#include "asterisk.h"
#include <math.h>
#include "asterisk/channel.h"
#include "asterisk/frame.h"
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/manager.h"
#include "asterisk/options.h"
#include "asterisk/astobj2.h"
#include "asterisk/pbx.h"
#include "asterisk/translate.h"
#include "asterisk/netsock2.h"
#include "asterisk/framehook.h"
Go to the source code of this file.
Data Structures | |
struct | ast_rtp_instance |
struct | ast_rtp_mime_type |
struct | engines |
struct | glues |
Functions | |
void | ast_rtp_codecs_packetization_set (struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs) |
Set codec packetization preferences. | |
int | ast_rtp_codecs_payload_code (struct ast_rtp_codecs *codecs, const int asterisk_format, const format_t code) |
Retrieve a payload based on whether it is an Asterisk format and the code. | |
void | ast_rtp_codecs_payload_formats (struct ast_rtp_codecs *codecs, format_t *astformats, int *nonastformats) |
Retrieve all formats that were found. | |
ast_rtp_payload_type | ast_rtp_codecs_payload_lookup (struct ast_rtp_codecs *codecs, int payload) |
Retrieve payload information by payload. | |
void | ast_rtp_codecs_payloads_clear (struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance) |
Clear payload information from an RTP instance. | |
void | ast_rtp_codecs_payloads_copy (struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance) |
Copy payload information from one RTP instance to another. | |
void | ast_rtp_codecs_payloads_default (struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance) |
Set payload information on an RTP instance to the default. | |
void | ast_rtp_codecs_payloads_set_m_type (struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload) |
Record payload information that was seen in an m= SDP line. | |
int | ast_rtp_codecs_payloads_set_rtpmap_type (struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options) |
Record payload information that was seen in an a=rtpmap: SDP line. | |
int | ast_rtp_codecs_payloads_set_rtpmap_type_rate (struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt, char *mimetype, char *mimesubtype, enum ast_rtp_options options, unsigned int sample_rate) |
Set payload type to a known MIME media type for a codec with a specific sample rate. | |
void | ast_rtp_codecs_payloads_unset (struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload) |
Remove payload information. | |
int | ast_rtp_engine_register2 (struct ast_rtp_engine *engine, struct ast_module *module) |
Register an RTP engine. | |
int | ast_rtp_engine_register_srtp (struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res) |
int | ast_rtp_engine_srtp_is_registered (void) |
int | ast_rtp_engine_unregister (struct ast_rtp_engine *engine) |
Unregister an RTP engine. | |
void | ast_rtp_engine_unregister_srtp (void) |
int | ast_rtp_glue_register2 (struct ast_rtp_glue *glue, struct ast_module *module) |
Register RTP glue. | |
int | ast_rtp_glue_unregister (struct ast_rtp_glue *glue) |
Unregister RTP glue. | |
int | ast_rtp_instance_activate (struct ast_rtp_instance *instance) |
Indicate to the RTP engine that packets are now expected to be sent/received on the RTP instance. | |
int | ast_rtp_instance_add_srtp_policy (struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy) |
Add or replace the SRTP policies for the given RTP instance. | |
format_t | ast_rtp_instance_available_formats (struct ast_rtp_instance *instance, format_t to_endpoint, format_t to_asterisk) |
Request the formats that can be transcoded. | |
enum ast_bridge_result | ast_rtp_instance_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms) |
Bridge two channels that use RTP instances. | |
void | ast_rtp_instance_change_source (struct ast_rtp_instance *instance) |
Indicate a new source of audio has dropped in and the ssrc should change. | |
int | ast_rtp_instance_destroy (struct ast_rtp_instance *instance) |
Destroy an RTP instance. | |
int | ast_rtp_instance_dtmf_begin (struct ast_rtp_instance *instance, char digit) |
Begin sending a DTMF digit. | |
int | ast_rtp_instance_dtmf_end (struct ast_rtp_instance *instance, char digit) |
Stop sending a DTMF digit. | |
int | ast_rtp_instance_dtmf_end_with_duration (struct ast_rtp_instance *instance, char digit, unsigned int duration) |
enum ast_rtp_dtmf_mode | ast_rtp_instance_dtmf_mode_get (struct ast_rtp_instance *instance) |
Get the DTMF mode of an RTP instance. | |
int | ast_rtp_instance_dtmf_mode_set (struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode) |
Set the DTMF mode that should be used. | |
int | ast_rtp_instance_early_bridge (struct ast_channel *c0, struct ast_channel *c1) |
Early bridge two channels that use RTP instances. | |
void | ast_rtp_instance_early_bridge_make_compatible (struct ast_channel *c0, struct ast_channel *c1) |
Make two channels compatible for early bridging. | |
int | ast_rtp_instance_fd (struct ast_rtp_instance *instance, int rtcp) |
Get the file descriptor for an RTP session (or RTCP). | |
ast_rtp_glue * | ast_rtp_instance_get_active_glue (struct ast_rtp_instance *instance) |
Get the RTP glue in use on an RTP instance. | |
int | ast_rtp_instance_get_and_cmp_local_address (struct ast_rtp_instance *instance, struct ast_sockaddr *address) |
Get the address of the local endpoint that we are sending RTP to, comparing its address to another. | |
int | ast_rtp_instance_get_and_cmp_remote_address (struct ast_rtp_instance *instance, struct ast_sockaddr *address) |
Get the address of the remote endpoint that we are sending RTP to, comparing its address to another. | |
ast_rtp_instance * | ast_rtp_instance_get_bridged (struct ast_rtp_instance *instance) |
Get the other RTP instance that an instance is bridged to. | |
ast_channel * | ast_rtp_instance_get_chan (struct ast_rtp_instance *instance) |
Get the channel that is associated with an RTP instance while in a bridge. | |
ast_rtp_codecs * | ast_rtp_instance_get_codecs (struct ast_rtp_instance *instance) |
Get the codecs structure of an RTP instance. | |
void * | ast_rtp_instance_get_data (struct ast_rtp_instance *instance) |
Get the data portion of an RTP instance. | |
ast_rtp_engine * | ast_rtp_instance_get_engine (struct ast_rtp_instance *instance) |
Get the RTP engine in use on an RTP instance. | |
void * | ast_rtp_instance_get_extended_prop (struct ast_rtp_instance *instance, int property) |
Get the value of an RTP instance extended property. | |
ast_rtp_glue * | ast_rtp_instance_get_glue (const char *type) |
Get the RTP glue that binds a channel to the RTP engine. | |
int | ast_rtp_instance_get_hold_timeout (struct ast_rtp_instance *instance) |
Get the RTP timeout value for when an RTP instance is on hold. | |
int | ast_rtp_instance_get_keepalive (struct ast_rtp_instance *instance) |
Get the RTP keepalive interval. | |
void | ast_rtp_instance_get_local_address (struct ast_rtp_instance *instance, struct ast_sockaddr *address) |
Get the local address that we are expecting RTP on. | |
int | ast_rtp_instance_get_prop (struct ast_rtp_instance *instance, enum ast_rtp_property property) |
Get the value of an RTP instance property. | |
char * | ast_rtp_instance_get_quality (struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size) |
Retrieve quality statistics about an RTP instance. | |
void | ast_rtp_instance_get_remote_address (struct ast_rtp_instance *instance, struct ast_sockaddr *address) |
Get the address of the remote endpoint that we are sending RTP to. | |
ast_srtp * | ast_rtp_instance_get_srtp (struct ast_rtp_instance *instance) |
Obtain the SRTP instance associated with an RTP instance. | |
int | ast_rtp_instance_get_stats (struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat) |
Retrieve statistics about an RTP instance. | |
int | ast_rtp_instance_get_timeout (struct ast_rtp_instance *instance) |
Get the RTP timeout value. | |
int | ast_rtp_instance_make_compatible (struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer) |
Request that the underlying RTP engine make two RTP instances compatible with eachother. | |
ast_rtp_instance * | ast_rtp_instance_new (const char *engine_name, struct sched_context *sched, const struct ast_sockaddr *sa, void *data) |
Create a new RTP instance. | |
ast_frame * | ast_rtp_instance_read (struct ast_rtp_instance *instance, int rtcp) |
Receive a frame over RTP. | |
int | ast_rtp_instance_sendcng (struct ast_rtp_instance *instance, int level) |
Send a comfort noise packet to the RTP instance. | |
int | ast_rtp_instance_set_alt_remote_address (struct ast_rtp_instance *instance, const struct ast_sockaddr *address) |
Set the address of an an alternate RTP address to receive from. | |
void | ast_rtp_instance_set_data (struct ast_rtp_instance *instance, void *data) |
Set the data portion of an RTP instance. | |
void | ast_rtp_instance_set_extended_prop (struct ast_rtp_instance *instance, int property, void *value) |
Set the value of an RTP instance extended property. | |
void | ast_rtp_instance_set_hold_timeout (struct ast_rtp_instance *instance, int timeout) |
Set the RTP timeout value for when the instance is on hold. | |
void | ast_rtp_instance_set_keepalive (struct ast_rtp_instance *instance, int interval) |
Set the RTP keepalive interval. | |
int | ast_rtp_instance_set_local_address (struct ast_rtp_instance *instance, const struct ast_sockaddr *address) |
Set the address that we are expecting to receive RTP on. | |
void | ast_rtp_instance_set_prop (struct ast_rtp_instance *instance, enum ast_rtp_property property, int value) |
Set the value of an RTP instance property. | |
int | ast_rtp_instance_set_qos (struct ast_rtp_instance *instance, int tos, int cos, const char *desc) |
Set QoS parameters on an RTP session. | |
int | ast_rtp_instance_set_read_format (struct ast_rtp_instance *instance, format_t format) |
Request that the underlying RTP engine provide audio frames in a specific format. | |
int | ast_rtp_instance_set_remote_address (struct ast_rtp_instance *instance, const struct ast_sockaddr *address) |
Set the address of the remote endpoint that we are sending RTP to. | |
void | ast_rtp_instance_set_stats_vars (struct ast_channel *chan, struct ast_rtp_instance *instance) |
Set standard statistics from an RTP instance on a channel. | |
void | ast_rtp_instance_set_timeout (struct ast_rtp_instance *instance, int timeout) |
Set the RTP timeout value. | |
int | ast_rtp_instance_set_write_format (struct ast_rtp_instance *instance, format_t format) |
Tell underlying RTP engine that audio frames will be provided in a specific format. | |
void | ast_rtp_instance_stop (struct ast_rtp_instance *instance) |
Stop an RTP instance. | |
void | ast_rtp_instance_stun_request (struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username) |
Request that the underlying RTP engine send a STUN BIND request. | |
void | ast_rtp_instance_update_source (struct ast_rtp_instance *instance) |
Indicate that the RTP marker bit should be set on an RTP stream. | |
int | ast_rtp_instance_write (struct ast_rtp_instance *instance, struct ast_frame *frame) |
Send a frame out over RTP. | |
char * | ast_rtp_lookup_mime_multiple2 (struct ast_str *buf, const format_t capability, const int asterisk_format, enum ast_rtp_options options) |
Convert formats into a string and put them into a buffer. | |
const char * | ast_rtp_lookup_mime_subtype2 (const int asterisk_format, const format_t code, enum ast_rtp_options options) |
Retrieve mime subtype information on a payload. | |
unsigned int | ast_rtp_lookup_sample_rate2 (int asterisk_format, format_t code) |
Get the sample rate associated with known RTP payload types. | |
int | ast_rtp_red_buffer (struct ast_rtp_instance *instance, struct ast_frame *frame) |
Buffer a frame in an RTP instance for RED. | |
int | ast_rtp_red_init (struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations) |
Initialize RED support on an RTP instance. | |
static void | instance_destructor (void *obj) |
static enum ast_bridge_result | local_bridge_loop (struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1) |
static enum ast_bridge_result | remote_bridge_loop (struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, struct ast_rtp_instance *vinstance0, struct ast_rtp_instance *vinstance1, struct ast_rtp_instance *tinstance0, struct ast_rtp_instance *tinstance1, struct ast_rtp_glue *glue0, struct ast_rtp_glue *glue1, format_t codec0, format_t codec1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1) |
static void | unref_instance_cond (struct ast_rtp_instance **instance) |
Conditionally unref an rtp instance. | |
Variables | |
static struct ast_rtp_mime_type | ast_rtp_mime_types [] |
ast_srtp_res * | res_srtp = NULL |
ast_srtp_policy_res * | res_srtp_policy = NULL |
static struct ast_rtp_payload_type | static_RTP_PT [AST_RTP_MAX_PT] |
Mapping between Asterisk codecs and rtp payload types. |
Definition in file rtp_engine.c.
void ast_rtp_codecs_packetization_set | ( | struct ast_rtp_codecs * | codecs, | |
struct ast_rtp_instance * | instance, | |||
struct ast_codec_pref * | prefs | |||
) |
Set codec packetization preferences.
codecs | Codecs structure to muck with | |
instance | Optionally the instance that the codecs structure belongs to | |
prefs | Codec packetization preferences |
ast_rtp_codecs_packetization_set(&codecs, NULL, &prefs);
This sets the packetization preferences pointed to by prefs on the codecs structure pointed to by codecs.
Definition at line 727 of file rtp_engine.c.
References ast_rtp_instance::codecs, ast_rtp_instance::engine, ast_rtp_engine::packetization_set, ast_rtp_codecs::pref, and prefs.
Referenced by __oh323_rtp_create(), check_peer_ok(), create_addr_from_peer(), gtalk_new(), jingle_new(), process_sdp_a_audio(), set_peer_capabilities(), start_rtp(), and transmit_response_with_sdp().
00728 { 00729 codecs->pref = *prefs; 00730 00731 if (instance && instance->engine->packetization_set) { 00732 instance->engine->packetization_set(instance, &instance->codecs.pref); 00733 } 00734 }
int ast_rtp_codecs_payload_code | ( | struct ast_rtp_codecs * | codecs, | |
const int | asterisk_format, | |||
const format_t | code | |||
) |
Retrieve a payload based on whether it is an Asterisk format and the code.
codecs | Codecs structure to look in | |
asterisk_format | Non-zero if the given code is an Asterisk format value | |
code | The format to look for |
Numerical | payload |
int payload = ast_rtp_codecs_payload_code(&codecs, 1, AST_FORMAT_ULAW);
This looks for the numerical payload for ULAW in the codecs structure.
Definition at line 654 of file rtp_engine.c.
References AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, and ast_rtp_codecs::payloads.
Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_rtp_dtmf_begin(), ast_rtp_write(), bridge_p2p_rtp_write(), multicast_rtp_write(), and start_rtp().
