Asterisk - The Open Source Telephony Project
21.4.1
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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_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. | |
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_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 | |
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_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 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_info * | ast_module_info = &__mod_info |
static struct ast_sip_session_sdp_handler | audio_sdp_handler |
SDP handler for 'audio' media stream. | |
static struct ast_sched_context * | sched |
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. | |
SIP SDP media stream handling.
Definition in file res_pjsip_sdp_rtp.c.
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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().
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Definition at line 2457 of file res_pjsip_sdp_rtp.c.