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Diffstat (limited to 'trunk/main/rtp.c')
| -rw-r--r-- | trunk/main/rtp.c | 4114 |
1 files changed, 0 insertions, 4114 deletions
diff --git a/trunk/main/rtp.c b/trunk/main/rtp.c deleted file mode 100644 index 5671f8a89..000000000 --- a/trunk/main/rtp.c +++ /dev/null @@ -1,4114 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 1999 - 2006, 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. - * - * \author Mark Spencer <markster@digium.com> - * - * \note RTP is defined in RFC 3550. - */ - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include <sys/time.h> -#include <signal.h> -#include <fcntl.h> - -#include "asterisk/rtp.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" - -#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 */ - -static int dtmftimeout = DEFAULT_DTMF_TIMEOUT; - -static int rtpstart; /*!< First port for RTP sessions (set in rtp.conf) */ -static int rtpend; /*!< 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 int stundebug; /*!< Are we debugging stun? */ -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 */ -}; - -/* Uncomment this to enable more intense native bridging, but note: this is currently buggy */ -/* #define P2P_INTENSE */ - -/*! - * \brief Structure representing a RTP session. - * - * RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]" - * - */ -/*! \brief The value of each payload format mapping: */ -struct rtpPayloadType { - int isAstFormat; /*!< whether the following code is an AST_FORMAT */ - int code; -}; - - -/*! \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; - int nat; - unsigned int flags; - struct sockaddr_in us; /*!< Socket representation of the local endpoint. */ - struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */ - 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; - ast_rtp_callback callback; -#ifdef P2P_INTENSE - ast_mutex_t bridge_lock; -#endif - struct rtpPayloadType current_RTP_PT[MAX_RTP_PT]; - int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */ - int rtp_lookup_code_cache_code; - int rtp_lookup_code_cache_result; - struct ast_rtcp *rtcp; - struct ast_codec_pref pref; - 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 */ -}; - -/* Forward declarations */ -static int ast_rtcp_write(const void *data); -static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw); -static int ast_rtcp_write_sr(const void *data); -static int ast_rtcp_write_rr(const void *data); -static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp); -static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp); -int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit); - -#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_HAS_DTMF (1 << 3) -#define FLAG_P2P_SENT_MARK (1 << 4) -#define FLAG_P2P_NEED_DTMF (1 << 5) -#define FLAG_CALLBACK_MODE (1 << 6) -#define FLAG_DTMF_COMPENSATE (1 << 7) -#define FLAG_HAS_STUN (1 << 8) - -/*! - * \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 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 */ - char quality[AST_MAX_USER_FIELD]; - double maxrxjitter; - double minrxjitter; - double maxrtt; - double minrtt; - int sendfur; -}; - -/*! - * \brief STUN support code - * - * This code provides some support for doing STUN transactions. - * Eventually it should be moved elsewhere as other protocols - * than RTP can benefit from it - e.g. SIP. - * STUN is described in RFC3489 and it is based on the exchange - * of UDP packets between a client and one or more servers to - * determine the externally visible address (and port) of the client - * once it has gone through the NAT boxes that connect it to the - * outside. - * The simplest request packet is just the header defined in - * struct stun_header, and from the response we may just look at - * one attribute, STUN_MAPPED_ADDRESS, that we find in the response. - * By doing more transactions with different server addresses we - * may determine more about the behaviour of the NAT boxes, of - * course - the details are in the RFC. - * - * All STUN packets start with a simple header made of a type, - * length (excluding the header) and a 16-byte random transaction id. - * Following the header we may have zero or more attributes, each - * structured as a type, length and a value (whose format depends - * on the type, but often contains addresses). - * Of course all fields are in network format. - */ - -typedef struct { unsigned int id[4]; } __attribute__((packed)) stun_trans_id; - -struct stun_header { - unsigned short msgtype; - unsigned short msglen; - stun_trans_id id; - unsigned char ies[0]; -} __attribute__((packed)); - -struct stun_attr { - unsigned short attr; - unsigned short len; - unsigned char value[0]; -} __attribute__((packed)); - -/* - * The format normally used for addresses carried by STUN messages. - */ -struct stun_addr { - unsigned char unused; - unsigned char family; - unsigned short port; - unsigned int addr; -} __attribute__((packed)); - -#define STUN_IGNORE (0) -#define STUN_ACCEPT (1) - -/*! \brief STUN message types - * 'BIND' refers to transactions used to determine the externally - * visible addresses. 'SEC' refers to transactions used to establish - * a session key for subsequent requests. - * 'SEC' functionality is not supported here. - */ - -#define STUN_BINDREQ 0x0001 -#define STUN_BINDRESP 0x0101 -#define STUN_BINDERR 0x0111 -#define STUN_SECREQ 0x0002 -#define STUN_SECRESP 0x0102 -#define STUN_SECERR 0x0112 - -/*! \brief Basic attribute types in stun messages. - * Messages can also contain custom attributes (codes above 0x7fff) - */ -#define STUN_MAPPED_ADDRESS 0x0001 -#define STUN_RESPONSE_ADDRESS 0x0002 -#define STUN_CHANGE_REQUEST 0x0003 -#define STUN_SOURCE_ADDRESS 0x0004 -#define STUN_CHANGED_ADDRESS 0x0005 -#define STUN_USERNAME 0x0006 -#define STUN_PASSWORD 0x0007 -#define STUN_MESSAGE_INTEGRITY 0x0008 -#define STUN_ERROR_CODE 0x0009 -#define STUN_UNKNOWN_ATTRIBUTES 0x000a -#define STUN_REFLECTED_FROM 0x000b - -/*! \brief helper function to print message names */ -static const char *stun_msg2str(int msg) -{ - switch (msg) { - case STUN_BINDREQ: - return "Binding Request"; - case STUN_BINDRESP: - return "Binding Response"; - case STUN_BINDERR: - return "Binding Error Response"; - case STUN_SECREQ: - return "Shared Secret Request"; - case STUN_SECRESP: - return "Shared Secret Response"; - case STUN_SECERR: - return "Shared Secret Error Response"; - } - return "Non-RFC3489 Message"; -} - -/*! \brief helper function to print attribute names */ -static const char *stun_attr2str(int msg) -{ - switch (msg) { - case STUN_MAPPED_ADDRESS: - return "Mapped Address"; - case STUN_RESPONSE_ADDRESS: - return "Response Address"; - case STUN_CHANGE_REQUEST: - return "Change Request"; - case STUN_SOURCE_ADDRESS: - return "Source Address"; - case STUN_CHANGED_ADDRESS: - return "Changed Address"; - case STUN_USERNAME: - return "Username"; - case STUN_PASSWORD: - return "Password"; - case STUN_MESSAGE_INTEGRITY: - return "Message Integrity"; - case STUN_ERROR_CODE: - return "Error Code"; - case STUN_UNKNOWN_ATTRIBUTES: - return "Unknown Attributes"; - case STUN_REFLECTED_FROM: - return "Reflected From"; - } - return "Non-RFC3489 Attribute"; -} - -/*! \brief here we store credentials extracted from a message */ -struct stun_state { - const char *username; - const char *password; -}; - -static int stun_process_attr(struct stun_state *state, struct stun_attr *attr) -{ - if (stundebug) - ast_verbose("Found STUN Attribute %s (%04x), length %d\n", - stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len)); - switch (ntohs(attr->attr)) { - case STUN_USERNAME: - state->username = (const char *) (attr->value); - break; - case STUN_PASSWORD: - state->password = (const char *) (attr->value); - break; - default: - if (stundebug) - ast_verbose("Ignoring STUN attribute %s (%04x), length %d\n", - stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len)); - } - return 0; -} - -/*! \brief append a string to an STUN message */ -static void append_attr_string(struct stun_attr **attr, int attrval, const char *s, int *len, int *left) -{ - int size = sizeof(**attr) + strlen(s); - if (*left > size) { - (*attr)->attr = htons(attrval); - (*attr)->len = htons(strlen(s)); - memcpy((*attr)->value, s, strlen(s)); - (*attr) = (struct stun_attr *)((*attr)->value + strlen(s)); - *len += size; - *left -= size; - } -} - -/*! \brief append an address to an STUN message */ -static void append_attr_address(struct stun_attr **attr, int attrval, struct sockaddr_in *sin, int *len, int *left) -{ - int size = sizeof(**attr) + 8; - struct stun_addr *addr; - if (*left > size) { - (*attr)->attr = htons(attrval); - (*attr)->len = htons(8); - addr = (struct stun_addr *)((*attr)->value); - addr->unused = 0; - addr->family = 0x01; - addr->port = sin->sin_port; - addr->addr = sin->sin_addr.s_addr; - (*attr) = (struct stun_attr *)((*attr)->value + 8); - *len += size; - *left -= size; - } -} - -/*! \brief wrapper to send an STUN message */ -static int stun_send(int s, struct sockaddr_in *dst, struct stun_header *resp) -{ - return sendto(s, resp, ntohs(resp->msglen) + sizeof(*resp), 0, - (struct sockaddr *)dst, sizeof(*dst)); -} - -/*! \brief helper function to generate a random request id */ -static void stun_req_id(struct stun_header *req) -{ - int x; - for (x = 0; x < 4; x++) - req->id.id[x] = ast_random(); -} - -size_t ast_rtp_alloc_size(void) -{ - return sizeof(struct ast_rtp); -} - -/*! \brief callback type to be invoked on stun responses. */ -typedef int (stun_cb_f)(struct stun_attr *attr, void *arg); - -/*! \brief handle an incoming STUN message. - * - * Do some basic sanity checks on packet size and content, - * try to extract a bit of information, and possibly reply. - * At the moment this only processes BIND requests, and returns - * the externally visible address of the request. - * If a callback is specified, invoke it with the attribute. - */ -static int stun_handle_packet(int s, struct sockaddr_in *src, - unsigned char *data, size_t len, stun_cb_f *stun_cb, void *arg) -{ - struct stun_header *hdr = (struct stun_header *)data; - struct stun_attr *attr; - struct stun_state st; - int ret = STUN_IGNORE; - int x; - - /* On entry, 'len' is the length of the udp payload. After the - * initial checks it becomes the size of unprocessed options, - * while 'data' is advanced accordingly. - */ - if (len < sizeof(struct stun_header)) { - ast_debug(1, "Runt STUN packet (only %d, wanting at least %d)\n", (int) len, (int) sizeof(struct stun_header)); - return -1; - } - len -= sizeof(struct stun_header); - data += sizeof(struct stun_header); - x = ntohs(hdr->msglen); /* len as advertised in the message */ - if (stundebug) - ast_verbose("STUN Packet, msg %s (%04x), length: %d\n", stun_msg2str(ntohs(hdr->msgtype)), ntohs(hdr->msgtype), x); - if (x > len) { - ast_debug(1, "Scrambled STUN packet length (got %d, expecting %d)\n", x, (int)len); - } else - len = x; - memset(&st, 0, sizeof(st)); - while (len) { - if (len < sizeof(struct stun_attr)) { - ast_debug(1, "Runt Attribute (got %d, expecting %d)\n", (int)len, (int) sizeof(struct stun_attr)); - break; - } - attr = (struct stun_attr *)data; - /* compute total attribute length */ - x = ntohs(attr->len) + sizeof(struct stun_attr); - if (x > len) { - ast_debug(1, "Inconsistent Attribute (length %d exceeds remaining msg len %d)\n", x, (int)len); - break; - } - if (stun_cb) - stun_cb(attr, arg); - if (stun_process_attr(&st, attr)) { - ast_debug(1, "Failed to handle attribute %s (%04x)\n", stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr)); - break; - } - /* Clear attribute id: in case previous entry was a string, - * this will act as the terminator for the string. - */ - attr->attr = 0; - data += x; - len -= x; - } - /* Null terminate any string. - * XXX NOTE, we write past the size of the buffer passed by the - * caller, so this is potentially dangerous. The only thing that - * saves us is that usually we read the incoming message in a - * much larger buffer in the struct ast_rtp - */ - *data = '\0'; - - /* Now prepare to generate a reply, which at the moment is done - * only for properly formed (len == 0) STUN_BINDREQ messages. - */ - if (len == 0) { - unsigned char respdata[1024]; - struct