Pluggable RTP Architecture. More...
#include "asterisk/astobj2.h"
#include "asterisk/frame.h"
#include "asterisk/netsock2.h"
#include "asterisk/sched.h"
#include "asterisk/res_srtp.h"
Go to the source code of this file.
Data Structures | |
struct | ast_rtp_codecs |
struct | ast_rtp_engine |
struct | ast_rtp_glue |
struct | ast_rtp_instance_stats |
struct | ast_rtp_payload_type |
Defines | |
#define | AST_RED_MAX_GENERATION 5 |
#define | AST_RTP_CISCO_DTMF (1 << 2) |
#define | AST_RTP_CN (1 << 1) |
#define | AST_RTP_DTMF (1 << 0) |
#define | ast_rtp_engine_register(engine) ast_rtp_engine_register2(engine, ast_module_info->self) |
#define | ast_rtp_glue_register(glue) ast_rtp_glue_register2(glue, ast_module_info->self) |
#define | AST_RTP_MAX AST_RTP_CISCO_DTMF |
#define | AST_RTP_MAX_PT 256 |
#define | AST_RTP_STAT_SET(current_stat, combined, placement, value) |
#define | AST_RTP_STAT_TERMINATOR(combined) |
Enumerations | |
enum | ast_rtp_dtmf_mode { AST_RTP_DTMF_MODE_NONE = 0, AST_RTP_DTMF_MODE_RFC2833, AST_RTP_DTMF_MODE_INBAND } |
enum | ast_rtp_glue_result { AST_RTP_GLUE_RESULT_FORBID = 0, AST_RTP_GLUE_RESULT_REMOTE, AST_RTP_GLUE_RESULT_LOCAL } |
enum | ast_rtp_instance_stat { AST_RTP_INSTANCE_STAT_ALL = 0, AST_RTP_INSTANCE_STAT_TXCOUNT, AST_RTP_INSTANCE_STAT_RXCOUNT, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, AST_RTP_INSTANCE_STAT_TXPLOSS, AST_RTP_INSTANCE_STAT_RXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, AST_RTP_INSTANCE_STAT_TXJITTER, AST_RTP_INSTANCE_STAT_RXJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_RTT, AST_RTP_INSTANCE_STAT_RTT, AST_RTP_INSTANCE_STAT_MAX_RTT, AST_RTP_INSTANCE_STAT_MIN_RTT, AST_RTP_INSTANCE_STAT_NORMDEVRTT, AST_RTP_INSTANCE_STAT_STDEVRTT, AST_RTP_INSTANCE_STAT_LOCAL_SSRC, AST_RTP_INSTANCE_STAT_REMOTE_SSRC } |
enum | ast_rtp_instance_stat_field { AST_RTP_INSTANCE_STAT_FIELD_QUALITY = 0, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT } |
enum | ast_rtp_options { AST_RTP_OPT_G726_NONSTANDARD = (1 << 0) } |
enum | ast_rtp_property { AST_RTP_PROPERTY_NAT = 0, AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, AST_RTP_PROPERTY_STUN, AST_RTP_PROPERTY_RTCP, AST_RTP_PROPERTY_MAX } |
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. | |
struct 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 *c_dst, struct ast_channel *c_src) |
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). | |
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. | |
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. | |
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. | |
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. | |
struct 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. | |
struct 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. | |
struct 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. | |
struct 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. | |
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. | |
struct 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 timeout) |
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. |
Pluggable RTP Architecture.
Definition in file rtp_engine.h.
#define AST_RED_MAX_GENERATION 5 |
Definition at line 82 of file rtp_engine.h.
Referenced by process_sdp_a_text().
#define AST_RTP_CISCO_DTMF (1 << 2) |
#define AST_RTP_CN (1 << 1) |
'Comfort Noise' (RFC3389)
Definition at line 220 of file rtp_engine.h.
Referenced by ast_rtp_read(), and ast_rtp_sendcng().
#define AST_RTP_DTMF (1 << 0) |
DTMF (RFC2833)
Definition at line 218 of file rtp_engine.h.
Referenced by add_noncodec_to_sdp(), add_sdp(), ast_rtp_dtmf_begin(), ast_rtp_read(), check_peer_ok(), create_addr(), create_addr_from_peer(), oh323_alloc(), oh323_request(), process_sdp(), sip_alloc(), and sip_dtmfmode().
#define ast_rtp_engine_register | ( | engine | ) | ast_rtp_engine_register2(engine, ast_module_info->self) |
Definition at line 423 of file rtp_engine.h.
Referenced by load_module().
#define ast_rtp_glue_register | ( | glue | ) | ast_rtp_glue_register2(glue, ast_module_info->self) |
Definition at line 476 of file rtp_engine.h.
Referenced by load_module().
#define AST_RTP_MAX AST_RTP_CISCO_DTMF |
Maximum RTP-specific code
Definition at line 224 of file rtp_engine.h.
Referenced by add_sdp(), and ast_rtp_lookup_mime_multiple2().
#define AST_RTP_MAX_PT 256 |
Definition at line 79 of file rtp_engine.h.
Referenced by ast_rtp_codecs_payload_code(), ast_rtp_codecs_payload_formats(), ast_rtp_codecs_payload_lookup(), ast_rtp_codecs_payloads_clear(), ast_rtp_codecs_payloads_copy(), ast_rtp_codecs_payloads_default(), ast_rtp_codecs_payloads_set_m_type(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_codecs_payloads_unset(), and process_sdp_a_audio().
