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asterisk/res/res_pjsip_sdp_rtp.c
Joshua Colp 4c535f5c30 core / pjsip: Add support for grouping streams together. 
In WebRTC streams (or media tracks in their world) can be grouped
together using the mslabel. This informs the browser that each
should be synchronized with each other.

This change extends the stream API so this information can
be stored with streams. The PJSIP support has been extended
to use the mslabel to determine grouped streams and store
this association on the streams. Finally when creating the
SDP the group information is used to cause each media stream
to use the same mslabel.

ASTERISK-27379

Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad
2017-10-30 17:10:03 -05:00

1973 lines
66 KiB
C
Raw Blame History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
* Kevin Harwell <kharwell@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
*
* \author Joshua Colp <jcolp@digium.com>
*
* \brief SIP SDP media stream handling
*/
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjmedia.h>
#include <pjlib.h>
#include "asterisk/utils.h"
#include "asterisk/module.h"
#include "asterisk/format.h"
#include "asterisk/format_cap.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/netsock2.h"
#include "asterisk/channel.h"
#include "asterisk/causes.h"
#include "asterisk/sched.h"
#include "asterisk/acl.h"
#include "asterisk/sdp_srtp.h"
#include "asterisk/dsp.h"
#include "asterisk/linkedlists.h" /* for AST_LIST_NEXT */
#include "asterisk/stream.h"
#include "asterisk/format_cache.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
/*! \brief Scheduler for RTCP purposes */
static struct ast_sched_context *sched;
/*! \brief Address for RTP */
static struct ast_sockaddr address_rtp;
static const char STR_AUDIO[] = "audio";
static const char STR_VIDEO[] = "video";
static int send_keepalive(const void *data)
{
struct ast_sip_session_media *session_media = (struct ast_sip_session_media *) data;
struct ast_rtp_instance *rtp = session_media->rtp;
int keepalive;
time_t interval;
int send_keepalive;
if (!rtp) {
return 0;
}
keepalive = ast_rtp_instance_get_keepalive(rtp);
if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
ast_debug(3, "Not sending RTP keepalive on RTP instance %p since direct media is in use\n", rtp);
return keepalive * 1000;
}
interval = time(NULL) - ast_rtp_instance_get_last_tx(rtp);
send_keepalive = interval >= keepalive;
ast_debug(3, "It has been %d seconds since RTP was last sent on instance %p. %sending keepalive\n",
(int) interval, rtp, send_keepalive ? "S" : "Not s");
if (send_keepalive) {
ast_rtp_instance_sendcng(rtp, 0);
return keepalive * 1000;
}
return (keepalive - interval) * 1000;
}
/*! \brief Check whether RTP is being received or not */
static int rtp_check_timeout(const void *data)
{
struct ast_sip_session_media *session_media = (struct ast_sip_session_media *)data;
struct ast_rtp_instance *rtp = session_media->rtp;
int elapsed;
struct ast_channel *chan;
if (!rtp) {
return 0;
}
elapsed = time(NULL) - ast_rtp_instance_get_last_rx(rtp);
if (elapsed < ast_rtp_instance_get_timeout(rtp)) {
return (ast_rtp_instance_get_timeout(rtp) - elapsed) * 1000;
}
chan = ast_channel_get_by_name(ast_rtp_instance_get_channel_id(rtp));
if (!chan) {
return 0;
}
ast_log(LOG_NOTICE, "Disconnecting channel '%s' for lack of RTP activity in %d seconds\n",
ast_channel_name(chan), elapsed);
ast_channel_lock(chan);
ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
ast_channel_unlock(chan);
ast_softhangup(chan, AST_SOFTHANGUP_DEV);
ast_channel_unref(chan);
return 0;
}
/*!
* \brief Enable RTCP on an RTP session.
*/
static void enable_rtcp(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *remote_media)
{
enum ast_rtp_instance_rtcp rtcp_type;
if (session->endpoint->media.rtcp_mux && session_media->remote_rtcp_mux) {
rtcp_type = AST_RTP_INSTANCE_RTCP_MUX;
} else {
rtcp_type = AST_RTP_INSTANCE_RTCP_STANDARD;
}
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RTCP, rtcp_type);
}
/*! \brief Internal function which creates an RTP instance */
static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
{
struct ast_rtp_engine_ice *ice;
struct ast_sockaddr temp_media_address;
struct ast_sockaddr *media_address = &address_rtp;
if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
if (ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0)) {
ast_debug(1, "Endpoint %s: Binding RTP media to %s\n",
ast_sorcery_object_get_id(session->endpoint),
session->endpoint->media.address);
media_address = &temp_media_address;
} else {
ast_debug(1, "Endpoint %s: RTP media address invalid: %s\n",
ast_sorcery_object_get_id(session->endpoint),
session->endpoint->media.address);
}
} else {
struct ast_sip_transport *transport;
transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
session->endpoint->transport);
if (transport) {
struct ast_sip_transport_state *trans_state;
trans_state = ast_sip_get_transport_state(ast_sorcery_object_get_id(transport));
if (trans_state) {
char hoststr[PJ_INET6_ADDRSTRLEN];
pj_sockaddr_print(&trans_state->host, hoststr, sizeof(hoststr), 0);
if (ast_sockaddr_parse(&temp_media_address, hoststr, 0)) {
ast_debug(1, "Transport %s bound to %s: Using it for RTP media.\n",
session->endpoint->transport, hoststr);
media_address = &temp_media_address;
} else {
ast_debug(1, "Transport %s bound to %s: Invalid for RTP media.\n",
session->endpoint->transport, hoststr);
}
ao2_ref(trans_state, -1);
}
ao2_ref(transport, -1);
}
}
if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, media_address, NULL))) {
ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->media.rtp.engine);
return -1;
}
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->media.rtp.symmetric);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_ASYMMETRIC_CODEC, session->endpoint->asymmetric_rtp_codec);
if (!session->endpoint->media.rtp.ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
ice->stop(session_media->rtp);
}
if (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
} else if (session->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
if (session_media->type == AST_MEDIA_TYPE_AUDIO &&
(session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
session->endpoint->media.cos_audio, "SIP RTP Audio");
} else if (session_media->type == AST_MEDIA_TYPE_VIDEO &&
(session->endpoint->media.tos_video || session->endpoint->media.cos_video)) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
session->endpoint->media.cos_video, "SIP RTP Video");
}
ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
return 0;
}
static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs,
struct ast_sip_session_media *session_media)
{
pjmedia_sdp_attr *attr;
pjmedia_sdp_rtpmap *rtpmap;
pjmedia_sdp_fmtp fmtp;
struct ast_format *format;
int i, num = 0, tel_event = 0;
char name[256];
char media[20];
char fmt_param[256];
enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
AST_RTP_OPT_G726_NONSTANDARD : 0;
ast_rtp_codecs_payloads_initialize(codecs);
/* Iterate through provided formats */
for (i = 0; i < stream->desc.fmt_count; ++i) {
/* The payload is kept as a string for things like t38 but for video it is always numerical */
ast_rtp_codecs_payloads_set_m_type(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]));
/* Look for the optional rtpmap attribute */
if (!(attr = pjmedia_sdp_media_find_attr2(stream, "rtpmap", &stream->desc.fmt[i]))) {
continue;
}
/* Interpret the attribute as an rtpmap */
if ((pjmedia_sdp_attr_to_rtpmap(session->inv_session->pool_prov, attr, &rtpmap)) != PJ_SUCCESS) {
continue;
}
ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