stun_header *resp = (struct stun_header *)respdata; - int resplen = 0; /* len excluding header */ - int respleft = sizeof(respdata) - sizeof(struct stun_header); - - resp->id = hdr->id; - resp->msgtype = 0; - resp->msglen = 0; - attr = (struct stun_attr *)resp->ies; - switch (ntohs(hdr->msgtype)) { - case STUN_BINDREQ: - if (stundebug) - ast_verbose("STUN Bind Request, username: %s\n", - st.username ? st.username : "<none>"); - if (st.username) - append_attr_string(&attr, STUN_USERNAME, st.username, &resplen, &respleft); - append_attr_address(&attr, STUN_MAPPED_ADDRESS, src, &resplen, &respleft); - resp->msglen = htons(resplen); - resp->msgtype = htons(STUN_BINDRESP); - stun_send(s, src, resp); - ret = STUN_ACCEPT; - break; - default: - if (stundebug) - ast_verbose("Dunno what to do with STUN message %04x (%s)\n", ntohs(hdr->msgtype), stun_msg2str(ntohs(hdr->msgtype))); - } - } - return ret; -} - -/*! \brief Extract the STUN_MAPPED_ADDRESS from the stun response. - * This is used as a callback for stun_handle_response - * when called from ast_stun_request. - */ -static int stun_get_mapped(struct stun_attr *attr, void *arg) -{ - struct stun_addr *addr = (struct stun_addr *)(attr + 1); - struct sockaddr_in *sa = (struct sockaddr_in *)arg; - - if (ntohs(attr->attr) != STUN_MAPPED_ADDRESS || ntohs(attr->len) != 8) - return 1; /* not us. */ - sa->sin_port = addr->port; - sa->sin_addr.s_addr = addr->addr; - return 0; -} - -/*! \brief Generic STUN request - * Send a generic stun request to the server specified, - * possibly waiting for a reply and filling the 'reply' field with - * the externally visible address. Note that in this case the request - * will be blocking. - * (Note, the interface may change slightly in the future). - * - * \param s the socket used to send the request - * \param dst the address of the STUN server - * \param username if non null, add the username in the request - * \param answer if non null, the function waits for a response and - * puts here the externally visible address. - * \return 0 on success, other values on error. - */ -int ast_stun_request(int s, struct sockaddr_in *dst, - const char *username, struct sockaddr_in *answer) -{ - struct stun_header *req; - unsigned char reqdata[1024]; - int reqlen, reqleft; - struct stun_attr *attr; - int res = 0; - int retry; - - req = (struct stun_header *)reqdata; - stun_req_id(req); - reqlen = 0; - reqleft = sizeof(reqdata) - sizeof(struct stun_header); - req->msgtype = 0; - req->msglen = 0; - attr = (struct stun_attr *)req->ies; - if (username) - append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft); - req->msglen = htons(reqlen); - req->msgtype = htons(STUN_BINDREQ); - for (retry = 0; retry < 3; retry++) { /* XXX make retries configurable */ - /* send request, possibly wait for reply */ - unsigned char reply_buf[1024]; - fd_set rfds; - struct timeval to = { 3, 0 }; /* timeout, make it configurable */ - struct sockaddr_in src; - socklen_t srclen; - - res = stun_send(s, dst, req); - if (res < 0) { - ast_log(LOG_WARNING, "ast_stun_request send #%d failed error %d, retry\n", - retry, res); - continue; - } - if (answer == NULL) - break; - FD_ZERO(&rfds); - FD_SET(s, &rfds); - res = ast_select(s + 1, &rfds, NULL, NULL, &to); - if (res <= 0) /* timeout or error */ - continue; - bzero(&src, sizeof(src)); - srclen = sizeof(src); - /* XXX pass -1 in the size, because stun_handle_packet might - * write past the end of the buffer. - */ - res = recvfrom(s, reply_buf, sizeof(reply_buf) - 1, - 0, (struct sockaddr *)&src, &srclen); - if (res < 0) { - ast_log(LOG_WARNING, "ast_stun_request recvfrom #%d failed error %d, retry\n", - retry, res); - continue; - } - bzero(answer, sizeof(struct sockaddr_in)); - stun_handle_packet(s, &src, reply_buf, res, - stun_get_mapped, answer); - res = 0; /* signal regular exit */ - break; - } - return res; -} - -/*! \brief send a STUN BIND request to the given destination. - * Optionally, add a username if specified. - */ -void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username) -{ - ast_stun_request(rtp->s, suggestion, username, NULL); -} - -/*! \brief List of current sessions */ -static AST_RWLIST_HEAD_STATIC(protos, ast_rtp_protocol); - -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; -} - -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; -} - -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 Put RTP timeout timers on hold during another transaction, like T.38 */ -void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp) -{ - rtp->rtptimeout = (-1) * rtp->rtptimeout; - rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout; -} - -/*! \brief Set rtp timeout */ -void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout) -{ - rtp->rtptimeout = timeout; -} - -/*! \brief Set rtp hold timeout */ -void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout) -{ - rtp->rtpholdtimeout = timeout; -} - -/*! \brief set RTP keepalive interval */ -void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period) -{ - rtp->rtpkeepalive = period; -} - -/*! \brief Get rtp timeout */ -int ast_rtp_get_rtptimeout(struct ast_rtp *rtp) -{ - if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ - return 0; - return rtp->rtptimeout; -} - -/*! \brief Get rtp hold timeout */ -int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp) -{ - if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */ - return 0; - return rtp->rtpholdtimeout; -} - -/*! \brief Get RTP keepalive interval */ -int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp) -{ - return rtp->rtpkeepalive; -} - -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; -} - -int ast_rtp_getnat(struct ast_rtp *rtp) -{ - return ast_test_flag(rtp, FLAG_NAT_ACTIVE); -} - -void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf) -{ - ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF); -} - -void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate) -{ - ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE); -} - -void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable) -{ - ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN); -} - -static void rtp_bridge_lock(struct ast_rtp *rtp) -{ -#ifdef P2P_INTENSE - ast_mutex_lock(&rtp->bridge_lock); -#endif - return; -} - -static void rtp_bridge_unlock(struct ast_rtp *rtp) -{ -#ifdef P2P_INTENSE - ast_mutex_unlock(&rtp->bridge_lock); -#endif - return; -} - -static struct ast_frame *send_dtmf(struct ast_rtp *rtp, enum ast_frame_type type) -{ - if (((ast_test_flag(rtp, FLAG_DTMF_COMPENSATE) && type == AST_FRAME_DTMF_END) || - (type == AST_FRAME_DTMF_BEGIN)) && ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) { - ast_debug(1, "Ignore potential DTMF echo from '%s'\n", ast_inet_ntoa(rtp->them.sin_addr)); - rtp->resp = 0; - rtp->dtmfsamples = 0; - return &ast_null_frame; - } - ast_debug(1, "Sending dtmf: %d (%c), at %s\n", rtp->resp, rtp->resp, ast_inet_ntoa(rtp->them.sin_addr)); - if (rtp->resp == 'X') { - rtp->f.frametype = AST_FRAME_CONTROL; - rtp->f.subclass = AST_CONTROL_FLASH; - } else { - rtp->f.frametype = type; - rtp->f.subclass = rtp->resp; - } - rtp->f.datalen = 0; - rtp->f.samples = 0; - rtp->f.mallocd = 0; - rtp->f.src = "RTP"; - 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 inline int rtcp_debug_test_addr(struct sockaddr_in *addr) -{ - if (rtcpdebug == 0) - 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 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; - unsigned char seq; - unsigned int flags; - unsigned int power; - - /* We should have at least 4 bytes in RTP data */ - if (len < 4) - return f; - - /* The format of Cisco RTP DTMF packet looks like next: - +0 - sequence number of DTMF RTP packet (begins from 1, - wrapped to 0) - +1 - set of flags - +1 (bit 0) - flaps by different DTMF digits delimited by audio - or repeated digit without audio??? - +2 (+4,+6,...) - power level? (rises from 0 to 32 at begin of tone - then falls to 0 at its end) - +3 (+5,+7,...) - detected DTMF digit (0..9,*,#,A-D,...) - Repeated DTMF information (bytes 4/5, 6/7) is history shifted right - by each new packet and thus provides some redudancy. - - Sample of Cisco RTP DTMF packet is (all data in hex): - 19 07 00 02 12 02 20 02 - showing end of DTMF digit '2'. - - The packets - 27 07 00 02 0A 02 20 02 - 28 06 20 02 00 02 0A 02 - shows begin of new digit '2' with very short pause (20 ms) after - previous digit '2'. Bit +1.0 flips at begin of new digit. - - Cisco RTP DTMF packets comes as replacement of audio RTP packets - so its uses the same sequencing and timestamping rules as replaced - audio packets. Repeat interval of DTMF packets is 20 ms and not rely - on audio framing parameters. Marker bit isn't used within stream of - DTMFs nor audio stream coming immediately after DTMF stream. Timestamps - are not sequential at borders between DTMF and audio streams, - */ - - seq = data[0]; - flags = data[1]; - power = data[2]; - event = data[3] & 0x1f; - - if (option_debug > 2 || rtpdebug) - ast_debug(0, "Cisco DTMF Digit: %02x (len=%d, seq=%d, flags=%02x, power=%d, history count=%d)\n", event, len, seq, flags, power, (len - 4) / 2); - 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 && power) || (rtp->resp && (rtp->resp != resp))) { - rtp->resp = resp; - /* Why we should care on DTMF compensation at reception? */ - if (!ast_test_flag(rtp, FLAG_DTMF_COMPENSATE)) { - f = send_dtmf(rtp, AST_FRAME_DTMF_BEGIN); - rtp->dtmfsamples = 0; - } - } else if ((rtp->resp == resp) && !power) { - f = send_dtmf(rtp, AST_FRAME_DTMF_END); - f->samples = rtp->dtmfsamples * 8; - rtp->resp = 0; - } else if (rtp->resp == resp) - rtp->dtmfsamples += 20 * 8; - 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 - * \param timestamp - * \returns - */ -static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp) -{ - unsigned int event; - unsigned int event_end; - unsigned int samples; - char resp = 0; - struct ast_frame *f = NULL; - - /* Figure out event, event end, and samples */ - event = ntohl(*((unsigned int *)(data))); - event >>= 24; - event_end = ntohl(*((unsigned int *)(data))); - event_end <<= 8; - event_end >>= 24; - samples = ntohl(*((unsigned int *)(data))); - samples &= 0xFFFF; - - /* Print out debug if turned on */ - if (rtpdebug || option_debug > 2) - ast_debug(0, "- RTP 2833 Event: %08x (len = %d)\n", event, len); - - /* Figure out what digit was pressed */ - 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'; - } else { - /* Not a supported event */ - ast_log(LOG_DEBUG, "Ignoring RTP 2833 Event: %08x. Not a DTMF Digit.\n", event); - return &ast_null_frame; - } - - if (ast_test_flag(rtp, FLAG_DTMF_COMPENSATE)) { - if ((rtp->lastevent != timestamp) || (rtp->resp && rtp->resp != resp)) { - rtp->resp = resp; - f = send_dtmf(rtp, AST_FRAME_DTMF_END); - f->len = 0; - rtp->lastevent = timestamp; - } - } else { - if ((!(rtp->resp) && (!(event_end & 0x80))) || (rtp->resp && rtp->resp != resp)) { - rtp->resp = resp; - f = send_dtmf(rtp, AST_FRAME_DTMF_BEGIN); - } else if ((event_end & 0x80) && (rtp->lastevent != seqno) && rtp->resp) { - f = send_dtmf(rtp, AST_FRAME_DTMF_END); - f->len = ast_tvdiff_ms(ast_samp2tv(samples, 8000), ast_tv(0, 0)); /* XXX hard coded 8kHz */ - rtp->resp = 0; - rtp->lastevent = seqno; - } - } - - rtp->dtmfcount = dtmftimeout; - rtp->dtmfsamples = samples; - - 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_debug(0, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len); - - if (!