#define AST_RTP_STAT_SET | ( | current_stat, | |||
combined, | |||||
placement, | |||||
value | ) |
Definition at line 296 of file rtp_engine.h.
Referenced by ast_rtp_get_stat().
#define AST_RTP_STAT_TERMINATOR | ( | combined | ) |
if (stat == combined) { \ return 0; \ }
Definition at line 304 of file rtp_engine.h.
Referenced by ast_rtp_get_stat().
enum ast_rtp_dtmf_mode |
RTP DTMF Modes
Definition at line 115 of file rtp_engine.h.
00115 { 00116 /*! No DTMF is being carried over the RTP stream */ 00117 AST_RTP_DTMF_MODE_NONE = 0, 00118 /*! DTMF is being carried out of band using RFC2833 */ 00119 AST_RTP_DTMF_MODE_RFC2833, 00120 /*! DTMF is being carried inband over the RTP stream */ 00121 AST_RTP_DTMF_MODE_INBAND, 00122 };
enum ast_rtp_glue_result |
Result codes when RTP glue is queried for information
Definition at line 125 of file rtp_engine.h.
00125 { 00126 /*! No remote or local bridging is permitted */ 00127 AST_RTP_GLUE_RESULT_FORBID = 0, 00128 /*! Move RTP stream to be remote between devices directly */ 00129 AST_RTP_GLUE_RESULT_REMOTE, 00130 /*! Perform RTP engine level bridging if possible */ 00131 AST_RTP_GLUE_RESULT_LOCAL, 00132 };
Statistics that can be retrieved from an RTP instance
AST_RTP_INSTANCE_STAT_ALL |
Retrieve all statistics |
AST_RTP_INSTANCE_STAT_TXCOUNT |
Retrieve number of packets transmitted |
AST_RTP_INSTANCE_STAT_RXCOUNT |
Retrieve number of packets received |
AST_RTP_INSTANCE_STAT_COMBINED_LOSS |
Retrieve ALL statistics relating to packet loss |
AST_RTP_INSTANCE_STAT_TXPLOSS |
Retrieve number of packets lost for transmitting |
AST_RTP_INSTANCE_STAT_RXPLOSS |
Retrieve number of packets lost for receiving |
AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS |
Retrieve maximum number of packets lost on remote side |
AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS |
Retrieve minimum number of packets lost on remote side |
AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS |
Retrieve average number of packets lost on remote side |
AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS |
Retrieve standard deviation of packets lost on remote side |
AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS |
Retrieve maximum number of packets lost on local side |
AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS |
Retrieve minimum number of packets lost on local side |
AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS |
Retrieve average number of packets lost on local side |
AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS |
Retrieve standard deviation of packets lost on local side |
AST_RTP_INSTANCE_STAT_COMBINED_JITTER |
Retrieve ALL statistics relating to jitter |
AST_RTP_INSTANCE_STAT_TXJITTER |
Retrieve jitter on transmitted packets |
AST_RTP_INSTANCE_STAT_RXJITTER |
Retrieve jitter on received packets |
AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER |
Retrieve maximum jitter on remote side |
AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER |
Retrieve minimum jitter on remote side |
AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER |
Retrieve average jitter on remote side |
AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER |
Retrieve standard deviation jitter on remote side |
AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER |
Retrieve maximum jitter on local side |
AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER |
Retrieve minimum jitter on local side |
AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER |
Retrieve average jitter on local side |
AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER |
Retrieve standard deviation jitter on local side |
AST_RTP_INSTANCE_STAT_COMBINED_RTT |
Retrieve ALL statistics relating to round trip time |
AST_RTP_INSTANCE_STAT_RTT |
Retrieve round trip time |
AST_RTP_INSTANCE_STAT_MAX_RTT |
Retrieve maximum round trip time |
AST_RTP_INSTANCE_STAT_MIN_RTT |
Retrieve minimum round trip time |
AST_RTP_INSTANCE_STAT_NORMDEVRTT |
Retrieve average round trip time |
AST_RTP_INSTANCE_STAT_STDEVRTT |
Retrieve standard deviation round trip time |
AST_RTP_INSTANCE_STAT_LOCAL_SSRC |
Retrieve local SSRC |
AST_RTP_INSTANCE_STAT_REMOTE_SSRC |
Retrieve remote SSRC |
Definition at line 147 of file rtp_engine.h.
