res_pjsip_sdp_rtp.c

/*
 * 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 Enable an RTP extension on an RTP session.
 */
static void enable_rtp_extension(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
	enum ast_rtp_extension extension, enum ast_rtp_extension_direction direction,
	const pjmedia_sdp_session *sdp)
{
	int id = -1;

	/* For a bundle group the local unique identifier space is shared across all streams within
	 * it.
	 */
	if (session_media->bundle_group != -1) {
		int index;

		for (index = 0; index < sdp->media_count; ++index) {
			struct ast_sip_session_media *other_session_media;
			int other_id;

			if (index >= AST_VECTOR_SIZE(&session->pending_media_state->sessions)) {
				break;
			}

			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;
			}

			other_id = ast_rtp_instance_extmap_get_id(other_session_media->rtp, extension);
			if (other_id == -1) {
				/* Worst case we have to fall back to the highest available free local unique identifier
				 * for the bundle group.
				 */
				other_id = ast_rtp_instance_extmap_count(other_session_media->rtp) + 1;
				if (id < other_id) {
					id = other_id;
				}
				continue;
			}

			id = other_id;
			break;
		}
	}

	ast_rtp_instance_extmap_enable(session_media->rtp, id, extension, direction);
}

/*! \brief Internal function which creates an RTP instance */
static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
	const pjmedia_sdp_session *sdp)
{
	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) {
		ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_RECV, session->endpoint->media.webrtc);
		ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_SEND, session->endpoint->media.webrtc);
		ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_REMB, session->endpoint->media.webrtc);
		if (session->endpoint->media.webrtc) {
			enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
		}
		if (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)
{
#ifndef HAVE_PJSIP_ENDPOINT_COMPACT_FORM
	extern pj_bool_t pjsip_use_compact_form;
#else
	pj_bool_t pjsip_use_compact_form = pjsip_cfg()->endpt.use_compact_form;
#endif
	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 (!session_media->remote_ice) {
		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 checks for ice attributes in an audio stream */
static void check_ice_support(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
				   const struct pjmedia_sdp_media *remote_stream)
{
	struct ast_rtp_engine_ice *ice;
	const pjmedia_sdp_attr *attr;
	unsigned int attr_i;

	if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
		session_media->remote_ice = 0;
		return;
	}

	/* Find all of the candidates */
	for (attr_i = 0; attr_i < remote_stream->attr_count; ++attr_i) {
		attr = remote_stream->attr[attr_i];
		if (!pj_strcmp2(&attr->name, "candidate")) {
			session_media->remote_ice = 1;
			break;
		}
	}

	if (attr_i == remote_stream->attr_count) {
		session_media->remote_ice = 0;
	}
}

/*! \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[33], 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, "%32s %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)) {