(ast_test_flag(rtp, FLAG_3389_WARNING))) { - 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(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) -{ - socklen_t len; - int position, i, packetwords; - int res; - struct sockaddr_in sin; - unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET]; - unsigned int *rtcpheader; - int pt; - struct timeval now; - unsigned int length; - int rc; - double rttsec; - uint64_t rtt = 0; - unsigned int dlsr; - unsigned int lsr; - unsigned int msw; - unsigned int lsw; - unsigned int comp; - struct ast_frame *f = &ast_null_frame; - - if (!rtp || !rtp->rtcp) - return &ast_null_frame; - - len = sizeof(sin); - - res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET, - 0, (struct sockaddr *)&sin, &len); - rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET); - - if (res < 0) { - if (errno == EBADF) - CRASH; - if (errno != EAGAIN) { - ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno)); - return NULL; - } - return &ast_null_frame; - } - - packetwords = res / 4; - - 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_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); - } - } - - ast_debug(1, "Got RTCP report of %d bytes\n", res); - - /* Process a compound packet */ - position = 0; - while (position < packetwords) { - i = position; - length = ntohl(rtcpheader[i]); - pt = (length & 0xff0000) >> 16; - rc = (length & 0x1f000000) >> 24; - length &= 0xffff; - - if ((i + length) > packetwords) { - if (option_debug || rtpdebug) - ast_log(LOG_DEBUG, "RTCP Read too short\n"); - return &ast_null_frame; - } - - if (rtcp_debug_test_addr(&sin)) { - ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); - ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown"); - ast_verbose("Reception reports: %d\n", rc); - ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]); - } - - i += 2; /* Advance past header and ssrc */ - - switch (pt) { - case RTCP_PT_SR: - gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */ - rtp->rtcp->spc = ntohl(rtcpheader[i+3]); - rtp->rtcp->soc = ntohl(rtcpheader[i + 4]); - rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/ - - if (rtcp_debug_test_addr(&sin)) { - ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096); - ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2])); - ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4])); - } - i += 5; - if (rc < 1) - break; - /* Intentional fall through */ - case RTCP_PT_RR: - /* Don't handle multiple reception reports (rc > 1) yet */ - /* Calculate RTT per RFC */ - gettimeofday(&now, NULL); - timeval2ntp(now, &msw, &lsw); - if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */ - comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16); - lsr = ntohl(rtcpheader[i + 4]); - dlsr = ntohl(rtcpheader[i + 5]); - rtt = comp - lsr - dlsr; - - /* Convert end to end delay to usec (keeping the calculation in 64bit space) - sess->ee_delay = (eedelay * 1000) / 65536; */ - if (rtt < 4294) { - rtt = (rtt * 1000000) >> 16; - } else { - rtt = (rtt * 1000) >> 16; - rtt *= 1000; - } - rtt = rtt / 1000.; - rttsec = rtt / 1000.; - - if (comp - dlsr >= lsr) { - rtp->rtcp->accumulated_transit += rttsec; - rtp->rtcp->rtt = rttsec; - if (rtp->rtcp->maxrtt<rttsec) - rtp->rtcp->maxrtt = rttsec; - if (rtp->rtcp->minrtt>rttsec) - rtp->rtcp->minrtt = rttsec; - } else if (rtcp_debug_test_addr(&sin)) { - ast_verbose("Internal RTCP NTP clock skew detected: " - "lsr=%u, now=%u, dlsr=%u (%d:%03dms), " - "diff=%d\n", - lsr, comp, dlsr, dlsr / 65536, - (dlsr % 65536) * 1000 / 65536, - dlsr - (comp - lsr)); - } - } - - rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]); - rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff; - if (rtcp_debug_test_addr(&sin)) { - ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24)); - ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost); - ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff)); - ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16); - ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter); - ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096); - ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0); - if (rtt) - ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt); - } - if (rtt) { - manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From %s:%d\r\n" - "PT: %d(%s)\r\n" - "ReceptionReports: %d\r\n" - "SenderSSRC: %u\r\n" - "FractionLost: %ld\r\n" - "PacketsLost: %d\r\n" - "HighestSequence: %ld\r\n" - "SequenceNumberCycles: %ld\r\n" - "IAJitter: %u\r\n" - "LastSR: %lu.%010lu\r\n" - "DLSR: %4.4f(sec)\r\n" - "RTT: %llu(sec)\r\n", - ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), - pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown", - rc, - rtcpheader[i + 1], - (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24), - rtp->rtcp->reported_lost, - (long) (ntohl(rtcpheader[i + 2]) & 0xffff), - (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16, - rtp->rtcp->reported_jitter, - (unsigned long) ntohl(rtcpheader[i + 4]) >> 16, ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096, - ntohl(rtcpheader[i + 5])/65536.0, - (unsigned long long)rtt); - } else { - manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From %s:%d\r\n" - "PT: %d(%s)\r\n" - "ReceptionReports: %d\r\n" - "SenderSSRC: %u\r\n" - "FractionLost: %ld\r\n" - "PacketsLost: %d\r\n" - "HighestSequence: %ld\r\n" - "SequenceNumberCycles: %ld\r\n" - "IAJitter: %u\r\n" - "LastSR: %lu.%010lu\r\n" - "DLSR: %4.4f(sec)\r\n", - ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), - pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown", - rc, - rtcpheader[i + 1], - (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24), - rtp->rtcp->reported_lost, - (long) (ntohl(rtcpheader[i + 2]) & 0xffff), - (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16, - rtp->rtcp->reported_jitter, - (unsigned long) ntohl(rtcpheader[i + 4]) >> 16, - ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096, - ntohl(rtcpheader[i + 5])/65536.0); - } - break; - case RTCP_PT_FUR: - if (rtcp_debug_test_addr(&sin)) - ast_verbose("Received an RTCP Fast Update Request\n"); - rtp->f.frametype = AST_FRAME_CONTROL; - rtp->f.subclass = AST_CONTROL_VIDUPDATE; - rtp->f.datalen = 0; - rtp->f.samples = 0; - rtp->f.mallocd = 0; - rtp->f.src = "RTP"; - f = &rtp->f; - break; - case RTCP_PT_SDES: - if (rtcp_debug_test_addr(&sin)) - ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); - break; - case RTCP_PT_BYE: - if (rtcp_debug_test_addr(&sin)) - ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); - break; - default: - ast_debug(1, "Unknown RTCP packet (pt=%d) received from %s:%d\n", pt, ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port)); - break; - } - position += (length + 1); - } - - return f; -} - -static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark) -{ - struct timeval now; - double transit; - double current_time; - double d; - double dtv; - double prog; - - if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) { - gettimeofday(&rtp->rxcore, NULL); - rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000; - /* map timestamp to a real time */ - rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */ - rtp->rxcore.tv_sec -= timestamp / 8000; - rtp->rxcore.tv_usec -= (timestamp % 8000) * 125; - /* Round to 0.1ms for nice, pretty timestamps */ - rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100; - if (rtp->rxcore.tv_usec < 0) { - /* Adjust appropriately if necessary */ - rtp->rxcore.tv_usec += 1000000; - rtp->rxcore.tv_sec -= 1; - } - } - - gettimeofday(&now,NULL); - /* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */ - tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000; - tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125; - if (tv->tv_usec >= 1000000) { - tv->tv_usec -= 1000000; - tv->tv_sec += 1; - } - prog = (double)((timestamp-rtp->seedrxts)/8000.); - dtv = (double)rtp->drxcore + (double)(prog); - current_time = (double)now.tv_sec + (double)now.tv_usec/1000000; - transit = current_time - dtv; - d = transit - rtp->rxtransit; - rtp->rxtransit = transit; - if (d<0) - d=-d; - rtp->rxjitter += (1./16.) * (d - rtp->rxjitter); - if (rtp->rtcp && rtp->rxjitter > rtp->rtcp->maxrxjitter) - rtp->rtcp->maxrxjitter = rtp->rxjitter; - if (rtp->rtcp && rtp->rxjitter < rtp->rtcp->minrxjitter) - rtp->rtcp->minrxjitter = rtp->rxjitter; -} - -/*! \brief Perform a Packet2Packet RTP write */ -static int bridge_p2p_rtp_write(struct ast_rtp *rtp, struct ast_rtp *bridged, unsigned int *rtpheader, int len, int hdrlen) -{ - int res = 0, payload = 0, bridged_payload = 0, mark; - struct rtpPayloadType rtpPT; - int reconstruct = ntohl(rtpheader[0]); - - /* Get fields from packet */ - payload = (reconstruct & 0x7f0000) >> 16; - mark = (((reconstruct & 0x800000) >> 23) != 0); - - /* Check what the payload value should be */ - rtpPT = ast_rtp_lookup_pt(rtp, payload); - - /* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */ - if (!bridged->current_RTP_PT[payload].code) - return -1; - - /* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */ - if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF) - return -1; - - /* Otherwise adjust bridged payload to match */ - bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code); - - /* If the mark bit has not been sent yet... do it now */ - if (!ast_test_flag(rtp, FLAG_P2P_SENT_MARK)) { - mark = 1; - ast_set_flag(rtp, FLAG_P2P_SENT_MARK); - } - - /* Reconstruct part of the packet */ - reconstruct &= 0xFF80FFFF; - reconstruct |= (bridged_payload << 16); - reconstruct |= (mark << 23); - rtpheader[0] = htonl(reconstruct); - - /* Send the packet back out */ - res = sendto(bridged->s, (void *)rtpheader, len, 0, (struct sockaddr *)&bridged->them, sizeof(bridged->them)); - if (res < 0) { - if (!bridged->nat || (bridged->nat && (ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) { - ast_debug(1, "RTP Transmission error of packet to %s:%d: %s\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), strerror(errno)); - } else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) { - 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(bridged->them.sin_addr), ntohs(bridged->them.sin_port)); - ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN); - } - return 0; - } else if (rtp_debug_test_addr(&bridged->them)) - ast_verbose("Sent RTP P2P packet to %s:%u (type %-2.2d, len %-6.6u)\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), bridged_payload, len - hdrlen); - - return 0; -} - -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 cc; - unsigned int ssrc; - unsigned int timestamp; - unsigned int *rtpheader; - struct rtpPayloadType rtpPT; - struct ast_rtp *bridged = NULL; - - /* If time is up, kill it */ - if (rtp->sending_digit) - ast_rtp_senddigit_continuation(rtp); - - 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); - - /* If strict RTP protection is enabled see if we need to learn this address or if the packet should be dropped */ - if (rtp->strict_rtp_state == STRICT_RTP_LEARN) { - /* Copy over address that this packet was received on */ - memcpy(&rtp->strict_rtp_address, &sin, sizeof(rtp->strict_rtp_address)); - /* Now move over to actually protecting the RTP port */ - rtp->strict_rtp_state = STRICT_RTP_CLOSED; - ast_debug(1, "Learned remote address is %s:%d for strict RTP purposes, now protecting the port.\n", ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port)); - } else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) { - /* If the address we previously learned doesn't match the address this packet came in on simply drop it */ - if ((rtp->strict_rtp_address.sin_addr.s_addr != sin.sin_addr.s_addr) || (rtp->strict_rtp_address.sin_port != sin.sin_port)) { - ast_debug(1, "Received RTP packet from %s:%d, dropping due to strict RTP protection. Expected it to be from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port)); - return &ast_null_frame; - } - } - - rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); - if (res < 0) { - if (errno == EBADF) - CRASH; - if (errno != EAGAIN) { - ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno)); - return NULL; - } - return &ast_null_frame; - } - - if (res < hdrlen) { - ast_log(LOG_WARNING, "RTP Read too short\n"); - return &ast_null_frame; - } - - /* Get fields */ - seqno = ntohl(rtpheader[0]); - - /* Check RTP version */ - version = (seqno & 0xC0000000) >> 30; - if (!version) { - /* If the two high bits are 0, this might be a - * STUN message, so process it. stun_handle_packet() - * answers to requests, and it returns STUN_ACCEPT - * if the request is valid. - */ - if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == STUN_ACCEPT) && - (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) { - memcpy(&rtp->them, &sin, sizeof(rtp->them)); - } - return &ast_null_frame; - } - -#if 0 /* Allow to receive RTP stream with closed transmission path */ - /* If we don't have the other side's address, then ignore this */ - if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) - return &ast_null_frame; -#endif - - /* Send to whoever send to us if NAT is turned on */ - if (rtp->nat) { - if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || - (rtp->them.sin_port != sin.sin_port)) { - rtp->them = sin; - if (rtp->rtcp) { - memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them)); - rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1); - } - rtp->rxseqno = 0; - ast_set_flag(rtp, FLAG_NAT_ACTIVE); - if (option_debug || rtpdebug) - ast_debug(0, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port)); - } - } - - /* If we are bridged to another RTP stream, send direct */ - if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen)) - return &ast_null_frame; - - if (version != 2) - return &ast_null_frame; - - payloadtype = (seqno & 0x7f0000) >> 16; - padding = seqno & (1 << 29); - mark = seqno & (1 << 23); - ext = seqno & (1 << 28); - cc = (seqno & 0xF000000) >> 24; - seqno &= 0xffff; - timestamp = ntohl(rtpheader[1]); - ssrc = ntohl(rtpheader[2]); - - if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) { - if (option_debug || rtpdebug) - ast_debug(0, "Forcing Marker bit, because SSRC has changed\n"); - mark = 1; - } - - rtp->rxssrc = ssrc; - - if (padding) { - /* Remove padding bytes */ - res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1]; - } - - if (cc) { - /* CSRC fields present */ - hdrlen += cc*4; - } - - if (ext) { - /* RTP Extension present */ - hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2; - hdrlen += 4; - if (option_debug) { - int profile; - profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16; - if (profile == 0x505a) - ast_debug(1, "Found Zfone extension in RTP stream - zrtp - not supported.