if (strcmp(name, "telephone-event") == 0) {
tel_event++;
}
ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL,
pj_strtoul(&stream->desc.fmt[i]), media, name, options, rtpmap->clock_rate);
/* Look for an optional associated fmtp attribute */
if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) {
continue;
}
if ((pjmedia_sdp_attr_get_fmtp(attr, &fmtp)) == PJ_SUCCESS) {
ast_copy_pj_str(fmt_param, &fmtp.fmt, sizeof(fmt_param));
if (sscanf(fmt_param, "%30d", &num) != 1) {
continue;
}
if ((format = ast_rtp_codecs_get_payload_format(codecs, num))) {
struct ast_format *format_parsed;
ast_copy_pj_str(fmt_param, &fmtp.fmt_param, sizeof(fmt_param));
format_parsed = ast_format_parse_sdp_fmtp(format, fmt_param);
if (format_parsed) {
ast_rtp_codecs_payload_replace_format(codecs, num, format_parsed);
ao2_ref(format_parsed, -1);
}
ao2_ref(format, -1);
}
}
}
if (!tel_event && (session->dtmf == AST_SIP_DTMF_AUTO)) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
if (session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
if (tel_event) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
} else {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_NONE);
}
}
/* Get the packetization, if it exists */
if ((attr = pjmedia_sdp_media_find_attr2(stream, "ptime", NULL))) {
unsigned long framing = pj_strtoul(pj_strltrim(&attr->value));
if (framing && session->endpoint->media.rtp.use_ptime) {
ast_rtp_codecs_set_framing(codecs, framing);
}
}
}
static int set_caps(struct ast_sip_session *session,
struct ast_sip_session_media *session_media,
struct ast_sip_session_media *session_media_transport,
const struct pjmedia_sdp_media *stream,
int is_offer, struct ast_stream *asterisk_stream)
{
RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, peer, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, joint, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, endpoint_caps, NULL, ao2_cleanup);
enum ast_media_type media_type = session_media->type;
struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
int fmts = 0;
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
ast_format_cap_count(session->direct_media_cap);
int dsp_features = 0;
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n",
ast_codec_media_type2str(session_media->type));
return -1;
}
/* get the endpoint capabilities */
if (direct_media_enabled) {
ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
} else {
ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
}
/* get the capabilities on the peer */
get_codecs(session, stream, &codecs, session_media);
ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
/* get the joint capabilities between peer and endpoint */
ast_format_cap_get_compatible(caps, peer, joint);
if (!ast_format_cap_count(joint)) {
struct ast_str *usbuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
struct ast_str *thembuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
ast_rtp_codecs_payloads_destroy(&codecs);
ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
ast_codec_media_type2str(session_media->type),
ast_format_cap_get_names(caps, &usbuf),
ast_format_cap_get_names(peer, &thembuf));
return -1;
}
if (is_offer) {
/*
* Setup rx payload type mapping to prefer the mapping
* from the peer that the RFC says we SHOULD use.
*/
ast_rtp_codecs_payloads_xover(&codecs, &codecs, NULL);
}
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
ast_stream_set_formats(asterisk_stream, joint);
/* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */
if (session_media_transport != session_media && session_media->bundled) {
int index;
for (index = 0; index < ast_format_cap_count(joint); ++index) {
struct ast_format *format = ast_format_cap_get_format(joint, index);
int rtp_code;
/* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for
* things as the format is guaranteed to have a payload already.
*/
rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0);
ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format);
ao2_ref(format, -1);
}
}
if (session->channel && ast_sip_session_is_pending_stream_default(session, asterisk_stream)) {
ast_channel_lock(session->channel);
ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, media_type);
if (session->endpoint->preferred_codec_only){
struct ast_format *preferred_fmt = ast_format_cap_get_format(joint, 0);
ast_format_cap_append(caps, preferred_fmt, 0);
ao2_ref(preferred_fmt, -1);
} else if (!session->endpoint->asymmetric_rtp_codec) {
struct ast_format *best;
/*
* If we don't allow the sending codec to be changed on our side
* then get the best codec from the joint capabilities of the media
* type and use only that. This ensures the core won't start sending
* out a format that we aren't currently sending.
*/
best = ast_format_cap_get_best_by_type(joint, media_type);
if (best) {
ast_format_cap_append(caps, best, ast_format_cap_get_framing(joint));
ao2_ref(best, -1);
}
} else {
ast_format_cap_append_from_cap(caps, joint, media_type);
}
/*
* Apply the new formats to the channel, potentially changing
* raw read/write formats and translation path while doing so.
*/
ast_channel_nativeformats_set(session->channel, caps);
if (media_type == AST_MEDIA_TYPE_AUDIO) {
ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
}
if ( ((session->dtmf == AST_SIP_DTMF_AUTO) || (session->dtmf == AST_SIP_DTMF_AUTO_INFO) )
&& (ast_rtp_instance_dtmf_mode_get(session_media->rtp) == AST_RTP_DTMF_MODE_RFC2833)
&& (session->dsp)) {
dsp_features = ast_dsp_get_features(session->dsp);
dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
if (dsp_features) {
ast_dsp_set_features(session->dsp, dsp_features);
} else {
ast_dsp_free(session->dsp);
session->dsp = NULL;
}
}
if (ast_channel_is_bridged(session->channel)) {
ast_channel_set_unbridged_nolock(session->channel, 1);
}
ast_channel_unlock(session->channel);
}
ast_rtp_codecs_payloads_destroy(&codecs);
return 0;
}
static pjmedia_sdp_attr* generate_rtpmap_attr(struct ast_sip_session *session, pjmedia_sdp_media *media, pj_pool_t *pool,
int rtp_code, int asterisk_format, struct ast_format *format, int code)
{
extern pj_bool_t pjsip_use_compact_form;
pjmedia_sdp_rtpmap rtpmap;
pjmedia_sdp_attr *attr = NULL;
char tmp[64];
enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
AST_RTP_OPT_G726_NONSTANDARD : 0;
snprintf(tmp, sizeof(tmp), "%d", rtp_code);
pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
if (rtp_code <= AST_RTP_PT_LAST_STATIC && pjsip_use_compact_form) {
return NULL;
}
rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1];
rtpmap.clock_rate = ast_rtp_lookup_sample_rate2(asterisk_format, format, code);
pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options));
if (!pj_stricmp2(&rtpmap.enc_name, "opus")) {
pj_cstr(&rtpmap.param, "2");
} else {
pj_cstr(&rtpmap.param, NULL);
}
pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);
return attr;
}
static pjmedia_sdp_attr* generate_fmtp_attr(pj_pool_t *pool, struct ast_format *format, int rtp_code)
{
struct ast_str *fmtp0 = ast_str_alloca(256);
pj_str_t fmtp1;
pjmedia_sdp_attr *attr = NULL;
char *tmp;
ast_format_generate_sdp_fmtp(format, rtp_code, &fmtp0);
if (ast_str_strlen(fmtp0)) {
tmp = ast_str_buffer(fmtp0) + ast_str_strlen(fmtp0) - 1;
/* remove any carriage return line feeds */
while (*tmp == '\r' || *tmp == '\n') --tmp;
*++tmp = '\0';
/* ast...generate gives us everything, just need value */
tmp = strchr(ast_str_buffer(fmtp0), ':');
if (tmp && tmp[1] != '\0') {
fmtp1 = pj_str(tmp + 1);
} else {
fmtp1 = pj_str(ast_str_buffer(fmtp0));
}
attr = pjmedia_sdp_attr_create(pool, "fmtp", &fmtp1);
}
return attr;
}
/*! \brief Function which adds ICE attributes to a media stream */
static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media,