00147 { 00148 /*! Retrieve all statistics */ 00149 AST_RTP_INSTANCE_STAT_ALL = 0, 00150 /*! Retrieve number of packets transmitted */ 00151 AST_RTP_INSTANCE_STAT_TXCOUNT, 00152 /*! Retrieve number of packets received */ 00153 AST_RTP_INSTANCE_STAT_RXCOUNT, 00154 /*! Retrieve ALL statistics relating to packet loss */ 00155 AST_RTP_INSTANCE_STAT_COMBINED_LOSS, 00156 /*! Retrieve number of packets lost for transmitting */ 00157 AST_RTP_INSTANCE_STAT_TXPLOSS, 00158 /*! Retrieve number of packets lost for receiving */ 00159 AST_RTP_INSTANCE_STAT_RXPLOSS, 00160 /*! Retrieve maximum number of packets lost on remote side */ 00161 AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, 00162 /*! Retrieve minimum number of packets lost on remote side */ 00163 AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, 00164 /*! Retrieve average number of packets lost on remote side */ 00165 AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, 00166 /*! Retrieve standard deviation of packets lost on remote side */ 00167 AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, 00168 /*! Retrieve maximum number of packets lost on local side */ 00169 AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, 00170 /*! Retrieve minimum number of packets lost on local side */ 00171 AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, 00172 /*! Retrieve average number of packets lost on local side */ 00173 AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, 00174 /*! Retrieve standard deviation of packets lost on local side */ 00175 AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, 00176 /*! Retrieve ALL statistics relating to jitter */ 00177 AST_RTP_INSTANCE_STAT_COMBINED_JITTER, 00178 /*! Retrieve jitter on transmitted packets */ 00179 AST_RTP_INSTANCE_STAT_TXJITTER, 00180 /*! Retrieve jitter on received packets */ 00181 AST_RTP_INSTANCE_STAT_RXJITTER, 00182 /*! Retrieve maximum jitter on remote side */ 00183 AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, 00184 /*! Retrieve minimum jitter on remote side */ 00185 AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, 00186 /*! Retrieve average jitter on remote side */ 00187 AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, 00188 /*! Retrieve standard deviation jitter on remote side */ 00189 AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, 00190 /*! Retrieve maximum jitter on local side */ 00191 AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, 00192 /*! Retrieve minimum jitter on local side */ 00193 AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, 00194 /*! Retrieve average jitter on local side */ 00195 AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, 00196 /*! Retrieve standard deviation jitter on local side */ 00197 AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, 00198 /*! Retrieve ALL statistics relating to round trip time */ 00199 AST_RTP_INSTANCE_STAT_COMBINED_RTT, 00200 /*! Retrieve round trip time */ 00201 AST_RTP_INSTANCE_STAT_RTT, 00202 /*! Retrieve maximum round trip time */ 00203 AST_RTP_INSTANCE_STAT_MAX_RTT, 00204 /*! Retrieve minimum round trip time */ 00205 AST_RTP_INSTANCE_STAT_MIN_RTT, 00206 /*! Retrieve average round trip time */ 00207 AST_RTP_INSTANCE_STAT_NORMDEVRTT, 00208 /*! Retrieve standard deviation round trip time */ 00209 AST_RTP_INSTANCE_STAT_STDEVRTT, 00210 /*! Retrieve local SSRC */ 00211 AST_RTP_INSTANCE_STAT_LOCAL_SSRC, 00212 /*! Retrieve remote SSRC */ 00213 AST_RTP_INSTANCE_STAT_REMOTE_SSRC, 00214 };
Field statistics that can be retrieved from an RTP instance
Definition at line 135 of file rtp_engine.h.
00135 { 00136 /*! Retrieve quality information */ 00137 AST_RTP_INSTANCE_STAT_FIELD_QUALITY = 0, 00138 /*! Retrieve quality information about jitter */ 00139 AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, 00140 /*! Retrieve quality information about packet loss */ 00141 AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, 00142 /*! Retrieve quality information about round trip time */ 00143 AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, 00144 };
enum ast_rtp_options |
Additional RTP options
Definition at line 109 of file rtp_engine.h.
00109 { 00110 /*! Remote side is using non-standard G.726 */ 00111 AST_RTP_OPT_G726_NONSTANDARD = (1 << 0), 00112 };
enum ast_rtp_property |
RTP Properties that can be set on an RTP instance
AST_RTP_PROPERTY_NAT |
Enable symmetric RTP support |
AST_RTP_PROPERTY_DTMF |
RTP instance will be carrying DTMF (using RFC2833) |
AST_RTP_PROPERTY_DTMF_COMPENSATE |
Expect unreliable DTMF from remote party |
AST_RTP_PROPERTY_STUN |
Enable STUN support |
AST_RTP_PROPERTY_RTCP |
Enable RTCP support |
AST_RTP_PROPERTY_MAX |
Maximum number of RTP properties supported.
|
Definition at line 88 of file rtp_engine.h.
00088 { 00089 /*! Enable symmetric RTP support */ 00090 AST_RTP_PROPERTY_NAT = 0, 00091 /*! RTP instance will be carrying DTMF (using RFC2833) */ 00092 AST_RTP_PROPERTY_DTMF, 00093 /*! Expect unreliable DTMF from remote party */ 00094 AST_RTP_PROPERTY_DTMF_COMPENSATE, 00095 /*! Enable STUN support */ 00096 AST_RTP_PROPERTY_STUN, 00097 /*! Enable RTCP support */ 00098 AST_RTP_PROPERTY_RTCP, 00099 00100 /*! 00101 * \brief Maximum number of RTP properties supported 00102 * 00103 * \note THIS MUST BE THE LAST ENTRY IN THIS ENUM. 00104 */ 00105 AST_RTP_PROPERTY_MAX, 00106 };
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 |
Example usage:
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, and ast_rtp_codecs::pref.
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 |
Example usage:
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_sendcng(), 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 |
Example usage:
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 | |||
) | [read] |
Retrieve payload information by payload.
codecs | Codecs structure to look in | |
payload | Numerical payload to look up |
Payload | information |
Example usage:
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, and ast_rtp_payload_type::code.
