SIP SDP media stream handling. More...

#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"
#include "asterisk/stream.h"
#include "asterisk/logger_category.h"
#include "asterisk/format_cache.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/res_pjsip_session_caps.h"

Include dependency graph for res_pjsip_sdp_rtp.c:

Go to the source code of this file.

Functions
static void	__reg_module (void)

static void	__unreg_module (void)

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)

static void	add_extmap_to_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, pj_pool_t pool, pjmedia_sdp_media media)

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)
	Function which adds ICE attributes to a media stream. More...

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)

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)

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)
	Function which adds ssrc attributes to a media stream. More...

static int	apply_cap_to_bundled (struct ast_sip_session_media session_media, struct ast_sip_session_media session_media_transport, struct ast_stream asterisk_stream, struct ast_format_cap joint)

static void	apply_dtls_attrib (struct ast_sip_session_media session_media, pjmedia_sdp_attr attr)

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)

struct ast_module *	AST_MODULE_SELF_SYM (void)

static void	change_outgoing_sdp_stream_media_address (pjsip_tx_data tdata, struct pjmedia_sdp_media stream, struct ast_sip_transport *transport)
	Function which updates the media stream with external media address, if applicable. More...

static enum ast_sip_session_media_encryption	check_endpoint_media_transport (struct ast_sip_endpoint endpoint, const struct pjmedia_sdp_media stream)
	Checks whether the encryption offered in SDP is compatible with the endpoint's configuration. More...

static void	check_ice_support (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_stream)
	Function which checks for ice attributes in an audio stream. More...

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)
	Function which creates an outgoing stream. More...

static int	create_rtp (struct ast_sip_session session, struct ast_sip_session_media session_media, const pjmedia_sdp_session *sdp)
	Internal function which creates an RTP instance. More...

static void	enable_rtcp (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_media)
	Enable RTCP on an RTP session. More...

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)
	Enable an RTP extension on an RTP session. More...

static pjmedia_sdp_attr *	generate_fmtp_attr (pj_pool_t pool, struct ast_format format, int rtp_code)

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)

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)

static enum ast_sip_session_media_encryption	get_media_encryption_type (pj_str_t transport, const struct pjmedia_sdp_media stream, unsigned int optimistic)
	figure out media transport encryption type from the media transport string More...

static int	load_module (void)
	Load the module. More...

static struct ast_frame *	media_session_rtcp_read_callback (struct ast_sip_session session, struct ast_sip_session_media session_media)

static struct ast_frame *	media_session_rtp_read_callback (struct ast_sip_session session, struct ast_sip_session_media session_media)

static int	media_session_rtp_write_callback (struct ast_sip_session session, struct ast_sip_session_media session_media, struct ast_frame *frame)

static int	media_stream_has_crypto (const struct pjmedia_sdp_media *stream)
	figure out if media stream has crypto lines for sdes More...

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)
	Function which negotiates an incoming media stream. More...

static int	parse_dtls_attrib (struct ast_sip_session_media session_media, const struct pjmedia_sdp_session sdp, const struct pjmedia_sdp_media *stream)

static void	process_extmap_attributes (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_stream)
	Function which processes extmap attributes in a stream. More...

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)
	Function which processes ICE attributes in an audio stream. More...

static void	process_ice_auth_attrb (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)

static void	process_ssrc_attributes (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_stream)
	Function which processes ssrc attributes in a stream. More...

static int	rtp_check_timeout (const void *data)
	Check whether RTP is being received or not. More...

static int	send_keepalive (const void *data)

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)

static void	set_ice_components (struct ast_sip_session session, struct ast_sip_session_media session_media)

static struct ast_format_cap *	set_incoming_call_offer_cap (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *stream)

static int	setup_dtls_srtp (struct ast_sip_session session, struct ast_sip_session_media session_media)

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)

static int	setup_sdes_srtp (struct ast_sip_session_media session_media, const struct pjmedia_sdp_media stream)

static int	setup_srtp (struct ast_sip_session_media *session_media)

static void	stream_destroy (struct ast_sip_session_media *session_media)
	Function which destroys the RTP instance when session ends. More...

static void	stream_stop (struct ast_sip_session_media *session_media)
	Function which stops the RTP instance. More...

static int	unload_module (void)
	Unloads the sdp RTP/AVP module from Asterisk. More...

static int	video_info_incoming_request (struct ast_sip_session session, struct pjsip_rx_data rdata)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "PJSIP SDP RTP/AVP stream handler" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "30ef0c93b36035ec78c9cfd712d36d9b" , .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_pjsip,res_pjsip_session", }

static struct ast_sockaddr	address_rtp
	Address for RTP. More...

static const struct ast_module_info *	ast_module_info = &__mod_info

static struct ast_sip_session_sdp_handler	audio_sdp_handler
	SDP handler for 'audio' media stream. More...

static struct ast_sched_context *	sched
	Scheduler for RTCP purposes. More...

static const char	STR_AUDIO [] = "audio"

static const char	STR_VIDEO [] = "video"

static struct ast_sip_session_supplement	video_info_supplement

static struct ast_sip_session_sdp_handler	video_sdp_handler
	SDP handler for 'video' media stream. More...

Detailed Description

SIP SDP media stream handling.

Author: Joshua Colp jcolp.nosp@m.@dig.nosp@m.ium.c.nosp@m.om

Definition in file res_pjsip_sdp_rtp.c.

Function Documentation

◆ __reg_module()

static void __reg_module ( void )

static

Definition at line 2390 of file res_pjsip_sdp_rtp.c.

◆ __unreg_module()

static void __unreg_module ( void )

static

Definition at line 2390 of file res_pjsip_sdp_rtp.c.

◆ add_crypto_to_stream()

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
	)

static

Definition at line 1580 of file res_pjsip_sdp_rtp.c.

References ast_free, AST_LIST_NEXT, AST_RTP_DTLS_CONNECTION_EXISTING, AST_RTP_DTLS_CONNECTION_NEW, AST_RTP_DTLS_HASH_SHA1, AST_RTP_DTLS_HASH_SHA256, AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_HOLDCONN, AST_RTP_DTLS_SETUP_PASSIVE, ast_rtp_instance_get_dtls(), ast_sdp_srtp_alloc(), ast_sdp_srtp_get_attrib(), AST_SIP_MEDIA_ENCRYPT_DTLS, AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_ENCRYPT_SDES, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_str_buffer(), ast_str_create, ast_str_set(), ast_rtp_dtls_cfg::default_setup, ast_sip_media_rtp_configuration::dtls_cfg, ast_sip_session_media::encryption, ast_sip_session::endpoint, ast_rtp_engine_dtls::get_connection, ast_rtp_engine_dtls::get_fingerprint, ast_rtp_engine_dtls::get_fingerprint_hash, ast_rtp_engine_dtls::get_setup, ast_sip_session::inv_session, ast_sip_endpoint::media, RAII_VAR, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, ast_sdp_srtp::sdp_srtp_list, setup_dtls_srtp(), ast_sip_session_media::srtp, ast_sip_media_rtp_configuration::srtp_tag_32, and tmp().