unsigned int include_candidates)
{
struct ast_rtp_engine_ice *ice;
struct ao2_container *candidates;
const char *username, *password;
pj_str_t stmp;
pjmedia_sdp_attr *attr;
struct ao2_iterator it_candidates;
struct ast_rtp_engine_ice_candidate *candidate;
if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
return;
}
if ((username = ice->get_ufrag(session_media->rtp))) {
attr = pjmedia_sdp_attr_create(pool, "ice-ufrag", pj_cstr(&stmp, username));
media->attr[media->attr_count++] = attr;
}
if ((password = ice->get_password(session_media->rtp))) {
attr = pjmedia_sdp_attr_create(pool, "ice-pwd", pj_cstr(&stmp, password));
media->attr[media->attr_count++] = attr;
}
if (!include_candidates) {
return;
}
candidates = ice->get_local_candidates(session_media->rtp);
if (!candidates) {
return;
}
it_candidates = ao2_iterator_init(candidates, 0);
for (; (candidate = ao2_iterator_next(&it_candidates)); ao2_ref(candidate, -1)) {
struct ast_str *attr_candidate = ast_str_create(128);
ast_str_set(&attr_candidate, -1, "%s %u %s %d %s ", candidate->foundation, candidate->id, candidate->transport,
candidate->priority, ast_sockaddr_stringify_addr_remote(&candidate->address));
ast_str_append(&attr_candidate, -1, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
switch (candidate->type) {
case AST_RTP_ICE_CANDIDATE_TYPE_HOST:
ast_str_append(&attr_candidate, -1, "host");
break;
case AST_RTP_ICE_CANDIDATE_TYPE_SRFLX:
ast_str_append(&attr_candidate, -1, "srflx");
break;
case AST_RTP_ICE_CANDIDATE_TYPE_RELAYED:
ast_str_append(&attr_candidate, -1, "relay");
break;
}
if (!ast_sockaddr_isnull(&candidate->relay_address)) {
ast_str_append(&attr_candidate, -1, " raddr %s rport", ast_sockaddr_stringify_addr_remote(&candidate->relay_address));
ast_str_append(&attr_candidate, -1, " %s", ast_sockaddr_stringify_port(&candidate->relay_address));
}
attr = pjmedia_sdp_attr_create(pool, "candidate", pj_cstr(&stmp, ast_str_buffer(attr_candidate)));
media->attr[media->attr_count++] = attr;
ast_free(attr_candidate);
}
ao2_iterator_destroy(&it_candidates);
ao2_ref(candidates, -1);
}
/*! \brief Function which processes ICE attributes in an audio stream */
static void process_ice_attributes(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_session *remote, const struct pjmedia_sdp_media *remote_stream)
{
struct ast_rtp_engine_ice *ice;
const pjmedia_sdp_attr *attr;
char attr_value[256];
unsigned int attr_i;
/* If ICE support is not enabled or available exit early */
if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
return;
}
attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-ufrag", NULL);
if (!attr) {
attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-ufrag", NULL);
}
if (attr) {
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
ice->set_authentication(session_media->rtp, attr_value, NULL);
} else {
return;
}
attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-pwd", NULL);
if (!attr) {
attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-pwd", NULL);
}
if (attr) {
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
ice->set_authentication(session_media->rtp, NULL, attr_value);
} else {
return;
}
if (pjmedia_sdp_media_find_attr2(remote_stream, "ice-lite", NULL)) {
ice->ice_lite(session_media->rtp);
}
/* Find all of the candidates */
for (attr_i = 0; attr_i < remote_stream->attr_count; ++attr_i) {
char foundation[32], transport[32], address[PJ_INET6_ADDRSTRLEN + 1], cand_type[6], relay_address[PJ_INET6_ADDRSTRLEN + 1] = "";
unsigned int port, relay_port = 0;
struct ast_rtp_engine_ice_candidate candidate = { 0, };
attr = remote_stream->attr[attr_i];
/* If this is not a candidate line skip it */
if (pj_strcmp2(&attr->name, "candidate")) {
continue;
}
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
if (sscanf(attr_value, "%31s %30u %31s %30u %46s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport,
(unsigned *)&candidate.priority, address, &port, cand_type, relay_address, &relay_port) < 7) {
/* Candidate did not parse properly */
continue;
}
if (session->endpoint->media.rtcp_mux && session_media->remote_rtcp_mux && candidate.id > 1) {
/* Remote side may have offered RTP and RTCP candidates. However, if we're using RTCP MUX,
* then we should ignore RTCP candidates.
*/
continue;
}
candidate.foundation = foundation;
candidate.transport = transport;
ast_sockaddr_parse(&candidate.address, address, PARSE_PORT_FORBID);
ast_sockaddr_set_port(&candidate.address, port);
if (!strcasecmp(cand_type, "host")) {
candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_HOST;
} else if (!strcasecmp(cand_type, "srflx")) {
candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX;
} else if (!strcasecmp(cand_type, "relay")) {
candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED;
} else {
continue;
}
if (!ast_strlen_zero(relay_address)) {
ast_sockaddr_parse(&candidate.relay_address, relay_address, PARSE_PORT_FORBID);
}
if (relay_port) {
ast_sockaddr_set_port(&candidate.relay_address, relay_port);
}
ice->add_remote_candidate(session_media->rtp, &candidate);
}
ice->set_role(session_media->rtp, pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_TRUE ?
AST_RTP_ICE_ROLE_CONTROLLING : AST_RTP_ICE_ROLE_CONTROLLED);
ice->start(session_media->rtp);
}
/*! \brief figure out if media stream has crypto lines for sdes */
static int media_stream_has_crypto(const struct pjmedia_sdp_media *stream)
{
int i;
for (i = 0; i < stream->attr_count; i++) {
pjmedia_sdp_attr *attr;
/* check the stream for the required crypto attribute */
attr = stream->attr[i];
if (pj_strcmp2(&attr->name, "crypto")) {
continue;
}
return 1;
}
return 0;
}
/*! \brief figure out media transport encryption type from the media transport string */
static enum ast_sip_session_media_encryption get_media_encryption_type(pj_str_t transport,
const struct pjmedia_sdp_media *stream, unsigned int *optimistic)
{
RAII_VAR(char *, transport_str, ast_strndup(transport.ptr, transport.slen), ast_free);
*optimistic = 0;
if (!transport_str) {
return AST_SIP_MEDIA_TRANSPORT_INVALID;
}
if (strstr(transport_str, "UDP/TLS")) {
return AST_SIP_MEDIA_ENCRYPT_DTLS;
} else if (strstr(transport_str, "SAVP")) {
return AST_SIP_MEDIA_ENCRYPT_SDES;
} else if (media_stream_has_crypto(stream)) {
*optimistic = 1;
return AST_SIP_MEDIA_ENCRYPT_SDES;
} else {
return AST_SIP_MEDIA_ENCRYPT_NONE;
}
}
/*!
* \brief Checks whether the encryption offered in SDP is compatible with the endpoint's configuration
* \internal
*
* \param endpoint_encryption Media encryption configured for the endpoint
* \param stream pjmedia_sdp_media stream description
*
* \retval AST_SIP_MEDIA_TRANSPORT_INVALID on encryption mismatch
* \retval The encryption requested in the SDP
*/
static enum ast_sip_session_media_encryption check_endpoint_media_transport(
struct ast_sip_endpoint *endpoint,
const struct pjmedia_sdp_media *stream)
{
enum ast_sip_session_media_encryption incoming_encryption;
char transport_end = stream->desc.transport.ptr[stream->desc.transport.slen - 1];
unsigned int optimistic;
if ((transport_end == 'F' && !endpoint->media.rtp.use_avpf)
|| (transport_end != 'F' && endpoint->media.rtp.use_avpf)) {
return AST_SIP_MEDIA_TRANSPORT_INVALID;
}
incoming_encryption = get_media_encryption_type(stream->desc.transport, stream, &optimistic);
if (incoming_encryption == endpoint->media.rtp.encryption) {
return incoming_encryption;
}
if (endpoint->media.rtp.force_avp ||
endpoint->media.rtp.encryption_optimistic) {
return incoming_encryption;
}
/* If an optimistic offer has been made but encryption is not enabled consider it as having
* no offer of crypto at all instead of invalid so the session proceeds.
