Pluggable RTP Architecture. More...

#include "asterisk.h"
#include <math.h>
#include "asterisk/channel.h"
#include "asterisk/frame.h"
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/manager.h"
#include "asterisk/options.h"
#include "asterisk/astobj2.h"
#include "asterisk/pbx.h"
#include "asterisk/translate.h"
#include "asterisk/netsock2.h"
#include "asterisk/_private.h"
#include "asterisk/framehook.h"

Include dependency graph for rtp_engine.c:

Go to the source code of this file.

Data Structures
struct	ast_rtp_instance
struct	ast_rtp_mime_type
struct	engines
struct	glues
Functions
static void	add_static_payload (int map, const struct ast_format *format, int rtp_code)
int	ast_rtp_codecs_find_payload_code (struct ast_rtp_codecs *codecs, int code)
	Search for a payload code in the ast_rtp_codecs structure.
struct ast_format *	ast_rtp_codecs_get_payload_format (struct ast_rtp_codecs *codecs, int payload)
	Retrieve the actual ast_format stored on the codecs structure for a specific payload.
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, int asterisk_format, const struct ast_format format, int 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, struct ast_format_cap 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_destroy (struct ast_rtp_codecs *codecs)
	Destroy the contents of an RTP codecs structure (but not the structure itself)
int	ast_rtp_codecs_payloads_initialize (struct ast_rtp_codecs *codecs)
	Initialize an RTP codecs structure.
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.
void	ast_rtp_dtls_cfg_copy (const struct ast_rtp_dtls_cfg src_cfg, struct ast_rtp_dtls_cfg dst_cfg)
	Copy contents of a DTLS configuration structure.
void	ast_rtp_dtls_cfg_free (struct ast_rtp_dtls_cfg *dtls_cfg)
	Free contents of a DTLS configuration structure.
int	ast_rtp_dtls_cfg_parse (struct ast_rtp_dtls_cfg dtls_cfg, const char name, const char *value)
	Parse DTLS related configuration options.
int	ast_rtp_engine_init ()
	initializes the rtp engine arrays
int	ast_rtp_engine_load_format (const struct ast_format *format)
	Custom formats declared in codecs.conf at startup must be communicated to the rtp_engine so their mime type can payload number can be initialized.
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_unload_format (const struct ast_format *format)
	Formats requiring the use of a format attribute interface must have that interface registered in order for the rtp engine to handle it correctly. If an attribute interface is unloaded, this function must be called to notify the rtp_engine.
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.
void	ast_rtp_instance_available_formats (struct ast_rtp_instance instance, struct ast_format_cap to_endpoint, struct ast_format_cap to_asterisk, struct ast_format_cap result)
	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_dtls *	ast_rtp_instance_get_dtls (struct ast_rtp_instance *instance)
	Obtain a pointer to the DTLS support present on 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.
struct ast_rtp_engine_ice *	ast_rtp_instance_get_ice (struct ast_rtp_instance *instance)
	Obtain a pointer to the ICE support present on an RTP instance.
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 ast_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 interval)
	Set the RTP keepalive interval.
int	ast_rtp_instance_set_local_address (struct ast_rtp_instance instance, const struct ast_sockaddr address)
	Set the address that we are expecting to receive RTP on.
void	ast_rtp_instance_set_prop (struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
	Set the value of an RTP instance property.
int	ast_rtp_instance_set_qos (struct ast_rtp_instance instance, int tos, int cos, const char desc)
	Set QoS parameters on an RTP session.
int	ast_rtp_instance_set_read_format (struct ast_rtp_instance instance, struct ast_format 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, struct ast_format 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, struct ast_format_cap ast_format_capability, int rtp_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, struct ast_format *format, int code, enum ast_rtp_options options)
	Retrieve mime subtype information on a payload.
unsigned int	ast_rtp_lookup_sample_rate2 (int asterisk_format, struct ast_format *format, int code)
	Get the sample rate associated with known RTP payload types.
int	ast_rtp_red_buffer (struct ast_rtp_instance instance, struct ast_frame frame)
	Buffer a frame in an RTP instance for RED.
int	ast_rtp_red_init (struct ast_rtp_instance instance, int buffer_time, int payloads, int generations)
	Initialize RED support on an RTP instance.
static void	instance_destructor (void *obj)
static enum ast_bridge_result	local_bridge_loop (struct ast_channel c0, struct ast_channel c1, struct ast_rtp_instance instance0, struct ast_rtp_instance instance1, int timeoutms, int flags, struct ast_frame fo, struct ast_channel rc, void pvt0, void pvt1)
static enum ast_bridge_result	remote_bridge_loop (struct ast_channel c0, struct ast_channel c1, struct ast_rtp_instance instance0, struct ast_rtp_instance instance1, struct ast_rtp_instance vinstance0, struct ast_rtp_instance vinstance1, struct ast_rtp_instance tinstance0, struct ast_rtp_instance tinstance1, struct ast_rtp_glue glue0, struct ast_rtp_glue glue1, struct ast_format_cap cap0, struct ast_format_cap cap1, int timeoutms, int flags, struct ast_frame fo, struct ast_channel rc, void pvt0, void pvt1)
static int	rtp_payload_type_add_ast (void obj, void arg, int flags)
static int	rtp_payload_type_add_nonast (void obj, void arg, int flags)
static int	rtp_payload_type_cmp (void obj, void arg, int flags)
static int	rtp_payload_type_find_format (void obj, void arg, int flags)
static int	rtp_payload_type_find_nonast_format (void obj, void arg, int flags)
static int	rtp_payload_type_hash (const void *obj, const int flags)
static void	set_next_mime_type (const struct ast_format format, int rtp_code, char type, char *subtype, unsigned int sample_rate)
static void	unref_instance_cond (struct ast_rtp_instance **instance)
	Conditionally unref an rtp instance.
Variables
static struct ast_rtp_mime_type	ast_rtp_mime_types [128]
static struct engines	engines
static struct glues	glues
static int	mime_types_len = 0
static ast_rwlock_t	mime_types_lock
struct ast_srtp_res *	res_srtp = NULL
struct ast_srtp_policy_res *	res_srtp_policy = NULL
static struct ast_rtp_payload_type	static_RTP_PT [AST_RTP_MAX_PT]
	Mapping between Asterisk codecs and rtp payload types.
static ast_rwlock_t	static_RTP_PT_lock