Referenced by create_outgoing_sdp_stream().

 {
  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 (session->inv_session->neg
  && 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;
 }

◆ add_extmap_to_stream()

static void add_extmap_to_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		pj_pool_t *	pool,
		pjmedia_sdp_media *	media
	)

static

Definition at line 1354 of file res_pjsip_sdp_rtp.c.

References AST_MEDIA_TYPE_VIDEO, AST_RTP_EXTENSION_DIRECTION_INACTIVE, AST_RTP_EXTENSION_DIRECTION_NONE, AST_RTP_EXTENSION_DIRECTION_RECVONLY, AST_RTP_EXTENSION_DIRECTION_SENDONLY, AST_RTP_EXTENSION_DIRECTION_SENDRECV, AST_RTP_EXTENSION_UNSUPPORTED, ast_rtp_instance_extmap_count(), ast_rtp_instance_extmap_get_direction(), ast_rtp_instance_extmap_get_extension(), ast_rtp_instance_extmap_get_uri(), ast_sip_session::endpoint, ast_sip_endpoint::media, ast_sip_session_media::rtp, ast_sip_session_media::type, and ast_sip_endpoint_media_configuration::webrtc.

Referenced by create_outgoing_sdp_stream().

 {
  int idx;
  char extmap_value[256];
 
  if (!session->endpoint->media.webrtc || session_media->type != AST_MEDIA_TYPE_VIDEO) {
  return;
  }
 
  /* RTP extension local unique identifiers start at '1' */
  for (idx = 1; idx <= ast_rtp_instance_extmap_count(session_media->rtp); ++idx) {
  enum ast_rtp_extension extension = ast_rtp_instance_extmap_get_extension(session_media->rtp, idx);
  const char *direction_str = "";
  pj_str_t stmp;
  pjmedia_sdp_attr *attr;
 
  /* If this is an unsupported RTP extension we can't place it into the SDP */
  if (extension == AST_RTP_EXTENSION_UNSUPPORTED) {
  continue;
  }
 
  switch (ast_rtp_instance_extmap_get_direction(session_media->rtp, idx)) {
  case AST_RTP_EXTENSION_DIRECTION_SENDRECV:
  /* Lack of a direction indicates sendrecv, so we leave it out */
  direction_str = "";
  break;
  case AST_RTP_EXTENSION_DIRECTION_SENDONLY:
  direction_str = "/sendonly";
  break;
  case AST_RTP_EXTENSION_DIRECTION_RECVONLY:
  direction_str = "/recvonly";
  break;
  case AST_RTP_EXTENSION_DIRECTION_NONE:
  /* It is impossible for a "none" direction extension to be negotiated but just in case
  * we treat it as inactive.
  */
  case AST_RTP_EXTENSION_DIRECTION_INACTIVE:
  direction_str = "/inactive";
  break;
  }
 
  snprintf(extmap_value, sizeof(extmap_value), "%d%s %s", idx, direction_str,
  ast_rtp_instance_extmap_get_uri(session_media->rtp, idx));
  attr = pjmedia_sdp_attr_create(pool, "extmap", pj_cstr(&stmp, extmap_value));
  pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
  }
 }

◆ add_ice_to_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
	)

static

Function which adds ICE attributes to a media stream.

Definition at line 675 of file res_pjsip_sdp_rtp.c.

References ast_rtp_engine_ice_candidate::address, ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_ref, ast_free, AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, ast_rtp_instance_get_ice(), ast_sockaddr_isnull(), ast_sockaddr_stringify_addr_remote(), ast_sockaddr_stringify_port(), ast_str_append(), ast_str_buffer(), ast_str_create, ast_str_set(), ast_sip_session::endpoint, ast_rtp_engine_ice_candidate::foundation, ast_rtp_engine_ice::get_local_candidates, ast_rtp_engine_ice::get_password, ast_rtp_engine_ice::get_ufrag, ast_sip_media_rtp_configuration::ice_support, ast_rtp_engine_ice_candidate::id, ast_sip_endpoint::media, password, ast_rtp_engine_ice_candidate::priority, ast_rtp_engine_ice_candidate::relay_address, ast_sip_session_media::remote_ice, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, ast_rtp_engine_ice::stop, ast_rtp_engine_ice_candidate::transport, and ast_rtp_engine_ice_candidate::type.

Referenced by create_outgoing_sdp_stream().

 {
  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) {
  ice->stop(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);
 }

◆ add_msid_to_stream()

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
	)

static

Definition at line 1272 of file res_pjsip_sdp_rtp.c.

References ast_codec_media_type2str(), ast_copy_string(), ast_debug, ast_stream_get_group(), ast_stream_get_metadata(), ast_stream_get_type(), ast_stream_set_metadata(), ast_strlen_zero, ast_uuid_generate_str(), AST_UUID_STR_LEN, AST_VECTOR_GET, ast_sip_session::endpoint, ast_sip_session_media::label, ast_sip_endpoint::media, ast_sip_session_media::mslabel, ast_sip_session::pending_media_state, and ast_sip_endpoint_media_configuration::webrtc.

Referenced by create_outgoing_sdp_stream().

 {
  pj_str_t stmp;
  pjmedia_sdp_attr *attr;
  char msid[(AST_UUID_STR_LEN * 2) + 2];
  const char *stream_label = ast_stream_get_metadata(stream, "SDP:LABEL");
 
  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));
  /* add for stream identification to replace stream_name */
  ast_stream_set_metadata(stream, "MSID:LABEL", session_media->label);
  }
 
  snprintf(msid, sizeof(msid), "%s %s", session_media->mslabel, session_media->label);
  ast_debug(3, "Stream msid: %p %s %s\n", stream,
  ast_codec_media_type2str(ast_stream_get_type(stream)), msid);
  attr = pjmedia_sdp_attr_create(pool, "msid", pj_cstr(&stmp, msid));
  pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 
  /* 'label' must come after 'msid' */
  if (!ast_strlen_zero(stream_label)) {
  ast_debug(3, "Stream Label: %p %s %s\n", stream,
  ast_codec_media_type2str(ast_stream_get_type(stream)), stream_label);
  attr = pjmedia_sdp_attr_create(pool, "label", pj_cstr(&stmp, stream_label));
  pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
  }
 }

◆ add_rtcp_fb_to_stream()

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
	)

static

Definition at line 1319 of file res_pjsip_sdp_rtp.c.

References AST_MEDIA_TYPE_VIDEO, ast_sip_session::endpoint, ast_sip_endpoint::media, ast_sip_session_media::type, and ast_sip_endpoint_media_configuration::webrtc.

Referenced by create_outgoing_sdp_stream().

 {
  pj_str_t stmp;
  pjmedia_sdp_attr *attr;
 
  if (!session->endpoint->media.webrtc) {
  return;
  }
 
  /* transport-cc is supposed to be for the entire transport, and any media sources so
  * while the header does not appear in audio streams and isn't negotiated there, we still
  * place this attribute in as Chrome does.
  */
  attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* transport-cc"));
  pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 
  if (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);
 
  attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* goog-remb"));
  pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 
  attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* nack"));
  pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 }

◆ add_ssrc_to_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
	)

static

Function which adds ssrc attributes to a media stream.

