA3000 will click instead of beeping... */ - rtpheader[1] = htonl(rtp->lastdigitts); -#endif - /* Make duration 800 (100ms) */ - rtpheader[3] |= htonl((800)); - /* Set the End bit */ - rtpheader[3] |= htonl((1 << 23)); - } - } - /* Increment the digit timestamp by 120ms, to ensure that digits - sent sequentially with no intervening non-digit packets do not - get sent with the same timestamp, and that sequential digits - have some 'dead air' in between them - */ - rtp->lastdigitts += 960; - /* Increment the sequence number to reflect the last packet - that was sent - */ - rtp->seqno++; - return 0; -} - -int ast_rtp_sendcng(struct ast_rtp *rtp, int level) -{ - unsigned int *rtpheader; - int hdrlen = 12; - int res; - int payload; - char data[256]; - char iabuf[INET_ADDRSTRLEN]; - level = 127 - (level & 0x7f); - payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN); - - /* If we have no peer, return immediately */ - if (!rtp->them.sin_addr.s_addr) - return 0; - - rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); - - /* Get a pointer to the header */ - rtpheader = (unsigned int *)data; - rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++)); - rtpheader[1] = htonl(rtp->lastts); - rtpheader[2] = htonl(rtp->ssrc); - data[12] = level; - if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { - res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); - if (res <0) - ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s:%d: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno)); - if(rtp_debug_test_addr(&rtp->them)) - ast_verbose("Sent Comfort Noise RTP packet to %s:%d (type %d, seq %d, ts %d, len %d)\n" - , ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); - - } - return 0; -} - -static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec) -{ - unsigned char *rtpheader; - char iabuf[INET_ADDRSTRLEN]; - int hdrlen = 12; - int res; - unsigned int ms; - int pred; - int mark = 0; - - ms = calc_txstamp(rtp, &f->delivery); - /* Default prediction */ - if (f->subclass < AST_FORMAT_MAX_AUDIO) { - pred = rtp->lastts + f->samples; - - /* Re-calculate last TS */ - rtp->lastts = rtp->lastts + ms * 8; - if (ast_tvzero(f->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 { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms); - mark = 1; - } - } - } else { - mark = f->subclass & 0x1; - pred = rtp->lastovidtimestamp + f->samples; - /* Re-calculate last TS */ - rtp->lastts = rtp->lastts + ms * 90; - /* If it's close to our prediction, go for it */ - if (ast_tvzero(f->delivery)) { - if (abs(rtp->lastts - pred) < 7200) { - rtp->lastts = pred; - rtp->lastovidtimestamp += f->samples; - } else { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, f->samples); - rtp->lastovidtimestamp = rtp->lastts; - } - } - } - /* If the timestamp for non-digit packets has moved beyond the timestamp - for digits, update the digit timestamp. - */ - if (rtp->lastts > rtp->lastdigitts) - rtp->lastdigitts = rtp->lastts; - - /* Get a pointer to the header */ - rtpheader = (unsigned char *)(f->data - 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 (rtp->them.sin_port && rtp->them.sin_addr.s_addr) { - res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them)); - if (res <0) { - if (!rtp->nat || (rtp->nat && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) { - ast_log(LOG_DEBUG, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno)); - } else if ((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) { - /* Only give this error message once if we are not RTP debugging */ - if (option_debug || rtpdebug) - ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port)); - ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN); - } - } - - if(rtp_debug_test_addr(&rtp->them)) - ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n" - , ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), codec, rtp->seqno, rtp->lastts,res - hdrlen); - } - - rtp->seqno++; - - return 0; -} - -int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f) -{ - struct ast_frame *f; - int codec; - int hdrlen = 12; - int subclass; - - - /* If we have no peer, return immediately */ - if (!rtp->them.sin_addr.s_addr) - return 0; - - /* If there is no data length, return immediately */ - if (!