00655 { 00656 int i; 00657 00658 for (i = 0; i < AST_RTP_MAX_PT; i++) { 00659 if (codecs->payloads[i].asterisk_format == asterisk_format && codecs->payloads[i].code == code) { 00660 return i; 00661 } 00662 } 00663 00664 for (i = 0; i < AST_RTP_MAX_PT; i++) { 00665 if (static_RTP_PT[i].asterisk_format == asterisk_format && static_RTP_PT[i].code == code) { 00666 return i; 00667 } 00668 } 00669 00670 return -1; 00671 }
void ast_rtp_codecs_payload_formats | ( | struct ast_rtp_codecs * | codecs, | |
format_t * | astformats, | |||
int * | nonastformats | |||
) |
Retrieve all formats that were found.
codecs | Codecs structure to look in | |
astformats | An integer to put the Asterisk formats in | |
nonastformats | An integer to put the non-Asterisk formats in |
int astformats, nonastformats;
ast_rtp_codecs_payload_Formats(&codecs, &astformats, &nonastformats);
This retrieves all the formats known about in the codecs structure and puts the Asterisk ones in the integer pointed to by astformats and the non-Asterisk ones in the integer pointed to by nonastformats.
Definition at line 636 of file rtp_engine.c.
References ast_debug, AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, and ast_rtp_codecs::payloads.
Referenced by gtalk_is_answered(), gtalk_newcall(), and process_sdp().
00637 { 00638 int i; 00639 00640 *astformats = *nonastformats = 0; 00641 00642 for (i = 0; i < AST_RTP_MAX_PT; i++) { 00643 if (codecs->payloads[i].code) { 00644 ast_debug(1, "Incorporating payload %d on %p\n", i, codecs); 00645 } 00646 if (codecs->payloads[i].asterisk_format) { 00647 *astformats |= codecs->payloads[i].code; 00648 } else { 00649 *nonastformats |= codecs->payloads[i].code; 00650 } 00651 } 00652 }
struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup | ( | struct ast_rtp_codecs * | codecs, | |
int | payload | |||
) |
Retrieve payload information by payload.
codecs | Codecs structure to look in | |
payload | Numerical payload to look up |
Payload | information |
struct ast_rtp_payload_type payload_type; payload_type = ast_rtp_codecs_payload_lookup(&codecs, 0);
This looks up the information for payload '0' from the codecs structure.
Definition at line 618 of file rtp_engine.c.
References AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, and static_RTP_PT.
Referenced by ast_rtp_read(), ast_rtp_sendcng(), bridge_p2p_rtp_write(), process_sdp_a_audio(), and setup_rtp_connection().
00619 { 00620 struct ast_rtp_payload_type result = { .asterisk_format = 0, }; 00621 00622 if (payload < 0 || payload >= AST_RTP_MAX_PT) { 00623 return result; 00624 } 00625 00626 result.asterisk_format = codecs->payloads[payload].asterisk_format; 00627 result.code = codecs->payloads[payload].code; 00628 00629 if (!result.code) { 00630 result = static_RTP_PT[payload]; 00631 } 00632 00633 return result; 00634 }
void ast_rtp_codecs_payloads_clear | ( | struct ast_rtp_codecs * | codecs, | |
struct ast_rtp_instance * | instance | |||
) |
Clear payload information from an RTP instance.
codecs | The codecs structure that payloads will be cleared from | |
instance | Optionally the instance that the codecs structure belongs to |
struct ast_rtp_codecs codecs; ast_rtp_codecs_payloads_clear(&codecs, NULL);
This clears the codecs structure and puts it into a pristine state.
Definition at line 488 of file rtp_engine.c.
References AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, ast_rtp_instance::engine, ast_rtp_engine::payload_set, and ast_rtp_codecs::payloads.
Referenced by gtalk_alloc(), and process_sdp().
00489 { 00490 int i; 00491 00492 for (i = 0; i < AST_RTP_MAX_PT; i++) { 00493 codecs->payloads[i].asterisk_format = 0; 00494 codecs->payloads[i].code = 0; 00495 if (instance && instance->engine && instance->engine->payload_set) { 00496 instance->engine->payload_set(instance, i, 0, 0); 00497 } 00498 } 00499 }
void ast_rtp_codecs_payloads_copy | ( | struct ast_rtp_codecs * | src, | |
struct ast_rtp_codecs * | dest, | |||
struct ast_rtp_instance * | instance | |||
) |
Copy payload information from one RTP instance to another.
src | The source codecs structure | |
dest | The destination codecs structure that the values from src will be copied to | |
instance | Optionally the instance that the dst codecs structure belongs to |
ast_rtp_codecs_payloads_copy(&codecs0, &codecs1, NULL);
This copies the payloads from the codecs0 structure to the codecs1 structure, overwriting any current values.
Definition at line 516 of file rtp_engine.c.
References ast_debug, AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, ast_rtp_instance::engine, ast_rtp_engine::payload_set, and ast_rtp_codecs::payloads.
Referenced by ast_rtp_instance_early_bridge_make_compatible(), and process_sdp().
00517 { 00518 int i; 00519 00520 for (i = 0; i < AST_RTP_MAX_PT; i++) { 00521 if (src->payloads[i].code) { 00522 ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest); 00523 dest->payloads[i].asterisk_format = src->payloads[i].asterisk_format; 00524 dest->payloads[i].code = src->payloads[i].code; 00525 if (instance && instance->engine && instance->engine->payload_set) { 00526 instance->engine->payload_set(instance, i, dest->payloads[i].asterisk_format, dest->payloads[i].code); 00527 } 00528 } 00529 } 00530 }
void ast_rtp_codecs_payloads_default | ( | struct ast_rtp_codecs * | codecs, | |
struct ast_rtp_instance * | instance | |||
) |
Set payload information on an RTP instance to the default.
codecs | The codecs structure to set defaults on | |
instance | Optionally the instance that the codecs structure belongs to |
struct ast_rtp_codecs codecs; ast_rtp_codecs_payloads_default(&codecs, NULL);
This sets the default payloads on the codecs structure.
Definition at line 501 of file rtp_engine.c.
References AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, ast_rtp_instance::engine, ast_rtp_engine::payload_set, ast_rtp_codecs::payloads, and static_RTP_PT.
00502 { 00503 int i; 00504 00505 for (i = 0; i < AST_RTP_MAX_PT; i++) { 00506 if (static_RTP_PT[i].code) { 00507 codecs->payloads[i].asterisk_format = static_RTP_PT[i].asterisk_format; 00508 codecs->payloads[i].code = static_RTP_PT[i].code; 00509 if (instance && instance->engine && instance->engine->payload_set) { 00510 instance->engine->payload_set(instance, i, codecs->payloads[i].asterisk_format, codecs->payloads[i].code); 00511 } 00512 } 00513 } 00514 }
void ast_rtp_codecs_payloads_set_m_type | ( | struct ast_rtp_codecs * | codecs, | |
struct ast_rtp_instance * | instance, | |||
int | payload | |||
) |
Record payload information that was seen in an m= SDP line.
codecs | The codecs structure to muck with | |
instance | Optionally the instance that the codecs structure belongs to | |
payload | Numerical payload that was seen in the m= SDP line |
ast_rtp_codecs_payloads_set_m_type(&codecs, NULL, 0);
This records that the numerical payload '0' was seen in the codecs structure.
Definition at line 532 of file rtp_engine.c.
References ast_debug, AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, ast_rtp_instance::engine, ast_rtp_engine::payload_set, ast_rtp_codecs::payloads, and static_RTP_PT.
Referenced by gtalk_is_answered(), gtalk_newcall(), jingle_newcall(), and process_sdp().
00533 { 00534 if (payload < 0 || payload >= AST_RTP_MAX_PT || !static_RTP_PT[payload].code) { 00535 return; 00536 } 00537 00538 codecs->payloads[payload].asterisk_format = static_RTP_PT[payload].asterisk_format; 00539 codecs->payloads[payload].code = static_RTP_PT[payload].code; 00540 00541 ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs); 00542 00543 if (instance && instance->engine && instance->engine->payload_set) { 00544 instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, codecs->payloads[payload].code); 00545 } 00546 }
int ast_rtp_codecs_payloads_set_rtpmap_type | ( | struct ast_rtp_codecs * | codecs, | |
struct ast_rtp_instance * | instance, | |||
int | payload, | |||
char * | mimetype, | |||
char * | mimesubtype, | |||
enum ast_rtp_options | options | |||
) |
Record payload information that was seen in an a=rtpmap: SDP line.
codecs | The codecs structure to muck with | |
instance | Optionally the instance that the codecs structure belongs to | |
payload | Numerical payload that was seen in the a=rtpmap: SDP line | |
mimetype | The string mime type that was seen | |
mimesubtype | The strin mime sub type that was seen | |
options | Optional options that may change the behavior of this specific payload |
0 | success | |
-1 | failure, invalid payload numbe | |
-2 | failure, unknown mimetype |
ast_rtp_codecs_payloads_set_rtpmap_type(&codecs, NULL, 0, "audio", "PCMU", 0);
This records that the numerical payload '0' was seen with mime type 'audio' and sub mime type 'PCMU' in the codecs structure.
Definition at line 597 of file rtp_engine.c.
References ast_rtp_codecs_payloads_set_rtpmap_type_rate().
Referenced by __oh323_rtp_create(), gtalk_is_answered(), gtalk_newcall(), jingle_newcall(), process_sdp(), set_dtmf_payload(), and setup_rtp_connection().
00598 { 00599 return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0); 00600 }
int ast_rtp_codecs_payloads_set_rtpmap_type_rate | ( | struct ast_rtp_codecs * | codecs, | |
struct ast_rtp_instance * | instance, | |||
int | pt, | |||
char * | mimetype, | |||
char * | mimesubtype, | |||
enum ast_rtp_options | options, | |||
unsigned int | sample_rate | |||
) |
Set payload type to a known MIME media type for a codec with a specific sample rate.
codecs | RTP structure to modify | |
instance | Optionally the instance that the codecs structure belongs to | |
pt | Payload type entry to modify | |
mimetype | top-level MIME type of media stream (typically "audio", "video", "text", etc.) | |
mimesubtype | MIME subtype of media stream (typically a codec name) | |
options | Zero or more flags from the ast_rtp_options enum | |
sample_rate | The sample rate of the media stream |
0 | on success | |
-1 | if the payload type is out of range | |
-2 | if the mimeType/mimeSubtype combination was not found |
Definition at line 548 of file rtp_engine.c.
References ARRAY_LEN, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_RTP_MAX_PT, ast_rtp_mime_types, AST_RTP_OPT_G726_NONSTANDARD, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, ast_rtp_instance::engine, ast_rtp_engine::payload_set, and ast_rtp_codecs::payloads.
Referenced by ast_rtp_codecs_payloads_set_rtpmap_type(), process_sdp_a_audio(), process_sdp_a_text(), and process_sdp_a_video().
00552 { 00553 unsigned int i; 00554 int found = 0; 00555 00556 if (pt < 0 || pt >= AST_RTP_MAX_PT) 00557 return -1; /* bogus payload type */ 00558 00559 for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) { 00560 const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i]; 00561 00562 if (strcasecmp(mimesubtype, t->subtype)) { 00563 continue; 00564 } 00565 00566 if (strcasecmp(mimetype, t->type)) { 00567 continue; 00568 } 00569 00570 /* if both sample rates have been supplied, and they don't match, 00571 * then this not a match; if one has not been supplied, then the 00572 * rates are not compared */ 00573 if (sample_rate && t->sample_rate && 00574 (sample_rate != t->sample_rate)) { 00575 continue; 00576 } 00577 00578 found = 1; 00579 codecs->payloads[pt] = t->payload_type; 00580 00581 if ((t->payload_type.code == AST_FORMAT_G726) && 00582 t->payload_type.asterisk_format && 00583 (options & AST_RTP_OPT_G726_NONSTANDARD)) { 00584 codecs->payloads[pt].code = AST_FORMAT_G726_AAL2; 00585 } 00586 00587 if (instance && instance->engine && instance->engine->payload_set) { 00588 instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, codecs->payloads[i].code); 00589 } 00590 00591 break; 00592 } 00593 00594 return (found ? 0 : -2); 00595 }
void ast_rtp_codecs_payloads_unset | ( | struct ast_rtp_codecs * | codecs, | |
struct ast_rtp_instance * | instance, | |||
int | payload | |||
) |
Remove payload information.
codecs | The codecs structure to muck with | |
instance | Optionally the instance that the codecs structure belongs to | |
payload | Numerical payload to unset |
ast_rtp_codecs_payloads_unset(&codecs, NULL, 0);
This clears the payload '0' from the codecs structure. It will be as if it was never set.
Definition at line 602 of file rtp_engine.c.
References ast_debug, AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, ast_rtp_instance::engine, ast_rtp_engine::payload_set, and ast_rtp_codecs::payloads.
Referenced by process_sdp_a_audio(), and process_sdp_a_video().
00603 { 00604 if (payload < 0 || payload >= AST_RTP_MAX_PT) { 00605 return; 00606 } 00607 00608 ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs); 00609 00610 codecs->payloads[payload].asterisk_format = 0; 00611 codecs->payloads[payload].code = 0; 00612 00613 if (instance && instance->engine && instance->engine->payload_set) { 00614 instance->engine->payload_set(instance, payload, 0, 0); 00615 } 00616 }
int ast_rtp_engine_register2 | ( | struct ast_rtp_engine * | engine, | |
struct ast_module * | module | |||
) |
Register an RTP engine.
engine | Structure of the RTP engine to register | |
module | Module that the RTP engine is part of |
0 | success | |
-1 | failure |
ast_rtp_engine_register2(&example_rtp_engine, NULL);
This registers the RTP engine declared as example_rtp_engine with the RTP engine core, but does not associate a module with it.
Definition at line 188 of file rtp_engine.c.
References ast_log(), AST_RWLIST_INSERT_TAIL, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_strlen_zero(), ast_verb, ast_rtp_engine::destroy, ast_rtp_engine::entry, LOG_WARNING, ast_rtp_engine::mod, ast_rtp_engine::name, ast_rtp_engine::new, ast_rtp_engine::read, and ast_rtp_engine::write.
00189 { 00190 struct ast_rtp_engine *current_engine; 00191 00192 /* Perform a sanity check on the engine structure to make sure it has the basics */ 00193 if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) { 00194 ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown"); 00195 return -1; 00196 } 00197 00198 /* Link owner module to the RTP engine for reference counting purposes */ 00199 engine->mod = module; 00200 00201 AST_RWLIST_WRLOCK(&engines); 00202 00203 /* Ensure that no two modules with the same name are registered at the same time */ 00204 AST_RWLIST_TRAVERSE(&engines, current_engine, entry) { 00205 if (!strcmp(current_engine->name, engine->name)) { 00206 ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name); 00207 AST_RWLIST_UNLOCK(&engines); 00208 return -1; 00209 } 00210 } 00211 00212 /* The engine survived our critique. Off to the list it goes to be used */ 00213 AST_RWLIST_INSERT_TAIL(&engines, engine, entry); 00214 00215 AST_RWLIST_UNLOCK(&engines); 00216 00217 ast_verb(2, "Registered RTP engine '%s'\n", engine->name); 00218 00219 return 0; 00220 }
int ast_rtp_engine_register_srtp | ( | struct ast_srtp_res * | srtp_res, | |
struct ast_srtp_policy_res * | policy_res | |||
) |
Definition at line 1767 of file rtp_engine.c.