Referenced by ast_rtp_read(), 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 |
Example usage:
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 |
Example usage:
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 |
Example usage:
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, and ast_rtp_codecs::payloads.
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 |
Example usage:
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, and ast_rtp_codecs::payloads.
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 |
Example usage:
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 |
This function 'fills in' an entry in the list of possible formats for a media stream associated with an RTP structure.
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, ast_rtp_mime_type::payload_type, ast_rtp_codecs::payloads, ast_rtp_mime_type::sample_rate, ast_rtp_mime_type::subtype, and ast_rtp_mime_type::type.
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 |
Example usage:
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 |
Example usage:
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 1803 of file rtp_engine.c.
Referenced by res_srtp_init().
01804 { 01805 if (res_srtp || res_srtp_policy) { 01806 return -1; 01807 } 01808 if (!srtp_res || !policy_res) { 01809 return -1; 01810 } 01811 01812 res_srtp = srtp_res; 01813 res_srtp_policy = policy_res; 01814 01815 return 0; 01816 }
int ast_rtp_engine_srtp_is_registered | ( | void | ) |
Definition at line 1824 of file rtp_engine.c.
Referenced by sdp_crypto_activate(), sdp_crypto_process(), sdp_crypto_setup(), set_crypto_policy(), and setup_srtp().
01825 { 01826 return res_srtp && res_srtp_policy; 01827 }
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 |
Example usage:
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 1818 of file rtp_engine.c.
Referenced by res_srtp_shutdown().
01819 { 01820 res_srtp = NULL; 01821 res_srtp_policy = NULL; 01822 }
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 |
Example usage:
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 |
Example usage:
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 |
Example usage:
ast_rtp_instance_activate(instance);
This tells the underlying RTP engine of instance that packets will now flow.
Definition at line 1744 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 1829 of file rtp_engine.c.
References ast_srtp_res::add_stream, ast_srtp_res::create, ast_srtp_res::replace, and ast_rtp_instance::srtp.
Referenced by sdp_crypto_activate().
01830 { 01831 int res = 0; 01832 01833 if (!res_srtp) { 01834 return -1; 01835 } 01836 01837 if (!instance->srtp) { 01838 res = res_srtp->create(&instance->srtp, instance, remote_policy); 01839 } else { 01840 res = res_srtp->replace(&instance->srtp, instance, remote_policy); 01841 } 01842 if (!res) { 01843 res = res_srtp->add_stream(instance->srtp, local_policy); 01844 } 01845 01846 return res; 01847 }
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 |
Example usage:
This sees if it is possible to have ulaw communicated to the endpoint but signed linear received into Asterisk.
Definition at line 1733 of file rtp_engine.c.
References ast_translate_available_formats(), ast_rtp_engine::available_formats, and ast_rtp_instance::engine.
Referenced by sip_call().
01734 { 01735 format_t formats; 01736 01737 if (instance->engine->available_formats && (formats = instance->engine->available_formats(instance, to_endpoint, to_asterisk))) { 01738 return formats; 01739 } 01740 01741 return ast_translate_available_formats(to_endpoint, to_asterisk); 01742 }
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 1274 of file rtp_engine.c.
References AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_check_hangup(), ast_codec_pref_getsize(), 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_instance::codecs, 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_rtp_codecs::pref, ast_channel::rawreadformat, ast_channel::rawwriteformat, remote_bridge_loop(), ast_sockaddr::ss, ast_channel::tech, ast_channel::tech_pvt, ast_channel_tech::type, and unref_instance_cond().
01275 { 01276 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, 01277 *vinstance0 = NULL, *vinstance1 = NULL, 01278 *tinstance0 = NULL, *tinstance1 = NULL; 01279 struct ast_rtp_glue *glue0, *glue1; 01280 struct ast_sockaddr addr1 = { {0, }, }, addr2 = { {0, }, }; 01281 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01282 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01283 enum ast_bridge_result res = AST_BRIDGE_FAILED; 01284 enum ast_rtp_dtmf_mode dmode; 01285 format_t codec0 = 0, codec1 = 0; 01286 int unlock_chans = 1; 01287 int read_ptime0, read_ptime1, write_ptime0, write_ptime1; 01288 01289 /* Lock both channels so we can look for the glue that binds them together */ 01290 ast_channel_lock(c0); 01291 while (ast_channel_trylock(c1)) { 01292 ast_channel_unlock(c0); 01293 usleep(1); 01294 ast_channel_lock(c0); 01295 } 01296 01297 /* Ensure neither channel got hungup during lock avoidance */ 01298 if (ast_check_hangup(c0) || ast_check_hangup(c1)) { 01299 ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name); 01300 goto done; 01301 } 01302 01303 /* Grab glue that binds