Definition at line 1202 of file res_pjsip_sdp_rtp.c.

References ast_rtp_instance_get_cname(), ast_rtp_instance_get_ssrc(), ast_sip_endpoint_media_configuration::bundle, ast_sip_session_media::bundle_group, ast_sip_session::endpoint, ast_sip_endpoint::media, ast_sip_session_media::rtp, and tmp().

Referenced by create_outgoing_sdp_stream().

 {
  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;
 }

◆ apply_cap_to_bundled()

static int apply_cap_to_bundled	(	struct ast_sip_session_media *	session_media,
		struct ast_sip_session_media *	session_media_transport,
		struct ast_stream *	asterisk_stream,
		struct ast_format_cap *	joint
	)

static

Definition at line 407 of file res_pjsip_sdp_rtp.c.

References ao2_ref, ast_format_cap_count(), ast_format_cap_get_format(), ast_rtp_codecs_payload_code(), ast_rtp_codecs_payload_set_rx(), ast_rtp_instance_get_codecs(), ast_stream_set_formats(), ast_sip_session_media::bundled, format, and ast_sip_session_media::rtp.

Referenced by negotiate_incoming_sdp_stream(), and set_caps().

 {
  if (!joint) {
  return -1;
  }
 
  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);
  }
  }
 
  return 0;
 }

◆ apply_dtls_attrib()

static void apply_dtls_attrib	(	struct ast_sip_session_media *	session_media,
		pjmedia_sdp_attr *	attr
	)

static

Definition at line 1050 of file res_pjsip_sdp_rtp.c.

References ast_copy_pj_str(), ast_log, AST_RTP_DTLS_HASH_SHA1, AST_RTP_DTLS_HASH_SHA256, AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_HOLDCONN, AST_RTP_DTLS_SETUP_PASSIVE, ast_rtp_instance_get_dtls(), LOG_WARNING, ast_rtp_engine_dtls::reset, ast_sip_session_media::rtp, ast_rtp_engine_dtls::set_fingerprint, ast_rtp_engine_dtls::set_setup, and value.

Referenced by parse_dtls_attrib().

 {
  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);
  }
  }
  }
 }

◆ apply_negotiated_sdp_stream()

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
	)

static

Definition at line 2065 of file res_pjsip_sdp_rtp.c.

 {
  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;
  SCOPE_ENTER(1, "%s Stream: %s\n", ast_sip_session_get_name(session),
  ast_str_tmp(128, ast_stream_to_str(asterisk_stream, &STR_TMP)));
 
  if (!session->channel) {
  SCOPE_EXIT_RTN_VALUE(1, "No channel\n");
  }
 
  /* 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) {
  SCOPE_EXIT_RTN_VALUE(-1, "Incompatible transport\n");
  }
 
  /* Create an RTP instance if need be */
  if (!session_media->rtp && create_rtp(session, session_media, local)) {
  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
  }
 
  process_ssrc_attributes(session, session_media, remote_stream);
  process_extmap_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.
  */
  SCOPE_EXIT_RTN_VALUE(-1, "Incompatible crypto\n");
  }
 
  if (!remote_stream->conn && !remote->conn) {
  SCOPE_EXIT_RTN_VALUE(1, "No connection info\n");
  }
 
  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 */
  SCOPE_EXIT_RTN_VALUE(-1, "Host invalid\n");
  }
 
  /* 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)) {
  SCOPE_EXIT_RTN_VALUE(-1, "set_caps failed\n");
  }
 
  /* 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);
  }
  SCOPE_EXIT_RTN_VALUE(1, "moh\n");
  }
 
  if (session_media->remotely_held_changed) {
  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_changed = 0;
  } else {
  /* 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_changed = 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);
  }
 
  SCOPE_EXIT_RTN_VALUE(1, "Handled\n");
 }

◆ AST_MODULE_SELF_SYM()

struct ast_module* AST_MODULE_SELF_SYM ( void )

Definition at line 2390 of file res_pjsip_sdp_rtp.c.

◆ change_outgoing_sdp_stream_media_address()

static void change_outgoing_sdp_stream_media_address	(	pjsip_tx_data *	tdata,
		struct pjmedia_sdp_media *	stream,
		struct ast_sip_transport *	transport
	)

static

Function which updates the media stream with external media address, if applicable.

Definition at line 2226 of file res_pjsip_sdp_rtp.c.

References ao2_cleanup, ast_copy_pj_str(), ast_debug, ast_sip_get_transport_state(), ast_sip_transport_is_nonlocal, ast_sockaddr_parse(), ast_sockaddr_stringify_addr_remote(), ast_sorcery_object_get_id(), host, PARSE_PORT_FORBID, and RAII_VAR.

 {
  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_addr_remote(&transport_state->external_media_address));
  pj_strdup2(tdata->pool, &stream->conn->addr, ast_sockaddr_stringify_addr_remote(&transport_state->external_media_address));
 }

◆ check_endpoint_media_transport()

static enum ast_sip_session_media_encryption check_endpoint_media_transport	(	struct ast_sip_endpoint *	endpoint,
		const struct pjmedia_sdp_media *	stream
	)

static

Checks whether the encryption offered in SDP is compatible with the endpoint's configuration.

Definition at line 970 of file res_pjsip_sdp_rtp.c.

References AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_sip_media_rtp_configuration::encryption, ast_sip_media_rtp_configuration::encryption_optimistic, ast_sip_media_rtp_configuration::force_avp, get_media_encryption_type(), ast_sip_endpoint::media, ast_sip_endpoint_media_configuration::rtp, and ast_sip_media_rtp_configuration::use_avpf.

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_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;
 }

◆ check_ice_support()

static void check_ice_support	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Function which checks for ice attributes in an audio stream.

Definition at line 750 of file res_pjsip_sdp_rtp.c.

References ast_rtp_instance_get_ice(), ast_sip_session::endpoint, ast_sip_media_rtp_configuration::ice_support, ast_sip_endpoint::media, ast_sip_session_media::remote_ice, ast_sip_session_media::rtp, and ast_sip_endpoint_media_configuration::rtp.

Referenced by negotiate_incoming_sdp_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;
  }
 }

◆ create_outgoing_sdp_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
	)

static

Function which creates an outgoing stream.

Definition at line 1704 of file res_pjsip_sdp_rtp.c.