References policy_res, res_srtp, res_srtp_policy, and srtp_res.
Referenced by res_srtp_init().
01768 { 01769 if (res_srtp || res_srtp_policy) { 01770 return -1; 01771 } 01772 if (!srtp_res || !policy_res) { 01773 return -1; 01774 } 01775 01776 res_srtp = srtp_res; 01777 res_srtp_policy = policy_res; 01778 01779 return 0; 01780 }
int ast_rtp_engine_srtp_is_registered | ( | void | ) |
Definition at line 1788 of file rtp_engine.c.
References res_srtp, and res_srtp_policy.
Referenced by sdp_crypto_activate(), sdp_crypto_process(), sdp_crypto_setup(), set_crypto_policy(), and setup_srtp().
01789 { 01790 return res_srtp && res_srtp_policy; 01791 }
int ast_rtp_engine_unregister | ( | struct ast_rtp_engine * | engine | ) |
Unregister an RTP engine.
engine | Structure of the RTP engine to unregister |
0 | success | |
-1 | failure |
ast_rtp_engine_unregister(&example_rtp_engine);
This unregisters the RTP engine declared as example_rtp_engine from the RTP engine core. If a module reference was provided when it was registered then this will only be called once the RTP engine is no longer in use.
Definition at line 222 of file rtp_engine.c.
References AST_RWLIST_REMOVE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_verb, ast_rtp_engine::entry, and ast_rtp_engine::name.
Referenced by load_module(), and unload_module().
00223 { 00224 struct ast_rtp_engine *current_engine = NULL; 00225 00226 AST_RWLIST_WRLOCK(&engines); 00227 00228 if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) { 00229 ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name); 00230 } 00231 00232 AST_RWLIST_UNLOCK(&engines); 00233 00234 return current_engine ? 0 : -1; 00235 }
void ast_rtp_engine_unregister_srtp | ( | void | ) |
Definition at line 1782 of file rtp_engine.c.
References res_srtp, and res_srtp_policy.
Referenced by res_srtp_shutdown().
01783 { 01784 res_srtp = NULL; 01785 res_srtp_policy = NULL; 01786 }
int ast_rtp_glue_register2 | ( | struct ast_rtp_glue * | glue, | |
struct ast_module * | module | |||
) |
Register RTP glue.
glue | The glue to register | |
module | Module that the RTP glue is part of |
0 | success | |
-1 | failure |
ast_rtp_glue_register2(&example_rtp_glue, NULL);
This registers the RTP glue declared as example_rtp_glue with the RTP engine core, but does not associate a module with it.
Definition at line 237 of file rtp_engine.c.
References ast_log(), AST_RWLIST_INSERT_TAIL, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_strlen_zero(), ast_verb, ast_rtp_glue::entry, LOG_WARNING, ast_rtp_glue::mod, and ast_rtp_glue::type.
00238 { 00239 struct ast_rtp_glue *current_glue = NULL; 00240 00241 if (ast_strlen_zero(glue->type)) { 00242 return -1; 00243 } 00244 00245 glue->mod = module; 00246 00247 AST_RWLIST_WRLOCK(&glues); 00248 00249 AST_RWLIST_TRAVERSE(&glues, current_glue, entry) { 00250 if (!strcasecmp(current_glue->type, glue->type)) { 00251 ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type); 00252 AST_RWLIST_UNLOCK(&glues); 00253 return -1; 00254 } 00255 } 00256 00257 AST_RWLIST_INSERT_TAIL(&glues, glue, entry); 00258 00259 AST_RWLIST_UNLOCK(&glues); 00260 00261 ast_verb(2, "Registered RTP glue '%s'\n", glue->type); 00262 00263 return 0; 00264 }
int ast_rtp_glue_unregister | ( | struct ast_rtp_glue * | glue | ) |
Unregister RTP glue.
glue | The glue to unregister |
0 | success | |
-1 | failure |
ast_rtp_glue_unregister(&example_rtp_glue);
This unregisters the RTP glue declared as example_rtp_gkue from the RTP engine core. If a module reference was provided when it was registered then this will only be called once the RTP engine is no longer in use.
Definition at line 266 of file rtp_engine.c.
References AST_RWLIST_REMOVE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_verb, ast_rtp_glue::entry, and ast_rtp_glue::type.
Referenced by load_module(), and unload_module().
00267 { 00268 struct ast_rtp_glue *current_glue = NULL; 00269 00270 AST_RWLIST_WRLOCK(&glues); 00271 00272 if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) { 00273 ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type); 00274 } 00275 00276 AST_RWLIST_UNLOCK(&glues); 00277 00278 return current_glue ? 0 : -1; 00279 }
int ast_rtp_instance_activate | ( | struct ast_rtp_instance * | instance | ) |
Indicate to the RTP engine that packets are now expected to be sent/received on the RTP instance.
instance | The RTP instance |
0 | success | |
-1 | failure |
ast_rtp_instance_activate(instance);
This tells the underlying RTP engine of instance that packets will now flow.
Definition at line 1708 of file rtp_engine.c.
References ast_rtp_engine::activate, and ast_rtp_instance::engine.
Referenced by handle_response_invite(), multicast_rtp_call(), and transmit_response_with_sdp().
int ast_rtp_instance_add_srtp_policy | ( | struct ast_rtp_instance * | instance, | |
struct ast_srtp_policy * | remote_policy, | |||
struct ast_srtp_policy * | local_policy | |||
) |
Add or replace the SRTP policies for the given RTP instance.
instance | the RTP instance | |
remote_policy | the remote endpoint's policy | |
local_policy | our policy for this RTP instance's remote endpoint |
0 | Success | |
non-zero | Failure |
Definition at line 1793 of file rtp_engine.c.
References ast_srtp_res::add_stream, ast_srtp_res::create, ast_srtp_res::replace, res_srtp, and ast_rtp_instance::srtp.
Referenced by sdp_crypto_activate().
01794 { 01795 int res = 0; 01796 01797 if (!res_srtp) { 01798 return -1; 01799 } 01800 01801 if (!instance->srtp) { 01802 res = res_srtp->create(&instance->srtp, instance, remote_policy); 01803 } else { 01804 res = res_srtp->replace(&instance->srtp, instance, remote_policy); 01805 } 01806 if (!res) { 01807 res = res_srtp->add_stream(instance->srtp, local_policy); 01808 } 01809 01810 return res; 01811 }
format_t ast_rtp_instance_available_formats | ( | struct ast_rtp_instance * | instance, | |
format_t | to_endpoint, | |||
format_t | to_asterisk | |||
) |
Request the formats that can be transcoded.
instance | The RTP instance | |
to_endpoint | Formats being sent/received towards the endpoint | |
to_asterisk | Formats being sent/received towards Asterisk |
supported | formats |
This sees if it is possible to have ulaw communicated to the endpoint but signed linear received into Asterisk.
Definition at line 1697 of file rtp_engine.c.
References ast_translate_available_formats(), ast_rtp_engine::available_formats, and ast_rtp_instance::engine.
Referenced by sip_call().
01698 { 01699 format_t formats; 01700 01701 if (instance->engine->available_formats && (formats = instance->engine->available_formats(instance, to_endpoint, to_asterisk))) { 01702 return formats; 01703 } 01704 01705 return ast_translate_available_formats(to_endpoint, to_asterisk); 01706 }
enum ast_bridge_result ast_rtp_instance_bridge | ( | struct ast_channel * | c0, | |
struct ast_channel * | c1, | |||
int | flags, | |||
struct ast_frame ** | fo, | |||
struct ast_channel ** | rc, | |||
int | timeoutms | |||
) |
Bridge two channels that use RTP instances.
c0 | First channel part of the bridge | |
c1 | Second channel part of the bridge | |
flags | Bridging flags | |
fo | If a frame needs to be passed up it is stored here | |
rc | Channel that passed the above frame up | |
timeoutms | How long the channels should be bridged for |
Bridge | result |
Definition at line 1255 of file rtp_engine.c.
References AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_check_hangup(), ast_debug, ast_getformatname(), ast_log(), AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_LOCAL, AST_RTP_GLUE_RESULT_REMOTE, ast_rtp_instance_dtmf_mode_get(), ast_rtp_instance_get_glue(), ast_rtp_instance_get_remote_address(), ast_sockaddr_is_ipv4_mapped(), ast_verb, ast_rtp_instance::chan, ast_rtp_engine::dtmf_compatible, ast_rtp_instance::engine, ast_rtp_glue::get_codec, ast_rtp_glue::get_rtp_info, ast_rtp_glue::get_vrtp_info, ast_rtp_instance::glue, ast_rtp_engine::local_bridge, local_bridge_loop(), LOG_WARNING, ast_channel::name, remote_bridge_loop(), ast_sockaddr::ss, ast_channel::tech, ast_channel::tech_pvt, ast_channel_tech::type, and unref_instance_cond().
01256 { 01257 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, 01258 *vinstance0 = NULL, *vinstance1 = NULL, 01259 *tinstance0 = NULL, *tinstance1 = NULL; 01260 struct ast_rtp_glue *glue0, *glue1; 01261 struct ast_sockaddr addr1 = { {0, }, }, addr2 = { {0, }, }; 01262 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01263 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01264 enum ast_bridge_result res = AST_BRIDGE_FAILED; 01265 enum ast_rtp_dtmf_mode dmode; 01266 format_t codec0 = 0, codec1 = 0; 01267 int unlock_chans = 1; 01268 01269 /* Lock both channels so we can look for the glue that binds them together */ 01270 ast_channel_lock(c0); 01271 while (ast_channel_trylock(c1)) { 01272 ast_channel_unlock(c0); 01273 usleep(1); 01274 ast_channel_lock(c0); 01275 } 01276 01277 /* Ensure neither channel got hungup during lock avoidance */ 01278 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 01279 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name); 01280 goto done; 01281 } 01282 01283 /* Grab glue that binds each channel to something using the RTP engine */ 01284 if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) { 01285 ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name); 01286 goto done; 01287 } 01288 01289 audio_glue0_res = glue0->get_rtp_info(c0, &instance0); 01290 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID; 01291 01292 audio_glue1_res = glue1->get_rtp_info(c1, &instance1); 01293 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID; 01294 01295 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */ 01296 if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01297 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01298 } 01299 if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01300 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01301 } 01302 01303 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */ 01304 if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) { 01305 res = AST_BRIDGE_FAILED_NOWARN; 01306 goto done; 01307 } 01308 01309 01310 /* If address families differ, force a local bridge */ 01311 ast_rtp_instance_get_remote_address(instance0, &addr1); 01312 ast_rtp_instance_get_remote_address(instance1, &addr2); 01313 01314 if (addr1.ss.ss_family != addr2.ss.ss_family || 01315 (ast_sockaddr_is_ipv4_mapped(&addr1) != ast_sockaddr_is_ipv4_mapped(&addr2))) { 01316 audio_glue0_res = AST_RTP_GLUE_RESULT_LOCAL; 01317 audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL; 01318 } 01319 01320 /* If we need to get DTMF see if we can do it outside of the RTP stream itself */ 01321 dmode = ast_rtp_instance_dtmf_mode_get(instance0); 01322 if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && dmode) { 01323 res = AST_BRIDGE_FAILED_NOWARN; 01324 goto done; 01325 } 01326 dmode = ast_rtp_instance_dtmf_mode_get(instance1); 01327 if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && dmode) { 01328 res = AST_BRIDGE_FAILED_NOWARN; 01329 goto done; 01330 } 01331 01332 /* If we have gotten to a local bridge make sure that both sides have the same local bridge callback and that they are DTMF compatible */ 01333 if ((audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) && ((instance0->engine->local_bridge != instance1->engine->local_bridge) || (instance0->engine->dtmf_compatible && !instance0->engine->dtmf_compatible(c0, instance0, c1, instance1)))) { 01334 res = AST_BRIDGE_FAILED_NOWARN; 01335 goto done; 01336 } 01337 01338 /* Make sure that codecs match */ 01339 codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0; 01340 codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0; 01341 if (codec0 && codec1 && !(codec0 & codec1)) { 01342 ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n", ast_getformatname(codec0), ast_getformatname(codec1)); 01343 res = AST_BRIDGE_FAILED_NOWARN; 01344 goto done; 01345 } 01346 01347 instance0->glue = glue0; 01348 instance1->glue = glue1; 01349 instance0->chan = c0; 01350 instance1->chan = c1; 01351 01352 /* Depending on the end result for bridging either do a local bridge or remote bridge */ 01353 if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) { 01354 ast_verb(3, "Locally bridging %s and %s\n", c0->name, c1->name); 01355 res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, c0->tech_pvt, c1->tech_pvt); 01356 } else { 01357 ast_verb(3, "Remotely bridging %s and %s\n", c0->name, c1->name); 01358 res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1, 01359 tinstance0, tinstance1, glue0, glue1, codec0, codec1, timeoutms, flags, 01360 fo, rc, c0->tech_pvt, c1->tech_pvt); 01361 } 01362 01363 instance0->glue = NULL; 01364 instance1->glue = NULL; 01365 instance0->chan = NULL; 01366 instance1->chan = NULL; 01367 01368 unlock_chans = 0; 01369 01370 done: 01371 if (unlock_chans) { 01372 ast_channel_unlock(c0); 01373 ast_channel_unlock(c1); 01374 } 01375 01376 unref_instance_cond(&instance0); 01377 unref_instance_cond(&instance1); 01378 unref_instance_cond(&vinstance0); 01379 unref_instance_cond(&vinstance1); 01380 unref_instance_cond(&tinstance0); 01381 unref_instance_cond(&tinstance1); 01382 01383 return res; 01384 }
void ast_rtp_instance_change_source | ( | struct ast_rtp_instance * | instance | ) |
Indicate a new source of audio has dropped in and the ssrc should change.
instance | Instance that the new media source is feeding into |
ast_rtp_instance_change_source(instance);
This indicates that the source of media that is feeding the instance pointed to by instance has changed and that the marker bit should be set and the SSRC updated.
Definition at line 767 of file rtp_engine.c.
References ast_rtp_engine::change_source, and ast_rtp_instance::engine.
Referenced by mgcp_indicate(), oh323_indicate(), sip_indicate(), and skinny_indicate().
00768 { 00769 if (instance->engine->change_source) { 00770 instance->engine->change_source(instance); 00771 } 00772 }
int ast_rtp_instance_destroy | ( | struct ast_rtp_instance * | instance | ) |
Destroy an RTP instance.
instance | The RTP instance to destroy |
0 | success | |
-1 | failure |
ast_rtp_instance_destroy(instance);
This destroys the RTP instance pointed to by instance. Once this function returns instance no longer points to valid memory and may not be used again.
Definition at line 301 of file rtp_engine.c.
References ao2_ref.