each channel to something using the RTP engine */ 01304 if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) { 01305 ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name); 01306 goto done; 01307 } 01308 01309 audio_glue0_res = glue0->get_rtp_info(c0, &instance0); 01310 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID; 01311 01312 audio_glue1_res = glue1->get_rtp_info(c1, &instance1); 01313 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID; 01314 01315 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */ 01316 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)) { 01317 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01318 } 01319 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)) { 01320 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01321 } 01322 01323 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */ 01324 if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) { 01325 res = AST_BRIDGE_FAILED_NOWARN; 01326 goto done; 01327 } 01328 01329 01330 /* If address families differ, force a local bridge */ 01331 ast_rtp_instance_get_remote_address(instance0, &addr1); 01332 ast_rtp_instance_get_remote_address(instance1, &addr2); 01333 01334 if (addr1.ss.ss_family != addr2.ss.ss_family || 01335 (ast_sockaddr_is_ipv4_mapped(&addr1) != ast_sockaddr_is_ipv4_mapped(&addr2))) { 01336 audio_glue0_res = AST_RTP_GLUE_RESULT_LOCAL; 01337 audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL; 01338 } 01339 01340 /* If we need to get DTMF see if we can do it outside of the RTP stream itself */ 01341 dmode = ast_rtp_instance_dtmf_mode_get(instance0); 01342 if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && dmode) { 01343 res = AST_BRIDGE_FAILED_NOWARN; 01344 goto done; 01345 } 01346 dmode = ast_rtp_instance_dtmf_mode_get(instance1); 01347 if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && dmode) { 01348 res = AST_BRIDGE_FAILED_NOWARN; 01349 goto done; 01350 } 01351 01352 /* 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 */ 01353 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)))) { 01354 res = AST_BRIDGE_FAILED_NOWARN; 01355 goto done; 01356 } 01357 01358 /* Make sure that codecs match */ 01359 codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0; 01360 codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0; 01361 if (codec0 && codec1 && !(codec0 & codec1)) { 01362 ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n", ast_getformatname(codec0), ast_getformatname(codec1)); 01363 res = AST_BRIDGE_FAILED_NOWARN; 01364 goto done; 01365 } 01366 01367 read_ptime0 = (ast_codec_pref_getsize(&instance0->codecs.pref, c0->rawreadformat)).cur_ms; 01368 read_ptime1 = (ast_codec_pref_getsize(&instance1->codecs.pref, c1->rawreadformat)).cur_ms; 01369 write_ptime0 = (ast_codec_pref_getsize(&instance0->codecs.pref, c0->rawwriteformat)).cur_ms; 01370 write_ptime1 = (ast_codec_pref_getsize(&instance1->codecs.pref, c1->rawwriteformat)).cur_ms; 01371 01372 if (read_ptime0 != write_ptime1 || read_ptime1 != write_ptime0) { 01373 ast_debug(1, "Packetization differs between RTP streams (%d != %d or %d != %d). Cannot native bridge in RTP\n", 01374 read_ptime0, write_ptime1, read_ptime1, write_ptime0); 01375 res = AST_BRIDGE_FAILED_NOWARN; 01376 goto done; 01377 } 01378 01379 instance0->glue = glue0; 01380 instance1->glue = glue1; 01381 instance0->chan = c0; 01382 instance1->chan = c1; 01383 01384 /* Depending on the end result for bridging either do a local bridge or remote bridge */ 01385 if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) { 01386 ast_verb(3, "Locally bridging %s and %s\n", c0->name, c1->name); 01387 res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, c0->tech_pvt, c1->tech_pvt); 01388 } else { 01389 ast_verb(3, "Remotely bridging %s and %s\n", c0->name, c1->name); 01390 res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1, 01391 tinstance0, tinstance1, glue0, glue1, codec0, codec1, timeoutms, flags, 01392 fo, rc, c0->tech_pvt, c1->tech_pvt); 01393 } 01394 01395 instance0->glue = NULL; 01396 instance1->glue = NULL; 01397 instance0->chan = NULL; 01398 instance1->chan = NULL; 01399 01400 unlock_chans = 0; 01401 01402 done: 01403 if (unlock_chans) { 01404 ast_channel_unlock(c0); 01405 ast_channel_unlock(c1); 01406 } 01407 01408 unref_instance_cond(&instance0); 01409 unref_instance_cond(&instance1); 01410 unref_instance_cond(&vinstance0); 01411 unref_instance_cond(&vinstance1); 01412 unref_instance_cond(&tinstance0); 01413 unref_instance_cond(&tinstance1); 01414 01415 return res; 01416 }
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 |
Example usage:
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 |
Example usage:
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 |
Example usage:
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 |
Example usage:
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().
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 |
Example usage:
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 |
Example usage:
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 1509 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::tech, ast_channel_tech::type, unref_instance_cond(), and ast_rtp_glue::update_peer.
