There are numerous problems with the current implementation of the RTP payload type mapping in Asterisk. It uses only one mapping structure to associate payload types to codecs. The single mapping is overkill if all of the payload type values are well known values. Dynamic payload type mappings do not work as well with the single mapping because RFC3264 allows each side of the link to negotiate different dynamic mappings for what they want to receive. Not only could you have the same codec mapped for sending and receiving on different payload types you could wind up with the same payload type mapped to different codecs for each direction. 1) An independent payload type mapping is needed for sending and receiving. 2) The receive mapping needs to keep track of previous mappings because of the slack to when negotiation happens and current packets in flight using the old mapping arrive. 3) The transmit mapping only needs to keep track of the current negotiated values since we are sending the packets and know when the switchover takes place. * Needed to create ast_rtp_codecs_payload_code_tx() and make some callers use the new function because ast_rtp_codecs_payload_code() was used for mappings in both directions. * Needed to create ast_rtp_codecs_payloads_xover() for cases where we need to pass preferred codec mappings to the peer channel for early media bridging or when we need to prefer the offered mapping that RFC3264 says we SHOULD use. * ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are the only new public functions created. All the others were only used for the tx or rx mapping direction so the function doxygen now reflects which direction the function operates. * chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing that makes no sense when processing an incoming SDP. We would be wiping out any mappings that we set for the possible outgoing SDP we sent earlier. ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac
325 lines
9.5 KiB
C
325 lines
9.5 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2009, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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* Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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*
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* \brief Multicast RTP Engine
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*
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* \author Joshua Colp <jcolp@digium.com>
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* \author Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
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*
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* \ingroup rtp_engines
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_REGISTER_FILE()
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#include <sys/time.h>
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#include <signal.h>
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#include <fcntl.h>
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#include <math.h>
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#include "asterisk/pbx.h"
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#include "asterisk/frame.h"
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#include "asterisk/channel.h"
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#include "asterisk/acl.h"
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#include "asterisk/config.h"
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#include "asterisk/lock.h"
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#include "asterisk/utils.h"
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#include "asterisk/cli.h"
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#include "asterisk/manager.h"
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#include "asterisk/unaligned.h"
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/format_cache.h"
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/*! Command value used for Linksys paging to indicate we are starting */
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#define LINKSYS_MCAST_STARTCMD 6
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/*! Command value used for Linksys paging to indicate we are stopping */
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#define LINKSYS_MCAST_STOPCMD 7
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/*! \brief Type of paging to do */
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enum multicast_type {
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/*! Simple multicast enabled client/receiver paging like Snom and Barix uses */
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MULTICAST_TYPE_BASIC = 0,
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/*! More advanced Linksys type paging which requires a start and stop packet */
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MULTICAST_TYPE_LINKSYS,
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};
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/*! \brief Structure for a Linksys control packet */
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struct multicast_control_packet {
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/*! Unique identifier for the control packet */
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uint32_t unique_id;
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/*! Actual command in the control packet */
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uint32_t command;
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/*! IP address for the RTP */
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uint32_t ip;
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/*! Port for the RTP */
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uint32_t port;
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};
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/*! \brief Structure for a multicast paging instance */
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struct multicast_rtp {
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/*! TYpe of multicast paging this instance is doing */
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enum multicast_type type;
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/*! Socket used for sending the audio on */
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int socket;
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/*! Synchronization source value, used when creating/sending the RTP packet */
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unsigned int ssrc;
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/*! Sequence number, used when creating/sending the RTP packet */
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uint16_t seqno;
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unsigned int lastts;
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struct timeval txcore;
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};
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/* Forward Declarations */
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static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data);
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static int multicast_rtp_activate(struct ast_rtp_instance *instance);
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static int multicast_rtp_destroy(struct ast_rtp_instance *instance);
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static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
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static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
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/* RTP Engine Declaration */
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static struct ast_rtp_engine multicast_rtp_engine = {
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.name = "multicast",
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.new = multicast_rtp_new,
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.activate = multicast_rtp_activate,
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.destroy = multicast_rtp_destroy,
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.write = multicast_rtp_write,
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.read = multicast_rtp_read,
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};
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/*! \brief Function called to create a new multicast instance */
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static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
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{
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struct multicast_rtp *multicast;
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const char *type = data;
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if (!(multicast = ast_calloc(1, sizeof(*multicast)))) {
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return -1;
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}
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if (!strcasecmp(type, "basic")) {
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multicast->type = MULTICAST_TYPE_BASIC;
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} else if (!strcasecmp(type, "linksys")) {
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multicast->type = MULTICAST_TYPE_LINKSYS;
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} else {
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ast_free(multicast);
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return -1;
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}
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if ((multicast->socket = socket(AF_INET, SOCK_DGRAM, 0)) < 0) {
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ast_free(multicast);
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return -1;
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}
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multicast->ssrc = ast_random();
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ast_rtp_instance_set_data(instance, multicast);
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return 0;
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}
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static int rtp_get_rate(struct ast_format *format)
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{
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return ast_format_cmp(format, ast_format_g722) == AST_FORMAT_CMP_EQUAL ?