*/
if (optimistic) {
return AST_SIP_MEDIA_ENCRYPT_NONE;
}
return AST_SIP_MEDIA_TRANSPORT_INVALID;
}
static int setup_srtp(struct ast_sip_session_media *session_media)
{
if (!session_media->srtp) {
session_media->srtp = ast_sdp_srtp_alloc();
if (!session_media->srtp) {
return -1;
}
}
if (!session_media->srtp->crypto) {
session_media->srtp->crypto = ast_sdp_crypto_alloc();
if (!session_media->srtp->crypto) {
return -1;
}
}
return 0;
}
static int setup_dtls_srtp(struct ast_sip_session *session,
struct ast_sip_session_media *session_media)
{
struct ast_rtp_engine_dtls *dtls;
if (!session->endpoint->media.rtp.dtls_cfg.enabled || !session_media->rtp) {
return -1;
}
dtls = ast_rtp_instance_get_dtls(session_media->rtp);
if (!dtls) {
return -1;
}
session->endpoint->media.rtp.dtls_cfg.suite = ((session->endpoint->media.rtp.srtp_tag_32) ? AST_AES_CM_128_HMAC_SHA1_32 : AST_AES_CM_128_HMAC_SHA1_80);
if (dtls->set_configuration(session_media->rtp, &session->endpoint->media.rtp.dtls_cfg)) {
ast_log(LOG_ERROR, "Attempted to set an invalid DTLS-SRTP configuration on RTP instance '%p'\n",
session_media->rtp);
return -1;
}
if (setup_srtp(session_media)) {
return -1;
}
return 0;
}
static void apply_dtls_attrib(struct ast_sip_session_media *session_media,
pjmedia_sdp_attr *attr)
{
struct ast_rtp_engine_dtls *dtls = ast_rtp_instance_get_dtls(session_media->rtp);
pj_str_t *value;
if (!attr->value.ptr || !dtls) {
return;
}
value = pj_strtrim(&attr->value);
if (!pj_strcmp2(&attr->name, "setup")) {
if (!pj_stricmp2(value, "active")) {
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_ACTIVE);
} else if (!pj_stricmp2(value, "passive")) {
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_PASSIVE);
} else if (!pj_stricmp2(value, "actpass")) {
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_ACTPASS);
} else if (!pj_stricmp2(value, "holdconn")) {
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_HOLDCONN);
} else {
ast_log(LOG_WARNING, "Unsupported setup attribute value '%*s'\n", (int)value->slen, value->ptr);
}
} else if (!pj_strcmp2(&attr->name, "connection")) {
if (!pj_stricmp2(value, "new")) {
dtls->reset(session_media->rtp);
} else if (!pj_stricmp2(value, "existing")) {
/* Do nothing */
} else {
ast_log(LOG_WARNING, "Unsupported connection attribute value '%*s'\n", (int)value->slen, value->ptr);
}
} else if (!pj_strcmp2(&attr->name, "fingerprint")) {
char hash_value[256], hash[32];
char fingerprint_text[value->slen + 1];
ast_copy_pj_str(fingerprint_text, value, sizeof(fingerprint_text));
if (sscanf(fingerprint_text, "%31s %255s", hash, hash_value) == 2) {
if (!strcasecmp(hash, "sha-1")) {
dtls->set_fingerprint(session_media->rtp, AST_RTP_DTLS_HASH_SHA1, hash_value);
} else if (!strcasecmp(hash, "sha-256")) {
dtls->set_fingerprint(session_media->rtp, AST_RTP_DTLS_HASH_SHA256, hash_value);
} else {
ast_log(LOG_WARNING, "Unsupported fingerprint hash type '%s'\n",
hash);
}
}
}
}
static int parse_dtls_attrib(struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_session *sdp,
const struct pjmedia_sdp_media *stream)
{
int i;
for (i = 0; i < sdp->attr_count; i++) {
apply_dtls_attrib(session_media, sdp->attr[i]);
}
for (i = 0; i < stream->attr_count; i++) {
apply_dtls_attrib(session_media, stream->attr[i]);
}
ast_set_flag(session_media->srtp, AST_SRTP_CRYPTO_OFFER_OK);
return 0;
}
static int setup_sdes_srtp(struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *stream)
{
int i;
for (i = 0; i < stream->attr_count; i++) {
pjmedia_sdp_attr *attr;
RAII_VAR(char *, crypto_str, NULL, ast_free);
/* check the stream for the required crypto attribute */
attr = stream->attr[i];
if (pj_strcmp2(&attr->name, "crypto")) {
continue;
}
crypto_str = ast_strndup(attr->value.ptr, attr->value.slen);
if (!crypto_str) {
return -1;
}
if (setup_srtp(session_media)) {
return -1;
}
if (!ast_sdp_crypto_process(session_media->rtp, session_media->srtp, crypto_str)) {
/* found a valid crypto attribute */
return 0;
}
ast_debug(1, "Ignoring crypto offer with unsupported parameters: %s\n", crypto_str);
}
/* no usable crypto attributes found */
return -1;
}
static int setup_media_encryption(struct ast_sip_session *session,
struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_session *sdp,
const struct pjmedia_sdp_media *stream)
{
switch (session_media->encryption) {
case AST_SIP_MEDIA_ENCRYPT_SDES:
if (setup_sdes_srtp(session_media, stream)) {
return -1;
}
break;
case AST_SIP_MEDIA_ENCRYPT_DTLS:
if (setup_dtls_srtp(session, session_media)) {
return -1;
}
if (parse_dtls_attrib(session_media, sdp, stream)) {
return -1;
}
break;
case AST_SIP_MEDIA_TRANSPORT_INVALID:
case AST_SIP_MEDIA_ENCRYPT_NONE:
break;
}
return 0;
}
static void set_ice_components(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
{
struct ast_rtp_engine_ice *ice;
ast_assert(session_media->rtp != NULL);
ice = ast_rtp_instance_get_ice(session_media->rtp);
if (!session->endpoint->media.rtp.ice_support || !ice) {
return;
}
if (session->endpoint->media.rtcp_mux && session_media->remote_rtcp_mux) {
/* We both support RTCP mux. Only one ICE component necessary */
ice->change_components(session_media->rtp, 1);
} else {
/* They either don't support RTCP mux or we don't know if they do yet. */
ice->change_components(session_media->rtp, 2);
}
}
/*! \brief Function which adds ssrc attributes to a media stream */
static void add_ssrc_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media)
{
pj_str_t stmp;
pjmedia_sdp_attr *attr;
char tmp[128];
if (!session->endpoint->media.bundle || session_media->bundle_group == -1) {
return;
}
snprintf(tmp, sizeof(tmp), "%u cname:%s", ast_rtp_instance_get_ssrc(session_media->rtp), ast_rtp_instance_get_cname(session_media->rtp));
attr = pjmedia_sdp_attr_create(pool, "ssrc", pj_cstr(&stmp, tmp));
media->attr[media->attr_count++] = attr;
}
/*! \brief Function which processes ssrc attributes in a stream */
static void process_ssrc_attributes(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *remote_stream)
{
int index;
if (!session->endpoint->media.bundle) {
return;
}
for (index = 0; index < remote_stream->attr_count; ++index) {
pjmedia_sdp_attr *attr = remote_stream->attr[index];
char attr_value[pj_strlen(&attr->value) + 1];
char *ssrc_attribute_name, *ssrc_attribute_value = NULL;
unsigned int ssrc;
/* We only care about ssrc attributes */
if (pj_strcmp2(&attr->name, "ssrc")) {
continue;
}
ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
if ((ssrc_attribute_name = strchr(attr_value, ' '))) {
/* This has an actual attribute */
*ssrc_attribute_name++ = '\0';
ssrc_attribute_value = strchr(ssrc_attribute_name, ':');
if (ssrc_attribute_value) {
/* Values are actually optional according to the spec */
*ssrc_attribute_value++ = '\0';
}
}
if (sscanf(attr_value, "%30u", &ssrc) < 1) {
continue;
}
/* If we are currently negotiating as a result of the remote side renegotiating then
* determine if the source for this stream has changed.