01510 { 01511 struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, 01512 *vinstance0 = NULL, *vinstance1 = NULL, 01513 *tinstance0 = NULL, *tinstance1 = NULL; 01514 struct ast_rtp_glue *glue0, *glue1; 01515 enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01516 enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01517 format_t codec0 = 0, codec1 = 0; 01518 int res = 0; 01519 01520 /* If there is no second channel just immediately bail out, we are of no use in that scenario */ 01521 if (!c1) { 01522 return -1; 01523 } 01524 01525 /* Lock both channels so we can look for the glue that binds them together */ 01526 ast_channel_lock(c0); 01527 while (ast_channel_trylock(c1)) { 01528 ast_channel_unlock(c0); 01529 usleep(1); 01530 ast_channel_lock(c0); 01531 } 01532 01533 /* Grab glue that binds each channel to something using the RTP engine */ 01534 if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) { 01535 ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name); 01536 goto done; 01537 } 01538 01539 audio_glue0_res = glue0->get_rtp_info(c0, &instance0); 01540 video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID; 01541 01542 audio_glue1_res = glue1->get_rtp_info(c1, &instance1); 01543 video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID; 01544 01545 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */ 01546 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)) { 01547 audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID; 01548 } 01549 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)) { 01550 audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID; 01551 } 01552 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)) { 01553 codec0 = glue0->get_codec(c0); 01554 } 01555 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)) { 01556 codec1 = glue1->get_codec(c1); 01557 } 01558 01559 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */ 01560 if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) { 01561 goto done; 01562 } 01563 01564 /* Make sure we have matching codecs */ 01565 if (!(codec0 & codec1)) { 01566 goto done; 01567 } 01568 01569 /* Bridge media early */ 01570 if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0)) { 01571 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); 01572 } 01573 01574 res = 0; 01575 01576 done: 01577 ast_channel_unlock(c0); 01578 ast_channel_unlock(c1); 01579 01580 unref_instance_cond(&instance0); 01581 unref_instance_cond(&instance1); 01582 unref_instance_cond(&vinstance0); 01583 unref_instance_cond(&vinstance1); 01584 unref_instance_cond(&tinstance0); 01585 unref_instance_cond(&tinstance1); 01586 01587 if (!res) { 01588 ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>"); 01589 } 01590 01591 return res; 01592 }
void ast_rtp_instance_early_bridge_make_compatible | ( | struct ast_channel * | c_dst, | |
struct ast_channel * | c_src | |||
) |
Make two channels compatible for early bridging.
c_dst | Destination channel to copy to | |
c_src | Source channel to copy from |
Definition at line 1423 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::tech, ast_channel_tech::type, unref_instance_cond(), and ast_rtp_glue::update_peer.
Referenced by dial_exec_full(), and do_forward().
01424 { 01425 struct ast_rtp_instance *instance_dst = NULL, *instance_src = NULL, 01426 *vinstance_dst = NULL, *vinstance_src = NULL, 01427 *tinstance_dst = NULL, *tinstance_src = NULL; 01428 struct ast_rtp_glue *glue_dst, *glue_src; 01429 enum ast_rtp_glue_result audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID; 01430 enum ast_rtp_glue_result audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_src_res = AST_RTP_GLUE_RESULT_FORBID; 01431 format_t codec_dst = 0, codec_src = 0; 01432 int res = 0; 01433 01434 /* Lock both channels so we can look for the glue that binds them together */ 01435 ast_channel_lock(c_dst); 01436 while (ast_channel_trylock(c_src)) { 01437 ast_channel_unlock(c_dst); 01438 usleep(1); 01439 ast_channel_lock(c_dst); 01440 } 01441 01442 /* Grab glue that binds each channel to something using the RTP engine */ 01443 if (!(glue_dst = ast_rtp_instance_get_glue(c_dst->tech->type)) || !(glue_src = ast_rtp_instance_get_glue(c_src->tech->type))) { 01444 ast_debug(1, "Can't find native functions for channel '%s'\n", glue_dst ? c_src->name : c_dst->name); 01445 goto done; 01446 } 01447 01448 audio_glue_dst_res = glue_dst->get_rtp_info(c_dst, &instance_dst); 01449 video_glue_dst_res = glue_dst->get_vrtp_info ? glue_dst->get_vrtp_info(c_dst, &vinstance_dst) : AST_RTP_GLUE_RESULT_FORBID; 01450 01451 audio_glue_src_res = glue_src->get_rtp_info(c_src, &instance_src); 01452 video_glue_src_res = glue_src->get_vrtp_info ? glue_src->get_vrtp_info(c_src, &vinstance_src) : AST_RTP_GLUE_RESULT_FORBID; 01453 01454 /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */ 01455 if (video_glue_dst_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01456 audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID; 01457 } 01458 if (video_glue_src_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE)) { 01459 audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID; 01460 } 01461 if (audio_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_dst_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_dst->get_codec) { 01462 codec_dst = glue_dst->get_codec(c_dst); 01463 } 01464 if (audio_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_src_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_src->get_codec) { 01465 codec_src = glue_src->get_codec(c_src); 01466 } 01467 01468 /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */ 01469 if (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE) { 01470 goto done; 01471 } 01472 01473 /* Make sure we have matching codecs */ 01474 if (!(codec_dst & codec_src)) { 01475 goto done; 01476 } 01477 01478 ast_rtp_codecs_payloads_copy(&instance_src->codecs, &instance_dst->codecs, instance_dst); 01479 01480 if (vinstance_dst && vinstance_src) { 01481 ast_rtp_codecs_payloads_copy(&vinstance_src->codecs, &vinstance_dst->codecs, vinstance_dst); 01482 } 01483 if (tinstance_dst && tinstance_src) { 01484 ast_rtp_codecs_payloads_copy(&tinstance_src->codecs, &tinstance_dst->codecs, tinstance_dst); 01485 } 01486 01487 if (glue_dst->update_peer(c_dst, instance_src, vinstance_src, tinstance_src, codec_src, 0)) { 01488 ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c_dst->name, c_src ? c_src->name : "<unspecified>"); 01489 } 01490 01491 res = 0; 01492 01493 done: 01494 ast_channel_unlock(c_dst); 01495 ast_channel_unlock(c_src); 01496 01497 unref_instance_cond(&instance_dst); 01498 unref_instance_cond(&instance_src); 01499 unref_instance_cond(&vinstance_dst); 01500 unref_instance_cond(&vinstance_src); 01501 unref_instance_cond(&tinstance_dst); 01502 unref_instance_cond(&tinstance_src); 01503 01504 if (!res) { 01505 ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", c_dst->name, c_src ? c_src->name : "<unspecified>"); 01506 } 01507 }
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 |
Example usage:
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 | ) | [read] |
Get the RTP glue in use on an RTP instance.
instance | The RTP instance |
pointer | to the glue |
Example:
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 1793 of file rtp_engine.c.