Referenced by __oh323_destroy(), __sip_destroy(), cleanup_connection(), destroy_endpoint(), gtalk_free_pvt(), jingle_free_pvt(), mgcp_hangup(), multicast_rtp_hangup(), multicast_rtp_request(), oh323_alloc(), skinny_hangup(), start_rtp(), unalloc_sub(), and unistim_hangup().
00302 { 00303 ao2_ref(instance, -1); 00304 00305 return 0; 00306 }
int ast_rtp_instance_dtmf_begin | ( | struct ast_rtp_instance * | instance, | |
char | digit | |||
) |
Begin sending a DTMF digit.
instance | The RTP instance to send the DTMF on | |
digit | What DTMF digit to send |
0 | success | |
-1 | failure |
ast_rtp_instance_dtmf_begin(instance, '1');
This starts sending the DTMF '1' on the RTP instance pointed to by instance. It will continue being sent until it is ended.
Definition at line 736 of file rtp_engine.c.
References ast_rtp_engine::dtmf_begin, and ast_rtp_instance::engine.
Referenced by gtalk_digit_begin(), mgcp_senddigit_begin(), oh323_digit_begin(), and sip_senddigit_begin().
00737 { 00738 return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1; 00739 }
int ast_rtp_instance_dtmf_end | ( | struct ast_rtp_instance * | instance, | |
char | digit | |||
) |
Stop sending a DTMF digit.
instance | The RTP instance to stop the DTMF on | |
digit | What DTMF digit to stop |
0 | success | |
-1 | failure |
ast_rtp_instance_dtmf_end(instance, '1');
This stops sending the DTMF '1' on the RTP instance pointed to by instance.
Definition at line 741 of file rtp_engine.c.
References ast_rtp_engine::dtmf_end, and ast_rtp_instance::engine.
Referenced by mgcp_senddigit_end(), and oh323_digit_end().
00742 { 00743 return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1; 00744 }
int ast_rtp_instance_dtmf_end_with_duration | ( | struct ast_rtp_instance * | instance, | |
char | digit, | |||
unsigned int | duration | |||
) |
Definition at line 745 of file rtp_engine.c.
References ast_rtp_engine::dtmf_end_with_duration, and ast_rtp_instance::engine.
Referenced by gtalk_digit_end(), and sip_senddigit_end().
00746 { 00747 return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1; 00748 }
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get | ( | struct ast_rtp_instance * | instance | ) |
Get the DTMF mode of an RTP instance.
instance | The RTP instance to get the DTMF mode of |
DTMF | mode |
enum ast_rtp_dtmf_mode dtmf_mode = ast_rtp_instance_dtmf_mode_get(instance);
This gets the DTMF mode set on the RTP instance pointed to by 'instance'.
Definition at line 755 of file rtp_engine.c.
References ast_rtp_engine::dtmf_mode_get, and ast_rtp_instance::engine.
Referenced by ast_rtp_instance_bridge().
00756 { 00757 return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0; 00758 }
int ast_rtp_instance_dtmf_mode_set | ( | struct ast_rtp_instance * | instance, | |
enum ast_rtp_dtmf_mode | dtmf_mode | |||
) |
Set the DTMF mode that should be used.
instance | the RTP instance to set DTMF mode on | |
dtmf_mode | The DTMF mode that is in use |
0 | success | |
-1 | failure |
This sets the RTP instance to use RFC2833 for DTMF transmission and receiving.
Definition at line 750 of file rtp_engine.c.
References ast_rtp_engine::dtmf_mode_set, and ast_rtp_instance::engine.
Referenced by enable_dsp_detect(), gtalk_alloc(), and sip_new().
00751 { 00752 return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0; 00753 }
int ast_rtp_instance_early_bridge | ( | struct ast_channel * | c0, | |
struct ast_channel * | c1 | |||
) |
Early bridge two channels that use RTP instances.
c0 | First channel part of the bridge | |
c1 | Second channel part of the bridge |
0 | success | |
-1 | failure |
Definition at line 1477 of file rtp_engine.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_debug, ast_log(), AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_REMOTE, ast_rtp_instance_get_glue(), ast_rtp_glue::get_codec, ast_rtp_glue::get_rtp_info, ast_rtp_glue::get_vrtp_info, LOG_WARNING, ast_channel::name, ast_channel::tech, ast_channel_tech::type, unref_instance_cond(), and ast_rtp_glue::update_peer.
01478 { 01479 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, 01480 *vinstance0 = NULL, *vinstance1 = NULL, 01481 *tinstance0 = NULL, *tinstance1 = NULL; 01482 struct ast_rtp_glue *glue0, *glue1; 01483 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01484 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01485 format_t codec0 = 0, codec1 = 0; 01486 int res = 0; 01487 01488 /* If there is no second channel just immediately bail out, we are of no use in that scenario */ 01489 if (!c1) { 01490 return -1; 01491 } 01492 01493 /* Lock both channels so we can look for the glue that binds them together */ 01494 ast_channel_lock(c0); 01495 while (ast_channel_trylock(c1)) { 01496 ast_channel_unlock(c0); 01497 usleep(1); 01498 ast_channel_lock(c0); 01499 } 01500 01501 /* Grab glue that binds each channel to something using the RTP engine */ 01502 if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) { 01503 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name); 01504 goto done; 01505 } 01506 01507 audio_glue0_res = glue0->get_rtp_info(c0, &instance0); 01508 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID; 01509 01510 audio_glue1_res = glue1->get_rtp_info(c1, &instance1); 01511 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID; 01512 01513 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */ 01514 if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01515 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01516 } 01517 if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01518 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01519 } 01520 if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) { 01521 codec0 = glue0->get_codec(c0); 01522 } 01523 if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) { 01524 codec1 = glue1->get_codec(c1); 01525 } 01526 01527 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */ 01528 if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) { 01529 goto done; 01530 } 01531 01532 /* Make sure we have matching codecs */ 01533 if (!(codec0 & codec1)) { 01534 goto done; 01535 } 01536 01537 /* Bridge media early */ 01538 if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0)) { 01539 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); 01540 } 01541 01542 res = 0; 01543 01544 done: 01545 ast_channel_unlock(c0); 01546 ast_channel_unlock(c1); 01547 01548 unref_instance_cond(&instance0); 01549 unref_instance_cond(&instance1); 01550 unref_instance_cond(&vinstance0); 01551 unref_instance_cond(&vinstance1); 01552 unref_instance_cond(&tinstance0); 01553 unref_instance_cond(&tinstance1); 01554 01555 if (!res) { 01556 ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); 01557 } 01558 01559 return res; 01560 }
void ast_rtp_instance_early_bridge_make_compatible | ( | struct ast_channel * | c0, | |
struct ast_channel * | c1 | |||
) |
Make two channels compatible for early bridging.
c0 | First channel part of the bridge | |
c1 | Second channel part of the bridge |
Definition at line 1391 of file rtp_engine.c.
References ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_debug, ast_log(), ast_rtp_codecs_payloads_copy(), AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_REMOTE, ast_rtp_instance_get_glue(), ast_rtp_instance::codecs, ast_rtp_glue::get_codec, ast_rtp_glue::get_rtp_info, ast_rtp_glue::get_vrtp_info, LOG_WARNING, ast_channel::name, ast_channel::tech, ast_channel_tech::type, unref_instance_cond(), and ast_rtp_glue::update_peer.
Referenced by dial_exec_full(), and do_forward().
01392 { 01393 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, 01394 *vinstance0 = NULL, *vinstance1 = NULL, 01395 *tinstance0 = NULL, *tinstance1 = NULL; 01396 struct ast_rtp_glue *glue0, *glue1; 01397 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01398 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01399 format_t codec0 = 0, codec1 = 0; 01400 int res = 0; 01401 01402 /* Lock both channels so we can look for the glue that binds them together */ 01403 ast_channel_lock(c0); 01404 while (ast_channel_trylock(c1)) { 01405 ast_channel_unlock(c0); 01406 usleep(1); 01407 ast_channel_lock(c0); 01408 } 01409 01410 /* Grab glue that binds each channel to something using the RTP engine */ 01411 if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) { 01412 ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name); 01413 goto done; 01414 } 01415 01416 audio_glue0_res = glue0->get_rtp_info(c0, &instance0); 01417 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID; 01418 01419 audio_glue1_res = glue1->get_rtp_info(c1, &instance1); 01420 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID; 01421 01422 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */ 01423 if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01424 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01425 } 01426 if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01427 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01428 } 01429 if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec) { 01430 codec0 = glue0->get_codec(c0); 01431 } 01432 if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec) { 01433 codec1 = glue1->get_codec(c1); 01434 } 01435 01436 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */ 01437 if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) { 01438 goto done; 01439 } 01440 01441 /* Make sure we have matching codecs */ 01442 if (!(codec0 & codec1)) { 01443 goto done; 01444 } 01445 01446 ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1); 01447 01448 if (vinstance0 && vinstance1) { 01449 ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1); 01450 } 01451 if (tinstance0 && tinstance1) { 01452 ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1); 01453 } 01454 01455 if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0)) { 01456 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); 01457 } 01458 01459 res = 0; 01460 01461 done: 01462 ast_channel_unlock(c0); 01463 ast_channel_unlock(c1); 01464 01465 unref_instance_cond(&instance0); 01466 unref_instance_cond(&instance1); 01467 unref_instance_cond(&vinstance0); 01468 unref_instance_cond(&vinstance1); 01469 unref_instance_cond(&tinstance0); 01470 unref_instance_cond(&tinstance1); 01471 01472 if (!res) { 01473 ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); 01474 } 01475 }
int ast_rtp_instance_fd | ( | struct ast_rtp_instance * | instance, | |
int | rtcp | |||
) |
Get the file descriptor for an RTP session (or RTCP).
instance | Instance to get the file descriptor for | |
rtcp | Whether to retrieve the file descriptor for RTCP or not |
fd | success | |
-1 | failure |
int rtp_fd = ast_rtp_instance_fd(instance, 0);
This retrieves the file descriptor for the socket carrying media on the instance pointed to by instance.
Definition at line 786 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::fd.
Referenced by __oh323_new(), __oh323_rtp_create(), __oh323_update_info(), gtalk_new(), jingle_new(), mgcp_new(), process_sdp(), sip_new(), sip_set_rtp_peer(), skinny_new(), start_rtp(), and unistim_new().
struct ast_rtp_glue* ast_rtp_instance_get_active_glue | ( | struct ast_rtp_instance * | instance | ) |
Get the RTP glue in use on an RTP instance.
instance | The RTP instance |
pointer | to the glue |
struct ast_rtp_glue *glue = ast_rtp_instance_get_active_glue(instance);
This gets the RTP glue currently in use on the RTP instance pointed to by 'instance'.
Definition at line 1757 of file rtp_engine.c.
References ast_rtp_instance::glue.
01758 { 01759 return instance->glue; 01760 }
int ast_rtp_instance_get_and_cmp_local_address | ( | struct ast_rtp_instance * | instance, | |
struct ast_sockaddr * | address | |||
) |
Get the address of the local endpoint that we are sending RTP to, comparing its address to another.
instance | The instance that we want to get the local address for | |
address | An initialized address that may be overwritten if the local address is different |
0 | address was not changed | |
1 | address was changed Example usage: |
struct ast_sockaddr address; int ret; ret = ast_rtp_instance_get_and_cmp_local_address(instance, &address);
This retrieves the current local address set on the instance pointed to by instance and puts the value into the address structure.
Definition at line 419 of file rtp_engine.c.
References ast_sockaddr_cmp(), ast_sockaddr_copy(), and ast_rtp_instance::local_address.
00421 { 00422 if (ast_sockaddr_cmp(address, &instance->local_address) != 0) { 00423 ast_sockaddr_copy(address, &instance->local_address); 00424 return 1; 00425 } 00426 00427 return 0; 00428 }
int ast_rtp_instance_get_and_cmp_remote_address | ( | struct ast_rtp_instance * | instance, | |
struct ast_sockaddr * | address | |||
) |
Get the address of the remote endpoint that we are sending RTP to, comparing its address to another.
instance | The instance that we want to get the remote address for | |
address | An initialized address that may be overwritten if the remote address is different |
0 | address was not changed | |
1 | address was changed Example usage: |
struct ast_sockaddr address; int ret; ret = ast_rtp_instance_get_and_cmp_remote_address(instance, &address);
This retrieves the current remote address set on the instance pointed to by instance and puts the value into the address structure.
Definition at line 436 of file rtp_engine.c.
References ast_sockaddr_cmp(), ast_sockaddr_copy(), and ast_rtp_instance::remote_address.
Referenced by sip_set_rtp_peer().
00438 { 00439 if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) { 00440 ast_sockaddr_copy(address, &instance->remote_address); 00441 return 1; 00442 } 00443 00444 return 0; 00445 }
struct ast_rtp_instance* ast_rtp_instance_get_bridged | ( | struct ast_rtp_instance * | instance | ) |
Get the other RTP instance that an instance is bridged to.
instance | The RTP instance that we want |
non-NULL | success | |
NULL | failure |
struct ast_rtp_instance *bridged = ast_rtp_instance_get_bridged(instance0);
This gets the RTP instance that instance0 is bridged to.
Definition at line 1386 of file rtp_engine.c.
References ast_rtp_instance::bridged.
Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and dialog_needdestroy().
01387 { 01388 return instance->bridged; 01389 }
struct ast_channel* ast_rtp_instance_get_chan | ( | struct ast_rtp_instance * | instance | ) |
Get the channel that is associated with an RTP instance while in a bridge.
instance | The RTP instance |
pointer | to the channel |
struct ast_channel *chan = ast_rtp_instance_get_chan(instance);
This gets the channel associated with the RTP instance pointed to by 'instance'.
Definition at line 1762 of file rtp_engine.c.
References ast_rtp_instance::chan.
01763 { 01764 return instance->chan; 01765 }
struct ast_rtp_codecs* ast_rtp_instance_get_codecs | ( | struct ast_rtp_instance * | instance | ) |
Get the codecs structure of an RTP instance.
instance | The RTP instance to get the codecs structure from |
struct ast_rtp_codecs *codecs = ast_rtp_instance_get_codecs(instance);
This gets the codecs structure on the RTP instance pointed to by 'instance'.
Definition at line 483 of file rtp_engine.c.
References ast_rtp_instance::codecs.
Referenced by __oh323_rtp_create(), add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_rtp_dtmf_begin(), ast_rtp_read(), ast_rtp_sendcng(), ast_rtp_write(), bridge_p2p_rtp_write(), check_peer_ok(), create_addr_from_peer(), gtalk_alloc(), gtalk_is_answered(), gtalk_new(), gtalk_newcall(), jingle_new(), jingle_newcall(), multicast_rtp_write(), process_sdp(), process_sdp_a_audio(), set_dtmf_payload(), set_peer_capabilities(), setup_rtp_connection(), start_rtp(), and transmit_response_with_sdp().