*/
if (pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_REMOTE_OFFER &&
session->active_media_state) {
struct ast_rtp_instance_stats stats = { 0, };
if (!ast_rtp_instance_get_stats(session_media->rtp, &stats, AST_RTP_INSTANCE_STAT_REMOTE_SSRC) &&
stats.remote_ssrc != ssrc) {
session_media->changed = 1;
}
}
ast_rtp_instance_set_remote_ssrc(session_media->rtp, ssrc);
}
}
static void add_msid_to_stream(struct ast_sip_session *session,
struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media,
struct ast_stream *stream)
{
pj_str_t stmp;
pjmedia_sdp_attr *attr;
char msid[(AST_UUID_STR_LEN * 2) + 2];
if (!session->endpoint->media.webrtc) {
return;
}
if (ast_strlen_zero(session_media->mslabel)) {
/* If this stream is grouped with another then use its media stream label if possible */
if (ast_stream_get_group(stream) != -1) {
struct ast_sip_session_media *group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, ast_stream_get_group(stream));
ast_copy_string(session_media->mslabel, group_session_media->mslabel, sizeof(session_media->mslabel));
}
if (ast_strlen_zero(session_media->mslabel)) {
ast_uuid_generate_str(session_media->mslabel, sizeof(session_media->mslabel));
}
}
if (ast_strlen_zero(session_media->label)) {
ast_uuid_generate_str(session_media->label, sizeof(session_media->label));
}
snprintf(msid, sizeof(msid), "%s %s", session_media->mslabel, session_media->label);
attr = pjmedia_sdp_attr_create(pool, "msid", pj_cstr(&stmp, msid));
pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
}
static void add_rtcp_fb_to_stream(struct ast_sip_session *session,
struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media)
{
pj_str_t stmp;
pjmedia_sdp_attr *attr;
if (!session->endpoint->media.webrtc || session_media->type != AST_MEDIA_TYPE_VIDEO) {
return;
}
/*
* For now just automatically add it the stream even though it hasn't
* necessarily been negotiated.
*/
attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* ccm fir"));
pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
}
/*! \brief Function which negotiates an incoming media stream */
static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
struct ast_sip_session_media *session_media, const pjmedia_sdp_session *sdp,
int index, struct ast_stream *asterisk_stream)
{
char host[NI_MAXHOST];
RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free);
pjmedia_sdp_media *stream = sdp->media[index];
struct ast_sip_session_media *session_media_transport;
enum ast_media_type media_type = session_media->type;
enum ast_sip_session_media_encryption encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
int res;
/* If no type formats have been configured reject this stream */
if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type)) {
ast_debug(3, "Endpoint has no codecs for media type '%s', declining stream\n",
ast_codec_media_type2str(session_media->type));
return 0;
}
/* Ensure incoming transport is compatible with the endpoint's configuration */
if (!session->endpoint->media.rtp.use_received_transport) {
encryption = check_endpoint_media_transport(session->endpoint, stream);
if (encryption == AST_SIP_MEDIA_TRANSPORT_INVALID) {
return -1;
}
}
ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
/* The provided host was actually invalid so we error out this negotiation */
return -1;
}
/* Using the connection information create an appropriate RTP instance */
if (!session_media->rtp && create_rtp(session, session_media)) {
return -1;
}
process_ssrc_attributes(session, session_media, stream);
session_media_transport = ast_sip_session_media_get_transport(session, session_media);
if (session_media_transport == session_media || !session_media->bundled) {
/* If this media session is carrying actual traffic then set up those aspects */
session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(stream, "rtcp-mux", NULL) != NULL);
set_ice_components(session, session_media);
enable_rtcp(session, session_media, stream);
res = setup_media_encryption(session, session_media, sdp, stream);
if (res) {
if (!session->endpoint->media.rtp.encryption_optimistic ||
!pj_strncmp2(&stream->desc.transport, "RTP/SAVP", 8)) {
/* If optimistic encryption is disabled and crypto should have been enabled
* but was not this session must fail. This must also fail if crypto was
* required in the offer but could not be set up.
*/
return -1;
}
/* There is no encryption, sad. */
session_media->encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
}
/* If we've been explicitly configured to use the received transport OR if
* encryption is on and crypto is present use the received transport.
* This is done in case of optimistic because it may come in as RTP/AVP or RTP/SAVP depending
* on the configuration of the remote endpoint (optimistic themselves or mandatory).
*/
if ((session->endpoint->media.rtp.use_received_transport) ||
((encryption == AST_SIP_MEDIA_ENCRYPT_SDES) && !res)) {
pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport);
}
} else {
/* This is bundled with another session, so mark it as such */
ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp);
enable_rtcp(session, session_media, stream);
}
if (set_caps(session, session_media, session_media_transport, stream, 1, asterisk_stream)) {
return 0;
}
return 1;
}
static int add_crypto_to_stream(struct ast_sip_session *session,
struct ast_sip_session_media *session_media,
pj_pool_t *pool, pjmedia_sdp_media *media)
{
pj_str_t stmp;
pjmedia_sdp_attr *attr;
enum ast_rtp_dtls_hash hash;
const char *crypto_attribute;
struct ast_rtp_engine_dtls *dtls;
struct ast_sdp_srtp *tmp;
static const pj_str_t STR_NEW = { "new", 3 };
static const pj_str_t STR_EXISTING = { "existing", 8 };
static const pj_str_t STR_ACTIVE = { "active", 6 };
static const pj_str_t STR_PASSIVE = { "passive", 7 };
static const pj_str_t STR_ACTPASS = { "actpass", 7 };
static const pj_str_t STR_HOLDCONN = { "holdconn", 8 };
enum ast_rtp_dtls_setup setup;
switch (session_media->encryption) {
case AST_SIP_MEDIA_ENCRYPT_NONE:
case AST_SIP_MEDIA_TRANSPORT_INVALID:
break;
case AST_SIP_MEDIA_ENCRYPT_SDES:
if (!session_media->srtp) {
session_media->srtp = ast_sdp_srtp_alloc();
if (!session_media->srtp) {
return -1;
}
}
tmp = session_media->srtp;
do {
crypto_attribute = ast_sdp_srtp_get_attrib(tmp,
0 /* DTLS running? No */,
session->endpoint->media.rtp.srtp_tag_32 /* 32 byte tag length? */);
if (!crypto_attribute) {
/* No crypto attribute to add, bad news */
return -1;
}
attr = pjmedia_sdp_attr_create(pool, "crypto",
pj_cstr(&stmp, crypto_attribute));
media->attr[media->attr_count++] = attr;
} while ((tmp = AST_LIST_NEXT(tmp, sdp_srtp_list)));
break;
case AST_SIP_MEDIA_ENCRYPT_DTLS:
if (setup_dtls_srtp(session, session_media)) {
return -1;
}
dtls = ast_rtp_instance_get_dtls(session_media->rtp);
if (!dtls) {
return -1;
}
switch (dtls->get_connection(session_media->rtp)) {
case AST_RTP_DTLS_CONNECTION_NEW:
attr = pjmedia_sdp_attr_create(pool, "connection", &STR_NEW);
media->attr[media->attr_count++] = attr;
break;
case AST_RTP_DTLS_CONNECTION_EXISTING:
attr = pjmedia_sdp_attr_create(pool, "connection", &STR_EXISTING);
media->attr[media->attr_count++] = attr;
break;
default:
break;
}
/* If this is an answer we need to use our current state, if it's an offer we need to use
* the configured value.