_f->datalen) - return 0; - - /* Make sure we have enough space for RTP header */ - if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { - ast_log(LOG_WARNING, "RTP can only send voice\n"); - return -1; - } - - subclass = _f->subclass; - if (_f->frametype == AST_FRAME_VIDEO) - subclass &= ~0x1; - - codec = ast_rtp_lookup_code(rtp, 1, subclass); - if (codec < 0) { - ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass)); - return -1; - } - - if (rtp->lasttxformat != subclass) { - /* New format, reset the smoother */ - if (option_debug) - ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass)); - rtp->lasttxformat = subclass; - if (rtp->smoother) - ast_smoother_free(rtp->smoother); - rtp->smoother = NULL; - } - - - switch(subclass) { - case AST_FORMAT_SLINEAR: - if (!rtp->smoother) { - rtp->smoother = ast_smoother_new(320); - } - if (!rtp->smoother) { - ast_log(LOG_WARNING, "Unable to create smoother :(\n"); - return -1; - } - ast_smoother_feed_be(rtp->smoother, _f); - - while((f = ast_smoother_read(rtp->smoother))) - ast_rtp_raw_write(rtp, f, codec); - break; - case AST_FORMAT_ULAW: - case AST_FORMAT_ALAW: - if (!rtp->smoother) { - rtp->smoother = ast_smoother_new(160); - } - if (!rtp->smoother) { - ast_log(LOG_WARNING, "Unable to create smoother :(\n"); - return -1; - } - ast_smoother_feed(rtp->smoother, _f); - - while((f = ast_smoother_read(rtp->smoother))) - ast_rtp_raw_write(rtp, f, codec); - break; - case AST_FORMAT_ADPCM: - case AST_FORMAT_G726: - if (!rtp->smoother) { - rtp->smoother = ast_smoother_new(80); - } - if (!rtp->smoother) { - ast_log(LOG_WARNING, "Unable to create smoother :(\n"); - return -1; - } - ast_smoother_feed(rtp->smoother, _f); - - while((f = ast_smoother_read(rtp->smoother))) - ast_rtp_raw_write(rtp, f, codec); - break; - case AST_FORMAT_G729A: - if (!rtp->smoother) { - rtp->smoother = ast_smoother_new(20); - if (rtp->smoother) - ast_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_G729); - } - if (!rtp->smoother) { - ast_log(LOG_WARNING, "Unable to create g729 smoother :(\n"); - return -1; - } - ast_smoother_feed(rtp->smoother, _f); - - while((f = ast_smoother_read(rtp->smoother))) - ast_rtp_raw_write(rtp, f, codec); - break; - case AST_FORMAT_GSM: - if (!rtp->smoother) { - rtp->smoother = ast_smoother_new(33); - } - if (!rtp->smoother) { - ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n"); - return -1; - } - ast_smoother_feed(rtp->smoother, _f); - while((f = ast_smoother_read(rtp->smoother))) - ast_rtp_raw_write(rtp, f, codec); - break; - case AST_FORMAT_ILBC: - if (!rtp->smoother) { - rtp->smoother = ast_smoother_new(50); - } - if (!rtp->smoother) { - ast_log(LOG_WARNING, "Unable to create ILBC smoother :(\n"); - return -1; - } - ast_smoother_feed(rtp->smoother, _f); - while((f = ast_smoother_read(rtp->smoother))) - ast_rtp_raw_write(rtp, f, codec); - break; - default: - ast_log(LOG_WARNING, "Not sure about sending format %s packets\n", ast_getformatname(subclass)); - /* fall through to... */ - case AST_FORMAT_H261: - case AST_FORMAT_H263: - case AST_FORMAT_H263_PLUS: - case AST_FORMAT_G723_1: - case AST_FORMAT_LPC10: - case AST_FORMAT_SPEEX: - /* Don't buffer outgoing frames; send them one-per-packet: */ - if (_f->offset < hdrlen) { - f = ast_frdup(_f); - } else { - f = _f; - } - ast_rtp_raw_write(rtp, f, codec); - } - - return 0; -} - -/*--- ast_rtp_proto_unregister: Unregister interface to channel driver */ -void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto) -{ - struct ast_rtp_protocol *cur, *prev; - - cur = protos; - prev = NULL; - while(cur) { - if (cur == proto) { - if (prev) - prev->next = proto->next; - else - protos = proto->next; - return; - } - prev = cur; - cur = cur->next; - } -} - -/*--- ast_rtp_proto_register: Register interface to channel driver */ -int ast_rtp_proto_register(struct ast_rtp_protocol *proto) -{ - struct ast_rtp_protocol *cur; - cur = protos; - while(cur) { - if (cur->type == proto->type) { - ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); - return -1; - } - cur = cur->next; - } - proto->next = protos; - protos = proto; - return 0; -} - -/*--- get_proto: Get channel driver interface structure */ -static struct ast_rtp_protocol *get_proto(struct ast_channel *chan) -{ - struct ast_rtp_protocol *cur; - - cur = protos; - while(cur) { - if (cur->type == chan->type) { - return cur; - } - cur = cur->next; - } - return NULL; -} - -/* ast_rtp_bridge: Bridge calls. If possible and allowed, initiate - re-invite so the peers exchange media directly outside - of Asterisk. */ -enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms) -{ - struct ast_frame *f; - struct ast_channel *who, *cs[3]; - struct ast_rtp *p0, *p1; /* Audio RTP Channels */ - struct ast_rtp *vp0, *vp1; /* Video RTP channels */ - struct ast_rtp_protocol *pr0, *pr1; - struct sockaddr_in ac0, ac1; - struct sockaddr_in vac0, vac1; - struct sockaddr_in t0, t1; - struct sockaddr_in vt0, vt1; - char iabuf[INET_ADDRSTRLEN]; - void *pvt0, *pvt1; - int codec0,codec1, oldcodec0, oldcodec1; - - memset(&vt0, 0, sizeof(vt0)); - memset(&vt1, 0, sizeof(vt1)); - memset(&vac0, 0, sizeof(vac0)); - memset(&vac1, 0, sizeof(vac1)); - - /* if need DTMF, cant native bridge */ - if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1)) - return AST_BRIDGE_FAILED_NOWARN; - - /* Lock channels */ - ast_mutex_lock(&c0->lock); - while(ast_mutex_trylock(&c1->lock)) { - ast_mutex_unlock(&c0->lock); - usleep(1); - ast_mutex_lock(&c0->lock); - } - - /* Find channel driver interfaces */ - pr0 = get_proto(c0); - pr1 = get_proto(c1); - if (!pr0) { - ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); - ast_mutex_unlock(&c0->lock); - ast_mutex_unlock(&c1->lock); - return AST_BRIDGE_FAILED; - } - if (!pr1) { - ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); - ast_mutex_unlock(&c0->lock); - ast_mutex_unlock(&c1->lock); - return AST_BRIDGE_FAILED; - } - - /* Get channel specific interface structures */ - pvt0 = c0->tech_pvt; - pvt1 = c1->tech_pvt; - - /* Get audio and video interface (if native bridge is possible) */ - p0 = pr0->get_rtp_info(c0); - if (pr0->get_vrtp_info) - vp0 = pr0->get_vrtp_info(c0); - else - vp0 = NULL; - p1 = pr1->get_rtp_info(c1); - if (pr1->get_vrtp_info) - vp1 = pr1->get_vrtp_info(c1); - else - vp1 = NULL; - - /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ - if (!p0 || !p1) { - /* Somebody doesn't want to play... */ - ast_mutex_unlock(&c0->lock); - ast_mutex_unlock(&c1->lock); - return AST_BRIDGE_FAILED_NOWARN; - } - /* Get codecs from both sides */ - if (pr0->get_codec) - codec0 = pr0->get_codec(c0); - else - codec0 = 0; - if (pr1->get_codec) - codec1 = pr1->get_codec(c1); - else - codec1 = 0; - if (pr0->get_codec && pr1->get_codec) { - /* Hey, we can't do reinvite if both parties speak different codecs */ - if (!(codec0 & codec1)) { - if (option_debug) - ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); - ast_mutex_unlock(&c0->lock); - ast_mutex_unlock(&c1->lock); - return AST_BRIDGE_FAILED_NOWARN; - } - } - - /* Ok, we should be able to redirect the media. Start with one channel */ - if (pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE))) - ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name); - else { - /* Store RTP peer */ - ast_rtp_get_peer(p1, &ac1); - if (vp1) - ast_rtp_get_peer(vp1, &vac1); - } - /* Then test the other channel */ - if (pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE))) - ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name); - else { - /* Store RTP peer */ - ast_rtp_get_peer(p0, &ac0); - if (vp0) - ast_rtp_get_peer(vp0, &vac0); - } - ast_mutex_unlock(&c0->lock); - ast_mutex_unlock(&c1->lock); - /* External