 {
  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 };
  static const pj_str_t STR_INACTIVE = { "inactive", 8 };
  static const pj_str_t STR_RECVONLY = { "recvonly", 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);
  SCOPE_ENTER(1, "%s Type: %s %s\n", ast_sip_session_get_name(session),
  ast_codec_media_type2str(media_type), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
 
  media = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_media));
  if (!media) {
  SCOPE_EXIT_RTN_VALUE(-1, "Pool alloc failure\n");
  }
  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);
 
  SCOPE_EXIT_RTN_VALUE(1, "Stream removed or no formats\n");
  }
 
  if (!session_media->rtp && create_rtp(session, session_media, sdp)) {
  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
  }
 
  /* 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)) {
  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't add crypto\n");
  }
 
  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) {
  SCOPE_EXIT_RTN_VALUE(-1, "Pool alloc failure\n");
  }
 
  /* 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));
  SCOPE_EXIT_RTN_VALUE(-1, "No local host ip\n");
  }
 
  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)) {
  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't add crypto\n");
  }
 
  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));
  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create caps\n");
  }
 
  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) {
  SCOPE_EXIT_RTN_VALUE(1, "No formats added to stream\n");
  }
 
  /* 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;
  }
 
  attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
  if (session_media->locally_held) {
  if (session_media->remotely_held) {
  attr->name = STR_INACTIVE; /* To place on hold a recvonly stream, send inactive */
  } else {
  attr->name = STR_SENDONLY; /* Send sendonly to initate a local hold */
  }
  } else {
  if (session_media->remotely_held) {
  attr->name = STR_RECVONLY; /* Remote has sent sendonly, reply recvonly */
  } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
  attr->name = STR_SENDONLY; /* Stream has requested sendonly */
  } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
  attr->name = STR_RECVONLY; /* Stream has requested recvonly */
  } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_INACTIVE) {
  attr->name = STR_INACTIVE; /* Stream has requested inactive */
  } else {
  attr->name = STR_SENDRECV; /* No hold in either direction */
  }
  }
  media->attr[media->attr_count++] = attr;
 
  /* If we've got rtcp-mux enabled, add it unless we received an offer without it */
  if (session->endpoint->media.rtcp_mux && session_media->remote_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_extmap_to_stream(session, session_media, pool, media);
 
  /* Add the media stream to the SDP */
  sdp->media[sdp->media_count++] = media;
 
  SCOPE_EXIT_RTN_VALUE(1, "RC: 1\n");
 }

◆ create_rtp()

static int create_rtp	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const pjmedia_sdp_session *	sdp
	)

static

Internal function which creates an RTP instance.

Definition at line 228 of file res_pjsip_sdp_rtp.c.

Referenced by apply_negotiated_sdp_stream(), create_outgoing_sdp_stream(), and negotiate_incoming_sdp_stream().

 {
  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_rtp(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_rtp(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_rtp(1, "Transport %s bound to %s: Using it for RTP media.\n",
  session->endpoint->transport, hoststr);
  media_address = &temp_media_address;
  } else {
  ast_debug_rtp(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);
  enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_TRANSPORT_WIDE_CC, 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;
 }

◆ enable_rtcp()

static void enable_rtcp	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_media
	)

static

Enable RTCP on an RTP session.

Definition at line 165 of file res_pjsip_sdp_rtp.c.

References AST_RTP_INSTANCE_RTCP_MUX, AST_RTP_INSTANCE_RTCP_STANDARD, ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_RTCP, ast_sip_session::endpoint, ast_sip_endpoint::media, ast_sip_session_media::remote_rtcp_mux, ast_sip_endpoint_media_configuration::rtcp_mux, and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream(), create_outgoing_sdp_stream(), and negotiate_incoming_sdp_stream().

 {
  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);
 }

◆ enable_rtp_extension()

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
	)

static

Enable an RTP extension on an RTP session.

Definition at line 182 of file res_pjsip_sdp_rtp.c.

References ast_rtp_instance_extmap_count(), ast_rtp_instance_extmap_enable(), ast_rtp_instance_extmap_get_id(), AST_VECTOR_GET, AST_VECTOR_SIZE, ast_sip_session_media::bundle_group, ast_sip_session::pending_media_state, and ast_sip_session_media::rtp.

Referenced by create_rtp().

 {
  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);
 }

◆ generate_fmtp_attr()

static pjmedia_sdp_attr* generate_fmtp_attr	(	pj_pool_t *	pool,
		struct ast_format *	format,
		int	rtp_code
	)

static

Definition at line 649 of file res_pjsip_sdp_rtp.c.

References ast_format_generate_sdp_fmtp(), ast_str_alloca, ast_str_buffer(), ast_str_strlen(), NULL, and tmp().

Referenced by create_outgoing_sdp_stream().

 {
  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;
 }

◆ generate_rtpmap_attr()

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
	)

static

Definition at line 614 of file res_pjsip_sdp_rtp.c.

References ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), AST_RTP_OPT_G726_NONSTANDARD, AST_RTP_PT_LAST_STATIC, ast_sip_session::endpoint, ast_sip_endpoint_media_configuration::g726_non_standard, ast_sip_endpoint::media, NULL, options, and tmp().

Referenced by create_outgoing_sdp_stream().

 {
 #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;
 }

◆ get_codecs()

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
	)

static

Definition at line 315 of file res_pjsip_sdp_rtp.c.

References ao2_ref, ast_copy_pj_str(), ast_format_parse_sdp_fmtp(), ast_rtp_codecs_get_payload_format(), ast_rtp_codecs_payload_replace_format(), ast_rtp_codecs_payloads_initialize(), ast_rtp_codecs_payloads_set_m_type(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_codecs_set_framing(), AST_RTP_DTMF_MODE_INBAND, AST_RTP_DTMF_MODE_NONE, AST_RTP_DTMF_MODE_RFC2833, ast_rtp_instance_dtmf_mode_set(), ast_rtp_instance_set_prop(), AST_RTP_OPT_G726_NONSTANDARD, AST_RTP_PROPERTY_DTMF, AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, ast_sip_session_get_name(), ast_sip_session::dtmf, ast_sip_session::endpoint, format, ast_sip_endpoint_media_configuration::g726_non_standard, ast_sip_session::inv_session, ast_sip_endpoint::media, name, NULL, options, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN, and ast_sip_media_rtp_configuration::use_ptime.

Referenced by set_caps(), and set_incoming_call_offer_cap().

 {
  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;
  SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
  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);
  ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
  }
 
  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);
  ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
  } else {
  ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_NONE);
  ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
  }
  }
 
 
  /* 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);
  }
  }
 
  SCOPE_EXIT_RTN();
 }

◆ get_media_encryption_type()

static enum ast_sip_session_media_encryption get_media_encryption_type	(	pj_str_t	transport,
		const struct pjmedia_sdp_media *	stream,
		unsigned int *	optimistic
	)

static

figure out media transport encryption type from the media transport string

Definition at line 938 of file res_pjsip_sdp_rtp.c.

References ast_free, AST_SIP_MEDIA_ENCRYPT_DTLS, AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_ENCRYPT_SDES, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_strndup, media_stream_has_crypto(), and RAII_VAR.

Referenced by check_endpoint_media_transport().

 {
  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;
  }
 }

◆ load_module()

static int load_module ( void )

static

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.

Definition at line 2347 of file res_pjsip_sdp_rtp.c.