*/
if (pjmedia_sdp_neg_get_state(session->inv_session->neg) != PJMEDIA_SDP_NEG_STATE_DONE) {
setup = dtls->get_setup(session_media->rtp);
} else {
setup = session->endpoint->media.rtp.dtls_cfg.default_setup;
}
switch (setup) {
case AST_RTP_DTLS_SETUP_ACTIVE:
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTIVE);
media->attr[media->attr_count++] = attr;
break;
case AST_RTP_DTLS_SETUP_PASSIVE:
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_PASSIVE);
media->attr[media->attr_count++] = attr;
break;
case AST_RTP_DTLS_SETUP_ACTPASS:
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTPASS);
media->attr[media->attr_count++] = attr;
break;
case AST_RTP_DTLS_SETUP_HOLDCONN:
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_HOLDCONN);
break;
default:
break;
}
hash = dtls->get_fingerprint_hash(session_media->rtp);
crypto_attribute = dtls->get_fingerprint(session_media->rtp);
if (crypto_attribute && (hash == AST_RTP_DTLS_HASH_SHA1 || hash == AST_RTP_DTLS_HASH_SHA256)) {
RAII_VAR(struct ast_str *, fingerprint, ast_str_create(64), ast_free);
if (!fingerprint) {
return -1;
}
if (hash == AST_RTP_DTLS_HASH_SHA1) {
ast_str_set(&fingerprint, 0, "SHA-1 %s", crypto_attribute);
} else {
ast_str_set(&fingerprint, 0, "SHA-256 %s", crypto_attribute);
}
attr = pjmedia_sdp_attr_create(pool, "fingerprint", pj_cstr(&stmp, ast_str_buffer(fingerprint)));
media->attr[media->attr_count++] = attr;
}
break;
}
return 0;
}
/*! \brief Function which creates an outgoing stream */
static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_session *remote, struct ast_stream *stream)
{
pj_pool_t *pool = session->inv_session->pool_prov;
static const pj_str_t STR_RTP_AVP = { "RTP/AVP", 7 };
static const pj_str_t STR_IN = { "IN", 2 };
static const pj_str_t STR_IP4 = { "IP4", 3};
static const pj_str_t STR_IP6 = { "IP6", 3};
static const pj_str_t STR_SENDRECV = { "sendrecv", 8 };
static const pj_str_t STR_SENDONLY = { "sendonly", 8 };
pjmedia_sdp_media *media;
const char *hostip = NULL;
struct ast_sockaddr addr;
char tmp[512];
pj_str_t stmp;
pjmedia_sdp_attr *attr;
int index = 0;
int noncodec = (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->dtmf == AST_SIP_DTMF_AUTO_INFO) ? AST_RTP_DTMF : 0;
int min_packet_size = 0, max_packet_size = 0;
int rtp_code;
RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
enum ast_media_type media_type = session_media->type;
struct ast_sip_session_media *session_media_transport;
pj_sockaddr ip;
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
ast_format_cap_count(session->direct_media_cap);
media = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_media));
if (!media) {
return -1;
}
pj_strdup2(pool, &media->desc.media, ast_codec_media_type2str(session_media->type));
/* If this is a removed (or declined) stream OR if no formats exist then construct a minimal stream in SDP */
if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED || !ast_stream_get_formats(stream) ||
!ast_format_cap_count(ast_stream_get_formats(stream))) {
media->desc.port = 0;
media->desc.port_count = 1;
if (remote && remote->media[ast_stream_get_position(stream)]) {
pjmedia_sdp_media *remote_media = remote->media[ast_stream_get_position(stream)];
int index;
media->desc.transport = remote_media->desc.transport;
/* Preserve existing behavior by copying the formats provided from the offer */
for (index = 0; index < remote_media->desc.fmt_count; ++index) {
media->desc.fmt[index] = remote_media->desc.fmt[index];
}
media->desc.fmt_count = remote_media->desc.fmt_count;
} else {
/* This is actually an offer so put a dummy payload in that is ignored and sane transport */
media->desc.transport = STR_RTP_AVP;
pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], "32");
}
sdp->media[sdp->media_count++] = media;
ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
return 1;
}
if (!session_media->rtp && create_rtp(session, session_media)) {
return -1;
}
/* If this stream has not been bundled already it is new and we need to ensure there is no SSRC conflict */
if (session_media->bundle_group != -1 && !session_media->bundled) {
for (index = 0; index < sdp->media_count; ++index) {
struct ast_sip_session_media *other_session_media;
other_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
if (!other_session_media->rtp || other_session_media->bundle_group != session_media->bundle_group) {
continue;
}
if (ast_rtp_instance_get_ssrc(session_media->rtp) == ast_rtp_instance_get_ssrc(other_session_media->rtp)) {
ast_rtp_instance_change_source(session_media->rtp);
/* Start the conflict check over again */
index = -1;
continue;
}
}
}
session_media_transport = ast_sip_session_media_get_transport(session, session_media);
if (session_media_transport == session_media || !session_media->bundled) {
set_ice_components(session, session_media);
enable_rtcp(session, session_media, NULL);
/* Crypto has to be added before setting the media transport so that SRTP is properly
* set up according to the configuration. This ends up changing the media transport.
*/
if (add_crypto_to_stream(session, session_media, pool, media)) {
return -1;
}
if (pj_strlen(&session_media->transport)) {
/* If a transport has already been specified use it */
media->desc.transport = session_media->transport;
} else {
media->desc.transport = pj_str(ast_sdp_get_rtp_profile(
/* Optimistic encryption places crypto in the normal RTP/AVP profile */
!session->endpoint->media.rtp.encryption_optimistic &&
(session_media->encryption == AST_SIP_MEDIA_ENCRYPT_SDES),
session_media->rtp, session->endpoint->media.rtp.use_avpf,
session->endpoint->media.rtp.force_avp));
}
media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn));
if (!media->conn) {
return -1;
}
/* Add connection level details */
if (direct_media_enabled) {
hostip = ast_sockaddr_stringify_fmt(&session_media->direct_media_addr, AST_SOCKADDR_STR_ADDR);
} else if (ast_strlen_zero(session->endpoint->media.address)) {
hostip = ast_sip_get_host_ip_string(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET());
} else {
hostip = session->endpoint->media.address;
}
if (ast_strlen_zero(hostip)) {
ast_log(LOG_ERROR, "No local host IP available for stream %s\n",
ast_codec_media_type2str(session_media->type));
return -1;
}
media->conn->net_type = STR_IN;
/* Assume that the connection will use IPv4 until proven otherwise */
media->conn->addr_type = STR_IP4;
pj_strdup2(pool, &media->conn->addr, hostip);
if ((pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &media->conn->addr, &ip) == PJ_SUCCESS) &&
(ip.addr.sa_family == pj_AF_INET6())) {
media->conn->addr_type = STR_IP6;
}
/* Add ICE attributes and candidates */
add_ice_to_stream(session, session_media, pool, media, 1);
ast_rtp_instance_get_local_address(session_media->rtp, &addr);
media->desc.port = direct_media_enabled ? ast_sockaddr_port(&session_media->direct_media_addr) : (pj_uint16_t) ast_sockaddr_port(&addr);
media->desc.port_count = 1;
} else {
pjmedia_sdp_media *bundle_group_stream = sdp->media[session_media_transport->stream_num];
/* As this is in a bundle group it shares the same details as the group instance */
media->desc.transport = bundle_group_stream->desc.transport;
media->conn = bundle_group_stream->conn;
media->desc.port = bundle_group_stream->desc.port;
if (add_crypto_to_stream(session, session_media_transport, pool, media)) {
return -1;
}
add_ice_to_stream(session, session_media_transport, pool, media, 0);
enable_rtcp(session, session_media, NULL);
}
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n",
ast_codec_media_type2str(session_media->type));
return -1;
}
if (direct_media_enabled) {
ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
} else {
ast_format_cap_append_from_cap(caps, ast_stream_get_formats(stream), media_type);
}
for (index = 0; index < ast_format_cap_count(caps); ++index) {
struct ast_format *format = ast_format_cap_get_format(caps, index);
if (ast_format_get_type(format) != media_type) {
ao2_ref(format, -1);
continue;
}
/* If this stream is not a transport we need to use the transport codecs structure for payload management to prevent
* conflicts.