00484 { 00485 return &instance->codecs; 00486 }
void* ast_rtp_instance_get_data | ( | struct ast_rtp_instance * | instance | ) |
Get the data portion of an RTP instance.
instance | The RTP instance we want the data portion from |
struct *blob = ast_rtp_instance_get_data(instance); (
This gets the data pointer on the RTP instance pointed to by 'instance'.
Definition at line 369 of file rtp_engine.c.
References ast_rtp_instance::data.
Referenced by __rtp_recvfrom(), __rtp_sendto(), ast_rtcp_read(), ast_rtcp_write(), ast_rtcp_write_rr(), ast_rtcp_write_sr(), ast_rtp_alt_remote_address_set(), ast_rtp_change_source(), ast_rtp_destroy(), ast_rtp_dtmf_begin(), ast_rtp_dtmf_continuation(), ast_rtp_dtmf_end_with_duration(), ast_rtp_dtmf_mode_get(), ast_rtp_dtmf_mode_set(), ast_rtp_fd(), ast_rtp_get_stat(), ast_rtp_local_bridge(), ast_rtp_prop_set(), ast_rtp_qos_set(), ast_rtp_raw_write(), ast_rtp_read(), ast_rtp_remote_address_set(), ast_rtp_sendcng(), ast_rtp_stop(), ast_rtp_stun_request(), ast_rtp_update_source(), ast_rtp_write(), bridge_p2p_rtp_write(), create_dtmf_frame(), multicast_rtp_activate(), multicast_rtp_destroy(), multicast_rtp_write(), process_cn_rfc3389(), process_dtmf_cisco(), process_dtmf_rfc2833(), red_write(), rtp_red_buffer(), and rtp_red_init().
00370 { 00371 return instance->data; 00372 }
struct ast_rtp_engine* ast_rtp_instance_get_engine | ( | struct ast_rtp_instance * | instance | ) |
Get the RTP engine in use on an RTP instance.
instance | The RTP instance |
pointer | to the engine |
struct ast_rtp_engine *engine = ast_rtp_instance_get_engine(instance);
This gets the RTP engine currently in use on the RTP instance pointed to by 'instance'.
Definition at line 1752 of file rtp_engine.c.
References ast_rtp_instance::engine.
01753 { 01754 return instance->engine; 01755 }
void* ast_rtp_instance_get_extended_prop | ( | struct ast_rtp_instance * | instance, | |
int | property | |||
) |
Get the value of an RTP instance extended property.
instance | The RTP instance to get the extended property on | |
property | The extended property to get |
Definition at line 460 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::extended_prop_get.
00461 { 00462 if (instance->engine->extended_prop_get) { 00463 return instance->engine->extended_prop_get(instance, property); 00464 } 00465 00466 return NULL; 00467 }
struct ast_rtp_glue* ast_rtp_instance_get_glue | ( | const char * | type | ) |
Get the RTP glue that binds a channel to the RTP engine.
type | Name of the glue we want |
non-NULL | success | |
NULL | failure |
struct ast_rtp_glue *glue = ast_rtp_instance_get_glue("Example");
This retrieves the RTP glue that has the name 'Example'.
Definition at line 791 of file rtp_engine.c.
References AST_RWLIST_RDLOCK, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, ast_rtp_glue::entry, and ast_rtp_glue::type.
Referenced by ast_rtp_instance_bridge(), ast_rtp_instance_early_bridge(), ast_rtp_instance_early_bridge_make_compatible(), and ast_rtp_instance_make_compatible().
00792 { 00793 struct ast_rtp_glue *glue = NULL; 00794 00795 AST_RWLIST_RDLOCK(&glues); 00796 00797 AST_RWLIST_TRAVERSE(&glues, glue, entry) { 00798 if (!strcasecmp(glue->type, type)) { 00799 break; 00800 } 00801 } 00802 00803 AST_RWLIST_UNLOCK(&glues); 00804 00805 return glue; 00806 }
int ast_rtp_instance_get_hold_timeout | ( | struct ast_rtp_instance * | instance | ) |
Get the RTP timeout value for when an RTP instance is on hold.
instance | The RTP instance |
timeout | value |
int timeout = ast_rtp_instance_get_hold_timeout(instance);
This gets the RTP hold timeout value for the RTP instance pointed to by 'instance'.
Definition at line 1742 of file rtp_engine.c.
References ast_rtp_instance::holdtimeout.
Referenced by check_rtp_timeout().
01743 { 01744 return instance->holdtimeout; 01745 }
int ast_rtp_instance_get_keepalive | ( | struct ast_rtp_instance * | instance | ) |
Get the RTP keepalive interval.
instance | The RTP instance |
period | Keepalive interval value |
int interval = ast_rtp_instance_get_keepalive(instance);
This gets the RTP keepalive interval value for the RTP instance pointed to by 'instance'.
Definition at line 1747 of file rtp_engine.c.
References ast_rtp_instance::keepalive.
Referenced by check_rtp_timeout().
01748 { 01749 return instance->keepalive; 01750 }
void ast_rtp_instance_get_local_address | ( | struct ast_rtp_instance * | instance, | |
struct ast_sockaddr * | address | |||
) |
Get the local address that we are expecting RTP on.
instance | The RTP instance to get the address from | |
address | The variable to store the address in |
struct ast_sockaddr address; ast_rtp_instance_get_local_address(instance, &address);
This gets the local address that we are expecting RTP on and stores it in the 'address' structure.
Definition at line 430 of file rtp_engine.c.
References ast_sockaddr_copy(), and ast_rtp_instance::local_address.
Referenced by add_sdp(), apply_directmedia_ha(), ast_rtp_prop_set(), external_rtp_create(), get_our_media_address(), gtalk_create_candidates(), handle_open_receive_channel_ack_message(), jingle_create_candidates(), multicast_send_control_packet(), oh323_set_rtp_peer(), sip_acf_channel_read(), skinny_set_rtp_peer(), and start_rtp().
00432 { 00433 ast_sockaddr_copy(address, &instance->local_address); 00434 }
int ast_rtp_instance_get_prop | ( | struct ast_rtp_instance * | instance, | |
enum ast_rtp_property | property | |||
) |
Get the value of an RTP instance property.
instance | The RTP instance to get the property from | |
property | The property to get |
Current | value of the property |
ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT);
This returns the current value of the NAT property on the instance pointed to by instance.
Definition at line 478 of file rtp_engine.c.
References ast_rtp_instance::properties.
Referenced by ast_rtcp_read(), ast_rtp_dtmf_compatible(), ast_rtp_raw_write(), ast_rtp_read(), bridge_p2p_rtp_write(), process_dtmf_cisco(), and process_dtmf_rfc2833().
00479 { 00480 return instance->properties[property]; 00481 }
char* ast_rtp_instance_get_quality | ( | struct ast_rtp_instance * | instance, | |
enum ast_rtp_instance_stat_field | field, | |||
char * | buf, | |||
size_t | size | |||
) |
Retrieve quality statistics about an RTP instance.
instance | Instance to get statistics on | |
field | What quality statistic to retrieve | |
buf | What buffer to put the result into | |
size | Size of the above buffer |
non-NULL | success | |
NULL | failure |
char quality[AST_MAX_USER_FIELD]; ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, &buf, sizeof(buf));
This retrieves general quality statistics and places a text representation into the buf pointed to by buf.
Definition at line 1577 of file rtp_engine.c.
References ast_rtp_instance_get_stats(), AST_RTP_INSTANCE_STAT_ALL, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, AST_RTP_INSTANCE_STAT_COMBINED_RTT, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, ast_rtp_instance_stats::local_maxjitter, ast_rtp_instance_stats::local_maxrxploss, ast_rtp_instance_stats::local_minjitter, ast_rtp_instance_stats::local_minrxploss, ast_rtp_instance_stats::local_normdevjitter, ast_rtp_instance_stats::local_normdevrxploss, ast_rtp_instance_stats::local_ssrc, ast_rtp_instance_stats::local_stdevjitter, ast_rtp_instance_stats::local_stdevrxploss, ast_rtp_instance_stats::maxrtt, ast_rtp_instance_stats::minrtt, ast_rtp_instance_stats::normdevrtt, ast_rtp_instance_stats::remote_maxjitter, ast_rtp_instance_stats::remote_maxrxploss, ast_rtp_instance_stats::remote_minjitter, ast_rtp_instance_stats::remote_minrxploss, ast_rtp_instance_stats::remote_normdevjitter, ast_rtp_instance_stats::remote_normdevrxploss, ast_rtp_instance_stats::remote_ssrc, ast_rtp_instance_stats::remote_stdevjitter, ast_rtp_instance_stats::remote_stdevrxploss, ast_rtp_instance_stats::rtt, ast_rtp_instance_stats::rxcount, ast_rtp_instance_stats::rxjitter, ast_rtp_instance_stats::rxploss, ast_rtp_instance_stats::stdevrtt, ast_rtp_instance_stats::txcount, ast_rtp_instance_stats::txjitter, and ast_rtp_instance_stats::txploss.
Referenced by ast_rtp_instance_set_stats_vars(), handle_request_bye(), sip_acf_channel_read(), and sip_hangup().
01578 { 01579 struct ast_rtp_instance_stats stats = { 0, }; 01580 enum ast_rtp_instance_stat stat; 01581 01582 /* Determine what statistics we will need to retrieve based on field passed in */ 01583 if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) { 01584 stat = AST_RTP_INSTANCE_STAT_ALL; 01585 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) { 01586 stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER; 01587 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) { 01588 stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS; 01589 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) { 01590 stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT; 01591 } else { 01592 return NULL; 01593 } 01594 01595 /* Attempt to actually retrieve the statistics we need to generate the quality string */ 01596 if (ast_rtp_instance_get_stats(instance, &stats, stat)) { 01597 return NULL; 01598 } 01599 01600 /* Now actually fill the buffer with the good information */ 01601 if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) { 01602 snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", 01603 stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt); 01604 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) { 01605 snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;", 01606 stats.local_minjitter, stats.local_maxjitter, stats.local_normdevjitter, sqrt(stats.local_stdevjitter), stats.remote_minjitter, stats.remote_maxjitter, stats.remote_normdevjitter, sqrt(stats.remote_stdevjitter)); 01607 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) { 01608 snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;", 01609 stats.local_minrxploss, stats.local_maxrxploss, stats.local_normdevrxploss, sqrt(stats.local_stdevrxploss), stats.remote_minrxploss, stats.remote_maxrxploss, stats.remote_normdevrxploss, sqrt(stats.remote_stdevrxploss)); 01610 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) { 01611 snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt); 01612 } 01613 01614 return buf; 01615 }
void ast_rtp_instance_get_remote_address | ( | struct ast_rtp_instance * | instance, | |
struct ast_sockaddr * | address | |||
) |
Get the address of the remote endpoint that we are sending RTP to.
instance | The instance that we want to get the remote address for | |
address | A structure to put the address into |
struct ast_sockaddr address; ast_rtp_instance_get_remote_address(instance, &address);
This retrieves the current remote address set on the instance pointed to by instance and puts the value into the address structure.
Definition at line 447 of file rtp_engine.c.
References ast_sockaddr_copy(), and ast_rtp_instance::remote_address.
Referenced by add_sdp(), apply_directmedia_ha(), ast_rtp_dtmf_begin(), ast_rtp_dtmf_continuation(), ast_rtp_dtmf_end_with_duration(), ast_rtp_instance_bridge(), ast_rtp_raw_write(), ast_rtp_read(), ast_rtp_sendcng(), ast_rtp_write(), bridge_p2p_rtp_write(), create_dtmf_frame(), gtalk_update_stun(), multicast_rtp_write(), multicast_send_control_packet(), oh323_set_rtp_peer(), process_cn_rfc3389(), process_dtmf_rfc2833(), process_sdp(), remote_bridge_loop(), sip_acf_channel_read(), skinny_set_rtp_peer(), and transmit_modify_with_sdp().
00449 { 00450 ast_sockaddr_copy(address, &instance->remote_address); 00451 }
struct ast_srtp* ast_rtp_instance_get_srtp | ( | struct ast_rtp_instance * | instance | ) |
Obtain the SRTP instance associated with an RTP instance.
instance | the RTP instance |
the | SRTP instance on success | |
NULL | if no SRTP instance exists |
Definition at line 1813 of file rtp_engine.c.
References ast_rtp_instance::srtp.
Referenced by __rtp_recvfrom(), __rtp_sendto(), and ast_rtp_change_source().
01814 { 01815 return instance->srtp; 01816 }
int ast_rtp_instance_get_stats | ( | struct ast_rtp_instance * | instance, | |
struct ast_rtp_instance_stats * | stats, | |||
enum ast_rtp_instance_stat | stat | |||
) |
Retrieve statistics about an RTP instance.
instance | Instance to get statistics on | |
stats | Structure to put results into | |
stat | What statistic(s) to retrieve |
0 | success | |
-1 | failure |
struct ast_rtp_instance_stats stats; ast_rtp_instance_get_stats(instance, &stats, AST_RTP_INSTANCE_STAT_ALL);
This retrieves all statistics the underlying RTP engine supports and puts the values into the stats structure.
Definition at line 1572 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::get_stat.
Referenced by ast_rtp_instance_get_quality(), ast_srtp_unprotect(), sdp_crypto_activate(), show_chanstats_cb(), and sip_acf_channel_read().
01573 { 01574 return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1; 01575 }
int ast_rtp_instance_get_timeout | ( | struct ast_rtp_instance * | instance | ) |
Get the RTP timeout value.
instance | The RTP instance |
timeout | value |
int timeout = ast_rtp_instance_get_timeout(instance);
This gets the RTP timeout value for the RTP instance pointed to by 'instance'.
Definition at line 1737 of file rtp_engine.c.
References ast_rtp_instance::timeout.
Referenced by check_rtp_timeout().
01738 { 01739 return instance->timeout; 01740 }
int ast_rtp_instance_make_compatible | ( | struct ast_channel * | chan, | |
struct ast_rtp_instance * | instance, | |||
struct ast_channel * | peer | |||
) |
Request that the underlying RTP engine make two RTP instances compatible with eachother.
chan | Our own Asterisk channel | |
instance | The first RTP instance | |
peer | The peer Asterisk channel |
0 | success | |
-1 | failure |
ast_rtp_instance_make_compatible(instance, peer);
This makes the RTP instance for 'peer' compatible with 'instance' and vice versa.
Definition at line 1661 of file rtp_engine.c.
References ao2_ref, ast_channel_lock, ast_channel_unlock, ast_rtp_instance_get_glue(), ast_rtp_instance::chan, ast_rtp_instance::engine, ast_rtp_glue::get_rtp_info, ast_rtp_instance::glue, ast_rtp_engine::make_compatible, ast_channel::tech, and ast_channel_tech::type.
Referenced by sip_setoption().
