Merge the pimp_my_sip branch into trunk.

The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.

SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.

API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 848 insertions, 0 deletions

diff --git a/res/res_sip_sdp_rtp.c b/res/res_sip_sdp_rtp.c
new file mode 100644
index 000000000..13e6aa1aa
--- /dev/null
+++ b/res/res_sip_sdp_rtp.c
@@ -0,0 +1,848 @@
+/*
+ * 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_sip</depend>
+	<depend>res_sip_session</depend>
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+#include <pjsip_ua.h>
+#include <pjmedia.h>
+#include <pjlib.h>
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/module.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/res_sip.h"
+#include "asterisk/res_sip_session.h"
+
+/*! \brief Scheduler for RTCP purposes */
+static struct ast_sched_context *sched;
+
+/*! \brief Address for IPv4 RTP */
+static struct ast_sockaddr address_ipv4;
+
+/*! \brief Address for IPv6 RTP */
+static struct ast_sockaddr address_ipv6;
+
+static const char STR_AUDIO[] = "audio";
+static const int FD_AUDIO = 0;
+
+static const char STR_VIDEO[] = "video";
+static const int FD_VIDEO = 2;
+
+/*! \brief Retrieves an ast_format_type based on the given stream_type */
+static enum ast_format_type stream_to_media_type(const char *stream_type)
+{
+	if (!strcasecmp(stream_type, STR_AUDIO)) {
+		return AST_FORMAT_TYPE_AUDIO;
+	} else if (!strcasecmp(stream_type, STR_VIDEO)) {
+		return AST_FORMAT_TYPE_VIDEO;
+	}
+
+	return 0;
+}
+
+/*! \brief Get the starting descriptor for a media type */
+static int media_type_to_fdno(enum ast_format_type media_type)
+{
+	switch (media_type) {
+	case AST_FORMAT_TYPE_AUDIO: return FD_AUDIO;
+	case AST_FORMAT_TYPE_VIDEO: return FD_VIDEO;
+	case AST_FORMAT_TYPE_TEXT:
+	case AST_FORMAT_TYPE_IMAGE: break;
+	}
+	return -1;
+}
+
+/*! \brief Remove all other cap types but the one given */
+static void format_cap_only_type(struct ast_format_cap *caps, enum ast_format_type media_type)
+{
+	int i = AST_FORMAT_INC;
+	while (i <= AST_FORMAT_TYPE_TEXT) {
+		if (i != media_type) {
+			ast_format_cap_remove_bytype(caps, i);
+		}
+		i += AST_FORMAT_INC;
+	}
+}
+
+/*! \brief Internal function which creates an RTP instance */
+static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
+{
+	struct ast_rtp_engine_ice *ice;
+
+	if (!(session_media->rtp = ast_rtp_instance_new("asterisk", sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
+		return -1;
+	}
+
+	ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RTCP, 1);
+	ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->rtp_symmetric);
+
+	ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session_media->rtp),
+					 session_media->rtp, &session->endpoint->prefs);
+
+	if (!session->endpoint->ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
+		ice->stop(session_media->rtp);
+	}
+
+	return 0;
+}
+
+static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs)
+{
+	pjmedia_sdp_attr *attr;
+	pjmedia_sdp_rtpmap *rtpmap;
+	pjmedia_sdp_fmtp fmtp;
+	struct ast_format *format;
+	int i, num = 0;
+	char name[256];
+	char media[20];
+	char fmt_param[256];
+
+	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));
+		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, 0, 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) {
+			sscanf(pj_strbuf(&fmtp.fmt), "%d", &num);
+			if ((format = ast_rtp_codecs_get_payload_format(codecs, num))) {
+				ast_copy_pj_str(fmt_param, &fmtp.fmt_param, sizeof(fmt_param));
+				ast_format_sdp_parse(format, fmt_param);
+			}
+		}
+	}
+}
+
+static int set_caps(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
+		    const struct pjmedia_sdp_media *stream)
+{
+	RAII_VAR(struct ast_format_cap *, caps, NULL, ast_format_cap_destroy);
+	RAII_VAR(struct ast_format_cap *, peer, NULL, ast_format_cap_destroy);
+	RAII_VAR(struct ast_format_cap *, joint, NULL, ast_format_cap_destroy);
+	enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
+	struct ast_rtp_codecs codecs;
+	struct ast_format fmt;
+	int fmts = 0;
+	int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
+		!ast_format_cap_is_empty(session->direct_media_cap);
+
+	if (!(caps = ast_format_cap_alloc_nolock()) ||
+	    !(peer = ast_format_cap_alloc_nolock())) {
+		ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", session_media->stream_type);
+		return -1;
+	}
+
+	/* get the endpoint capabilities */
+	if (direct_media_enabled) {
+		ast_format_cap_joint_copy(session->endpoint->codecs, session->direct_media_cap, caps);
+	} else {