\n"); - else - ast_debug(1, "Found unknown RTP Extensions %x\n", profile); - } - } - - if (res < hdrlen) { - ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen); - return &ast_null_frame; - } - - rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */ - - if (rtp->rxcount==1) { - /* This is the first RTP packet successfully received from source */ - rtp->seedrxseqno = seqno; - } - - /* Do not schedule RR if RTCP isn't run */ - if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) { - /* Schedule transmission of Receiver Report */ - rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); - } - if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */ - rtp->cycles += RTP_SEQ_MOD; - - rtp->lastrxseqno = seqno; - - if (rtp->themssrc==0) - rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */ - - if (rtp_debug_test_addr(&sin)) - ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", - ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen); - - rtpPT = ast_rtp_lookup_pt(rtp, payloadtype); - if (!rtpPT.isAstFormat) { - struct ast_frame *f = NULL; - - /* 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 RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration); - } - f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp); - } else if (rtpPT.code == AST_RTP_CISCO_DTMF) { - /* It's really special -- process it the Cisco way */ - if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) { - f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); - rtp->lastevent = seqno; - } - } else if (rtpPT.code == AST_RTP_CN) { - /* Comfort Noise */ - f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); - } else { - ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr)); - } - return f ? f : &ast_null_frame; - } - rtp->lastrxformat = rtp->f.subclass = rtpPT.code; - rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT; - - if (!rtp->lastrxts) - rtp->lastrxts = timestamp; - - rtp->rxseqno = seqno; - - /* Record received timestamp as last received now */ - rtp->lastrxts = timestamp; - - 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; - rtp->f.seqno = seqno; - if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) { - 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); - /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */ - ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO); - rtp->f.ts = timestamp / 8; - rtp->f.len = rtp->f.samples / 8; - } else if(rtp->f.subclass & AST_FORMAT_VIDEO_MASK) { - /* 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; - /* Pass the RTP marker bit as bit 0 in the subclass field. - * This is ok because subclass is actually a bitmask, and - * the low bits represent audio formats, that are not - * involved here since we deal with video. - */ - if (mark) - rtp->f.subclass |= 0x1; - } else { - /* TEXT -- samples is # of samples vs. 1000 */ - if (!rtp->lastitexttimestamp) - rtp->lastitexttimestamp = timestamp; - rtp->f.samples = timestamp - rtp->lastitexttimestamp; - rtp->lastitexttimestamp = timestamp; - rtp->f.delivery.tv_sec = 0; - rtp->f.delivery.tv_usec = 0; - } - 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_ULAW}, "audio", "G711U"}, - {{1, AST_FORMAT_ALAW}, "audio", "PCMA"}, - {{1, AST_FORMAT_ALAW}, "audio", "G711A"}, - {{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_G729A}, "audio", "G729A"}, - {{1, AST_FORMAT_SPEEX}, "audio", "speex"}, - {{1, AST_FORMAT_ILBC}, "audio", "iLBC"}, - {{1, AST_FORMAT_G722}, "audio", "G722"}, - {{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32"}, - {{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"}, - {{1, AST_FORMAT_H264}, "video", "H264"}, - {{1, AST_FORMAT_MP4_VIDEO}, "video", "MP4V-ES"}, - {{1, AST_FORMAT_T140}, "text", "T140"}, -}; - -/*! - * \brief Mapping between Asterisk codecs and rtp payload types - * - * 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 - * - * See http://www.iana.org/assignments/rtp-parameters for a list of - * assigned values - */ -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}, - [9] = {1, AST_FORMAT_G722}, - [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}, - [97] = {1, AST_FORMAT_ILBC}, - [98] = {1, AST_FORMAT_H263_PLUS}, - [99] = {1, AST_FORMAT_H264}, - [101] = {0, AST_RTP_DTMF}, - [102] = {1, AST_FORMAT_T140}, /* Real time text chat */ - [103] = {1, AST_FORMAT_H263_PLUS}, - [104] = {1, AST_FORMAT_MP4_VIDEO}, - [110] = {1, AST_FORMAT_SPEEX}, - [111] = {1, AST_FORMAT_G726}, - [112] = {1, AST_FORMAT_G726_AAL2}, - [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */ -}; - -void ast_rtp_pt_clear(struct ast_rtp* rtp) -{ - int i; - - if (!rtp) - return; - - rtp_bridge_lock(rtp); - - 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; - - rtp_bridge_unlock(rtp); -} - -void ast_rtp_pt_default(struct ast_rtp* rtp) -{ - int i; - - rtp_bridge_lock(rtp); - - /* 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; - - rtp_bridge_unlock(rtp); -} - -void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src) -{ - unsigned int i; - - rtp_bridge_lock(dest); - rtp_bridge_lock(src); - - for (i=0; i < MAX_RTP_PT; ++i) { - dest->current_RTP_PT[i].isAstFormat = - src->current_RTP_PT[i].isAstFormat; - dest->current_RTP_PT[i].code = - src->current_RTP_PT[i].code; - } - dest->rtp_lookup_code_cache_isAstFormat = 0; - dest->rtp_lookup_code_cache_code = 0; - dest->rtp_lookup_code_cache_result = 0; - - rtp_bridge_unlock(src); - rtp_bridge_unlock(dest); -} - -/*! \brief Get channel driver interface structure */ -static struct ast_rtp_protocol *get_proto(struct ast_channel *chan) -{ - struct ast_rtp_protocol *cur = NULL; - - AST_RWLIST_RDLOCK(&protos); - AST_RWLIST_TRAVERSE(&protos, cur, list) { - if (cur->type == chan->tech->type) - break; - } - AST_RWLIST_UNLOCK(&protos); - - return cur; -} - -int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1) -{ - // dest = c0, src = c1 - struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ - struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ - struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */ - struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; - enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED; - enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED; - int srccodec, destcodec, nat_active = 0; - - /* Lock channels */ - ast_channel_lock(c0); - if (c1) { - while (ast_channel_trylock(c1)) { - ast_channel_unlock(c0); - usleep(1); - ast_channel_lock(c0); - } - } - - /* Find channel driver interfaces */ - destpr = get_proto(c0); - if (c1) - srcpr = get_proto(c1); - if (!destpr) { - ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c0->name); - ast_channel_unlock(c0); - if (c1) - ast_channel_unlock(c1); - return -1; - } - if (!srcpr) { - ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c1 ? c1->name : "<unspecified>"); - ast_channel_unlock(c0); - if (c1) - ast_channel_unlock(c1); - return -1; - } - - /* Get audio, video and text interface (if native bridge is possible) */ - audio_dest_res = destpr->get_rtp_info(c0, &destp); - video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(c0, &vdestp) : AST_RTP_GET_FAILED; - text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(c0, &tdestp) : AST_RTP_GET_FAILED; - if (srcpr) { - audio_src_res = srcpr->get_rtp_info(c1, &srcp); - video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(c1, &vsrcp) : AST_RTP_GET_FAILED; - text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED; - } - - /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ - if (audio_dest_res != AST_RTP_TRY_NATIVE) { - /* Somebody doesn't want to play... */ - ast_channel_unlock(c0); - if (c1) - ast_channel_unlock(c1); - return -1; - } - if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec) - srccodec = srcpr->get_codec(c1); - else - srccodec = 0; - if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec) - destcodec = destpr->get_codec(c0); - else - destcodec = 0; - /* Ensure we have at least one matching codec */ - if (!(srccodec & destcodec)) { - ast_channel_unlock(c0); - if (c1) - ast_channel_unlock(c1); - return 0; - } - /* Consider empty media as non-existant */ - if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) - srcp = NULL; - if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE))) - nat_active = 1; - /* Bridge media early */ - if (destpr->set_rtp_peer(c0, srcp, vsrcp, tsrcp, srccodec, nat_active)) - ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); - ast_channel_unlock(c0); - if (c1) - ast_channel_unlock(c1); - ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); - return 0; -} - -int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media) -{ - struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */ - struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */ - struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */ - struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; - enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED; - enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED; - int srccodec, destcodec; - - /* Lock channels */ - ast_channel_lock(dest); - while (ast_channel_trylock(src)) { - ast_channel_unlock(dest); - usleep(1); - ast_channel_lock(dest); - } - - /* Find channel driver interfaces */ - if (!(destpr = get_proto(dest))) { - ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", dest->name); - ast_channel_unlock(dest); - ast_channel_unlock(src); - return 0; - } - if (!