References ast_rtp_instance::glue.
01794 { 01795 return instance->glue; 01796 }
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 | ) | [read] |
Get the other RTP instance that an instance is bridged to.
instance | The RTP instance that we want |
non-NULL | success | |
NULL | failure |
Example usage:
struct ast_rtp_instance *bridged = ast_rtp_instance_get_bridged(instance0);
This gets the RTP instance that instance0 is bridged to.
Definition at line 1418 of file rtp_engine.c.
References ast_rtp_instance::bridged.
Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and dialog_needdestroy().
01419 { 01420 return instance->bridged; 01421 }
struct ast_channel* ast_rtp_instance_get_chan | ( | struct ast_rtp_instance * | instance | ) | [read] |
Get the channel that is associated with an RTP instance while in a bridge.
instance | The RTP instance |
pointer | to the channel |
Example:
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 1798 of file rtp_engine.c.
References ast_rtp_instance::chan.
01799 { 01800 return instance->chan; 01801 }
struct ast_rtp_codecs* ast_rtp_instance_get_codecs | ( | struct ast_rtp_instance * | instance | ) | [read] |
Get the codecs structure of an RTP instance.
instance | The RTP instance to get the codecs structure from |
Example usage:
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 |
Example usage:
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 | ) | [read] |
Get the RTP engine in use on an RTP instance.
instance | The RTP instance |
pointer | to the engine |
Example usage:
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 1788 of file rtp_engine.c.
References ast_rtp_instance::engine.
01789 { 01790 return instance->engine; 01791 }
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 | ) | [read] |
Get the RTP glue that binds a channel to the RTP engine.
type | Name of the glue we want |
non-NULL | success | |
NULL | failure |
Example usage:
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(), ast_rtp_instance_make_compatible(), and remote_bridge_loop().
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 |
Example usage:
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 1778 of file rtp_engine.c.
References ast_rtp_instance::holdtimeout.
Referenced by check_rtp_timeout().
01779 { 01780 return instance->holdtimeout; 01781 }
int ast_rtp_instance_get_keepalive | ( | struct ast_rtp_instance * | instance | ) |
Get the RTP keepalive interval.
instance | The RTP instance |
period | Keepalive interval value |
Example usage:
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 1783 of file rtp_engine.c.
References ast_rtp_instance::keepalive.
Referenced by check_rtp_timeout().
01784 { 01785 return instance->keepalive; 01786 }
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 |
Example usage:
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(), start_rtp(), and unistim_set_rtp_peer().
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 |
Example usage:
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 |
Example usage:
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 1609 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().
01610 { 01611 struct ast_rtp_instance_stats stats = { 0, }; 01612 enum ast_rtp_instance_stat stat; 01613 01614 /* Determine what statistics we will need to retrieve based on field passed in */ 01615 if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) { 01616 stat = AST_RTP_INSTANCE_STAT_ALL; 01617 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) { 01618 stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER; 01619 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) { 01620 stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS; 01621 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) { 01622 stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT; 01623 } else { 01624 return NULL; 01625 } 01626 01627 /* Attempt to actually retrieve the statistics we need to generate the quality string */ 01628 if (ast_rtp_instance_get_stats(instance, &stats, stat)) { 01629 return NULL; 01630 } 01631 01632 /* Now actually fill the buffer with the good information */ 01633 if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) { 01634 snprintf(buf, size, "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", 01635 stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.rxjitter, stats.rxcount, stats.txjitter, stats.txcount, stats.txploss, stats.rtt); 01636 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) { 01637 snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;", 01638 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)); 01639 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) { 01640 snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;", 01641 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)); 01642 } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) { 01643 snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt); 01644 } 01645 01646 return buf; 01647 }
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 |
Example usage:
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(), handle_response_invite(), 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(), transmit_modify_with_sdp(), and unistim_set_rtp_peer().
00449 { 00450 ast_sockaddr_copy(address, &instance->remote_address); 00451 }
struct ast_srtp* ast_rtp_instance_get_srtp | ( | struct ast_rtp_instance * | instance | ) | [read] |
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 1849 of file rtp_engine.c.
References ast_rtp_instance::srtp.
Referenced by __rtp_recvfrom(), __rtp_sendto(), and ast_rtp_change_source().
01850 { 01851 return instance->srtp; 01852 }
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 |
Example usage:
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 1604 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().
int ast_rtp_instance_get_timeout | ( | struct ast_rtp_instance * | instance | ) |
Get the RTP timeout value.
instance | The RTP instance |
timeout | value |
Example usage:
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 1773 of file rtp_engine.c.
References ast_rtp_instance::timeout.
Referenced by check_rtp_timeout().
01774 { 01775 return instance->timeout; 01776 }
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 |
Example usage:
ast_rtp_instance_make_compatible(instance, peer);
This makes the RTP instance for 'peer' compatible with 'instance' and vice versa.