RTP Bridge up, now loop and see if something happes that force us to take the - media back to Asterisk */ - cs[0] = c0; - cs[1] = c1; - cs[2] = NULL; - oldcodec0 = codec0; - oldcodec1 = codec1; - for (;;) { - /* Check if something changed... */ - if ((c0->tech_pvt != pvt0) || - (c1->tech_pvt != pvt1) || - (c0->masq || c0->masqr || c1->masq || c1->masqr)) { - ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n"); - if (c0->tech_pvt == pvt0) { - if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name); - } - if (c1->tech_pvt == pvt1) { - if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name); - } - return AST_BRIDGE_RETRY; - } - /* Now check if they have changed address */ - ast_rtp_get_peer(p1, &t1); - ast_rtp_get_peer(p0, &t0); - if (pr0->get_codec) - codec0 = pr0->get_codec(c0); - if (pr1->get_codec) - codec1 = pr1->get_codec(c1); - if (vp1) - ast_rtp_get_peer(vp1, &vt1); - if (vp0) - ast_rtp_get_peer(vp0, &vt0); - if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) { - if (option_debug > 1) { - ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", - c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t1.sin_addr), ntohs(t1.sin_port), codec1); - ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n", - c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vt1.sin_addr), ntohs(vt1.sin_port), codec1); - ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", - c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1); - ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", - c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1); - } - if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE))) - ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name); - memcpy(&ac1, &t1, sizeof(ac1)); - memcpy(&vac1, &vt1, sizeof(vac1)); - oldcodec1 = codec1; - } - if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) { - if (option_debug) { - ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", - c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t0.sin_addr), ntohs(t0.sin_port), codec0); - ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n", - c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vt0.sin_addr), ntohs(vt0.sin_port), codec0); - ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", - c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0); - ast_log(LOG_DEBUG, "Oooh, '%s' wasv %s:%d/(format %d)\n", - c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vac0.sin_addr), ntohs(vac0.sin_port), oldcodec0); - } - if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE))) - ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name); - memcpy(&ac0, &t0, sizeof(ac0)); - memcpy(&vac0, &vt0, sizeof(vac0)); - oldcodec0 = codec0; - } - who = ast_waitfor_n(cs, 2, &timeoutms); - if (!who) { - if (!timeoutms) - return AST_BRIDGE_RETRY; - if (option_debug) - ast_log(LOG_DEBUG, "Ooh, empty read...\n"); - /* check for hangup / whentohangup */ - if (ast_check_hangup(c0) || ast_check_hangup(c1)) - break; - continue; - } - f = ast_read(who); - if (!f || ((f->frametype == AST_FRAME_DTMF) && - (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || - ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) { - *fo = f; - *rc = who; - if (option_debug) - ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup"); - if ((c0->tech_pvt == pvt0) && (!c0->_softhangup)) { - if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name); - } - if ((c1->tech_pvt == pvt1) && (!c1->_softhangup)) { - if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name); - } - return AST_BRIDGE_COMPLETE; - } else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) { - if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) || - (f->subclass == AST_CONTROL_VIDUPDATE)) { - ast_indicate(who == c0 ? c1 : c0, f->subclass); - ast_frfree(f); - } else { - *fo = f; - *rc = who; - ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name); - return