Detailed Description

Pluggable RTP Architecture.

Author:: Joshua Colp <jcolp@digium.com>

Definition in file rtp_engine.c.

Function Documentation

static void add_static_payload	(	int	map,
		const struct ast_format *	format,
		int	rtp_code
	)		`[static]`

Definition at line 2191 of file rtp_engine.c.

References ast_format_copy(), ast_getformatname(), ast_log(), ast_rwlock_unlock, ast_rwlock_wrlock, ast_rtp_payload_type::asterisk_format, LOG_WARNING, map, ast_rtp_payload_type::rtp_code, and static_RTP_PT.

Referenced by ast_rtp_engine_init(), and ast_rtp_engine_load_format().

{
   int x;
   ast_rwlock_wrlock(&static_RTP_PT_lock);
   if (map < 0) {
      /* find next available dynamic payload slot */
      for (x = 96; x < 127; x++) {
         if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
            map = x;
            break;
         }
      }
   }

   if (map < 0) {
      ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n" ,ast_getformatname(format));
      ast_rwlock_unlock(&static_RTP_PT_lock);
      return;
   }

   if (format) {
      static_RTP_PT[map].asterisk_format = 1;
      ast_format_copy(&static_RTP_PT[map].format, format);
   } else {
      static_RTP_PT[map].rtp_code = rtp_code;
   }
   ast_rwlock_unlock(&static_RTP_PT_lock);
}

int ast_rtp_codecs_find_payload_code	(	struct ast_rtp_codecs *	codecs,
		int	code
	)

Search for a payload code in the ast_rtp_codecs structure.

Parameters:

codecs	Codecs structure to look in
code	The format to look for

Return values:

Numerical payload or -1 if unable to find payload in codecs

Example usage:

 int payload = ast_rtp_codecs_payload_code(&codecs, 0);

This looks for the numerical payload for ULAW in the codecs structure.

Definition at line 773 of file rtp_engine.c.

References ao2_find, ao2_ref, OBJ_KEY, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_codecs::payloads, and type.

Referenced by bridge_p2p_rtp_write().

{
   struct ast_rtp_payload_type *type;
   int res = -1;

   /* Search the payload type in the codecs passed */
   if ((type = ao2_find(codecs->payloads, &code, OBJ_NOLOCK | OBJ_KEY)))
   {
      res = type->payload;
      ao2_ref(type, -1);
      return res;
   }

   return res;
}

struct ast_format* ast_rtp_codecs_get_payload_format	(	struct ast_rtp_codecs *	codecs,
		int	payload
	)		`[read]`

Retrieve the actual ast_format stored on the codecs structure for a specific payload.

Parameters:

codecs	Codecs structure to look in
payload	Numerical payload to look up

Return values:

pointer	to format structure on success
NULL	on failure

Since:: 10.0

Definition at line 673 of file rtp_engine.c.

