Asterisk - The Open Source Telephony Project  21.4.1
Functions | Variables
res_pjsip_sdp_rtp.c File Reference

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"

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.
 
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.
 
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_moduleAST_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.
 
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.
 
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.
 
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.
 
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.
 
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.
 
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.
 
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 pjmedia_sdp_attr * generate_rtpmap_attr2 (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, int sample_rate)
 
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, struct ast_format_cap *astformats)
 
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
 
static int load_module (void)
 Load the module. More...
 
static struct ast_framemedia_session_rtcp_read_callback (struct ast_sip_session *session, struct ast_sip_session_media *session_media)
 
static struct ast_framemedia_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
 
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.
 
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.
 
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.
 
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.
 
static int rtp_check_timeout (const void *data)
 Check whether RTP is being received or not.
 
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_capset_incoming_call_offer_cap (struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_media *stream)
 
static void set_session_media_remotely_held (struct ast_sip_session_media *session_media, const struct ast_sip_session *session, const pjmedia_sdp_media *media, const struct ast_stream *stream, const struct ast_sockaddr *addrs)
 
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.
 
static void stream_stop (struct ast_sip_session_media *session_media)
 Function which stops the RTP instance.
 
static int unload_module (void)
 Unloads the sdp RTP/AVP module from Asterisk.
 
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 = "da6642af068ee5e6490c5b1d2cc1d238" , .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.
 
static const struct ast_module_infoast_module_info = &__mod_info
 
static struct ast_sip_session_sdp_handler audio_sdp_handler
 SDP handler for 'audio' media stream.
 
static struct ast_sched_contextsched
 Scheduler for RTCP purposes.
 
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.
 

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

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 2486 of file res_pjsip_sdp_rtp.c.

References address_rtp, ast_check_ipv6(), AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, ast_sched_context_create(), ast_sched_start_thread(), ast_sockaddr_parse(), and unload_module().

2487 {
2488  if (ast_check_ipv6()) {
2489  ast_sockaddr_parse(&address_rtp, "::", 0);
2490  } else {
2491  ast_sockaddr_parse(&address_rtp, "0.0.0.0", 0);
2492  }
2493 
2494  if (!(sched = ast_sched_context_create())) {
2495  ast_log(LOG_ERROR, "Unable to create scheduler context.\n");
2496  goto end;
2497  }
2498 
2500  ast_log(LOG_ERROR, "Unable to create scheduler context thread.\n");
2501  goto end;
2502  }
2503 
2504  if (ast_sip_session_register_sdp_handler(&audio_sdp_handler, STR_AUDIO)) {
2505  ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_AUDIO);
2506  goto end;
2507  }
2508 
2509  if (ast_sip_session_register_sdp_handler(&video_sdp_handler, STR_VIDEO)) {
2510  ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_VIDEO);
2511  goto end;
2512  }
2513 
2514  ast_sip_session_register_supplement(&video_info_supplement);
2515 
2516  return AST_MODULE_LOAD_SUCCESS;
2517 end:
2518  unload_module();
2519 
2520  return AST_MODULE_LOAD_DECLINE;
2521 }
int ast_sched_start_thread(struct ast_sched_context *con)
Start a thread for processing scheduler entries.
Definition: sched.c:197
int ast_sockaddr_parse(struct ast_sockaddr *addr, const char *str, int flags)
Parse an IPv4 or IPv6 address string.
Definition: netsock2.c:230
Definition: sched.c:76
static struct ast_sockaddr address_rtp
Address for RTP.
static int unload_module(void)
Unloads the sdp RTP/AVP module from Asterisk.
static struct ast_sip_session_sdp_handler video_sdp_handler
SDP handler for 'video' media stream.
int ast_check_ipv6(void)
Test that an OS supports IPv6 Networking.
Definition: utils.c:2792
struct ast_sched_context * ast_sched_context_create(void)
Create a scheduler context.
Definition: sched.c:238
static struct ast_sip_session_sdp_handler audio_sdp_handler
SDP handler for 'audio' media stream.
Module has failed to load, may be in an inconsistent state.
Definition: module.h:78

Variable Documentation

struct ast_sip_session_supplement video_info_supplement
static
Initial value:
= {
.method = "INFO",
.incoming_request = video_info_incoming_request,
}

Definition at line 2457 of file res_pjsip_sdp_rtp.c.