(srcpr = get_proto(src))) { - ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", src->name); - ast_channel_unlock(dest); - ast_channel_unlock(src); - return 0; - } - - /* Get audio and video interface (if native bridge is possible) */ - audio_dest_res = destpr->get_rtp_info(dest, &destp); - video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED; - text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(dest, &tdestp) : AST_RTP_GET_FAILED; - audio_src_res = srcpr->get_rtp_info(src, &srcp); - video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; - text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(src, &tsrcp) : AST_RTP_GET_FAILED; - - /* Ensure we have at least one matching codec */ - if (srcpr->get_codec) - srccodec = srcpr->get_codec(src); - else - srccodec = 0; - if (destpr->get_codec) - destcodec = destpr->get_codec(dest); - else - destcodec = 0; - - /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ - if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) { - /* Somebody doesn't want to play... */ - ast_channel_unlock(dest); - ast_channel_unlock(src); - return 0; - } - ast_rtp_pt_copy(destp, srcp); - if (vdestp && vsrcp) - ast_rtp_pt_copy(vdestp, vsrcp); - if (tdestp && tsrcp) - ast_rtp_pt_copy(tdestp, tsrcp); - if (media) { - /* Bridge early */ - if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) - ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name); - } - ast_channel_unlock(dest); - ast_channel_unlock(src); - ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name); - return 1; -} - -/*! \brief Make a note of a RTP payload 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 || static_RTP_PT[pt].code == 0) - return; /* bogus payload type */ - - rtp_bridge_lock(rtp); - rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; - rtp_bridge_unlock(rtp); -} - -/*! \brief remove setting from payload type list if the rtpmap header indicates - an unknown media type */ -void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt) -{ - rtp_bridge_lock(rtp); - rtp->current_RTP_PT[pt].isAstFormat = 0; - rtp->current_RTP_PT[pt].code = 0; - rtp_bridge_unlock(rtp); -} - -/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in - * an SDP "a=rtpmap:" line. - * \return 0 if the MIME type was found and set, -1 if it wasn't found - */ -int ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt, - char *mimeType, char *mimeSubtype, - enum ast_rtp_options options) -{ - unsigned int i; - int found = 0; - - if (pt < 0 || pt > MAX_RTP_PT) - return -1; /* bogus payload type */ - - rtp_bridge_lock(rtp); - - for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { - if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 && - strcasecmp(mimeType, mimeTypes[i].type) == 0) { - found = 1; - rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType; - if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) && - mimeTypes[i].payloadType.isAstFormat && - (options & AST_RTP_OPT_G726_NONSTANDARD)) - rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2; - break; - } - } - - rtp_bridge_unlock(rtp); - - return (found ? 0 : -1); -} - -/*! \brief 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; - - rtp_bridge_lock(rtp); - - *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; - } - } - - rtp_bridge_unlock(rtp); - - return; -} - -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 negotiated codecs */ - rtp_bridge_lock(rtp); - result = rtp->current_RTP_PT[pt]; - rtp_bridge_unlock(rtp); - - /* If it doesn't exist, check our static RTP type list, just in case */ - if (!result.code) - result = static_RTP_PT[pt]; - - return result; -} - -/*! \brief 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 = 0; - - rtp_bridge_lock(rtp); - - 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 */ - pt = rtp->rtp_lookup_code_cache_result; - rtp_bridge_unlock(rtp); - return pt; - } - - /* 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; - rtp_bridge_unlock(rtp); - 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; - rtp_bridge_unlock(rtp); - return pt; - } - } - - rtp_bridge_unlock(rtp); - - return -1; -} - -const char *ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code, - enum ast_rtp_options options) -{ - unsigned int i; - - for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) { - if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) { - if (isAstFormat && - (code == AST_FORMAT_G726_AAL2) && - (options & AST_RTP_OPT_G726_NONSTANDARD)) - return "G726-32"; - else - return mimeTypes[i].subtype; - } - } - - return ""; -} - -char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability, - const int isAstFormat, enum ast_rtp_options options) -{ - 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, options); - - snprintf(end, size, "%s|", name); - len = strlen(end); - end += len; - size -= len; - } - } - - if (start == end) - ast_copy_string(start, "nothing)", size); - else if (size > 1) - *(end -1) = ')'; - - return buf; -} - -/*! \brief Open RTP or RTCP socket for a session. - * Print a message on failure. - */ -static int rtp_socket(const char *type) -{ - int s = socket(AF_INET, SOCK_DGRAM, 0); - if (s < 0) { - if (type == NULL) - type = "RTP/RTCP"; - ast_log(LOG_WARNING, "Unable to allocate %s socket: %s\n", type, strerror(errno)); - } else { - long 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; - - if (!(rtcp = ast_calloc(1, sizeof(*rtcp)))) - return NULL; - rtcp->s = rtp_socket("RTCP"); - rtcp->us.sin_family = AF_INET; - rtcp->them.sin_family = AF_INET; - - if (rtcp->s < 0) { - ast_free(rtcp); - return NULL; - } - - return rtcp; -} - -/*! - * \brief Initialize a new RTP structure. - * - */ -void ast_rtp_new_init(struct ast_rtp *rtp) -{ -#ifdef P2P_INTENSE - ast_mutex_init(&rtp->bridge_lock); -#endif - - rtp->them.sin_family = AF_INET; - rtp->us.sin_family = AF_INET; - rtp->ssrc = ast_random(); - rtp->seqno = ast_random() & 0xffff; - ast_set_flag(rtp, FLAG_HAS_DTMF); - rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN); - - return; -} - -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 startplace; - - if (!(rtp = ast_calloc(1, sizeof(*rtp)))) - return NULL; - - ast_rtp_new_init(rtp); - - rtp->s = rtp_socket("RTP"); - if (rtp->s < 0) - goto fail; - if (sched && rtcpenable) { - rtp->sched = sched; - rtp->rtcp = ast_rtcp_new(); - } - - /* - * Try to bind the RTP port, x, and possibly the RTCP port, x+1 as well. - * Start from a random (even, by RTP spec) port number, and - * iterate until success or no ports are available. - * Note that the requirement of RTP port being even, or RTCP being the - * next one, cannot be enforced in presence of a NAT box because the - * mapping is not under our control. - */ - x = (ast_random() % (rtpend-rtpstart)) + rtpstart; - x = x & ~1; /* make it an even number */ - startplace = x; /* remember the starting point */ - /* this is constant across the loop */ - rtp->us.sin_addr = addr; - if (rtp->rtcp) - rtp->rtcp->us.sin_addr = addr; - for (;;) { - rtp->us.sin_port = htons(x); - if (!bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) { - /* bind succeeded, if no rtcp then we are done */ - if (!rtp->rtcp) - break; - /* have rtcp, try to bind it */ - rtp->rtcp->us.sin_port = htons(x + 1); - if (!bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))) - break; /* success again, we are really done */ - /* - * RTCP bind failed, so close and recreate the - * already bound RTP socket for the next round. - */ - close(rtp->s); - rtp->s = rtp_socket("RTP"); - if (rtp->s < 0) - goto fail; - } - /* - * If we get here, there was an error in one of the bind() - * calls, so make sure it is nothing unexpected. - */ - if (errno != EADDRINUSE) { - /* We got an error that wasn't expected, abort! */ - ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno)); - goto fail; - } - /* - * One of the ports is in use. For the next iteration, - * increment by two and handle wraparound. - * If we reach the starting point, then declare failure. - */ - x += 2; - if (x > rtpend) - x = (rtpstart + 1) & ~1; - if (x == startplace) { - ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n"); - goto fail; - } - } - rtp->sched = sched; - rtp->io = io; - if (callbackmode) { - rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp); - ast_set_flag(rtp, FLAG_CALLBACK_MODE); - } - ast_rtp_pt_default(rtp); - return rtp; - -fail: - if (rtp->s >= 0) - close(rtp->s); - if (rtp->rtcp) { - close(rtp->rtcp->s); - ast_free(rtp->rtcp); - } - ast_free(rtp); - return NULL; -} - -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_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc) -{ - return ast_netsock_set_qos(rtp->s, tos, cos, desc); -} - -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; - /* If strict RTP protection is enabled switch back to the learn state so we don't drop packets from above */ - if (strictrtp) - rtp->strict_rtp_state = STRICT_RTP_LEARN; -} - -int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them) -{ - if ((them->sin_family != AF_INET) || - (them->sin_port != rtp->them.sin_port) || - (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) { - them->sin_family = AF_INET; - them->sin_port = rtp->them.sin_port; - them->sin_addr = rtp->them.sin_addr; - return 1; - } - return 0; -} - -void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us) -{ - *us = rtp->us; -} - -struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp) -{ - struct ast_rtp *bridged = NULL; - - rtp_bridge_lock(rtp); - bridged = rtp->bridged; - rtp_bridge_unlock(rtp); - - return bridged; -} - -void ast_rtp_stop(struct ast_rtp *rtp) -{ - AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); - - 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->rtcp->them.sin_addr)); - memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port)); - } - - ast_clear_flag(rtp, FLAG_P2P_SENT_MARK); -} - -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->lastitexttimestamp = 0; - rtp->lastotexttimestamp = 0; - rtp->lasteventseqn = 0; - rtp->lastevent = 0; - rtp->lasttxformat = 0; - rtp->lastrxformat = 0; - rtp->dtmfcount = 0; - rtp->dtmfsamples = 0; - rtp->seqno = 0; - rtp->rxseqno = 0; -} - -char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual) -{ - /* - *ssrc our ssrc - *themssrc their ssrc - *lp lost packets - *rxjitter our calculated jitter(rx) - *rxcount no. received packets - *txjitter reported jitter of the other end - *txcount transmitted packets - *rlp remote lost packets - *rtt round trip time - */ - - if (qual && rtp) { - qual->local_ssrc = rtp->ssrc; - qual->local_jitter = rtp->rxjitter; - qual->local_count = rtp->rxcount; - qual->remote_ssrc = rtp->themssrc; - qual->remote_count = rtp->txcount; - if (rtp->rtcp) { - qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior; - qual->remote_lostpackets = rtp->rtcp->reported_lost; - qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0; - qual->rtt = rtp->rtcp->rtt; - } - } - if (rtp->rtcp) { - snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), - "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", - rtp->ssrc, - rtp->themssrc, - rtp->rtcp->expected_prior - rtp->rtcp->received_prior, - rtp->rxjitter, - rtp->rxcount, - (double)rtp->rtcp->reported_jitter / 65536.0, - rtp->txcount, - rtp->rtcp->reported_lost, - rtp->rtcp->rtt); - return rtp->rtcp->quality; - } else - return "<Unknown> - RTP/RTCP has already been destroyed"; -} - -void ast_rtp_destroy(struct ast_rtp *rtp) -{ - if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) { - /*Print some info on the call here */ - ast_verbose(" RTP-stats\n"); - ast_verbose("* Our Receiver:\n"); - ast_verbose(" SSRC: %u\n", rtp->themssrc); - ast_verbose(" Received packets: %u\n", rtp->rxcount); - ast_verbose(" Lost packets: %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior); - ast_verbose(" Jitter: %.4f\n", rtp->rxjitter); - ast_verbose(" Transit: %.4f\n", rtp->rxtransit); - ast_verbose(" RR-count: %u\n", rtp->rtcp->rr_count); - ast_verbose("* Our Sender:\n"); - ast_verbose(" SSRC: %u\n", rtp->ssrc); - ast_verbose(" Sent packets: %u\n", rtp->txcount); - ast_verbose(" Lost packets: %u\n", rtp->rtcp->reported_lost); - ast_verbose(" Jitter: %u\n", rtp->rtcp->reported_jitter / (unsigned int)65536.0); - ast_verbose(" SR-count: %u\n", rtp->rtcp->sr_count); - ast_verbose(" RTT: %f\n", rtp->rtcp->rtt); - } - - manager_event(EVENT_FLAG_REPORTING, "RTPReceiverStat", "SSRC: %u\r\n" - "ReceivedPackets: %u\r\n" - "LostPackets: %u\r\n" - "Jitter: %.4f\r\n" - "Transit: %.4f\r\n" - "RRCount: %u\r\n", - rtp->themssrc, - rtp->rxcount, - rtp->rtcp->expected_prior - rtp->rtcp->received_prior, - rtp->rxjitter, - rtp->rxtransit, - rtp->rtcp->rr_count); - manager_event(EVENT_FLAG_REPORTING, "RTPSenderStat", "SSRC: %u\r\n" - "SentPackets: %u\r\n" - "LostPackets: %u\r\n" - "Jitter: %u\r\n" - "SRCount: %u\r\n" - "RTT: %f\r\n", - rtp->ssrc, - rtp->txcount, - rtp->rtcp->reported_lost, - rtp->rtcp->reported_jitter, - rtp->rtcp->sr_count, - rtp->rtcp->rtt); - 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) { - AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid); - close(rtp->rtcp->s); - ast_free(rtp->rtcp); - rtp->rtcp=NULL; - } -#ifdef P2P_INTENSE - ast_mutex_destroy(&rtp->bridge_lock); -#endif - ast_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; -} - -/*! \brief Send begin frames for DTMF */ -int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit) -{ - unsigned int *rtpheader; - int hdrlen = 12, res = 0, i = 0, payload = 0; - char data[256]; - - 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 0; - } - - /* If