*/
if (session_media_transport != session_media) {
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media_transport->rtp), 1, format, 0)) == -1) {
ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
ao2_ref(format, -1);
continue;
}
/* Our instance has to match the payload number though */
ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media->rtp), rtp_code, format);
} else {
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0)) == -1) {
ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
ao2_ref(format, -1);
continue;
}
}
if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
media->attr[media->attr_count++] = attr;
}
if ((attr = generate_fmtp_attr(pool, format, rtp_code))) {
media->attr[media->attr_count++] = attr;
}
if (ast_format_get_maximum_ms(format) &&
((ast_format_get_maximum_ms(format) < max_packet_size) || !max_packet_size)) {
max_packet_size = ast_format_get_maximum_ms(format);
}
ao2_ref(format, -1);
if (media->desc.fmt_count == PJMEDIA_MAX_SDP_FMT) {
break;
}
}
/* Add non-codec formats */
if (ast_sip_session_is_pending_stream_default(session, stream) && media_type != AST_MEDIA_TYPE_VIDEO
&& media->desc.fmt_count < PJMEDIA_MAX_SDP_FMT) {
for (index = 1LL; index <= AST_RTP_MAX; index <<= 1) {
if (!(noncodec & index)) {
continue;
}
rtp_code = ast_rtp_codecs_payload_code(
ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index);
if (rtp_code == -1) {
continue;
}
if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 0, NULL, index))) {
media->attr[media->attr_count++] = attr;
}
if (index == AST_RTP_DTMF) {
snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code);
attr = pjmedia_sdp_attr_create(pool, "fmtp", pj_cstr(&stmp, tmp));
media->attr[media->attr_count++] = attr;
}
if (media->desc.fmt_count == PJMEDIA_MAX_SDP_FMT) {
break;
}
}
}
/* If no formats were actually added to the media stream don't add it to the SDP */
if (!media->desc.fmt_count) {
return 1;
}
/* If ptime is set add it as an attribute */
min_packet_size = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(session_media->rtp));
if (!min_packet_size) {
min_packet_size = ast_format_cap_get_framing(caps);
}
if (min_packet_size) {
snprintf(tmp, sizeof(tmp), "%d", min_packet_size);
attr = pjmedia_sdp_attr_create(pool, "ptime", pj_cstr(&stmp, tmp));
media->attr[media->attr_count++] = attr;
}
if (max_packet_size) {
snprintf(tmp, sizeof(tmp), "%d", max_packet_size);
attr = pjmedia_sdp_attr_create(pool, "maxptime", pj_cstr(&stmp, tmp));
media->attr[media->attr_count++] = attr;
}
/* Add the sendrecv attribute - we purposely don't keep track because pjmedia-sdp will automatically change our offer for us */
attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
attr->name = !session_media->locally_held ? STR_SENDRECV : STR_SENDONLY;
media->attr[media->attr_count++] = attr;
/* If we've got rtcp-mux enabled, just unconditionally offer it in all SDPs */
if (session->endpoint->media.rtcp_mux) {
attr = pjmedia_sdp_attr_create(pool, "rtcp-mux", NULL);
pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
}
add_ssrc_to_stream(session, session_media, pool, media);
add_msid_to_stream(session, session_media, pool, media, stream);
add_rtcp_fb_to_stream(session, session_media, pool, media);
/* Add the media stream to the SDP */
sdp->media[sdp->media_count++] = media;
return 1;
}
static struct ast_frame *media_session_rtp_read_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
{
struct ast_frame *f;
if (!session_media->rtp) {
return &ast_null_frame;
}
f = ast_rtp_instance_read(session_media->rtp, 0);
if (!f) {
return NULL;
}
ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
return f;
}
static struct ast_frame *media_session_rtcp_read_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
{
struct ast_frame *f;
if (!session_media->rtp) {
return &ast_null_frame;
}
f = ast_rtp_instance_read(session_media->rtp, 1);
if (!f) {
return NULL;
}
ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
return f;
}
static int media_session_rtp_write_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media, struct ast_frame *frame)
{
if (!session_media->rtp) {
return 0;
}
return ast_rtp_instance_write(session_media->rtp, frame);
}
static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *local,
const struct pjmedia_sdp_session *remote, int index, struct ast_stream *asterisk_stream)
{
RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free);
struct pjmedia_sdp_media *remote_stream = remote->media[index];
enum ast_media_type media_type = session_media->type;
char host[NI_MAXHOST];
int res;
struct ast_sip_session_media *session_media_transport;
if (!session->channel) {
return 1;
}
/* Ensure incoming transport is compatible with the endpoint's configuration */
if (!session->endpoint->media.rtp.use_received_transport &&
check_endpoint_media_transport(session->endpoint, remote_stream) == AST_SIP_MEDIA_TRANSPORT_INVALID) {
return -1;
}
/* Create an RTP instance if need be */
if (!session_media->rtp && create_rtp(session, session_media)) {
return -1;
}
process_ssrc_attributes(session, session_media, remote_stream);
session_media_transport = ast_sip_session_media_get_transport(session, session_media);
if (session_media_transport == session_media || !session_media->bundled) {
session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(remote_stream, "rtcp-mux", NULL) != NULL);
set_ice_components(session, session_media);
enable_rtcp(session, session_media, remote_stream);
res = setup_media_encryption(session, session_media, remote, remote_stream);
if (!session->endpoint->media.rtp.encryption_optimistic && res) {
/* If optimistic encryption is disabled and crypto should have been enabled but was not
* this session must fail.