01662 { 01663 struct ast_rtp_glue *glue; 01664 struct ast_rtp_instance *peer_instance = NULL; 01665 int res = -1; 01666 01667 if (!instance->engine->make_compatible) { 01668 return -1; 01669 } 01670 01671 ast_channel_lock(peer); 01672 01673 if (!(glue = ast_rtp_instance_get_glue(peer->tech->type))) { 01674 ast_channel_unlock(peer); 01675 return -1; 01676 } 01677 01678 glue->get_rtp_info(peer, &peer_instance); 01679 01680 if (!peer_instance || peer_instance->engine != instance->engine) { 01681 ast_channel_unlock(peer); 01682 ao2_ref(peer_instance, -1); 01683 peer_instance = NULL; 01684 return -1; 01685 } 01686 01687 res = instance->engine->make_compatible(chan, instance, peer, peer_instance); 01688 01689 ast_channel_unlock(peer); 01690 01691 ao2_ref(peer_instance, -1); 01692 peer_instance = NULL; 01693 01694 return res; 01695 }
struct ast_rtp_instance* ast_rtp_instance_new | ( | const char * | engine_name, | |
struct sched_context * | sched, | |||
const struct ast_sockaddr * | sa, | |||
void * | data | |||
) |
Create a new RTP instance.
engine_name | Name of the engine to use for the RTP instance | |
sched | Scheduler context that the RTP engine may want to use | |
sa | Address we want to bind to | |
data | Unique data for the engine |
non-NULL | success | |
NULL | failure |
struct ast_rtp_instance *instance = NULL; instance = ast_rtp_instance_new(NULL, sched, &sin, NULL);
This creates a new RTP instance using the default engine and asks the RTP engine to bind to the address given in the address structure.
Definition at line 308 of file rtp_engine.c.
References ao2_alloc, ao2_ref, ast_debug, ast_log(), ast_module_ref(), ast_module_unref(), AST_RWLIST_FIRST, AST_RWLIST_RDLOCK, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, ast_sockaddr_copy(), ast_strlen_zero(), ast_rtp_engine::entry, instance_destructor(), LOG_ERROR, ast_rtp_engine::mod, ast_rtp_engine::name, and sched.
Referenced by __oh323_rtp_create(), dialog_initialize_rtp(), gtalk_alloc(), jingle_alloc(), multicast_rtp_request(), and start_rtp().
00311 { 00312 struct ast_sockaddr address = {{0,}}; 00313 struct ast_rtp_instance *instance = NULL; 00314 struct ast_rtp_engine *engine = NULL; 00315 00316 AST_RWLIST_RDLOCK(&engines); 00317 00318 /* If an engine name was specified try to use it or otherwise use the first one registered */ 00319 if (!ast_strlen_zero(engine_name)) { 00320 AST_RWLIST_TRAVERSE(&engines, engine, entry) { 00321 if (!strcmp(engine->name, engine_name)) { 00322 break; 00323 } 00324 } 00325 } else { 00326 engine = AST_RWLIST_FIRST(&engines); 00327 } 00328 00329 /* If no engine was actually found bail out now */ 00330 if (!engine) { 00331 ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n"); 00332 AST_RWLIST_UNLOCK(&engines); 00333 return NULL; 00334 } 00335 00336 /* Bump up the reference count before we return so the module can not be unloaded */ 00337 ast_module_ref(engine->mod); 00338 00339 AST_RWLIST_UNLOCK(&engines); 00340 00341 /* Allocate a new RTP instance */ 00342 if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) { 00343 ast_module_unref(engine->mod); 00344 return NULL; 00345 } 00346 instance->engine = engine; 00347 ast_sockaddr_copy(&instance->local_address, sa); 00348 ast_sockaddr_copy(&address, sa); 00349 00350 ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance); 00351 00352 /* And pass it off to the engine to setup */ 00353 if (instance->engine->new(instance, sched, &address, data)) { 00354 ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance); 00355 ao2_ref(instance, -1); 00356 return NULL; 00357 } 00358 00359 ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance); 00360 00361 return instance; 00362 }
struct ast_frame* ast_rtp_instance_read | ( | struct ast_rtp_instance * | instance, | |
int | rtcp | |||
) |
Receive a frame over RTP.
instance | The RTP instance to receive frame on | |
rtcp | Whether to read in RTCP or not |
non-NULL | success | |
NULL | failure |
struct ast_frame *frame; frame = ast_rtp_instance_read(instance, 0);
This asks the RTP engine to read in RTP from the instance and return it as an Asterisk frame.
Definition at line 379 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::read.
Referenced by gtalk_rtp_read(), jingle_rtp_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), sip_rtp_read(), skinny_rtp_read(), and unistim_rtp_read().
int ast_rtp_instance_sendcng | ( | struct ast_rtp_instance * | instance, | |
int | level | |||
) |
Send a comfort noise packet to the RTP instance.
instance | The RTP instance | |
level | Magnitude of the noise level |
0 | Success | |
non-zero | Failure |
Definition at line 1818 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::sendcng.
Referenced by check_rtp_timeout().
01819 { 01820 if (instance->engine->sendcng) { 01821 return instance->engine->sendcng(instance, level); 01822 } 01823 01824 return -1; 01825 }
int ast_rtp_instance_set_alt_remote_address | ( | struct ast_rtp_instance * | instance, | |
const struct ast_sockaddr * | address | |||
) |
Set the address of an an alternate RTP address to receive from.
instance | The RTP instance to change the address on | |
address | Address to set it to |
0 | success | |
-1 | failure |
ast_rtp_instance_set_alt_remote_address(instance, &address);
This changes the alternate remote address that RTP will be sent to on instance to the address given in the sin structure.
Definition at line 405 of file rtp_engine.c.
References ast_rtp_instance::alt_remote_address, ast_rtp_engine::alt_remote_address_set, ast_sockaddr_copy(), and ast_rtp_instance::engine.
Referenced by handle_request_invite().
00407 { 00408 ast_sockaddr_copy(&instance->alt_remote_address, address); 00409 00410 /* oink */ 00411 00412 if (instance->engine->alt_remote_address_set) { 00413 instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address); 00414 } 00415 00416 return 0; 00417 }
void ast_rtp_instance_set_data | ( | struct ast_rtp_instance * | instance, | |
void * | data | |||
) |
Set the data portion of an RTP instance.
instance | The RTP instance to manipulate | |
data | Pointer to data |
ast_rtp_instance_set_data(instance, blob);
This sets the data pointer on the RTP instance pointed to by 'instance' to blob.
Definition at line 364 of file rtp_engine.c.
References ast_rtp_instance::data.
Referenced by multicast_rtp_new().
void ast_rtp_instance_set_extended_prop | ( | struct ast_rtp_instance * | instance, | |
int | property, | |||
void * | value | |||
) |
Set the value of an RTP instance extended property.
instance | The RTP instance to set the extended property on | |
property | The extended property to set | |
value | The value to set the extended property to |
Definition at line 453 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::extended_prop_set.
00454 { 00455 if (instance->engine->extended_prop_set) { 00456 instance->engine->extended_prop_set(instance, property, value); 00457 } 00458 }
void ast_rtp_instance_set_hold_timeout | ( | struct ast_rtp_instance * | instance, | |
int | timeout | |||
) |
Set the RTP timeout value for when the instance is on hold.
instance | The RTP instance | |
timeout | Value to set the timeout to |
ast_rtp_instance_set_hold_timeout(instance, 5000);
This sets the RTP hold timeout value on 'instance' to be 5000.
Definition at line 1727 of file rtp_engine.c.
References ast_rtp_instance::holdtimeout.
Referenced by check_rtp_timeout(), and dialog_initialize_rtp().
01728 { 01729 instance->holdtimeout = timeout; 01730 }
void ast_rtp_instance_set_keepalive | ( | struct ast_rtp_instance * | instance, | |
int | timeout | |||
) |
Set the RTP keepalive interval.
instance | The RTP instance | |
period | Value to set the keepalive interval to |
ast_rtp_instance_set_keepalive(instance, 5000);
This sets the RTP keepalive interval on 'instance' to be 5000.
Definition at line 1732 of file rtp_engine.c.
References ast_rtp_instance::keepalive.
Referenced by dialog_initialize_rtp().
01733 { 01734 instance->keepalive = interval; 01735 }
int ast_rtp_instance_set_local_address | ( | struct ast_rtp_instance * | instance, | |
const struct ast_sockaddr * | address | |||
) |
Set the address that we are expecting to receive RTP on.
instance | The RTP instance to change the address on | |
address | Address to set it to |
0 | success | |
-1 | failure |
ast_rtp_instance_set_local_address(instance, &sin);
This changes the local address that RTP is expected on to the address given in the sin structure.
Definition at line 384 of file rtp_engine.c.
References ast_sockaddr_copy(), and ast_rtp_instance::local_address.
Referenced by ast_rtp_new().
00386 { 00387 ast_sockaddr_copy(&instance->local_address, address); 00388 return 0; 00389 }
void ast_rtp_instance_set_prop | ( | struct ast_rtp_instance * | instance, | |
enum ast_rtp_property | property, | |||
int | value | |||
) |
Set the value of an RTP instance property.
instance | The RTP instance to set the property on | |
property | The property to modify | |
value | The value to set the property to |
ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_NAT, 1);
This enables the AST_RTP_PROPERTY_NAT property on the instance pointed to by instance.
Definition at line 469 of file rtp_engine.c.
References ast_rtp_instance::engine, ast_rtp_engine::prop_set, and ast_rtp_instance::properties.
Referenced by __oh323_rtp_create(), create_addr_from_peer(), dialog_initialize_rtp(), do_setnat(), gtalk_alloc(), handle_request_invite(), oh323_rtp_read(), process_sdp(), sip_dtmfmode(), sip_set_rtp_peer(), and start_rtp().
00470 { 00471 instance->properties[property] = value; 00472 00473 if (instance->engine->prop_set) { 00474 instance->engine->prop_set(instance, property, value); 00475 } 00476 }
int ast_rtp_instance_set_qos | ( | struct ast_rtp_instance * | instance, | |
int | tos, | |||
int | cos, | |||
const char * | desc | |||
) |
Set QoS parameters on an RTP session.
instance | Instance to set the QoS parameters on | |
tos | Terms of service value | |
cos | Class of service value | |
desc | What is setting the QoS values |
0 | success | |
-1 | failure |
ast_rtp_instance_set_qos(instance, 0, 0, "Example");
This sets the TOS and COS values to 0 on the instance pointed to by instance.
Definition at line 774 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::qos.
Referenced by __oh323_rtp_create(), dialog_initialize_rtp(), and start_rtp().
00775 { 00776 return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1; 00777 }
int ast_rtp_instance_set_read_format | ( | struct ast_rtp_instance * | instance, | |
format_t | format | |||
) |
Request that the underlying RTP engine provide audio frames in a specific format.
instance | The RTP instance to change read format on | |
format | Format that frames are wanted in |
0 | success | |
-1 | failure |
ast_rtp_instance_set_read_format(instance, AST_FORMAT_ULAW);
This requests that the RTP engine provide audio frames in the ULAW format.
Definition at line 1651 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::set_read_format.
Referenced by sip_new(), and sip_setoption().
01652 { 01653 return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1; 01654 }
int ast_rtp_instance_set_remote_address | ( | struct ast_rtp_instance * | instance, | |
const struct ast_sockaddr * | address | |||
) |
Set the address of the remote endpoint that we are sending RTP to.
instance | The RTP instance to change the address on | |
address | Address to set it to |
0 | success | |
-1 | failure |
ast_rtp_instance_set_remote_address(instance, &sin);
This changes the remote address that RTP will be sent to on instance to the address given in the sin structure.
Definition at line 391 of file rtp_engine.c.
References ast_sockaddr_copy(), ast_rtp_instance::engine, ast_rtp_instance::remote_address, and ast_rtp_engine::remote_address_set.
Referenced by ast_rtp_read(), ast_rtp_stop(), handle_open_receive_channel_ack_message(), multicast_rtp_request(), process_sdp(), setup_rtp_connection(), and start_rtp().
00393 { 00394 ast_sockaddr_copy(&instance->remote_address, address); 00395 00396 /* moo */ 00397 00398 if (instance->engine->remote_address_set) { 00399 instance->engine->remote_address_set(instance, &instance->remote_address); 00400 } 00401 00402 return 0; 00403 }
void ast_rtp_instance_set_stats_vars | ( | struct ast_channel * | chan, | |
struct ast_rtp_instance * | instance | |||
) |
Set standard statistics from an RTP instance on a channel.
chan | Channel to set the statistics on | |
instance | The RTP instance that statistics will be retrieved from |
ast_rtp_instance_set_stats_vars(chan, rtp);
This retrieves standard statistics from the RTP instance rtp and sets it on the channel pointed to by chan.
Definition at line 1617 of file rtp_engine.c.
References ast_bridged_channel(), AST_MAX_USER_FIELD, ast_rtp_instance_get_quality(), AST_RTP_INSTANCE_STAT_FIELD_QUALITY, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, ast_channel::bridge, pbx_builtin_setvar_helper(), and quality.
Referenced by handle_request_bye(), and sip_hangup().
01618 { 01619 char quality_buf[AST_MAX_USER_FIELD], *quality; 01620 struct ast_channel *bridge = ast_bridged_channel(chan); 01621 01622 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) { 01623 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality); 01624 if (bridge) { 01625 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality); 01626 } 01627 } 01628 01629 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) { 01630 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality); 01631 if (bridge) { 01632 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality); 01633 } 01634 } 01635 01636 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) { 01637 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality); 01638 if (bridge) { 01639 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality); 01640 } 01641 } 01642 01643 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) { 01644 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality); 01645 if (bridge) { 01646 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality); 01647 } 01648 } 01649 }
void ast_rtp_instance_set_timeout | ( | struct ast_rtp_instance * | instance, | |
int | timeout | |||
) |
Set the RTP timeout value.
instance | The RTP instance | |
timeout | Value to set the timeout to |
ast_rtp_instance_set_timeout(instance, 5000);
This sets the RTP timeout value on 'instance' to be 5000.
Definition at line 1722 of file rtp_engine.c.
References ast_rtp_instance::timeout.
Referenced by check_rtp_timeout(), and dialog_initialize_rtp().
01723 { 01724 instance->timeout = timeout; 01725 }
int ast_rtp_instance_set_write_format | ( | struct ast_rtp_instance * | instance, | |
format_t | format | |||
) |
Tell underlying RTP engine that audio frames will be provided in a specific format.
instance | The RTP instance to change write format on | |
format | Format that frames will be provided in |
0 | success | |
-1 | failure |
ast_rtp_instance_set_write_format(instance, AST_FORMAT_ULAW);
This tells the underlying RTP engine that audio frames will be provided to it in ULAW format.
Definition at line 1656 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::set_write_format.
Referenced by sip_new(), and sip_setoption().
01657 { 01658 return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1; 01659 }
void ast_rtp_instance_stop | ( | struct ast_rtp_instance * | instance | ) |
Stop an RTP instance.
instance | Instance that media is no longer going to at this time |
ast_rtp_instance_stop(instance);
This tells the RTP engine being used for the instance pointed to by instance that media is no longer going to it at this time, but may in the future.
Definition at line 779 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::stop.