References address_rtp, ast_check_ipv6(), ast_log, AST_MODFLAG_LOAD_ORDER, AST_MODPRI_CHANNEL_DRIVER, AST_MODULE_INFO(), AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, AST_MODULE_SUPPORT_CORE, ast_sched_context_create(), ast_sched_start_thread(), ast_sip_session_register_sdp_handler(), ast_sip_session_register_supplement, ast_sockaddr_parse(), ASTERISK_GPL_KEY, end, LOG_ERROR, STR_AUDIO, STR_VIDEO, and unload_module().

 {
  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;
 }

◆ media_session_rtcp_read_callback()

static struct ast_frame* media_session_rtcp_read_callback	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 2038 of file res_pjsip_sdp_rtp.c.

References ast_null_frame, ast_rtp_instance_read(), ast_rtp_instance_set_last_rx(), NULL, and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream().

 {
  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;
 }

◆ media_session_rtp_read_callback()

static struct ast_frame* media_session_rtp_read_callback	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 2020 of file res_pjsip_sdp_rtp.c.

References ast_null_frame, ast_rtp_instance_read(), ast_rtp_instance_set_last_rx(), NULL, and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream().

 {
  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;
 }

◆ media_session_rtp_write_callback()

static int media_session_rtp_write_callback	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		struct ast_frame *	frame
	)

static

Definition at line 2056 of file res_pjsip_sdp_rtp.c.

References ast_rtp_instance_write(), and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream().

 {
  if (!session_media->rtp) {
  return 0;
  }
 
  return ast_rtp_instance_write(session_media->rtp, frame);
 }

◆ media_stream_has_crypto()

static int media_stream_has_crypto ( const struct pjmedia_sdp_media * stream )

static

figure out if media stream has crypto lines for sdes

Definition at line 918 of file res_pjsip_sdp_rtp.c.

Referenced by get_media_encryption_type().

 {
  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;
 }

◆ negotiate_incoming_sdp_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
	)

static

Function which negotiates an incoming media stream.

Definition at line 1462 of file res_pjsip_sdp_rtp.c.

 {
  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;
  struct ast_format_cap *joint;
  int res;
  SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
  /* 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));
  SCOPE_EXIT_RTN_VALUE(0, "Endpoint has no codecs\n");
  }
 
  /* 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) {
  SCOPE_EXIT_RTN_VALUE(-1, "Incompatible transport\n");
  }
  }
 
  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 */
  SCOPE_EXIT_RTN_VALUE(-1, "Invalid host\n");
  }
 
  /* Using the connection information create an appropriate RTP instance */
  if (!session_media->rtp && create_rtp(session, session_media, sdp)) {
  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
  }
 
  process_ssrc_attributes(session, session_media, stream);
  process_extmap_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.
  */
  SCOPE_EXIT_RTN_VALUE(-1, "Incompatible crypto\n");
  }
  /* 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 ICE support is enabled find all the needed attributes */
  check_ice_support(session, session_media, stream);
 
  /* If ICE support is enabled then check remote ICE started? */
  if (session_media->remote_ice) {
  process_ice_auth_attrb(session, session_media, sdp, stream);
  }
 
  if (ast_sip_session_is_pending_stream_default(session, asterisk_stream) && media_type == AST_MEDIA_TYPE_AUDIO) {
  /* Check if incomming SDP is changing the remotely held state */
  if (ast_sockaddr_isnull(addrs) ||
  ast_sockaddr_is_any(addrs) ||
  pjmedia_sdp_media_find_attr2(stream, "sendonly", NULL) ||
  pjmedia_sdp_media_find_attr2(stream, "inactive", NULL)) {
  if (!session_media->remotely_held) {
  session_media->remotely_held = 1;
  session_media->remotely_held_changed = 1;
  }
  } else if (session_media->remotely_held) {
  session_media->remotely_held = 0;
  session_media->remotely_held_changed = 1;
  }
  }
 
  joint = set_incoming_call_offer_cap(session, session_media, stream);
  res = apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream, joint);
  ao2_cleanup(joint);
  if (res != 0) {
  SCOPE_EXIT_RTN_VALUE(0, "Something failed\n");
  }
 
  SCOPE_EXIT_RTN_VALUE(1);
 }

◆ parse_dtls_attrib()

static int parse_dtls_attrib	(	struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_session *	sdp,
		const struct pjmedia_sdp_media *	stream
	)

static

Definition at line 1099 of file res_pjsip_sdp_rtp.c.

References apply_dtls_attrib(), ast_set_flag, AST_SRTP_CRYPTO_OFFER_OK, and ast_sip_session_media::srtp.

Referenced by setup_media_encryption().

 {
  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;
 }

◆ process_extmap_attributes()

static void process_extmap_attributes	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Function which processes extmap attributes in a stream.

Definition at line 1404 of file res_pjsip_sdp_rtp.c.

References ast_copy_pj_str(), AST_MEDIA_TYPE_VIDEO, AST_RTP_EXTENSION_DIRECTION_INACTIVE, AST_RTP_EXTENSION_DIRECTION_RECVONLY, AST_RTP_EXTENSION_DIRECTION_SENDONLY, AST_RTP_EXTENSION_DIRECTION_SENDRECV, ast_rtp_instance_extmap_clear(), ast_rtp_instance_extmap_negotiate(), ast_strlen_zero, ast_sip_session::endpoint, id, ast_sip_endpoint::media, ast_sip_session_media::rtp, ast_sip_session_media::type, and ast_sip_endpoint_media_configuration::webrtc.

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_stream().

 {
  int index;
 
  if (!session->endpoint->media.webrtc || session_media->type != AST_MEDIA_TYPE_VIDEO) {
  return;
  }
 
  ast_rtp_instance_extmap_clear(session_media->rtp);
 
  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 *uri;
  int id;
  char direction_str[10] = "";
  char *attributes;
  enum ast_rtp_extension_direction direction = AST_RTP_EXTENSION_DIRECTION_SENDRECV;
 
  /* We only care about extmap attributes */
  if (pj_strcmp2(&attr->name, "extmap")) {
  continue;
  }
 
  ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
 
  /* Split the combined unique identifier and direction away from the URI and attributes for easier parsing */
  uri = strchr(attr_value, ' ');
  if (ast_strlen_zero(uri)) {
  continue;
  }
  *uri++ = '\0';
 
  if ((sscanf(attr_value, "%30d%9s", &id, direction_str) < 1) || (id < 1)) {
  /* We require at a minimum the unique identifier */
  continue;
  }
 
  /* Convert from the string to the internal representation */
  if (!strcasecmp(direction_str, "/sendonly")) {
  direction = AST_RTP_EXTENSION_DIRECTION_SENDONLY;
  } else if (!strcasecmp(direction_str, "/recvonly")) {
  direction = AST_RTP_EXTENSION_DIRECTION_RECVONLY;
  } else if (!strcasecmp(direction_str, "/inactive")) {
  direction = AST_RTP_EXTENSION_DIRECTION_INACTIVE;
  }
 
  attributes = strchr(uri, ' ');
  if (!ast_strlen_zero(attributes)) {
  *attributes++ = '\0';
  }
 
  ast_rtp_instance_extmap_negotiate(session_media->rtp, id, direction, uri, attributes);
  }
 }

◆ process_ice_attributes()

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
	)

static

Function which processes ICE attributes in an audio stream.