+		ast_format_cap_copy(caps, session->endpoint->codecs);
+	}
+	format_cap_only_type(caps, media_type);
+
+	/* get the capabilities on the peer */
+	get_codecs(session, stream, &codecs);
+	ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
+
+	/* get the joint capabilities between peer and endpoint */
+	if (!(joint = ast_format_cap_joint(caps, peer))) {
+		char usbuf[64], thembuf[64];
+
+		ast_rtp_codecs_payloads_destroy(&codecs);
+
+		ast_getformatname_multiple(usbuf, sizeof(usbuf), caps);
+		ast_getformatname_multiple(thembuf, sizeof(thembuf), peer);
+		ast_log(LOG_WARNING, "No joint capabilities between our configuration(%s) and incoming SDP(%s)\n", usbuf, thembuf);
+		return -1;
+	}
+
+	ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
+				     session_media->rtp);
+
+	ast_format_cap_copy(caps, session->req_caps);
+	ast_format_cap_remove_bytype(caps, media_type);
+	ast_format_cap_append(caps, joint);
+	ast_format_cap_append(session->req_caps, caps);
+
+	if (session->channel) {
+		ast_format_cap_copy(caps, ast_channel_nativeformats(session->channel));
+		ast_format_cap_remove_bytype(caps, media_type);
+		ast_format_cap_append(caps, joint);
+
+		/* Apply the new formats to the channel, potentially changing read/write formats while doing so */
+		ast_format_cap_append(ast_channel_nativeformats(session->channel), caps);
+		ast_codec_choose(&session->endpoint->prefs, caps, 0, &fmt);
+		ast_format_copy(ast_channel_rawwriteformat(session->channel), &fmt);
+		ast_format_copy(ast_channel_rawreadformat(session->channel), &fmt);
+		ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
+		ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
+	}
+
+	ast_rtp_codecs_payloads_destroy(&codecs);
+	return 1;
+}
+
+static pjmedia_sdp_attr* generate_rtpmap_attr(pjmedia_sdp_media *media, pj_pool_t *pool, int rtp_code,
+					      int asterisk_format, struct ast_format *format, int code)
+{
+	pjmedia_sdp_rtpmap rtpmap;
+	pjmedia_sdp_attr *attr = NULL;
+	char tmp[64];
+
+	snprintf(tmp, sizeof(tmp), "%d", rtp_code);
+	pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
+	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, 0));
+	rtpmap.param.slen = 0;
+
+	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_sdp_generate(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) {
+			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)
+{
+	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->ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp)) ||
+		!(candidates = ice->get_local_candidates(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;
+	}
+
+	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 %d %s %d %s ", candidate->foundation, candidate->id, candidate->transport,
+					candidate->priority, ast_sockaddr_stringify_host(&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_host(&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);
+}
+
+/*! \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->ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
+		return;
+	}
+
+	if ((attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-ufrag", NULL))) {
+		ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
+		ice->set_authentication(session_media->rtp, attr_value, NULL);
+	}
+
+	if ((attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-pwd", NULL))) {
+		ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
+		ice->set_authentication(session_media->rtp, NULL, attr_value);
+	}
+
+	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] = "";
+		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,
+			&candidate.priority, address, &port, cand_type, relay_address, &relay_port) < 7) {
+			/* Candidate did not parse properly */
+			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->start(session_media->rtp);
+}
+
+static void apply_packetization(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
+			 const struct pjmedia_sdp_media *remote_stream)
+{
+	pjmedia_sdp_attr *attr;
+	pj_str_t value;
+	unsigned long framing;
+	int codec;
+	struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(session_media->rtp)->pref;
+
+	/* Apply packetization if available and configured to do so */
+	if (!session->endpoint->use_ptime || !(attr = pjmedia_sdp_media_find_attr2(remote_stream, "ptime", NULL))) {
+		return;
+	}
+
+	value = attr->value;
+	framing = pj_strtoul(pj_strltrim(&value));
+
+	for (codec = 0; codec < AST_RTP_MAX_PT; codec++) {
+		struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(
+											   session_media->rtp), codec);
+
+		if (!format.asterisk_format) {
+			continue;
+		}
+