Referenced by process_sdp(), setup_rtp_connection(), and stop_media_flows().
void ast_rtp_instance_stun_request | ( | struct ast_rtp_instance * | instance, | |
struct ast_sockaddr * | suggestion, | |||
const char * | username | |||
) |
Request that the underlying RTP engine send a STUN BIND request.
instance | The RTP instance | |
suggestion | The suggested destination | |
username | Optionally a username for the request |
ast_rtp_instance_stun_request(instance, NULL, NULL);
This requests that the RTP engine send a STUN BIND request on the session pointed to by 'instance'.
Definition at line 1713 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::stun_request.
Referenced by gtalk_update_stun(), and jingle_update_stun().
01716 { 01717 if (instance->engine->stun_request) { 01718 instance->engine->stun_request(instance, suggestion, username); 01719 } 01720 }
void ast_rtp_instance_update_source | ( | struct ast_rtp_instance * | instance | ) |
Indicate that the RTP marker bit should be set on an RTP stream.
instance | Instance that the new media source is feeding into |
ast_rtp_instance_update_source(instance);
This indicates that the source of media that is feeding the instance pointed to by instance has been updated and that the marker bit should be set.
Definition at line 760 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::update_source.
Referenced by mgcp_indicate(), oh323_indicate(), sip_answer(), sip_indicate(), sip_write(), and skinny_indicate().
00761 { 00762 if (instance->engine->update_source) { 00763 instance->engine->update_source(instance); 00764 } 00765 }
int ast_rtp_instance_write | ( | struct ast_rtp_instance * | instance, | |
struct ast_frame * | frame | |||
) |
Send a frame out over RTP.
instance | The RTP instance to send frame out on | |
frame | the frame to send out |
0 | success | |
-1 | failure |
ast_rtp_instance_write(instance, frame);
This gives the frame pointed to by frame to the RTP engine being used for the instance and asks that it be transmitted to the current remote address set on the RTP instance.
Definition at line 374 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::write.
Referenced by gtalk_write(), jingle_write(), mgcp_write(), multicast_rtp_write(), oh323_write(), sip_write(), skinny_write(), and unistim_write().
char* ast_rtp_lookup_mime_multiple2 | ( | struct ast_str * | buf, | |
const format_t | capability, | |||
const int | asterisk_format, | |||
enum ast_rtp_options | options | |||
) |
Convert formats into a string and put them into a buffer.
buf | Buffer to put the mime output into | |
capability | Formats that we are looking up | |
asterisk_format | Non-zero if the given capability are Asterisk format capabilities | |
options | Additional options that may change the result |
non-NULL | success | |
NULL | failure |
char buf[256] = ""; char *mime = ast_rtp_lookup_mime_multiple2(&buf, sizeof(buf), AST_FORMAT_ULAW | AST_FORMAT_ALAW, 1, 0);
This returns the mime values for ULAW and ALAW in the buffer pointed to by buf.
Definition at line 703 of file rtp_engine.c.
References ast_rtp_lookup_mime_subtype2(), AST_RTP_MAX, ast_str_append(), ast_str_buffer(), format, and name.
Referenced by process_sdp().
00704 { 00705 format_t format; 00706 int found = 0; 00707 00708 if (!buf) { 00709 return NULL; 00710 } 00711 00712 ast_str_append(&buf, 0, "0x%llx (", (unsigned long long) capability); 00713 00714 for (format = 1; format < AST_RTP_MAX; format <<= 1) { 00715 if (capability & format) { 00716 const char *name = ast_rtp_lookup_mime_subtype2(asterisk_format, format, options); 00717 ast_str_append(&buf, 0, "%s|", name); 00718 found = 1; 00719 } 00720 } 00721 00722 ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)"); 00723 00724 return ast_str_buffer(buf); 00725 }
const char* ast_rtp_lookup_mime_subtype2 | ( | const int | asterisk_format, | |
const format_t | code, | |||
enum ast_rtp_options | options | |||
) |
Retrieve mime subtype information on a payload.
asterisk_format | Non-zero if the given code is an Asterisk format value | |
code | Format to look up | |
options | Additional options that may change the result |
Mime | subtype success | |
NULL | failure |
const char *subtype = ast_rtp_lookup_mime_subtype2(1, AST_FORMAT_ULAW, 0);
This looks up the mime subtype for the ULAW format.
Definition at line 673 of file rtp_engine.c.
References ARRAY_LEN, AST_FORMAT_G726_AAL2, ast_rtp_mime_types, AST_RTP_OPT_G726_NONSTANDARD, ast_rtp_payload_type::asterisk_format, ast_rtp_mime_type::payload_type, and ast_rtp_mime_type::subtype.
Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_rtp_lookup_mime_multiple2(), transmit_connect(), transmit_connect_with_sdp(), transmit_modify_request(), and transmit_modify_with_sdp().
00674 { 00675 int i; 00676 00677 for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); i++) { 00678 if (ast_rtp_mime_types[i].payload_type.code == code && ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format) { 00679 if (asterisk_format && (code == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) { 00680 return "G726-32"; 00681 } else { 00682 return ast_rtp_mime_types[i].subtype; 00683 } 00684 } 00685 } 00686 00687 return ""; 00688 }
unsigned int ast_rtp_lookup_sample_rate2 | ( | int | asterisk_format, | |
format_t | code | |||
) |
Get the sample rate associated with known RTP payload types.
asterisk_format | True if the value in the 'code' parameter is an AST_FORMAT value | |
code | Format code, either from AST_FORMAT list or from AST_RTP list |
Definition at line 690 of file rtp_engine.c.
References ARRAY_LEN, ast_rtp_mime_types, ast_rtp_payload_type::asterisk_format, ast_rtp_mime_type::payload_type, and ast_rtp_mime_type::sample_rate.
Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_tcodec_to_sdp(), and add_vcodec_to_sdp().
00691 { 00692 unsigned int i; 00693 00694 for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) { 00695 if ((ast_rtp_mime_types[i].payload_type.code == code) && (ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format)) { 00696 return ast_rtp_mime_types[i].sample_rate; 00697 } 00698 } 00699 00700 return 0; 00701 }
int ast_rtp_red_buffer | ( | struct ast_rtp_instance * | instance, | |
struct ast_frame * | frame | |||
) |
Buffer a frame in an RTP instance for RED.
instance | The instance to buffer the frame on | |
frame | Frame that we want to buffer |
0 | success | |
-1 | failure |
Definition at line 1567 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::red_buffer.
Referenced by sip_write().
01568 { 01569 return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1; 01570 }
int ast_rtp_red_init | ( | struct ast_rtp_instance * | instance, | |
int | buffer_time, | |||
int * | payloads, | |||
int | generations | |||
) |
Initialize RED support on an RTP instance.
instance | The instance to initialize RED support on | |
buffer_time | How long to buffer before sending | |
payloads | Payload values | |
generations | Number of generations |
0 | success | |
-1 | failure |
Definition at line 1562 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::red_init.
Referenced by process_sdp().
01563 { 01564 return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1; 01565 }
static void instance_destructor | ( | void * | obj | ) | [static] |
Definition at line 281 of file rtp_engine.c.
References ast_debug, ast_module_unref(), ast_rtp_instance::data, ast_srtp_res::destroy, ast_rtp_engine::destroy, ast_rtp_instance::engine, ast_rtp_engine::mod, ast_rtp_engine::name, res_srtp, and ast_rtp_instance::srtp.
Referenced by ast_rtp_instance_new().
00282 { 00283 struct ast_rtp_instance *instance = obj; 00284 00285 /* Pass us off to the engine to destroy */ 00286 if (instance->data && instance->engine->destroy(instance)) { 00287 ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance); 00288 return; 00289 } 00290 00291 if (instance->srtp) { 00292 res_srtp->destroy(instance->srtp); 00293 } 00294 00295 /* Drop our engine reference */ 00296 ast_module_unref(instance->engine->mod); 00297 00298 ast_debug(1, "Destroyed RTP instance '%p'\n", instance); 00299 }
static enum ast_bridge_result local_bridge_loop | ( | struct ast_channel * | c0, | |
struct ast_channel * | c1, | |||
struct ast_rtp_instance * | instance0, | |||
struct ast_rtp_instance * | instance1, | |||
int | timeoutms, | |||
int | flags, | |||
struct ast_frame ** | fo, | |||
struct ast_channel ** | rc, | |||
void * | pvt0, | |||
void * | pvt1 | |||
) | [static] |
Definition at line 808 of file rtp_engine.c.
References AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, AST_BRIDGE_RETRY, ast_channel_unlock, ast_debug, ast_framehook_list_is_empty(), ast_frfree, ast_poll_channel_add(), ast_read(), ast_channel::audiohooks, ast_rtp_instance::bridged, ast_rtp_instance::engine, ast_channel::framehooks, ast_rtp_engine::local_bridge, ast_channel::masq, ast_channel::masqr, ast_channel::monitor, ast_channel::name, ast_channel::rawreadformat, ast_channel::rawwriteformat, and ast_channel::tech_pvt.
Referenced by ast_rtp_instance_bridge().
00809 { 00810 enum ast_bridge_result res = AST_BRIDGE_FAILED; 00811 struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, }; 00812 struct ast_frame *fr = NULL; 00813 00814 /* Start locally bridging both instances */ 00815 if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) { 00816 ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c0->name, c1->name); 00817 ast_channel_unlock(c0); 00818 ast_channel_unlock(c1); 00819 return AST_BRIDGE_FAILED_NOWARN; 00820 } 00821 if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) { 00822 ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c1->name, c0->name); 00823 if (instance0->engine->local_bridge) { 00824 instance0->engine->local_bridge(instance0, NULL); 00825 } 00826 ast_channel_unlock(c0); 00827 ast_channel_unlock(c1); 00828 return AST_BRIDGE_FAILED_NOWARN; 00829 } 00830 00831 ast_channel_unlock(c0); 00832 ast_channel_unlock(c1); 00833 00834 instance0->bridged = instance1; 00835 instance1->bridged = instance0; 00836 00837 ast_poll_channel_add(c0, c1); 00838 00839 /* Hop into a loop waiting for a frame from either channel */ 00840 cs[0] = c0; 00841 cs[1] = c1; 00842 cs[2] = NULL; 00843 for (;;) { 00844 /* If the underlying formats have changed force this bridge to break */ 00845 if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) { 00846 ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n"); 00847 res = AST_BRIDGE_FAILED_NOWARN; 00848 break; 00849 } 00850 /* Check if anything changed */ 00851 if ((c0->tech_pvt != pvt0) || 00852 (c1->tech_pvt != pvt1) || 00853 (c0->masq || c0->masqr || c1->masq || c1->masqr) || 00854 (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks) || 00855 (!ast_framehook_list_is_empty(c0->framehooks) || !ast_framehook_list_is_empty(c1->framehooks))) { 00856 ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n"); 00857 /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */ 00858 if ((c0->masq || c0->masqr) && (fr = ast_read(c0))) { 00859 ast_frfree(fr); 00860 } 00861 if ((c1->masq || c1->masqr) && (fr = ast_read(c1))) { 00862 ast_frfree(fr); 00863 } 00864 res = AST_BRIDGE_RETRY; 00865 break; 00866 } 00867 /* Wait on a channel to feed us a frame */ 00868 if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) { 00869 if (!timeoutms) { 00870 res = AST_BRIDGE_RETRY; 00871 break; 00872 } 00873 ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n"); 00874 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 00875 break; 00876 } 00877 continue; 00878 } 00879 /* Read in frame from channel */ 00880 fr = ast_read(who); 00881 other = (who == c0) ? c1 : c0; 00882 /* Depending on the frame we may need to break out of our bridge */ 00883 if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) && 00884 ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) | 00885 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) { 00886 /* Record received frame and who */ 00887 *fo = fr; 00888 *rc = who; 00889 ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup"); 00890 res = AST_BRIDGE_COMPLETE; 00891 break; 00892 } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) { 00893 if ((fr->subclass.integer == AST_CONTROL_HOLD) || 00894 (fr->subclass.integer == AST_CONTROL_UNHOLD) || 00895 (fr->subclass.integer == AST_CONTROL_VIDUPDATE) || 00896 (fr->subclass.integer == AST_CONTROL_SRCUPDATE) || 00897 (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) || 00898 (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) { 00899 /* If we are going on hold, then break callback mode and P2P bridging */ 00900 if (fr->subclass.integer == AST_CONTROL_HOLD) { 00901 if (instance0->engine->local_bridge) { 00902 instance0->engine->local_bridge(instance0, NULL); 00903 } 00904 if (instance1->engine->local_bridge) { 00905 instance1->engine->local_bridge(instance1, NULL); 00906 } 00907 instance0->bridged = NULL; 00908 instance1->bridged = NULL; 00909 } else if (fr->subclass.integer == AST_CONTROL_UNHOLD) { 00910 if (instance0->engine->local_bridge) { 00911 instance0->engine->local_bridge(instance0, instance1); 00912 } 00913 if (instance1->engine->local_bridge) { 00914 instance1->engine->local_bridge(instance1, instance0); 00915 } 00916 instance0->bridged = instance1; 00917 instance1->bridged = instance0; 00918 } 00919 /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */ 00920 if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) { 00921 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen); 00922 } 00923 ast_frfree(fr); 00924 } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) { 00925 if (ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) { 00926 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen); 00927 } 00928 ast_frfree(fr); 00929 } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) { 00930 if (ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) { 00931 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen); 00932 } 00933 ast_frfree(fr); 00934 } else { 00935 *fo = fr; 00936 *rc = who; 00937 ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, who->name); 00938 res = AST_BRIDGE_COMPLETE; 00939 break; 00940 } 00941 } else { 00942 if ((fr->frametype == AST_FRAME_DTMF_BEGIN) || 00943 (fr->frametype == AST_FRAME_DTMF_END) || 00944 (fr->frametype == AST_FRAME_VOICE) || 00945 (fr->frametype == AST_FRAME_VIDEO) || 00946 (fr->frametype == AST_FRAME_IMAGE) || 00947 (fr->frametype == AST_FRAME_HTML) || 00948 (fr->frametype == AST_FRAME_MODEM) || 00949 (fr->frametype == AST_FRAME_TEXT)) { 00950 ast_write(other, fr); 00951 } 00952 00953 ast_frfree(fr); 00954 } 00955 /* Swap priority */ 00956 cs[2] = cs[0]; 00957 cs[0] = cs[1]; 00958 cs[1] = cs[2]; 00959 } 00960 00961 /* Stop locally bridging both instances */ 00962 if (instance0->engine->local_bridge) { 00963 instance0->engine->local_bridge(instance0, NULL); 00964 } 00965 if (instance1->engine->local_bridge) { 00966 instance1->engine->local_bridge(instance1, NULL); 00967 } 00968 00969 instance0->bridged = NULL; 00970 instance1->bridged = NULL; 00971 00972 ast_poll_channel_del(c0, c1); 00973 00974 return res; 00975 }
static enum ast_bridge_result remote_bridge_loop | ( | struct ast_channel * | c0, | |
struct ast_channel * | c1, | |||
struct ast_rtp_instance * | instance0, | |||
struct ast_rtp_instance * | instance1, | |||
struct ast_rtp_instance * | vinstance0, | |||
struct ast_rtp_instance * | vinstance1, | |||
struct ast_rtp_instance * | tinstance0, | |||
struct ast_rtp_instance * | tinstance1, | |||
struct ast_rtp_glue * | glue0, | |||
struct ast_rtp_glue * | glue1, | |||
format_t | codec0, | |||
format_t | codec1, | |||
int | timeoutms, | |||
int | flags, | |||
struct ast_frame ** | fo, | |||
struct ast_channel ** | rc, | |||
void * | pvt0, | |||
void * | pvt1 | |||
) | [static] |
Definition at line 977 of file rtp_engine.c.