Definition at line 814 of file res_pjsip_sdp_rtp.c.

References ast_rtp_engine_ice::add_remote_candidate, ast_rtp_engine_ice_candidate::address, ast_copy_pj_str(), ast_debug_ice, AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, AST_RTP_ICE_ROLE_CONTROLLED, AST_RTP_ICE_ROLE_CONTROLLING, ast_rtp_instance_get_ice(), ast_sockaddr_parse(), ast_sockaddr_set_port, ast_strlen_zero, ast_sip_session::endpoint, ast_rtp_engine_ice_candidate::foundation, ast_rtp_engine_ice::ice_lite, ast_sip_media_rtp_configuration::ice_support, ast_rtp_engine_ice_candidate::id, ast_sip_session::inv_session, ast_sip_endpoint::media, NULL, PARSE_PORT_FORBID, ast_rtp_engine_ice_candidate::priority, ast_rtp_engine_ice_candidate::relay_address, ast_sip_session_media::remote_rtcp_mux, ast_sip_endpoint_media_configuration::rtcp_mux, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, ast_rtp_engine_ice::set_authentication, ast_rtp_engine_ice::set_role, ast_rtp_engine_ice::start, ast_rtp_engine_ice_candidate::transport, and ast_rtp_engine_ice_candidate::type.

Referenced by apply_negotiated_sdp_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;
  }
 
  ast_debug_ice(2, "(%p) ICE process attributes\n", session_media->rtp);
 
  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 {
  ast_debug_ice(2, "(%p) ICE no, or invalid ice-ufrag\n", session_media->rtp);
  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 {
  ast_debug_ice(2, "(%p) ICE no, or invalid ice-pwd\n", session_media->rtp);
  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);
 }

◆ process_ice_auth_attrb()

static void process_ice_auth_attrb	(	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
	)

static

Definition at line 776 of file res_pjsip_sdp_rtp.c.

References ast_copy_pj_str(), ast_rtp_instance_get_ice(), ast_sip_session::endpoint, ast_sip_media_rtp_configuration::ice_support, ast_sip_endpoint::media, NULL, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, and ast_rtp_engine_ice::set_authentication.

Referenced by negotiate_incoming_sdp_stream().

 {
  struct ast_rtp_engine_ice *ice;
  const pjmedia_sdp_attr *ufrag_attr, *passwd_attr;
  char ufrag_attr_value[256];
  char passwd_attr_value[256];
 
  /* 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;
  }
 
  ufrag_attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-ufrag", NULL);
  if (!ufrag_attr) {
  ufrag_attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-ufrag", NULL);
  }
  if (ufrag_attr) {
  ast_copy_pj_str(ufrag_attr_value, (pj_str_t*)&ufrag_attr->value, sizeof(ufrag_attr_value));
  } else {
  return;
  }
  passwd_attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-pwd", NULL);
  if (!passwd_attr) {
  passwd_attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-pwd", NULL);
  }
  if (passwd_attr) {
  ast_copy_pj_str(passwd_attr_value, (pj_str_t*)&passwd_attr->value, sizeof(passwd_attr_value));
  } else {
  return;
  }
 
  if (ufrag_attr && passwd_attr) {
  ice->set_authentication(session_media->rtp, ufrag_attr_value, passwd_attr_value);
  }
 }

◆ process_ssrc_attributes()

static void process_ssrc_attributes	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Function which processes ssrc attributes in a stream.

Definition at line 1218 of file res_pjsip_sdp_rtp.c.

References ast_sip_session::active_media_state, ast_copy_pj_str(), ast_rtp_instance_get_stats(), ast_rtp_instance_set_remote_ssrc(), AST_RTP_INSTANCE_STAT_REMOTE_SSRC, ast_sip_endpoint_media_configuration::bundle, ast_sip_session_media::changed, ast_sip_session::endpoint, ast_sip_session::inv_session, ast_sip_endpoint::media, NULL, ast_rtp_instance_stats::remote_ssrc, and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_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);
  break;
  }
 }

◆ rtp_check_timeout()

static int rtp_check_timeout ( const void * data )

static

Check whether RTP is being received or not.

Definition at line 105 of file res_pjsip_sdp_rtp.c.

References AST_CAUSE_REQUESTED_CHAN_UNAVAIL, ast_channel_get_by_name(), ast_channel_hangupcause_set(), ast_channel_lock, ast_channel_name(), ast_channel_unlock, ast_channel_unref, ast_codec_media_type2str(), ast_debug_rtp, ast_log, ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_last_rx(), ast_rtp_instance_get_timeout(), ast_sockaddr_isnull(), ast_softhangup(), AST_SOFTHANGUP_DEV, ast_sip_session_media::direct_media_addr, LOG_NOTICE, NULL, ast_sip_session_media::rtp, timeout, and ast_sip_session_media::type.

Referenced by apply_negotiated_sdp_stream().

 {
  struct ast_sip_session_media *session_media = (struct ast_sip_session_media *)data;
  struct ast_rtp_instance *rtp = session_media->rtp;
  int elapsed;
  int timeout;
  struct ast_channel *chan;
 
  if (!rtp) {
  return 0;
  }
 
  chan = ast_channel_get_by_name(ast_rtp_instance_get_channel_id(rtp));
  if (!chan) {
  return 0;
  }
 
  /* Get channel lock to make sure that we access a consistent set of values
  * (last_rx and direct_media_addr) - the lock is held when values are modified
  * (see send_direct_media_request()/check_for_rtp_changes() in chan_pjsip.c). We
  * are trying to avoid a situation where direct_media_addr has been reset but the
  * last-rx time was not set yet.
  */
  ast_channel_lock(chan);
 
  elapsed = time(NULL) - ast_rtp_instance_get_last_rx(rtp);
  timeout = ast_rtp_instance_get_timeout(rtp);
  if (elapsed < timeout) {
  ast_channel_unlock(chan);
  ast_channel_unref(chan);
  return (timeout - elapsed) * 1000;
  }
 
  /* Last RTP packet was received too long ago
  * - disconnect channel unless direct media is in use.
  */
  if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
  ast_debug_rtp(3, "(%p) RTP not disconnecting channel '%s' for lack of %s RTP activity in %d seconds "
  "since direct media is in use\n", rtp, ast_channel_name(chan),
  ast_codec_media_type2str(session_media->type), elapsed);
  ast_channel_unlock(chan);
  ast_channel_unref(chan);
  return timeout * 1000; /* recheck later, direct media may have ended then */
  }
 
  ast_log(LOG_NOTICE, "Disconnecting channel '%s' for lack of %s RTP activity in %d seconds\n",
  ast_channel_name(chan), ast_codec_media_type2str(session_media->type), elapsed);
 
  ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
  ast_softhangup(chan, AST_SOFTHANGUP_DEV);
 
  ast_channel_unlock(chan);
  ast_channel_unref(chan);
 
  return 0;
 }

◆ send_keepalive()

static int send_keepalive ( const void * data )

static

Definition at line 71 of file res_pjsip_sdp_rtp.c.