+		ast_codec_pref_setsize(pref, &format.format, framing);
+	}
+
+	ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session_media->rtp),
+					 session_media->rtp, pref);
+}
+
+/*! \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 struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
+{
+	char host[NI_MAXHOST];
+	RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
+	enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
+
+	/* If no type formats have been configured reject this stream */
+	if (!ast_format_cap_has_type(session->endpoint->codecs, media_type)) {
+		return 0;
+	}
+
+	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, ast_sockaddr_is_ipv6(addrs))) {
+		return -1;
+	}
+
+	return set_caps(session, session_media, stream);
+}
+
+/*! \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)
+{
+	pj_pool_t *pool = session->inv_session->pool_prov;
+	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_RTP_AVP = { "RTP/AVP", 7 };
+	static const pj_str_t STR_SENDRECV = { "sendrecv", 8 };
+	pjmedia_sdp_media *media;
+	char hostip[PJ_INET6_ADDRSTRLEN+2];
+	struct ast_sockaddr addr;
+	char tmp[512];
+	pj_str_t stmp;
+	pjmedia_sdp_attr *attr;
+	int index = 0, min_packet_size = 0, noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0;
+	int rtp_code;
+	struct ast_format format;
+	struct ast_format compat_format;
+	RAII_VAR(struct ast_format_cap *, caps, NULL, ast_format_cap_destroy);
+	enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
+
+	int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
+		!ast_format_cap_is_empty(session->direct_media_cap);
+
+	if (!ast_format_cap_has_type(session->endpoint->codecs, media_type)) {
+		/* If no type formats are configured don't add a stream */
+		return 0;
+	} else if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->rtp_ipv6)) {
+		return -1;
+	}
+
+	if (!(media = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_media))) ||
+		!(media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn)))) {
+		return -1;
+	}
+
+	/* TODO: This should eventually support SRTP */
+	media->desc.media = pj_str(session_media->stream_type);
+	media->desc.transport = STR_RTP_AVP;
+
+	/* Add connection level details */
+	if (direct_media_enabled) {
+		ast_copy_string(hostip, ast_sockaddr_stringify_fmt(&session_media->direct_media_addr, AST_SOCKADDR_STR_ADDR), sizeof(hostip));
+	} else if (ast_strlen_zero(session->endpoint->external_media_address)) {
+		pj_sockaddr localaddr;
+
+		if (pj_gethostip(session->endpoint->rtp_ipv6 ? pj_AF_INET6() : pj_AF_INET(), &localaddr)) {
+			return -1;
+		}
+		pj_sockaddr_print(&localaddr, hostip, sizeof(hostip), 2);
+	} else {
+		ast_copy_string(hostip, session->endpoint->external_media_address, sizeof(hostip));
+	}
+
+	media->conn->net_type = STR_IN;
+	media->conn->addr_type = session->endpoint->rtp_ipv6 ? STR_IP6 : STR_IP4;
+	pj_strdup2(pool, &media->conn->addr, hostip);
+	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;
+
+	/* Add ICE attributes and candidates */
+	add_ice_to_stream(session, session_media, pool, media);
+
+	if (!(caps = ast_format_cap_alloc_nolock())) {
+		ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", session_media->stream_type);
+		return -1;
+	}
+
+	if (direct_media_enabled) {
+		ast_format_cap_joint_copy(session->endpoint->codecs, session->direct_media_cap, caps);
+	} else if (ast_format_cap_is_empty(session->req_caps)) {
+		ast_format_cap_copy(caps, session->endpoint->codecs);
+	} else {
+		ast_format_cap_joint_copy(session->endpoint->codecs, session->req_caps, caps);
+	}
+
+	for (index = 0; ast_codec_pref_index(&session->endpoint->prefs, index, &format); ++index) {
+		struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(session_media->rtp)->pref;
+
+		if (AST_FORMAT_GET_TYPE(format.id) != media_type) {
+			continue;
+		}
+
+		if (!ast_format_cap_get_compatible_format(caps, &format, &compat_format)) {
+			continue;
+		}
+
+		if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, &compat_format, 0)) == -1) {
+			return -1;
+		}
+
+		if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 1, &compat_format, 0))) {
+			continue;
+		}
+
+		media->attr[media->attr_count++] = attr;
+
+		if ((attr = generate_fmtp_attr(pool, &compat_format, rtp_code))) {
+			media->attr[media->attr_count++] = attr;
+		}
+
+		if (pref && media_type != AST_FORMAT_TYPE_VIDEO) {
+			struct ast_format_list fmt = ast_codec_pref_getsize(pref, &compat_format);
+			if (fmt.cur_ms && ((fmt.cur_ms < min_packet_size) || !min_packet_size)) {
+				min_packet_size = fmt.cur_ms;
+			}
+		}
+	}
+
+	/* Add non-codec formats */