References AST_BRIDGE_COMPLETE, AST_BRIDGE_FAILED, AST_BRIDGE_RETRY, ast_channel_connected_line_macro(), ast_channel_redirecting_macro(), ast_channel_unlock, ast_check_hangup(), AST_CONTROL_CONNECTED_LINE, AST_CONTROL_HOLD, AST_CONTROL_REDIRECTING, AST_CONTROL_SRCUPDATE, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_UNHOLD, AST_CONTROL_UPDATE_RTP_PEER, AST_CONTROL_VIDUPDATE, ast_debug, AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_framehook_list_is_empty(), ast_frfree, ast_getformatname(), ast_indicate_data(), ast_log(), ast_poll_channel_add(), ast_read(), ast_rtp_instance_get_remote_address(), ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_waitfor_n(), ast_write(), ast_channel::audiohooks, ast_rtp_instance::bridged, ast_frame::data, ast_frame::datalen, ast_channel::framehooks, ast_frame::frametype, ast_rtp_glue::get_codec, ast_frame_subclass::integer, LOG_WARNING, ast_channel::masq, ast_channel::masqr, ast_channel::monitor, ast_channel::name, ast_frame::ptr, ast_frame::subclass, t1, ast_channel::tech_pvt, and ast_rtp_glue::update_peer.
Referenced by ast_rtp_instance_bridge().
00981 { 00982 enum ast_bridge_result res = AST_BRIDGE_FAILED; 00983 struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, }; 00984 format_t oldcodec0 = codec0, oldcodec1 = codec1; 00985 struct ast_sockaddr ac1 = {{0,}}, vac1 = {{0,}}, tac1 = {{0,}}, ac0 = {{0,}}, vac0 = {{0,}}, tac0 = {{0,}}; 00986 struct ast_sockaddr t1 = {{0,}}, vt1 = {{0,}}, tt1 = {{0,}}, t0 = {{0,}}, vt0 = {{0,}}, tt0 = {{0,}}; 00987 struct ast_frame *fr = NULL; 00988 00989 /* Test the first channel */ 00990 if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0))) { 00991 ast_rtp_instance_get_remote_address(instance1, &ac1); 00992 if (vinstance1) { 00993 ast_rtp_instance_get_remote_address(vinstance1, &vac1); 00994 } 00995 if (tinstance1) { 00996 ast_rtp_instance_get_remote_address(tinstance1, &tac1); 00997 } 00998 } else { 00999 ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name); 01000 } 01001 01002 /* Test the second channel */ 01003 if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0))) { 01004 ast_rtp_instance_get_remote_address(instance0, &ac0); 01005 if (vinstance0) { 01006 ast_rtp_instance_get_remote_address(instance0, &vac0); 01007 } 01008 if (tinstance0) { 01009 ast_rtp_instance_get_remote_address(instance0, &tac0); 01010 } 01011 } else { 01012 ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name); 01013 } 01014 01015 ast_channel_unlock(c0); 01016 ast_channel_unlock(c1); 01017 01018 instance0->bridged = instance1; 01019 instance1->bridged = instance0; 01020 01021 ast_poll_channel_add(c0, c1); 01022 01023 /* Go into a loop handling any stray frames that may come in */ 01024 cs[0] = c0; 01025 cs[1] = c1; 01026 cs[2] = NULL; 01027 for (;;) { 01028 /* Check if anything changed */ 01029 if ((c0->tech_pvt != pvt0) || 01030 (c1->tech_pvt != pvt1) || 01031 (c0->masq || c0->masqr || c1->masq || c1->masqr) || 01032 (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks) || 01033 (!ast_framehook_list_is_empty(c0->framehooks) || !ast_framehook_list_is_empty(c1->framehooks))) { 01034 ast_debug(1, "Oooh, something is weird, backing out\n"); 01035 res = AST_BRIDGE_RETRY; 01036 break; 01037 } 01038 01039 /* Check if they have changed their address */ 01040 ast_rtp_instance_get_remote_address(instance1, &t1); 01041 if (vinstance1) { 01042 ast_rtp_instance_get_remote_address(vinstance1, &vt1); 01043 } 01044 if (tinstance1) { 01045 ast_rtp_instance_get_remote_address(tinstance1, &tt1); 01046 } 01047 if (glue1->get_codec) { 01048 codec1 = glue1->get_codec(c1); 01049 } 01050 01051 ast_rtp_instance_get_remote_address(instance0, &t0); 01052 if (vinstance0) { 01053 ast_rtp_instance_get_remote_address(vinstance0, &vt0); 01054 } 01055 if (tinstance0) { 01056 ast_rtp_instance_get_remote_address(tinstance0, &tt0); 01057 } 01058 if (glue0->get_codec) { 01059 codec0 = glue0->get_codec(c0); 01060 } 01061 01062 if ((ast_sockaddr_cmp(&t1, &ac1)) || 01063 (vinstance1 && ast_sockaddr_cmp(&vt1, &vac1)) || 01064 (tinstance1 && ast_sockaddr_cmp(&tt1, &tac1)) || 01065 (codec1 != oldcodec1)) { 01066 ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n", 01067 c1->name, ast_sockaddr_stringify(&t1), 01068 ast_getformatname(codec1)); 01069 ast_debug(1, "Oooh, '%s' changed end vaddress to %s (format %s)\n", 01070 c1->name, ast_sockaddr_stringify(&vt1), 01071 ast_getformatname(codec1)); 01072 ast_debug(1, "Oooh, '%s' changed end taddress to %s (format %s)\n", 01073 c1->name, ast_sockaddr_stringify(&tt1), 01074 ast_getformatname(codec1)); 01075 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n", 01076 c1->name, ast_sockaddr_stringify(&ac1), 01077 ast_getformatname(oldcodec1)); 01078 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n", 01079 c1->name, ast_sockaddr_stringify(&vac1), 01080 ast_getformatname(oldcodec1)); 01081 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n", 01082 c1->name, ast_sockaddr_stringify(&tac1), 01083 ast_getformatname(oldcodec1)); 01084 if (glue0->update_peer(c0, 01085 ast_sockaddr_isnull(&t1) ? NULL : instance1, 01086 ast_sockaddr_isnull(&vt1) ? NULL : vinstance1, 01087 ast_sockaddr_isnull(&tt1) ? NULL : tinstance1, 01088 codec1, 0)) { 01089 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name); 01090 } 01091 ast_sockaddr_copy(&ac1, &t1); 01092 ast_sockaddr_copy(&vac1, &vt1); 01093 ast_sockaddr_copy(&tac1, &tt1); 01094 oldcodec1 = codec1; 01095 } 01096 if ((ast_sockaddr_cmp(&t0, &ac0)) || 01097 (vinstance0 && ast_sockaddr_cmp(&vt0, &vac0)) || 01098 (tinstance0 && ast_sockaddr_cmp(&tt0, &tac0)) || 01099 (codec0 != oldcodec0)) { 01100 ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n", 01101 c0->name, ast_sockaddr_stringify(&t0), 01102 ast_getformatname(codec0)); 01103 ast_debug(1, "Oooh, '%s' was %s/(format %s)\n", 01104 c0->name, ast_sockaddr_stringify(&ac0), 01105 ast_getformatname(oldcodec0)); 01106 if (glue1->update_peer(c1, t0.len ? instance0 : NULL, 01107 vt0.len ? vinstance0 : NULL, 01108 tt0.len ? tinstance0 : NULL, 01109 codec0, 0)) { 01110 ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name); 01111 } 01112 ast_sockaddr_copy(&ac0, &t0); 01113 ast_sockaddr_copy(&vac0, &vt0); 01114 ast_sockaddr_copy(&tac0, &tt0); 01115 oldcodec0 = codec0; 01116 } 01117 01118 /* Wait for frame to come in on the channels */ 01119 if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) { 01120 if (!timeoutms) { 01121 res = AST_BRIDGE_RETRY; 01122 break; 01123 } 01124 ast_debug(1, "Ooh, empty read...\n"); 01125 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 01126 break; 01127 } 01128 continue; 01129 } 01130 fr = ast_read(who); 01131 other = (who == c0) ? c1 : c0; 01132 if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) && 01133 (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 01134 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) { 01135 /* Break out of bridge */ 01136 *fo = fr; 01137 *rc = who; 01138 ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup"); 01139 res = AST_BRIDGE_COMPLETE; 01140 break; 01141 } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) { 01142 if ((fr->subclass.integer == AST_CONTROL_HOLD) || 01143 (fr->subclass.integer == AST_CONTROL_UNHOLD) || 01144 (fr->subclass.integer == AST_CONTROL_VIDUPDATE) || 01145 (fr->subclass.integer == AST_CONTROL_SRCUPDATE) || 01146 (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) || 01147 (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) { 01148 if (fr->subclass.integer == AST_CONTROL_HOLD) { 01149 /* If we someone went on hold we want the other side to reinvite back to us */ 01150 if (who == c0) { 01151 glue1->update_peer(c1, NULL, NULL, NULL, 0, 0); 01152 } else { 01153 glue0->update_peer(c0, NULL, NULL, NULL, 0, 0); 01154 } 01155 } else if (fr->subclass.integer == AST_CONTROL_UNHOLD || 01156 fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER) { 01157 /* If they went off hold they should go back to being direct, or if we have 01158 * been told to force a peer update, go ahead and do it. */ 01159 if (who == c0) { 01160 glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0); 01161 } else { 01162 glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0); 01163 } 01164 } 01165 /* Update local address information */ 01166 ast_rtp_instance_get_remote_address(instance0, &t0); 01167 ast_sockaddr_copy(&ac0, &t0); 01168 ast_rtp_instance_get_remote_address(instance1, &t1); 01169 ast_sockaddr_copy(&ac1, &t1); 01170 /* Update codec information */ 01171 if (glue0->get_codec && c0->tech_pvt) { 01172 oldcodec0 = codec0 = glue0->get_codec(c0); 01173 } 01174 if (glue1->get_codec && c1->tech_pvt) { 01175 oldcodec1 = codec1 = glue1->get_codec(c1); 01176 } 01177 /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */ 01178 if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) { 01179 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen); 01180 } 01181 ast_frfree(fr); 01182 } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) { 01183 if (ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) { 01184 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen); 01185 } 01186 ast_frfree(fr); 01187 } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) { 01188 if (ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) { 01189 ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen); 01190 } 01191 ast_frfree(fr); 01192 } else { 01193 *fo = fr; 01194 *rc = who; 01195 ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, who->name); 01196 return AST_BRIDGE_COMPLETE; 01197 } 01198 } else { 01199 if ((fr->frametype == AST_FRAME_DTMF_BEGIN) || 01200 (fr->frametype == AST_FRAME_DTMF_END) || 01201 (fr->frametype == AST_FRAME_VOICE) || 01202 (fr->frametype == AST_FRAME_VIDEO) || 01203 (fr->frametype == AST_FRAME_IMAGE) || 01204 (fr->frametype == AST_FRAME_HTML) || 01205 (fr->frametype == AST_FRAME_MODEM) || 01206 (fr->frametype == AST_FRAME_TEXT)) { 01207 ast_write(other, fr); 01208 } 01209 ast_frfree(fr); 01210 } 01211 /* Swap priority */ 01212 cs[2] = cs[0]; 01213 cs[0] = cs[1]; 01214 cs[1] = cs[2]; 01215 } 01216 01217 if (ast_test_flag(c0, AST_FLAG_ZOMBIE)) { 01218 ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", c0->name); 01219 } else if (c0->tech_pvt != pvt0) { 01220 ast_debug(1, "Channel c0->'%s' pvt changed, in bridge with c1->'%s'\n", c0->name, c1->name); 01221 } else if (glue0 != ast_rtp_instance_get_glue(c0->tech->type)) { 01222 ast_debug(1, "Channel c0->'%s' technology changed, in bridge with c1->'%s'\n", c0->name, c1->name); 01223 } else if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) { 01224 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name); 01225 } 01226 if (ast_test_flag(c1, AST_FLAG_ZOMBIE)) { 01227 ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", c1->name); 01228 } else if (c1->tech_pvt != pvt1) { 01229 ast_debug(1, "Channel c1->'%s' pvt changed, in bridge with c0->'%s'\n", c1->name, c0->name); 01230 } else if (glue1 != ast_rtp_instance_get_glue(c1->tech->type)) { 01231 ast_debug(1, "Channel c1->'%s' technology changed, in bridge with c0->'%s'\n", c1->name, c0->name); 01232 } else if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) { 01233 ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name); 01234 } 01235 01236 instance0->bridged = NULL; 01237 instance1->bridged = NULL; 01238 01239 ast_poll_channel_del(c0, c1); 01240 01241 return res; 01242 }
static void unref_instance_cond | ( | struct ast_rtp_instance ** | instance | ) | [static] |
Conditionally unref an rtp instance.
Definition at line 1247 of file rtp_engine.c.
References ao2_ref.
Referenced by ast_rtp_instance_bridge(), ast_rtp_instance_early_bridge(), and ast_rtp_instance_early_bridge_make_compatible().
01248 { 01249 if (*instance) { 01250 ao2_ref(*instance, -1); 01251 *instance = NULL; 01252 } 01253 }
struct ast_rtp_mime_type ast_rtp_mime_types[] [static] |
The following array defines the MIME Media type (and subtype) for each of our codecs, or RTP-specific data type.
Referenced by ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_lookup_mime_subtype2(), and ast_rtp_lookup_sample_rate2().
struct ast_srtp_res* res_srtp = NULL |
Definition at line 48 of file rtp_engine.c.
struct ast_srtp_policy_res* res_srtp_policy = NULL |
Definition at line 49 of file rtp_engine.c.
Referenced by ast_rtp_engine_register_srtp(), ast_rtp_engine_srtp_is_registered(), ast_rtp_engine_unregister_srtp(), sdp_crypto_activate(), and set_crypto_policy().
struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] [static] |
Mapping between Asterisk codecs and rtp payload types.
Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s: also, our own choices for dynamic payload types. This is our master table for transmission
See http://www.iana.org/assignments/rtp-parameters for a list of assigned values
Definition at line 147 of file rtp_engine.c.
Referenced by ast_rtp_codecs_payload_lookup(), ast_rtp_codecs_payloads_default(), and ast_rtp_codecs_payloads_set_m_type().