References ast_debug_rtp, ast_rtp_instance_get_keepalive(), ast_rtp_instance_get_last_tx(), ast_rtp_instance_sendcng(), ast_sockaddr_isnull(), ast_sip_session_media::direct_media_addr, keepalive, NULL, and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream().

 {
  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_rtp(3, "(%p) RTP not sending keepalive 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_rtp(3, "(%p) RTP it has been %d seconds since RTP was last sent. %sending keepalive\n",
  rtp, (int) interval, send_keepalive ? "S" : "Not s");
 
  if (send_keepalive) {
  ast_rtp_instance_sendcng(rtp, 0);
  return keepalive * 1000;
  }
 
  return (keepalive - interval) * 1000;
 }

◆ set_caps()

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
	)

static

Definition at line 485 of file res_pjsip_sdp_rtp.c.

References ao2_cleanup, ao2_ref, apply_cap_to_bundled(), ast_channel_is_bridged(), ast_channel_lock, ast_channel_nativeformats(), ast_channel_nativeformats_set(), ast_channel_readformat(), ast_channel_set_unbridged_nolock(), ast_channel_unlock, ast_channel_writeformat(), ast_codec_media_type2str(), ast_dsp_free(), ast_dsp_get_features(), ast_dsp_set_features(), ast_format_cap_alloc, ast_format_cap_append, ast_format_cap_append_from_cap(), ast_format_cap_count(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_format_cap_get_best_by_type(), ast_format_cap_get_compatible(), ast_format_cap_get_format(), ast_format_cap_get_framing(), ast_format_cap_get_names(), AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_remove_by_type(), ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_UNKNOWN, AST_RTP_CODECS_NULL_INIT, ast_rtp_codecs_payload_formats(), ast_rtp_codecs_payloads_copy(), ast_rtp_codecs_payloads_destroy(), ast_rtp_codecs_payloads_xover(), AST_RTP_DTMF_MODE_RFC2833, ast_rtp_instance_dtmf_mode_get(), ast_rtp_instance_get_codecs(), ast_set_read_format(), ast_set_write_format(), AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, ast_sip_session_get_name(), ast_sip_session_is_pending_stream_default(), ast_sockaddr_isnull(), ast_str_alloca, ast_sip_endpoint::asymmetric_rtp_codec, ast_sip_session::channel, ast_sip_endpoint_media_configuration::codecs, ast_sip_session_media::direct_media_addr, ast_sip_session::direct_media_cap, ast_sip_session::dsp, DSP_FEATURE_DIGIT_DETECT, ast_sip_session::dtmf, ast_sip_session::endpoint, get_codecs(), LOG_ERROR, LOG_NOTICE, ast_sip_endpoint::media, NULL, ast_sip_endpoint::preferred_codec_only, RAII_VAR, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, and ast_sip_session_media::type.

Referenced by apply_negotiated_sdp_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);
  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;
  SCOPE_ENTER(1, "%s %s\n", ast_sip_session_get_name(session), is_offer ? "OFFER" : "ANSWER");
 
  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));
  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create %s capabilities\n",
  ast_codec_media_type2str(session_media->type));
  }
 
  /* 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));
  SCOPE_EXIT_RTN_VALUE(-1, "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));
  }
 
  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);
 
  apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream, joint);
 
  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);
  SCOPE_EXIT_RTN_VALUE(0);
 }

◆ set_ice_components()

static void set_ice_components	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 1181 of file res_pjsip_sdp_rtp.c.

References ast_assert, ast_rtp_instance_get_ice(), ast_rtp_engine_ice::change_components, ast_sip_session::endpoint, ast_sip_media_rtp_configuration::ice_support, ast_sip_endpoint::media, NULL, ast_sip_session_media::remote_rtcp_mux, ast_sip_endpoint_media_configuration::rtcp_mux, ast_sip_session_media::rtp, and ast_sip_endpoint_media_configuration::rtp.

Referenced by apply_negotiated_sdp_stream(), create_outgoing_sdp_stream(), and negotiate_incoming_sdp_stream().

 {
  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);
  }
 }

◆ set_incoming_call_offer_cap()

static struct ast_format_cap* set_incoming_call_offer_cap	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	stream
	)

static

Definition at line 438 of file res_pjsip_sdp_rtp.c.

References ao2_cleanup, ao2_ref, ast_codec_media_type2str(), ast_format_cap_alloc, ast_format_cap_empty(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_log, AST_RTP_CODECS_NULL_INIT, ast_rtp_codecs_payload_formats(), ast_rtp_codecs_payloads_copy(), ast_rtp_codecs_payloads_destroy(), ast_rtp_codecs_payloads_xover(), ast_rtp_instance_get_codecs(), ast_sip_session_create_joint_call_cap(), ast_sip_session_get_name(), get_codecs(), LOG_ERROR, NULL, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, and ast_sip_session_media::type.

Referenced by negotiate_incoming_sdp_stream().

 {
  struct ast_format_cap *incoming_call_offer_cap;
  struct ast_format_cap *remote;
  struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
  int fmts = 0;
  SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
 
  remote = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
  if (!remote) {
  ast_log(LOG_ERROR, "Failed to allocate %s incoming remote capabilities\n",
  ast_codec_media_type2str(session_media->type));
  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't allocate caps\n");
  }
 
  /* Get the peer's capabilities*/
  get_codecs(session, stream, &codecs, session_media);
  ast_rtp_codecs_payload_formats(&codecs, remote, &fmts);
 
  incoming_call_offer_cap = ast_sip_session_create_joint_call_cap(
  session, session_media->type, remote);
 
  ao2_ref(remote, -1);
 
  if (!incoming_call_offer_cap || ast_format_cap_empty(incoming_call_offer_cap)) {
  ao2_cleanup(incoming_call_offer_cap);
  ast_rtp_codecs_payloads_destroy(&codecs);
  SCOPE_EXIT_RTN_VALUE(NULL, "No incoming call offer caps\n");
  }
 
  /*
  * 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_rtp_codecs_payloads_destroy(&codecs);
 
  SCOPE_EXIT_RTN_VALUE(incoming_call_offer_cap);
 }

◆ setup_dtls_srtp()

static int setup_dtls_srtp	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 1023 of file res_pjsip_sdp_rtp.c.

References AST_AES_CM_128_HMAC_SHA1_32, AST_AES_CM_128_HMAC_SHA1_80, ast_log, ast_rtp_instance_get_dtls(), ast_sip_media_rtp_configuration::dtls_cfg, ast_rtp_dtls_cfg::enabled, ast_sip_session::endpoint, LOG_ERROR, ast_sip_endpoint::media, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, ast_rtp_engine_dtls::set_configuration, setup_srtp(), ast_sip_media_rtp_configuration::srtp_tag_32, and ast_rtp_dtls_cfg::suite.

Referenced by add_crypto_to_stream(), and setup_media_encryption().