+	if (media_type != AST_FORMAT_TYPE_VIDEO) {
+		for (index = 1LL; index <= AST_RTP_MAX; index <<= 1) {
+			if (!(noncodec & index) || (rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp),
+											   0, NULL, index)) == -1) {
+				continue;
+			}
+
+			if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 0, NULL, index))) {
+				continue;
+			}
+
+			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 ptime is set add it as an attribute */
+	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;
+	}
+
+	/* 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 = STR_SENDRECV;
+	media->attr[media->attr_count++] = attr;
+
+	/* Add the media stream to the SDP */
+	sdp->media[sdp->media_count++] = media;
+
+	return 1;
+}
+
+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_media *local_stream,
+				       const struct pjmedia_sdp_session *remote, const struct pjmedia_sdp_media *remote_stream)
+{
+	RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
+	enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
+	char host[NI_MAXHOST];
+	int fdno;
+
+	if (!session->channel) {
+		return 1;
+	}
+
+	/* Create an RTP instance if need be */
+	if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->rtp_ipv6)) {
+		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);
+
+	if (set_caps(session, session_media, local_stream) < 1) {
+		return -1;
+	}
+
+	if (media_type == AST_FORMAT_TYPE_AUDIO) {
+		apply_packetization(session, session_media, remote_stream);
+	}
+
+	if ((fdno = media_type_to_fdno(media_type)) < 0) {
+		return -1;
+	}
+	ast_channel_set_fd(session->channel, fdno, ast_rtp_instance_fd(session_media->rtp, 0));
+	ast_channel_set_fd(session->channel, fdno + 1, ast_rtp_instance_fd(session_media->rtp, 1));
+
+	/* If ICE support is enabled find all the needed attributes */
+	process_ice_attributes(session, session_media, remote, remote_stream);
+
+	/* audio stream handles music on hold */
+	if (media_type != AST_FORMAT_TYPE_AUDIO) {
+		return 1;
+	}
+
+	/* Music on hold for audio streams only */
+	if (session_media->held &&
+	    (!ast_sockaddr_isnull(addrs) ||
+	     !pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL))) {
+		/* The remote side has taken us off hold */
+		ast_queue_control(session->channel, AST_CONTROL_UNHOLD);
+		ast_queue_frame(session->channel, &ast_null_frame);
+		session_media->held = 0;
+	} else if (ast_sockaddr_isnull(addrs) ||
+		   ast_sockaddr_is_any(addrs) ||
+		   pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
+		/* The remote side has put us on hold */
+		ast_queue_control_data(session->channel, AST_CONTROL_HOLD, S_OR(session->endpoint->mohsuggest, NULL),
+				       !ast_strlen_zero(session->endpoint->mohsuggest) ? strlen(session->endpoint->mohsuggest) + 1 : 0);
+		ast_rtp_instance_stop(session_media->rtp);
+		ast_queue_frame(session->channel, &ast_null_frame);
+		session_media->held = 1;
+	} else {
+		/* The remote side has not changed state, but make sure the instance is active */
+		ast_rtp_instance_activate(session_media->rtp);
+	}
+
+	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)
+{
+	char host[NI_MAXHOST];
+	struct ast_sockaddr addr = { { 0, } };
+
+	ast_copy_pj_str(host, &stream->conn->addr, sizeof(host));
+	ast_sockaddr_parse(&addr, host, PARSE_PORT_FORBID);
+
+	/* Is the address within the SDP inside the same network? */
+	if (ast_apply_ha(transport->localnet, &addr) == AST_SENSE_ALLOW) {
+		return;
+	}
+
+	pj_strdup2(tdata->pool, &stream->conn->addr, transport->external_media_address);
+}
+
+/*! \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) {
+		ast_rtp_instance_stop(session_media->rtp);
+		ast_rtp_instance_destroy(session_media->rtp);
+	}
+}
+
+/*! \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_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_destroy = stream_destroy,
+};
+
+static int video_info_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+	struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+	pjsip_tx_data *tdata;
+
+	if (pj_strcmp2(&rdata->msg_info.msg->body->content_type.type, "application") ||
+	    pj_strcmp2(&rdata->msg_info.msg->body->content_type.subtype, "media_control+xml")) {
+
+		return 0;
+	}
+
+	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)
+{
+	ast_sockaddr_parse(&address_ipv4, "0.0.0.0", 0);
+	ast_sockaddr_parse(&address_ipv6, "::", 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_FAILURE;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP SDP RTP/AVP stream handler",
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_CHANNEL_DRIVER,
+	);