Definition at line 1693 of file rtp_engine.c.
References ao2_ref, ast_channel_lock, ast_channel_unlock, ast_log(), ast_rtp_instance_get_glue(), ast_rtp_instance::engine, ast_rtp_glue::get_rtp_info, LOG_ERROR, ast_rtp_engine::make_compatible, ast_channel::tech, ast_rtp_glue::type, and ast_channel_tech::type.
Referenced by sip_setoption().
01694 { 01695 struct ast_rtp_glue *glue; 01696 struct ast_rtp_instance *peer_instance = NULL; 01697 int res = -1; 01698 01699 if (!instance->engine->make_compatible) { 01700 return -1; 01701 } 01702 01703 ast_channel_lock(peer); 01704 01705 if (!(glue = ast_rtp_instance_get_glue(peer->tech->type))) { 01706 ast_channel_unlock(peer); 01707 return -1; 01708 } 01709 01710 glue->get_rtp_info(peer, &peer_instance); 01711 if (!peer_instance) { 01712 ast_log(LOG_ERROR, "Unable to get_rtp_info for peer type %s\n", glue->type); 01713 ast_channel_unlock(peer); 01714 return -1; 01715 } 01716 if (peer_instance->engine != instance->engine) { 01717 ast_log(LOG_ERROR, "Peer engine mismatch for type %s\n", glue->type); 01718 ast_channel_unlock(peer); 01719 ao2_ref(peer_instance, -1); 01720 return -1; 01721 } 01722 01723 res = instance->engine->make_compatible(chan, instance, peer, peer_instance); 01724 01725 ast_channel_unlock(peer); 01726 01727 ao2_ref(peer_instance, -1); 01728 peer_instance = NULL; 01729 01730 return res; 01731 }
struct ast_rtp_instance* ast_rtp_instance_new | ( | const char * | engine_name, | |
struct sched_context * | sched, | |||
const struct ast_sockaddr * | sa, | |||
void * | data | |||
) | [read] |
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 |
Example usage:
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_instance::engine, ast_rtp_engine::entry, instance_destructor(), ast_rtp_instance::local_address, LOG_ERROR, ast_rtp_engine::mod, ast_rtp_engine::name, and ast_rtp_engine::new.
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 | |||
) | [read] |
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 |
Example usage:
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 1854 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::sendcng.
Referenced by check_rtp_timeout().
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 |
Example usage:
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 |
Example usage:
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 ast_rtp_new(), and 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 |
Example usage:
ast_rtp_instance_set_hold_timeout(instance, 5000);
This sets the RTP hold timeout value on 'instance' to be 5000.
Definition at line 1763 of file rtp_engine.c.
References ast_rtp_instance::holdtimeout.
Referenced by check_rtp_timeout(), and dialog_initialize_rtp().
01764 { 01765 instance->holdtimeout = timeout; 01766 }
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 |
Example usage:
ast_rtp_instance_set_keepalive(instance, 5000);
This sets the RTP keepalive interval on 'instance' to be 5000.
Definition at line 1768 of file rtp_engine.c.
References ast_rtp_instance::keepalive.
Referenced by dialog_initialize_rtp().
01769 { 01770 instance->keepalive = interval; 01771 }
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 |
Example usage:
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 |
Example usage:
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().
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 |
Example usage:
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().
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 |
Example usage:
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 1683 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::set_read_format.
Referenced by sip_new(), and sip_setoption().
01684 { 01685 return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1; 01686 }
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 |
Example usage:
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 |
Example usage:
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 1649 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().
01650 { 01651 char quality_buf[AST_MAX_USER_FIELD], *quality; 01652 struct ast_channel *bridge = ast_bridged_channel(chan); 01653 01654 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) { 01655 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality); 01656 if (bridge) { 01657 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality); 01658 } 01659 } 01660 01661 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) { 01662 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality); 01663 if (bridge) { 01664 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality); 01665 } 01666 } 01667 01668 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) { 01669 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality); 01670 if (bridge) { 01671 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality); 01672 } 01673 } 01674 01675 if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) { 01676 pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality); 01677 if (bridge) { 01678 pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality); 01679 } 01680 } 01681 }
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 |
Example usage:
ast_rtp_instance_set_timeout(instance, 5000);
This sets the RTP timeout value on 'instance' to be 5000.
Definition at line 1758 of file rtp_engine.c.
References ast_rtp_instance::timeout.
Referenced by check_rtp_timeout(), and dialog_initialize_rtp().
01759 { 01760 instance->timeout = timeout; 01761 }
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 |
Example usage:
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 1688 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::set_write_format.
Referenced by sip_new(), and sip_setoption().
01689 { 01690 return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1; 01691 }
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 |
Example usage:
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 |
Example usage:
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 1749 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().
01752 { 01753 if (instance->engine->stun_request) { 01754 instance->engine->stun_request(instance, suggestion, username); 01755 } 01756 }
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 |
Example usage:
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 |
Example usage:
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 |
Example usage:
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 |
Example usage:
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 1599 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::red_buffer.
Referenced by sip_write().
01600 { 01601 return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1; 01602 }
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 1594 of file rtp_engine.c.
References ast_rtp_instance::engine, and ast_rtp_engine::red_init.
Referenced by process_sdp().