we have no peer, return immediately */ - if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) - return 0; - - payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF); - - rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); - rtp->send_duration = 160; - - /* 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); - - 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 *) &rtp->them, sizeof(rtp->them)); - if (res < 0) - ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n", - ast_inet_ntoa(rtp->them.sin_addr), - ntohs(rtp->them.sin_port), strerror(errno)); - if (rtp_debug_test_addr(&rtp->them)) - 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(rtp->them.sin_addr), - ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); - /* Increment sequence number */ - rtp->seqno++; - /* Increment duration */ - rtp->send_duration += 160; - /* Clear marker bit and set seqno */ - rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno)); - } - - /* Since we received a begin, we can safely store the digit and disable any compensation */ - rtp->sending_digit = 1; - rtp->send_digit = digit; - rtp->send_payload = payload; - - return 0; -} - -/*! \brief Send continuation frame for DTMF */ -static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp) -{ - unsigned int *rtpheader; - int hdrlen = 12, res = 0; - char data[256]; - - if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) - return 0; - - /* Setup packet to send */ - rtpheader = (unsigned int *)data; - 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)); - - /* Transmit */ - 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(rtp->them.sin_addr), - ntohs(rtp->them.sin_port), strerror(errno)); - if (rtp_debug_test_addr(&rtp->them)) - 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(rtp->them.sin_addr), - ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); - - /* Increment sequence number */ - rtp->seqno++; - /* Increment duration */ - rtp->send_duration += 160; - - return 0; -} - -/*! \brief Send end packets for DTMF */ -int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit) -{ - unsigned int *rtpheader; - int hdrlen = 12, res = 0, i = 0; - char data[256]; - - /* If no address, then bail out */ - if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port) - return 0; - - 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 0; - } - - rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); - - rtpheader = (unsigned int *)data; - 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)); - /* Set end bit */ - rtpheader[3] |= htonl((1 << 23)); - rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno)); - /* Send 3 termination packets */ - for (i = 0; i < 3; i++) { - 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(rtp->them.sin_addr), - ntohs(rtp->them.sin_port), strerror(errno)); - if (rtp_debug_test_addr(&rtp->them)) - 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(rtp->them.sin_addr), - ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); - } - rtp->sending_digit = 0; - rtp->send_digit = 0; - /* Increment lastdigitts */ - rtp->lastdigitts += 960; - rtp->seqno++; - - return res; -} - -/*! \brief Public function: Send an H.261 fast update request, some devices need this rather than SIP XML */ -int ast_rtcp_send_h261fur(void *data) -{ - struct ast_rtp *rtp = data; - int res; - - rtp->rtcp->sendfur = 1; - res = ast_rtcp_write(data); - - 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)); - - if (rtp->rtcp->sendfur) { - rtcpheader[13] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1); - rtcpheader[14] = htonl(rtp->ssrc); /* Our SSRC */ - len += 8; - rtp->rtcp->sendfur = 0; - } - - /* 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 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; - - 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 (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); - - if (rtp->rtcp->sendfur) { - rtcpheader[8] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1); /* Header from page 36 in RFC 3550 */ - rtcpheader[9] = htonl(rtp->ssrc); /* Our SSRC */ - len += 8; - rtp->rtcp->sendfur = 0; - } - - /*! \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 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; -} - -/*! \brief generate comfort noice (CNG) */ -int ast_rtp_sendcng(struct ast_rtp *rtp, int level) -{ - unsigned int *rtpheader; - int hdrlen = 12; - int res; - int payload; - char data[256]; - 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(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:%u (type %d, seq %u, ts %u, len %d)\n" - , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen); - - } - return 0; -} - -/*! \brief Write RTP packet with audio or video media frames into UDP packet */ -static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec) -{ - unsigned char *rtpheader; - 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_AUDIO_MASK) { - 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 { - ast_debug(3, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms); - mark = 1; - } - } - } else if(f->subclass & AST_FORMAT_VIDEO_MASK) { - 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 { - 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, f->samples); - rtp->lastovidtimestamp = rtp->lastts; - } - } - } else { - pred = rtp->lastotexttimestamp + 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->lastotexttimestamp += f->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, f->samples); - rtp->lastotexttimestamp = 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; - - if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO)) - rtp->lastts = f->ts * 8; - - /* 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_debug(1, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.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(rtp->them.sin_addr), ntohs(rtp->them.sin_port)); - ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN); - } - } else { - rtp->txcount++; - rtp->txoctetcount +=(res - hdrlen); - - if (rtp->rtcp && rtp->rtcp->schedid < 1) - rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp); - } - - if (rtp_debug_test_addr(&rtp->them)) - ast_verbose("Sent RTP packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", - ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), codec, rtp->seqno, rtp->lastts,res - hdrlen); - } - - rtp->seqno++; - - return 0; -} - -int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs) -{ - int x; - for (x = 0; x < 32; x++) { /* Ugly way */ - rtp->pref.order[x] = prefs->order[x]; - rtp->pref.framing[x] = prefs->framing[x]; - } - if (rtp->smoother) - ast_smoother_free(rtp->smoother); - rtp->smoother = NULL; - return 0; -} - -struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp) -{ - return &rtp->pref; -} - -int ast_rtp_codec_getformat(int pt) -{ - if (pt < 0 || pt > MAX_RTP_PT) - return 0; /* bogus payload type */ - - if (static_RTP_PT[pt].isAstFormat) - return static_RTP_PT[pt].code; - else - 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) && (_f->frametype != AST_FRAME_TEXT)) { - ast_log(LOG_WARNING, "RTP can only send voice, video and text\n"); - return -1; - } - - /* The bottom bit of a video subclass contains the marker bit */ - 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 */ - ast_debug(1, "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; - } - - if (!rtp->smoother && subclass != AST_FORMAT_SPEEX && subclass != AST_FORMAT_G723_1) { - struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass); - if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */ - if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) { - ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms)); - return -1; - } - if (fmt.flags) - ast_smoother_set_flags(rtp->smoother, fmt.flags); - ast_debug(1, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms)); - } - } - if (rtp->smoother) { - if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) { - ast_smoother_feed_be(rtp->smoother, _f); - } else { - ast_smoother_feed(rtp->smoother, _f); - } - - while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) - ast_rtp_raw_write(rtp, f, codec); - } else { - /* Don't buffer outgoing frames; send them one-per-packet: */ - if (_f->offset < hdrlen) - f = ast_frdup(_f); /*! \bug XXX this might never be free'd. Why do we do this? */ - else - f = _f; - if (f->data) - ast_rtp_raw_write(rtp, f, codec); - if (f != _f) - ast_frfree(f); - } - - return 0; -} - -/*! \brief Unregister interface to channel driver */ -void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto) -{ - AST_RWLIST_WRLOCK(&protos); - AST_RWLIST_REMOVE(&protos, proto, list); - AST_RWLIST_UNLOCK(&protos); -} - -/*! \brief Register interface to channel driver */ -int ast_rtp_proto_register(struct ast_rtp_protocol *proto) -{ - struct ast_rtp_protocol *cur; - - AST_RWLIST_WRLOCK(&protos); - AST_RWLIST_TRAVERSE(&protos, cur, list) { - if (!strcmp(cur->type, proto->type)) { - ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type); - AST_RWLIST_UNLOCK(&protos); - return -1; - } - } - AST_RWLIST_INSERT_HEAD(&protos, proto, list); - AST_RWLIST_UNLOCK(&protos); - - return 0; -} - -/*! \brief Bridge loop for true native bridge (reinvite) */ -static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, struct ast_rtp *tp0, struct ast_rtp *tp1, struct ast_rtp_protocol *pr0, struct ast_rtp_protocol *pr1, int codec0, int codec1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1) -{ - struct ast_frame *fr = NULL; - struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, }; - int oldcodec0 = codec0, oldcodec1 = codec1; - struct sockaddr_in ac1 = {0,}, vac1 = {0,}, tac1 = {0,}, ac0 = {0,}, vac0 = {0,}, tac0 = {0,}; - struct sockaddr_in t1 = {0,}, vt1 = {0,}, tt1 = {0,}, t0 = {0,}, vt0 = {0,}, tt0 = {0,}; - - /* Set it up so audio goes directly between the two endpoints */ - - /* Test the first channel */ - if (!(pr0->set_rtp_peer(c0, p1, vp1, tp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))) { - ast_rtp_get_peer(p1, &ac1); - if (vp1) - ast_rtp_get_peer(vp1, &vac1); - if (tp1) - ast_rtp_get_peer(tp1, &tac1); - } else - ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name); - - /* Test the second channel */ - if (!(pr1->set_rtp_peer(c1, p0, vp0, tp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))) { - ast_rtp_get_peer(p0, &ac0); - if (vp0) - ast_rtp_get_peer(vp0, &vac0); - if (tp0) - ast_rtp_get_peer(tp0, &tac0); - } else - ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name); - - /* Now we can unlock and move into our loop */ - ast_channel_unlock(c0); - ast_channel_unlock(c1); - - ast_poll_channel_add(c0, c1); - - /* Throw our channels into the structure and enter the loop */ - cs[0] = c0; - cs[1] = c1; - cs[2] = NULL; - for (;;) { - /* Check if anything changed */ - if ((c0->tech_pvt != pvt0) || - (c1->tech_pvt != pvt1) || - (c0->masq || c0->masqr || c1->masq || c1->masqr) || - (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) { - ast_debug(1, "Oooh, something is weird, backing out\n"); - if (c0->tech_pvt == pvt0) - if (pr0->set_rtp_peer(c0, NULL, 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, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name); - ast_poll_channel_del(c0, c1); - return AST_BRIDGE_RETRY; - } - - /* Check if they have changed their address */ - ast_rtp_get_peer(p1, &t1); - if (vp1) - ast_rtp_get_peer(vp1, &vt1); - if (tp1) - ast_rtp_get_peer(tp1, &tt1); - if (pr1->get_codec) - codec1 = pr1->get_codec(c1); - ast_rtp_get_peer(p0, &t0); - if (vp0) - ast_rtp_get_peer(vp0, &vt0); - if (tp0) - ast_rtp_get_peer(tp0, &tt0); - if (pr0->get_codec) - codec0 = pr0->get_codec(c0); - if ((inaddrcmp(&t1, &ac1)) || - (vp1 && inaddrcmp(&vt1, &vac1)) || - (tp1 && inaddrcmp(&tt1, &tac1)) || - (codec1 != oldcodec1)) { - ast_debug(2, "Oooh, '%s' changed end address to %s:%d (format %d)\n", - c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1); - ast_debug(2, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n", - c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1); - ast_debug(2, "Oooh, '%s' changed end taddress to %s:%d (format %d)\n", - c1->name, ast_inet_ntoa(tt1.sin_addr), ntohs(tt1.sin_port), codec1); - ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n", - c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1); - ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n", - c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1); - ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n", - c1->name, ast_inet_ntoa(tac1.sin_addr), ntohs(tac1.sin_port), oldcodec1); - if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, tt1.sin_addr.s_addr ? tp1 : 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)); - memcpy(&tac1, &tt1, sizeof(tac1)); - oldcodec1 = codec1; - } - if ((inaddrcmp(&t0, &ac0)) || - (vp0 && inaddrcmp(&vt0, &vac0)) || - (tp0 && inaddrcmp(&tt0, &tac0))) { - ast_debug(2, "Oooh, '%s' changed end address to %s:%d (format %d)\n", - c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0); - ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n", - c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0); - if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, tt0.sin_addr.s_addr ? tp0 : 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)); - memcpy(&tac0, &tt0, sizeof(tac0)); - oldcodec0 = codec0; - } - - /* Wait for frame to come in on the channels */ - if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) { - if (!timeoutms) { - if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name); - if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name); - return AST_BRIDGE_RETRY; - } - ast_debug(1, "Ooh, empty read...