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8000 : ast_format_get_sample_rate(format);
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}
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static unsigned int calc_txstamp(struct multicast_rtp *rtp, struct timeval *delivery)
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{
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struct timeval t;
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long ms;
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if (ast_tvzero(rtp->txcore)) {
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rtp->txcore = ast_tvnow();
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rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
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}
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t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
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if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
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ms = 0;
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}
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rtp->txcore = t;
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return (unsigned int) ms;
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}
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/*! \brief Helper function which populates a control packet with useful information and sends it */
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static int multicast_send_control_packet(struct ast_rtp_instance *instance, struct multicast_rtp *multicast, int command)
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{
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struct multicast_control_packet control_packet = { .unique_id = htonl((u_long)time(NULL)),
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.command = htonl(command),
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};
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struct ast_sockaddr control_address, remote_address;
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ast_rtp_instance_get_local_address(instance, &control_address);
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ast_rtp_instance_get_remote_address(instance, &remote_address);
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/* Ensure the user of us have given us both the control address and destination address */
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if (ast_sockaddr_isnull(&control_address) ||
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ast_sockaddr_isnull(&remote_address)) {
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return -1;
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}
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/* The protocol only supports IPv4. */
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if (ast_sockaddr_is_ipv6(&remote_address)) {
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ast_log(LOG_WARNING, "Cannot send control packet for IPv6 "
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"remote address.\n");
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return -1;
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}
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control_packet.ip = htonl(ast_sockaddr_ipv4(&remote_address));
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control_packet.port = htonl(ast_sockaddr_port(&remote_address));
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/* Based on a recommendation by Brian West who did the FreeSWITCH implementation we send control packets twice */
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ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address);
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ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address);
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return 0;
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}
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/*! \brief Function called to indicate that audio is now going to flow */
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static int multicast_rtp_activate(struct ast_rtp_instance *instance)
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{
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struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
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if (multicast->type != MULTICAST_TYPE_LINKSYS) {
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return 0;
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}
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return multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STARTCMD);
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}
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/*! \brief Function called to destroy a multicast instance */
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static int multicast_rtp_destroy(struct ast_rtp_instance *instance)
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{
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struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
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if (multicast->type == MULTICAST_TYPE_LINKSYS) {
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multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STOPCMD);
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}
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close(multicast->socket);
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ast_free(multicast);
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return 0;
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}
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/*! \brief Function called to broadcast some audio on a multicast instance */
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static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
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{
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struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
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struct ast_frame *f = frame;
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struct ast_sockaddr remote_address;
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int hdrlen = 12, res = 0, codec;
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unsigned char *rtpheader;
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unsigned int ms = calc_txstamp(multicast, &frame->delivery);
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int rate = rtp_get_rate(frame->subclass.format) / 1000;
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/* We only accept audio, nothing else */
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if (frame->frametype != AST_FRAME_VOICE) {
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return 0;
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}
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/* Grab the actual payload number for when we create the RTP packet */
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codec = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance),
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1, frame->subclass.format, 0);
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if (codec < 0) {
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return -1;
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}
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/* If we do not have space to construct an RTP header duplicate the frame so we get some */
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if (frame->offset < hdrlen) {
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f = ast_frdup(frame);
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}
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/* Calucate last TS */
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multicast->lastts = multicast->lastts + ms * rate;
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/* Construct an RTP header for our packet */
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rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
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put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno)));
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if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO)) {
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put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
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} else {
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put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
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}
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put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
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/* Increment sequence number and wrap to 0 if it overflows 16 bits. */
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multicast->seqno = 0xFFFF & (multicast->seqno + 1);
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/* Finally send it out to the eager phones listening for us */
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ast_rtp_instance_get_remote_address(instance, &remote_address);
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if (ast_sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, &remote_address) < 0) {
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ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s: %s\n",
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ast_sockaddr_stringify(&remote_address),
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strerror(errno));
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res = -1;
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}
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/* If we were forced to duplicate the frame free the new one */
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if (frame != f) {
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ast_frfree(f);
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}
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return res;
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}
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/*! \brief Function called to read from a multicast instance */
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static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
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{
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return &ast_null_frame;
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}
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static int load_module(void)
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{
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if (ast_rtp_engine_register(&multicast_rtp_engine)) {
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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static int unload_module(void)
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{
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ast_rtp_engine_unregister(&multicast_rtp_engine);
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return 0;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
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.support_level = AST_MODULE_SUPPORT_CORE,
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_CHANNEL_DEPEND,
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);
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