AST_BRIDGE_COMPLETE; - } - } else { - if ((f->frametype == AST_FRAME_DTMF) || - (f->frametype == AST_FRAME_VOICE) || - (f->frametype == AST_FRAME_VIDEO)) { - /* Forward voice or DTMF frames if they happen upon us */ - if (who == c0) { - ast_write(c1, f); - } else if (who == c1) { - ast_write(c0, f); - } - } - ast_frfree(f); - } - /* Swap priority not that it's a big deal at this point */ - cs[2] = cs[0]; - cs[0] = cs[1]; - cs[1] = cs[2]; - - } - return AST_BRIDGE_FAILED; -} - -static int rtp_do_debug_ip(int fd, int argc, char *argv[]) -{ - struct hostent *hp; - struct ast_hostent ahp; - char iabuf[INET_ADDRSTRLEN]; - int port = 0; - char *p, *arg; - - if (argc != 4) - return RESULT_SHOWUSAGE; - arg = argv[3]; - p = strstr(arg, ":"); - if (p) { - *p = '\0'; - p++; - port = atoi(p); - } - hp = ast_gethostbyname(arg, &ahp); - if (hp == NULL) - return RESULT_SHOWUSAGE; - rtpdebugaddr.sin_family = AF_INET; - memcpy(&rtpdebugaddr.sin_addr, hp->h_addr, sizeof(rtpdebugaddr.sin_addr)); - rtpdebugaddr.sin_port = htons(port); - if (port == 0) - ast_cli(fd, "RTP Debugging Enabled for IP: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtpdebugaddr.sin_addr)); - else - ast_cli(fd, "RTP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtpdebugaddr.sin_addr), port); - rtpdebug = 1; - return RESULT_SUCCESS; -} - -static int rtp_do_debug(int fd, int argc, char *argv[]) -{ - if(argc != 2) { - if(argc != 4) - return RESULT_SHOWUSAGE; - return rtp_do_debug_ip(fd, argc, argv); - } - rtpdebug = 1; - memset(&rtpdebugaddr,0,sizeof(rtpdebugaddr)); - ast_cli(fd, "RTP Debugging Enabled\n"); - return RESULT_SUCCESS; -} - -static int rtp_no_debug(int fd, int argc, char *argv[]) -{ - if(argc !=3) - return RESULT_SHOWUSAGE; - rtpdebug = 0; - ast_cli(fd,"RTP Debugging Disabled\n"); - return RESULT_SUCCESS; -} - -static char debug_usage[] = - "Usage: rtp debug [ip host[:port]]\n" - " Enable dumping of all RTP packets to and from host.\n"; - -static char no_debug_usage[] = - "Usage: rtp no debug\n" - " Disable all RTP debugging\n"; - -static struct ast_cli_entry cli_debug_ip = -{{ "rtp", "debug", "ip", NULL } , rtp_do_debug, "Enable RTP debugging on IP", debug_usage }; - -static struct ast_cli_entry cli_debug = -{{ "rtp", "debug", NULL } , rtp_do_debug, "Enable RTP debugging", debug_usage }; - -static struct ast_cli_entry cli_no_debug = -{{ "rtp", "no", "debug", NULL } , rtp_no_debug, "Disable RTP debugging", no_debug_usage }; - -void ast_rtp_reload(void) -{ - struct ast_config *cfg; - char *s; - - rtpstart = 5000; - rtpend = 31000; - dtmftimeout = DEFAULT_DTMF_TIMEOUT; - cfg = ast_config_load("rtp.conf"); - if (cfg) { - if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) { - rtpstart = atoi(s); - if (rtpstart < 1024) - rtpstart = 1024; - if (rtpstart > 65535) - rtpstart = 65535; - } - if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) { - rtpend = atoi(s); - if (rtpend < 1024) - rtpend = 1024; - if (rtpend > 65535) - rtpend = 65535; - } - if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) { -#ifdef SO_NO_CHECK - if (ast_false(s)) - nochecksums = 1; - else - nochecksums = 0; -#else - if (ast_false(s)) - ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n"); -#endif - } - if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { - dtmftimeout = atoi(s); - if ((dtmftimeout < 0) || (dtmftimeout > 20000)) { - ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", - dtmftimeout, DEFAULT_DTMF_TIMEOUT); - dtmftimeout = DEFAULT_DTMF_TIMEOUT; - }; - } - ast_config_destroy(cfg); - } - if (rtpstart >= rtpend) { - ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n"); - rtpstart = 5000; - rtpend = 31000; - } - if (option_verbose > 1) - ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); - -} - -/*--- ast_rtp_init: Initialize the RTP system in Asterisk */ -void ast_rtp_init(void) -{ - ast_cli_register(&cli_debug); - ast_cli_register(&cli_debug_ip); - ast_cli_register(&cli_no_debug); - ast_rtp_reload(); -} |