 {
  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;
 }

◆ setup_media_encryption()

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
	)

static

Definition at line 1154 of file res_pjsip_sdp_rtp.c.

References AST_SIP_MEDIA_ENCRYPT_DTLS, AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_ENCRYPT_SDES, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_sip_session_media::encryption, parse_dtls_attrib(), setup_dtls_srtp(), and setup_sdes_srtp().

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_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;
 }

◆ setup_sdes_srtp()

static int setup_sdes_srtp	(	struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	stream
	)

static

Definition at line 1118 of file res_pjsip_sdp_rtp.c.

References ast_free, ast_log, ast_sdp_crypto_process(), ast_strndup, LOG_WARNING, NULL, RAII_VAR, ast_sip_session_media::rtp, setup_srtp(), and ast_sip_session_media::srtp.

Referenced by setup_media_encryption().

 {
  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_log(LOG_WARNING, "Ignoring crypto offer with unsupported parameters: %s\n", crypto_str);
  }
 
  /* no usable crypto attributes found */
  return -1;
 }

◆ setup_srtp()

static int setup_srtp ( struct ast_sip_session_media * session_media )

static

Definition at line 1004 of file res_pjsip_sdp_rtp.c.

References ast_sdp_crypto_alloc(), ast_sdp_srtp_alloc(), ast_sdp_srtp::crypto, and ast_sip_session_media::srtp.

Referenced by setup_dtls_srtp(), and setup_sdes_srtp().

 {
  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;
 }

◆ stream_destroy()

static void stream_destroy ( struct ast_sip_session_media * session_media )

static

Function which destroys the RTP instance when session ends.

Definition at line 2264 of file res_pjsip_sdp_rtp.c.

References ast_rtp_instance_destroy(), NULL, ast_sip_session_media::rtp, and stream_stop().

 {
  if (session_media->rtp) {
  stream_stop(session_media);
  ast_rtp_instance_destroy(session_media->rtp);
  }
  session_media->rtp = NULL;
 }

◆ stream_stop()

static void stream_stop ( struct ast_sip_session_media * session_media )

static

Function which stops the RTP instance.

Definition at line 2252 of file res_pjsip_sdp_rtp.c.

References ast_rtp_instance_stop(), AST_SCHED_DEL, ast_sip_session_media::keepalive_sched_id, ast_sip_session_media::rtp, and ast_sip_session_media::timeout_sched_id.

Referenced by stream_destroy().

 {
  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);
 }

◆ unload_module()

static int unload_module ( void )

static

Unloads the sdp RTP/AVP module from Asterisk.

Definition at line 2324 of file res_pjsip_sdp_rtp.c.

References ast_sched_context_destroy(), ast_sip_session_unregister_sdp_handler(), ast_sip_session_unregister_supplement(), STR_AUDIO, and STR_VIDEO.

Referenced by load_module().

 {
  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;
 }

◆ video_info_incoming_request()

static int video_info_incoming_request	(	struct ast_sip_session *	session,
		struct pjsip_rx_data *	rdata
	)

static

Definition at line 2295 of file res_pjsip_sdp_rtp.c.

References AST_CONTROL_VIDUPDATE, ast_queue_control(), ast_sip_is_content_type(), ast_sip_session::channel, ast_sip_session::inv_session, and NULL.

 {
  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;
 }

Variable Documentation

◆ __mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "PJSIP SDP RTP/AVP stream handler" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "30ef0c93b36035ec78c9cfd712d36d9b" , .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_pjsip,res_pjsip_session", }

static

Definition at line 2390 of file res_pjsip_sdp_rtp.c.

◆ address_rtp

struct ast_sockaddr address_rtp

static

Address for RTP.

Definition at line 66 of file res_pjsip_sdp_rtp.c.

Referenced by create_rtp(), and load_module().

◆ ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 2390 of file res_pjsip_sdp_rtp.c.

◆ audio_sdp_handler

struct ast_sip_session_sdp_handler audio_sdp_handler

static

SDP handler for 'audio' media stream.

Definition at line 2274 of file res_pjsip_sdp_rtp.c.

◆ sched

struct ast_sched_context* sched

static

Scheduler for RTCP purposes.

Definition at line 63 of file res_pjsip_sdp_rtp.c.

◆ STR_AUDIO

const char STR_AUDIO[] = "audio"

static

Definition at line 68 of file res_pjsip_sdp_rtp.c.

Referenced by load_module(), and unload_module().

◆ STR_VIDEO

const char STR_VIDEO[] = "video"

static

Definition at line 69 of file res_pjsip_sdp_rtp.c.

Referenced by load_module(), and unload_module().

◆ video_info_supplement

struct ast_sip_session_supplement video_info_supplement

static

Initial value:

= {
 .method = "INFO",
 .incoming_request = video_info_incoming_request,
}

Definition at line 2318 of file res_pjsip_sdp_rtp.c.

◆ video_sdp_handler

struct ast_sip_session_sdp_handler video_sdp_handler

static

SDP handler for 'video' media stream.

Definition at line 2285 of file res_pjsip_sdp_rtp.c.

Functions

Variables

Detailed Description

Function Documentation

◆ __reg_module()

◆ __unreg_module()

◆ add_crypto_to_stream()

◆ add_extmap_to_stream()

◆ add_ice_to_stream()

◆ add_msid_to_stream()

◆ add_rtcp_fb_to_stream()

◆ add_ssrc_to_stream()

◆ apply_cap_to_bundled()

◆ apply_dtls_attrib()

◆ apply_negotiated_sdp_stream()

◆ AST_MODULE_SELF_SYM()

◆ change_outgoing_sdp_stream_media_address()

◆ check_endpoint_media_transport()

◆ check_ice_support()

◆ create_outgoing_sdp_stream()

◆ create_rtp()

◆ enable_rtcp()

◆ enable_rtp_extension()

◆ generate_fmtp_attr()

◆ generate_rtpmap_attr()

◆ get_codecs()

◆ get_media_encryption_type()

◆ load_module()

◆ media_session_rtcp_read_callback()

◆ media_session_rtp_read_callback()

◆ media_session_rtp_write_callback()

◆ media_stream_has_crypto()

◆ negotiate_incoming_sdp_stream()

◆ parse_dtls_attrib()

◆ process_extmap_attributes()

◆ process_ice_attributes()

◆ process_ice_auth_attrb()

◆ process_ssrc_attributes()

◆ rtp_check_timeout()

◆ send_keepalive()

◆ set_caps()

◆ set_ice_components()

◆ set_incoming_call_offer_cap()

◆ setup_dtls_srtp()

◆ setup_media_encryption()

◆ setup_sdes_srtp()

◆ setup_srtp()

◆ stream_destroy()

◆ stream_stop()

◆ unload_module()

◆ video_info_incoming_request()

Variable Documentation

◆ __mod_info

◆ address_rtp

◆ ast_module_info

◆ audio_sdp_handler

◆ sched

◆ STR_AUDIO

◆ STR_VIDEO

◆ video_info_supplement

◆ video_sdp_handler