*/
return -1;
}
if (!remote_stream->conn && !remote->conn) {
return 1;
}
ast_copy_pj_str(host, remote_stream->conn ? &remote_stream->conn->addr : &remote->conn->addr, sizeof(host));
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
/* The provided host was actually invalid so we error out this negotiation */
return -1;
}
/* Apply connection information to the RTP instance */
ast_sockaddr_set_port(addrs, remote_stream->desc.port);
ast_rtp_instance_set_remote_address(session_media->rtp, addrs);
ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback);
ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 0),
media_session_rtp_read_callback);
if (!session->endpoint->media.rtcp_mux || !session_media->remote_rtcp_mux) {
ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 1),
media_session_rtcp_read_callback);
}
/* If ICE support is enabled find all the needed attributes */
process_ice_attributes(session, session_media, remote, remote_stream);
} else {
/* This is bundled with another session, so mark it as such */
ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp);
ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback);
enable_rtcp(session, session_media, remote_stream);
}
if (set_caps(session, session_media, session_media_transport, remote_stream, 0, asterisk_stream)) {
return 1;
}
/* Set the channel uniqueid on the RTP instance now that it is becoming active */
ast_channel_lock(session->channel);
ast_rtp_instance_set_channel_id(session_media->rtp, ast_channel_uniqueid(session->channel));
ast_channel_unlock(session->channel);
/* Ensure the RTP instance is active */
ast_rtp_instance_set_stream_num(session_media->rtp, ast_stream_get_position(asterisk_stream));
ast_rtp_instance_activate(session_media->rtp);
/* audio stream handles music on hold */
if (media_type != AST_MEDIA_TYPE_AUDIO) {
if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
}
return 1;
}
if (ast_sockaddr_isnull(addrs) ||
ast_sockaddr_is_any(addrs) ||
pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL) ||
pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
if (!session_media->remotely_held) {
/* The remote side has put us on hold */
ast_queue_hold(session->channel, session->endpoint->mohsuggest);
ast_rtp_instance_stop(session_media->rtp);
ast_queue_frame(session->channel, &ast_null_frame);
session_media->remotely_held = 1;
}
} else if (session_media->remotely_held) {
/* The remote side has taken us off hold */
ast_queue_unhold(session->channel);
ast_queue_frame(session->channel, &ast_null_frame);
session_media->remotely_held = 0;
} else if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
}
/* This purposely resets the encryption to the configured in case it gets added later */
session_media->encryption = session->endpoint->media.rtp.encryption;
if (session->endpoint->media.rtp.keepalive > 0 &&
session_media->type == AST_MEDIA_TYPE_AUDIO) {
ast_rtp_instance_set_keepalive(session_media->rtp, session->endpoint->media.rtp.keepalive);
/* Schedule the initial keepalive early in case this is being used to punch holes through
* a NAT. This way there won't be an awkward delay before media starts flowing in some
* scenarios.
*/
AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
session_media->keepalive_sched_id = ast_sched_add_variable(sched, 500, send_keepalive,
session_media, 1);
}
/* As the channel lock is not held during this process the scheduled item won't block if
* it is hanging up the channel at the same point we are applying this negotiated SDP.
*/
AST_SCHED_DEL(sched, session_media->timeout_sched_id);
/* Due to the fact that we only ever have one scheduled timeout item for when we are both
* off hold and on hold we don't need to store the two timeouts differently on the RTP
* instance itself.
*/
ast_rtp_instance_set_timeout(session_media->rtp, 0);
if (session->endpoint->media.rtp.timeout && !session_media->remotely_held) {
ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout);
} else if (session->endpoint->media.rtp.timeout_hold && session_media->remotely_held) {
ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout_hold);
}
if (ast_rtp_instance_get_timeout(session_media->rtp)) {
session_media->timeout_sched_id = ast_sched_add_variable(sched,
ast_rtp_instance_get_timeout(session_media->rtp) * 1000, rtp_check_timeout,
session_media, 1);
}
return 1;
}
/*! \brief Function which updates the media stream with external media address, if applicable */
static void change_outgoing_sdp_stream_media_address(pjsip_tx_data *tdata, struct pjmedia_sdp_media *stream, struct ast_sip_transport *transport)
{
RAII_VAR(struct ast_sip_transport_state *, transport_state, ast_sip_get_transport_state(ast_sorcery_object_get_id(transport)), ao2_cleanup);
char host[NI_MAXHOST];
struct ast_sockaddr our_sdp_addr = { { 0, } };
/* If the stream has been rejected there will be no connection line */
if (!stream->conn || !transport_state) {
return;
}
ast_copy_pj_str(host, &stream->conn->addr, sizeof(host));
ast_sockaddr_parse(&our_sdp_addr, host, PARSE_PORT_FORBID);
/* Reversed check here. We don't check the remote endpoint being
* in our local net, but whether our outgoing session IP is
* local. If it is not, we won't do rewriting. No localnet
* configured? Always rewrite. */
if (ast_sip_transport_is_nonlocal(transport_state, &our_sdp_addr) && transport_state->localnet) {
return;
}
ast_debug(5, "Setting media address to %s\n", ast_sockaddr_stringify_host(&transport_state->external_media_address));
pj_strdup2(tdata->pool, &stream->conn->addr, ast_sockaddr_stringify_host(&transport_state->external_media_address));
}
/*! \brief Function which stops the RTP instance */
static void stream_stop(struct ast_sip_session_media *session_media)
{
if (!session_media->rtp) {
return;
}
AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
AST_SCHED_DEL(sched, session_media->timeout_sched_id);
ast_rtp_instance_stop(session_media->rtp);
}
/*! \brief Function which destroys the RTP instance when session ends */
static void stream_destroy(struct ast_sip_session_media *session_media)
{
if (session_media->rtp) {
stream_stop(session_media);
ast_rtp_instance_destroy(session_media->rtp);
}
session_media->rtp = NULL;
}
/*! \brief SDP handler for 'audio' media stream */
static struct ast_sip_session_sdp_handler audio_sdp_handler = {
.id = STR_AUDIO,
.negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
.create_outgoing_sdp_stream = create_outgoing_sdp_stream,
.apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
.change_outgoing_sdp_stream_media_address = change_outgoing_sdp_stream_media_address,
.stream_stop = stream_stop,
.stream_destroy = stream_destroy,
};
/*! \brief SDP handler for 'video' media stream */
static struct ast_sip_session_sdp_handler video_sdp_handler = {
.id = STR_VIDEO,
.negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
.create_outgoing_sdp_stream = create_outgoing_sdp_stream,
.apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
.change_outgoing_sdp_stream_media_address = change_outgoing_sdp_stream_media_address,
.stream_stop = stream_stop,
.stream_destroy = stream_destroy,
};
static int video_info_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
struct pjsip_transaction *tsx;
pjsip_tx_data *tdata;
if (!session->channel
|| !ast_sip_is_content_type(&rdata->msg_info.msg->body->content_type,
"application",
"media_control+xml")) {
return 0;
}
tsx = pjsip_rdata_get_tsx(rdata);
ast_queue_control(session->channel, AST_CONTROL_VIDUPDATE);
if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, 200, NULL, &tdata) == PJ_SUCCESS) {
pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
}
return 0;
}
static struct ast_sip_session_supplement video_info_supplement = {
.method = "INFO",
.incoming_request = video_info_incoming_request,
};
/*! \brief Unloads the sdp RTP/AVP module from Asterisk */
static int unload_module(void)
{
ast_sip_session_unregister_supplement(&video_info_supplement);
ast_sip_session_unregister_sdp_handler(&video_sdp_handler, STR_VIDEO);
ast_sip_session_unregister_sdp_handler(&audio_sdp_handler, STR_AUDIO);
if (sched) {
ast_sched_context_destroy(sched);
}
return 0;
}
/*!
* \brief Load the module
*
* Module loading including tests for configuration or dependencies.
* This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
* or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
* tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
* configuration file or other non-critical problem return
* AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
*/
static int load_module(void)
{
CHECK_PJSIP_SESSION_MODULE_LOADED();
if (ast_check_ipv6()) {
ast_sockaddr_parse(&address_rtp, "::", 0);
} else {
ast_sockaddr_parse(&address_rtp, "0.0.0.0", 0);
}
if (!(sched = ast_sched_context_create())) {
ast_log(LOG_ERROR, "Unable to create scheduler context.\n");
goto end;
}
if (ast_sched_start_thread(sched)) {
ast_log(LOG_ERROR, "Unable to create scheduler context thread.\n");
goto end;
}
if (ast_sip_session_register_sdp_handler(&audio_sdp_handler, STR_AUDIO)) {
ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_AUDIO);
goto end;
}
if (ast_sip_session_register_sdp_handler(&video_sdp_handler, STR_VIDEO)) {
ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_VIDEO);
goto end;
}
ast_sip_session_register_supplement(&video_info_supplement);
return AST_MODULE_LOAD_SUCCESS;
end:
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP SDP RTP/AVP stream handler",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);
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