\n"); - if (ast_check_hangup(c0) || ast_check_hangup(c1)) - break; - continue; - } - fr = ast_read(who); - other = (who == c0) ? c1 : c0; - if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) && - (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || - ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) { - /* Break out of bridge */ - *fo = fr; - *rc = who; - ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup"); - if (c0->tech_pvt == pvt0) - if (pr0->set_rtp_peer(c0, NULL, 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, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name); - ast_poll_channel_del(c0, c1); - return AST_BRIDGE_COMPLETE; - } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) { - if ((fr->subclass == AST_CONTROL_HOLD) || - (fr->subclass == AST_CONTROL_UNHOLD) || - (fr->subclass == AST_CONTROL_VIDUPDATE)) { - if (fr->subclass == AST_CONTROL_HOLD) { - /* If we someone went on hold we want the other side to reinvite back to us */ - if (who == c0) - pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0); - else - pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0); - } else if (fr->subclass == AST_CONTROL_UNHOLD) { - /* If they went off hold they should go back to being direct */ - if (who == c0) - pr1->set_rtp_peer(c1, p0, vp0, tp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)); - else - pr0->set_rtp_peer(c0, p1, vp1, tp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)); - } - /* Update local address information */ - ast_rtp_get_peer(p0, &t0); - memcpy(&ac0, &t0, sizeof(ac0)); - ast_rtp_get_peer(p1, &t1); - memcpy(&ac1, &t1, sizeof(ac1)); - /* Update codec information */ - if (pr0->get_codec && c0->tech_pvt) - oldcodec0 = codec0 = pr0->get_codec(c0); - if (pr1->get_codec && c1->tech_pvt) - oldcodec1 = codec1 = pr1->get_codec(c1); - ast_indicate_data(other, fr->subclass, fr->data, fr->datalen); - ast_frfree(fr); - } else { - *fo = fr; - *rc = who; - ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name); - return AST_BRIDGE_COMPLETE; - } - } else { - if ((fr->frametype == AST_FRAME_DTMF_BEGIN) || - (fr->frametype == AST_FRAME_DTMF_END) || - (fr->frametype == AST_FRAME_VOICE) || - (fr->frametype == AST_FRAME_VIDEO) || - (fr->frametype == AST_FRAME_IMAGE) || - (fr->frametype == AST_FRAME_HTML) || - (fr->frametype == AST_FRAME_MODEM) || - (fr->frametype == AST_FRAME_TEXT)) { - ast_write(other, fr); - } - ast_frfree(fr); - } - /* Swap priority */ -#ifndef HAVE_EPOLL - cs[2] = cs[0]; - cs[0] = cs[1]; - cs[1] = cs[2]; -#endif - } - - ast_poll_channel_del(c0, c1); - - if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name); - if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0)) - ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name); - - return AST_BRIDGE_FAILED; -} - -/*! \brief P2P RTP Callback */ -#ifdef P2P_INTENSE -static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata) -{ - int res = 0, hdrlen = 12; - struct sockaddr_in sin; - socklen_t len; - unsigned int *header; - struct ast_rtp *rtp = cbdata, *bridged = NULL; - - if (!rtp) - return 1; - - len = sizeof(sin); - if ((res = recvfrom(fd, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 0, (struct sockaddr *)&sin, &len)) < 0) - return 1; - - header = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET); - - /* If NAT support is turned on, then see if we need to change their address */ - if ((rtp->nat) && - ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) || - (rtp->them.sin_port != sin.sin_port))) { - rtp->them = sin; - rtp->rxseqno = 0; - ast_set_flag(rtp, FLAG_NAT_ACTIVE); - if (option_debug || rtpdebug) - ast_debug(0, "P2P RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port)); - } - - /* Write directly out to other RTP stream if bridged */ - if ((bridged = ast_rtp_get_bridged(rtp))) - bridge_p2p_rtp_write(rtp, bridged, header, res, hdrlen); - - return 1; -} - -/*! \brief Helper function to switch a channel and RTP stream into callback mode */ -static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod) -{ - /* If we need DTMF, are looking for STUN, or we have no IO structure then we can't do direct callback */ - if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || ast_test_flag(rtp, FLAG_HAS_STUN) || !rtp->io) - return 0; - - /* If the RTP structure is already in callback mode, remove it temporarily */ - if (rtp->ioid) { - ast_io_remove(rtp->io, rtp->ioid); - rtp->ioid = NULL; - } - - /* Steal the file descriptors from the channel */ - chan->fds[0] = -1; - - /* Now, fire up callback mode */ - iod[0] = ast_io_add(rtp->io, ast_rtp_fd(rtp), p2p_rtp_callback, AST_IO_IN, rtp); - - return 1; -} -#else -static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod) -{ - return 0; -} -#endif - -/*! \brief Helper function to switch a channel and RTP stream out of callback mode */ -static int p2p_callback_disable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod) -{ - ast_channel_lock(chan); - - /* Remove the callback from the IO context */ - ast_io_remove(rtp->io, iod[0]); - - /* Restore file descriptors */ - chan->fds[0] = ast_rtp_fd(rtp); - ast_channel_unlock(chan); - - /* Restore callback mode if previously used */ - if (ast_test_flag(rtp, FLAG_CALLBACK_MODE)) - rtp->ioid = ast_io_add(rtp->io, ast_rtp_fd(rtp), rtpread, AST_IO_IN, rtp); - - return 0; -} - -/*! \brief Helper function that sets what an RTP structure is bridged to */ -static void p2p_set_bridge(struct ast_rtp *rtp0, struct ast_rtp *rtp1) -{ - rtp_bridge_lock(rtp0); - rtp0->bridged = rtp1; - rtp_bridge_unlock(rtp0); - - return; -} - -/*! \brief Bridge loop for partial native bridge (packet2packet) - - In p2p mode, Asterisk is a very basic RTP proxy, just forwarding whatever - rtp/rtcp we get in to the channel. - \note this currently only works for Audio -*/ -static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1) -{ - struct ast_frame *fr = NULL; - struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, }; - int *p0_iod[2] = {NULL, NULL}, *p1_iod[2] = {NULL, NULL}; - int p0_callback = 0, p1_callback = 0; - enum ast_bridge_result res = AST_BRIDGE_FAILED; - - /* Okay, setup each RTP structure to do P2P forwarding */ - ast_clear_flag(p0, FLAG_P2P_SENT_MARK); - p2p_set_bridge(p0, p1); - ast_clear_flag(p1, FLAG_P2P_SENT_MARK); - p2p_set_bridge(p1, p0); - - /* Activate callback modes if possible */ - p0_callback = p2p_callback_enable(c0, p0, &p0_iod[0]); - p1_callback = p2p_callback_enable(c1, p1, &p1_iod[0]); - - /* Now let go of the channel locks and be on our way */ - ast_channel_unlock(c0); - ast_channel_unlock(c1); - - ast_poll_channel_add(c0, c1); - - /* Go into a loop forwarding frames until we don't need to anymore */ - cs[0] = c0; - cs[1] = c1; - cs[2] = NULL; - for (;;) { - /* If the underlying formats have changed force this bridge to break */ - if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) { - ast_debug(3, "p2p-rtp-bridge: Oooh, formats changed, backing out\n"); - res = AST_BRIDGE_FAILED_NOWARN; - break; - } - /* Check if anything changed */ - if ((c0->tech_pvt != pvt0) || - (c1->tech_pvt != pvt1) || - (c0->masq || c0->masqr || c1->masq || c1->masqr) || - (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) { - ast_debug(3, "p2p-rtp-bridge: Oooh, something is weird, backing out\n"); - /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */ - if ((c0->masq || c0->masqr) && (fr = ast_read(c0))) - ast_frfree(fr); - if ((c1->masq || c1->masqr) && (fr = ast_read(c1))) - ast_frfree(fr); - res = AST_BRIDGE_RETRY; - break; - } - /* Wait on a channel to feed us a frame */ - if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) { - if (!timeoutms) { - res = AST_BRIDGE_RETRY; - break; - } - if (option_debug > 2) - ast_log(LOG_NOTICE, "p2p-rtp-bridge: Ooh, empty read...\n"); - if (ast_check_hangup(c0) || ast_check_hangup(c1)) - break; - continue; - } - /* Read in frame from channel */ - fr = ast_read(who); - other = (who == c0) ? c1 : c0; - /* Depending on the frame we may need to break out of our bridge */ - if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) && - ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) | - ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) { - /* Record received frame and who */ - *fo = fr; - *rc = who; - ast_debug(3, "p2p-rtp-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup"); - res = AST_BRIDGE_COMPLETE; - break; - } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) { - if ((fr->subclass == AST_CONTROL_HOLD) || - (fr->subclass == AST_CONTROL_UNHOLD) || - (fr->subclass == AST_CONTROL_VIDUPDATE)) { - /* If we are going on hold, then break callback mode and P2P bridging */ - if (fr->subclass == AST_CONTROL_HOLD) { - if (p0_callback) - p0_callback = p2p_callback_disable(c0, p0, &p0_iod[0]); - if (p1_callback) - p1_callback = p2p_callback_disable(c1, p1, &p1_iod[0]); - p2p_set_bridge(p0, NULL); - p2p_set_bridge(p1, NULL); - } else if (fr->subclass == AST_CONTROL_UNHOLD) { - /* If we are off hold, then go back to callback mode and P2P bridging */ - ast_clear_flag(p0, FLAG_P2P_SENT_MARK); - p2p_set_bridge(p0, p1); - ast_clear_flag(p1, FLAG_P2P_SENT_MARK); - p2p_set_bridge(p1, p0); - p0_callback = p2p_callback_enable(c0, p0, &p0_iod[0]); - p1_callback = p2p_callback_enable(c1, p1, &p1_iod[0]); - } - ast_indicate_data(other, fr->subclass, fr->data, fr->datalen); - ast_frfree(fr); - } else { - *fo = fr; - *rc = who; - ast_debug(3, "p2p-rtp-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name); - res = AST_BRIDGE_COMPLETE; - break; - } - } else { - if ((fr->frametype == AST_FRAME_DTMF_BEGIN) || - (fr->frametype == AST_FRAME_DTMF_END) || - (fr->frametype == AST_FRAME_VOICE) || - (fr->frametype == AST_FRAME_VIDEO) || - (fr->frametype == AST_FRAME_IMAGE) || - (fr->frametype == AST_FRAME_HTML) || - (fr->frametype == AST_FRAME_MODEM) || - (fr->frametype == AST_FRAME_TEXT)) { - ast_write(other, fr); - } - - ast_frfree(fr); - } - /* Swap priority */ -#ifndef HAVE_EPOLL - cs[2] = cs[0]; - cs[0] = cs[1]; - cs[1] = cs[2]; -#endif - } - - /* If we are totally avoiding the core, then restore our link to it */ - if (p0_callback) - p0_callback = p2p_callback_disable(c0, p0, &p0_iod[0]); - if (p1_callback) - p1_callback = p2p_callback_disable(c1, p1, &p1_iod[0]); - - /* Break out of the direct bridge */ - p2p_set_bridge(p0, NULL); - p2p_set_bridge(p1, NULL); - - ast_poll_channel_del(c0, c1); - - return res; -} - -/*! \page AstRTPbridge The Asterisk RTP bridge - The RTP bridge is called from the channel drivers that are using the RTP - subsystem in Asterisk - like SIP, H.323 and Jingle/Google Talk. - - This bridge aims to offload the Asterisk server by setting up - the media stream directly between the endpoints, keeping the - signalling in Asterisk. - - It checks with the channel driver, using a callback function, if - there are possibilities for a remote bridge. - - If this fails, the bridge hands off to the core bridge. Reasons - can be NAT support needed, DTMF features in audio needed by - the PBX for transfers or spying/monitoring on channels. - - If transcoding is needed - we can't do a remote bridge. - If only NAT support is needed, we're using Asterisk in - RTP proxy mode with the p2p RTP bridge, basically - forwarding incoming audio packets to the outbound - stream on a network level. - - References: - - ast_rtp_bridge() - - ast_channel_early_bridge() - - ast_channel_bridge() - - rtp.c - - rtp.h -*/ -/*! \brief 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_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ - struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */ - struct ast_rtp *tp0 = NULL, *tp1 = NULL; /* Text RTP channels */ - struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL; - enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED, text_p0_res = AST_RTP_GET_FAILED; - enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED, text_p1_res = AST_RTP_GET_FAILED; - enum ast_bridge_result res = AST_BRIDGE_FAILED; - int codec0 = 0, codec1 = 0; - void *pvt0 = NULL, *pvt1 = NULL; - - /* Lock channels */ - ast_channel_lock(c0); - while (ast_channel_trylock(c1)) { - ast_channel_unlock(c0); - usleep(1); - ast_channel_lock(c0); - } - - /* Find channel driver interfaces */ - if (!(pr0 = get_proto(c0))) { - ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name); - ast_channel_unlock(c0); - ast_channel_unlock(c1); - return AST_BRIDGE_FAILED; - } - if (!(pr1 = get_proto(c1))) { - ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name); - ast_channel_unlock(c0); - ast_channel_unlock(c1); - 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) */ - audio_p0_res = pr0->get_rtp_info(c0, &p0); - video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED; - text_p0_res = pr0->get_trtp_info ? pr0->get_trtp_info(c0, &vp0) : AST_RTP_GET_FAILED; - audio_p1_res = pr1->get_rtp_info(c1, &p1); - video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED; - text_p1_res = pr1->get_trtp_info ? pr1->get_trtp_info(c1, &vp1) : AST_RTP_GET_FAILED; - - /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */ - if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE)) - audio_p0_res = AST_RTP_GET_FAILED; - if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE)) - audio_p1_res = AST_RTP_GET_FAILED; - - /* Check if a bridge is possible (partial/native) */ - if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) { - /* Somebody doesn't want to play... */ - ast_channel_unlock(c0); - ast_channel_unlock(c1); - return AST_BRIDGE_FAILED_NOWARN; - } - - /* If we need to feed DTMF frames into the core then only do a partial native bridge */ - if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) { - ast_set_flag(p0, FLAG_P2P_NEED_DTMF); - audio_p0_res = AST_RTP_TRY_PARTIAL; - } - - if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) { - ast_set_flag(p1, FLAG_P2P_NEED_DTMF); - audio_p1_res = AST_RTP_TRY_PARTIAL; - } - - /* If both sides are not using the same method of DTMF transmission - * (ie: one is RFC2833, other is INFO... then we can not do direct media. - * -------------------------------------------------- - * | DTMF Mode | HAS_DTMF | Accepts Begin Frames | - * |-----------|------------|-----------------------| - * | Inband | False | True | - * | RFC2833 | True | True | - * | SIP INFO | False | False | - * -------------------------------------------------- - * However, if DTMF from both channels is being monitored by the core, then - * we can still do packet-to-packet bridging, because passing through the - * core will handle DTMF mode translation. - */ - if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) || - (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) { - if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) { - ast_channel_unlock(c0); - ast_channel_unlock(c1); - return AST_BRIDGE_FAILED_NOWARN; - } - audio_p0_res = AST_RTP_TRY_PARTIAL; - audio_p1_res = AST_RTP_TRY_PARTIAL; - } - - /* If the core will need to compensate and the P2P bridge will need to feed up DTMF frames then we can not reliably do so yet, so do not P2P bridge */ - if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF) && ast_test_flag(p0, FLAG_DTMF_COMPENSATE)) || - (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF) && ast_test_flag(p1, FLAG_DTMF_COMPENSATE))) { - ast_channel_unlock(c0); - ast_channel_unlock(c1); - return AST_BRIDGE_FAILED_NOWARN; - } - - /* Get codecs from both sides */ - codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0; - codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0; - if (codec0 && codec1 && !(codec0 & codec1)) { - /* Hey, we can't do native bridging if both parties speak different codecs */ - ast_debug(3, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1); - ast_channel_unlock(c0); - ast_channel_unlock(c1); - return AST_BRIDGE_FAILED_NOWARN; - } - - /* If either side can only do a partial bridge, then don't try for a true native bridge */ - if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) { - struct ast_format_list fmt0, fmt1; - - /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */ - if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) { - ast_debug(1, "Cannot packet2packet bridge - raw formats are incompatible\n"); - ast_channel_unlock(c0); - ast_channel_unlock(c1); - return AST_BRIDGE_FAILED_NOWARN; - } - /* They must also be using the same packetization */ - fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat); - fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat); - if (fmt0.cur_ms != fmt1.cur_ms) { - ast_debug(1, "Cannot packet2packet bridge - packetization settings prevent it\n"); - ast_channel_unlock(c0); - ast_channel_unlock(c1); - return AST_BRIDGE_FAILED_NOWARN; - } - - ast_verb(3, "Packet2Packet bridging %s and %s\n", c0->name, c1->name); - res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1); - } else { - ast_verb(3, "Native bridging %s and %s\n", c0->name, c1->name); - res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, tp0, tp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1); - } - - return res; -} - -static char *rtp_do_debug_ip(struct ast_cli_args *a) -{ - struct hostent *hp; - struct ast_hostent ahp; - int port = 0; - char *p, *arg; - - arg = a->argv[3]; - p = strstr(arg, ":"); - if (p) { - *p = '\0'; - p++; - port = atoi(p); - } - hp = ast_gethostbyname(arg, &ahp); - if (hp == NULL) { - ast_cli(a->fd, "Lookup failed for '%s'\n", arg); - return CLI_FAILURE; - } - 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(a->fd, "RTP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtpdebugaddr.sin_addr)); - else - ast_cli(a->fd, "RTP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtpdebugaddr.sin_addr), port); - rtpdebug = 1; - return CLI_SUCCESS; -} - -static char *rtcp_do_debug_ip(struct ast_cli_args *a) -{ - struct hostent *hp; - struct ast_hostent ahp; - int port = 0; - char *p, *arg; - - arg = a->argv[3]; - p = strstr(arg, ":"); - if (p) { - *p = '\0'; - p++; - port = atoi(p); - } - hp = ast_gethostbyname(arg, &ahp); - if (hp == NULL) { - ast_cli(a->fd, "Lookup failed for '%s'\n", arg); - return CLI_FAILURE; - } - rtcpdebugaddr.sin_family = AF_INET; - memcpy(&rtcpdebugaddr.sin_addr, hp->h_addr, sizeof(rtcpdebugaddr.sin_addr)); - rtcpdebugaddr.sin_port = htons(port); - if (port == 0) - ast_cli(a->fd, "RTCP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr)); - else - ast_cli(a->fd, "RTCP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr), port); - rtcpdebug = 1; - return CLI_SUCCESS; -} - -static char *handle_cli_rtp_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a) -{ - switch (cmd) { - case CLI_INIT: - e->command = "rtp debug [off|ip]"; - e->usage = - "Usage: rtp debug [off]|[ip host[:port]]\n" - " Enable/Disable dumping of all RTP packets. If 'ip' is\n" - " specified, limit the dumped packets to those to and from\n" - " the specified 'host' with optional port.\n"; - return NULL; - case CLI_GENERATE: - return NULL; - } - - if (a->argc < 2 || a->argc > 4) - return CLI_SHOWUSAGE; - if (a->argc == 2) { - rtpdebug = 1; - memset(&rtpdebugaddr, 0, sizeof(rtpdebugaddr)); - ast_cli(a->fd, "RTP Debugging Enabled\n"); - } else if (a->argc == 3) { - if (strncasecmp(a->argv[2], "off", 3)) - return CLI_SHOWUSAGE; - rtpdebug = 0; - ast_cli(a->fd, "RTP Debugging Disabled\n"); - } else { - if (strncasecmp(a->argv[2], "ip", 2)) - return CLI_SHOWUSAGE; - return rtp_do_debug_ip(a); - } - - return CLI_SUCCESS; -} - -static char *handle_cli_rtcp_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a) -{ - switch (cmd) { - case CLI_INIT: - e->command = "rtcp debug [off|ip]"; - e->usage = - "Usage: rtcp debug [off]|[ip host[:port]]\n" - " Enable/Disable dumping of all RTCP packets. If 'ip' is\n" - " specified, limit the dumped packets to those to and from\n" - " the specified 'host' with optional port.\n"; - return NULL; - case CLI_GENERATE: - return NULL; - } - - if (a->argc < 2 || a->argc > 4) - return CLI_SHOWUSAGE; - if (a->argc == 2) { - rtcpdebug = 1; - memset(&rtcpdebugaddr, 0, sizeof(rtcpdebugaddr)); - ast_cli(a->fd, "RTCP Debugging Enabled\n"); - } else if (a->argc == 3) { - if (strncasecmp(a->argv[2], "off", 3)) - return CLI_SHOWUSAGE; - rtcpdebug = 0; - ast_cli(a->fd, "RTCP Debugging Disabled\n"); - } else { - if (strncasecmp(a->argv[2], "ip", 2)) - return CLI_SHOWUSAGE; - return rtcp_do_debug_ip(a); - } - - return CLI_SUCCESS; -} - -static char *handle_cli_rtcp_stats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a) -{ - switch (cmd) { - case CLI_INIT: - e->command = "rtcp stats [off]"; - e->usage = - "Usage: rtcp stats [off]\n" - " Enable/Disable dumping of RTCP stats.\n"; - return NULL; - case CLI_GENERATE: - return NULL; - } - - if (a->argc < 2 || a->argc > 3) - return CLI_SHOWUSAGE; - if (a->argc == 3 && strncasecmp(a->argv[2], "off", 3)) - return CLI_SHOWUSAGE; - - rtcpstats = (a->argc == 3) ? 0 : 1; - ast_cli(a->fd, "RTCP Stats %s\n", rtcpstats ? "Enabled" : "Disabled"); - return CLI_SUCCESS; -} - -static char *handle_cli_stun_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a) -{ - switch (cmd) { - case CLI_INIT: - e->command = "stun debug [off]"; - e->usage = - "Usage: stun debug [off]\n" - " Enable/Disable STUN (Simple Traversal of UDP through NATs)\n" - " debugging\n"; - return NULL; - case CLI_GENERATE: - return NULL; - } - - if (a->argc < 2 || a->argc > 3) - return CLI_SHOWUSAGE; - if (a->argc == 3 && strncasecmp(a->argv[2], "off", 3)) - return CLI_SHOWUSAGE; - - stundebug = (a->argc == 3) ? 0 : 1; - ast_cli(a->fd, "STUN Debugging %s\n", stundebug ? "Enabled" : "Disabled"); - return CLI_SUCCESS; -} - -static struct ast_cli_entry cli_rtp[] = { - AST_CLI_DEFINE(handle_cli_rtp_debug, "Enable/Disable RTP debugging"), - AST_CLI_DEFINE(handle_cli_rtcp_debug, "Enable/Disable RTCP debugging"), - AST_CLI_DEFINE(handle_cli_rtcp_stats, "Enable/Disable RTCP stats"), - AST_CLI_DEFINE(handle_cli_stun_debug, "Enable/Disable STUN debugging"), -}; - -static int __ast_rtp_reload(int reload) -{ - struct ast_config *cfg; - const char *s; - struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 }; - - if ((cfg = ast_config_load("rtp.conf", config_flags)) == CONFIG_STATUS_FILEUNCHANGED) - return 0; - - rtpstart = 5000; - rtpend = 31000; - dtmftimeout = DEFAULT_DTMF_TIMEOUT; - strictrtp = STRICT_RTP_OPEN; - 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", "rtcpinterval"))) { - rtcpinterval = atoi(s); - if (rtcpinterval == 0) - rtcpinterval = 0; /* Just so we're clear... it's zero */ - if (rtcpinterval < RTCP_MIN_INTERVALMS) - rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */ - if (rtcpinterval > RTCP_MAX_INTERVALMS) - rtcpinterval = RTCP_MAX_INTERVALMS; - } - 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; - }; - } - if ((s = ast_variable_retrieve(cfg, "general", "strictrtp"))) { - strictrtp = ast_true(s); - } - 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; - } - ast_verb(2, "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend); - return 0; -} - -int ast_rtp_reload(void) -{ - return __ast_rtp_reload(1); -} - -/*! \brief Initialize the RTP system in Asterisk */ -void ast_rtp_init(void) -{ - ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry)); - __ast_rtp_reload(0); -} - |