References ao2_find, ao2_ref, AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, format, ast_rtp_payload_type::format, OBJ_KEY, OBJ_NOLOCK, ast_rtp_codecs::payloads, and type.

Referenced by process_sdp_a_audio(), and process_sdp_a_video().

{
   struct ast_rtp_payload_type *type;
   struct ast_format *format;

   if (payload < 0 || payload >= AST_RTP_MAX_PT) {
      return NULL;
   }

   if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
      return NULL;
   }

   format = type->asterisk_format ? &type->format : NULL;

   ao2_ref(type, -1);

   return format;
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 874 of file rtp_engine.c.

References ast_rtp_instance::codecs, ast_rtp_instance::engine, ast_rtp_engine::packetization_set, ast_rtp_codecs::pref, and prefs.

Referenced by __oh323_rtp_create(), check_peer_ok(), create_addr_from_peer(), gtalk_new(), jingle_enable_video(), jingle_new(), process_sdp_a_audio(), set_peer_capabilities(), start_rtp(), and transmit_response_with_sdp().

{
   codecs->pref = *prefs;

   if (instance && instance->engine->packetization_set) {
      instance->engine->packetization_set(instance, &instance->codecs.pref);
   }
}

int ast_rtp_codecs_payload_code	(	struct ast_rtp_codecs *	codecs,
		int	asterisk_format,
		const struct ast_format *	format,
		int	code
	)

Retrieve a payload based on whether it is an Asterisk format and the code.

Parameters:

codecs	Codecs structure to look in
asterisk_format	Non-zero if the given Asterisk format is present
format	Asterisk format to look for
code	The format to look for

Return values:

Numerical payload

Example usage:

 int payload = ast_rtp_codecs_payload_code(&codecs, 1, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0), 0);

This looks for the numerical payload for ULAW in the codecs structure.

Since:: 1.8

Definition at line 742 of file rtp_engine.c.

References ao2_callback, ao2_ref, ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, AST_RTP_MAX_PT, ast_rwlock_rdlock, ast_rwlock_unlock, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_codecs::payloads, ast_rtp_payload_type::rtp_code, rtp_payload_type_find_format(), rtp_payload_type_find_nonast_format(), static_RTP_PT, and type.

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(), jingle_add_payloads_to_description(), multicast_rtp_write(), and send_start_rtp().

{
   struct ast_rtp_payload_type *type;
   int i, res = -1;

   if (asterisk_format && format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_format, (void*)format))) {
      res = type->payload;
      ao2_ref(type, -1);
      return res;
   } else if (!asterisk_format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_nonast_format, (void*)&code))) {
      res = type->payload;
      ao2_ref(type, -1);
      return res;
   }

   ast_rwlock_rdlock(&static_RTP_PT_lock);
   for (i = 0; i < AST_RTP_MAX_PT; i++) {
      if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
         (ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
         res = i;
         break;
      } else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
         (static_RTP_PT[i].rtp_code == code)) {
         res = i;
         break;
      }
   }
   ast_rwlock_unlock(&static_RTP_PT_lock);

   return res;
}

void ast_rtp_codecs_payload_formats	(	struct ast_rtp_codecs *	codecs,
		struct ast_format_cap *	astformats,
		int *	nonastformats
	)

Retrieve all formats that were found.

Parameters:

codecs	Codecs structure to look in
astformats	A capabilities structure to put the Asterisk formats in.
nonastformats	An integer to put the non-Asterisk formats in

Example usage:

 struct ast_format_cap *astformats = ast_format_cap_alloc_nolock()
 int 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.

Since:: 1.8

Definition at line 717 of file rtp_engine.c.

References ao2_callback, ast_format_cap_remove_all(), OBJ_MULTIPLE, OBJ_NODATA, OBJ_NOLOCK, ast_rtp_codecs::payloads, rtp_payload_type_add_ast(), and rtp_payload_type_add_nonast().

Referenced by gtalk_is_answered(), gtalk_newcall(), jingle_interpret_description(), and process_sdp().

{
   ast_format_cap_remove_all(astformats);
   *nonastformats = 0;

   ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_ast, astformats);
   ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_nonast, nonastformats);
}

struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup	(	struct ast_rtp_codecs *	codecs,
		int	payload
	)		`[read]`

Retrieve payload information by payload.

Parameters:

codecs	Codecs structure to look in
payload	Numerical payload to look up

Return values:

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.

Since:: 1.8

Definition at line 650 of file rtp_engine.c.

References ao2_find, ao2_ref, AST_RTP_MAX_PT, ast_rwlock_rdlock, ast_rwlock_unlock, ast_rtp_payload_type::asterisk_format, OBJ_KEY, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_payload_type::rtp_code, static_RTP_PT, and type.

Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), process_sdp_a_audio(), and setup_rtp_connection().

{
   struct ast_rtp_payload_type result = { .asterisk_format = 0, }, *type;

   if (payload < 0 || payload >= AST_RTP_MAX_PT) {
      return result;
   }

   if ((type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
      result = *type;
      ao2_ref(type, -1);
   }

   if (!result.rtp_code && !result.asterisk_format) {
      ast_rwlock_rdlock(&static_RTP_PT_lock);
      result = static_RTP_PT[payload];
      ast_rwlock_unlock(&static_RTP_PT_lock);
   }

   return result;
}

void ast_rtp_codecs_payloads_clear	(	struct ast_rtp_codecs *	codecs,
		struct ast_rtp_instance *	instance
	)

Clear payload information from an RTP instance.

Parameters:

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.

Since:: 1.8

Definition at line 451 of file rtp_engine.c.

References ast_rtp_codecs_payloads_destroy(), ast_rtp_codecs_payloads_initialize(), AST_RTP_MAX_PT, ast_rtp_instance::engine, and ast_rtp_engine::payload_set.

Referenced by gtalk_alloc(), and process_sdp().

{
   ast_rtp_codecs_payloads_destroy(codecs);

   if (instance && instance->engine && instance->engine->payload_set) {
      int i;
      for (i = 0; i < AST_RTP_MAX_PT; i++) {
         instance->engine->payload_set(instance, i, 0, NULL, 0);
      }
   }

   ast_rtp_codecs_payloads_initialize(codecs);
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 499 of file rtp_engine.c.

References ao2_alloc, ao2_find, ao2_link_flags, ao2_ref, ast_debug, AST_RTP_MAX_PT, ast_rtp_payload_type::asterisk_format, ast_rtp_instance::engine, ast_rtp_payload_type::format, OBJ_KEY, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_engine::payload_set, ast_rtp_codecs::payloads, ast_rtp_payload_type::rtp_code, and type.

Referenced by ast_rtp_instance_early_bridge_make_compatible(), jingle_interpret_description(), and process_sdp().

{
   int i;
   struct ast_rtp_payload_type *type;

   for (i = 0; i < AST_RTP_MAX_PT; i++) {
      struct ast_rtp_payload_type *new_type;

      if (!(type = ao2_find(src->payloads, &i, OBJ_KEY | OBJ_NOLOCK))) {
         continue;
      }

      if (!(new_type = ao2_alloc(sizeof(*new_type), NULL))) {
         continue;
      }

      ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);

      new_type->payload = i;
      *new_type = *type;

      ao2_link_flags(dest->payloads, new_type, OBJ_NOLOCK);

      ao2_ref(new_type, -1);

      if (instance && instance->engine && instance->engine->payload_set) {
         instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
      }

      ao2_ref(type, -1);
   }
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 465 of file rtp_engine.c.

References ao2_alloc, ao2_link_flags, ao2_ref, ast_format_copy(), AST_RTP_MAX_PT, ast_rwlock_rdlock, ast_rwlock_unlock, ast_rtp_payload_type::asterisk_format, ast_rtp_instance::engine, ast_rtp_payload_type::format, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_engine::payload_set, ast_rtp_codecs::payloads, ast_rtp_payload_type::rtp_code, static_RTP_PT, and type.

{
   int i;

   ast_rwlock_rdlock(&static_RTP_PT_lock);
   for (i = 0; i < AST_RTP_MAX_PT; i++) {
      if (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
         struct ast_rtp_payload_type *type;

         if (!(type = ao2_alloc(sizeof(*type), NULL))) {
            /* Unfortunately if this occurs the payloads container will not contain all possible default payloads
             * but we err on the side of doing what we can in the hopes that the extreme memory conditions which
             * caused this to occur will go away.
             */
            continue;
         }

         type->payload = i;
         type->asterisk_format = static_RTP_PT[i].asterisk_format;
         type->rtp_code = static_RTP_PT[i].rtp_code;
         ast_format_copy(&type->format, &static_RTP_PT[i].format);

         ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);

         if (instance && instance->engine && instance->engine->payload_set) {
            instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
         }

         ao2_ref(type, -1);
      }
   }
   ast_rwlock_unlock(&static_RTP_PT_lock);
}

void ast_rtp_codecs_payloads_destroy ( struct ast_rtp_codecs * codecs )

Destroy the contents of an RTP codecs structure (but not the structure itself)

Parameters:

codecs The codecs structure to destroy the contents of

Example usage:

 struct ast_rtp_codecs codecs;
 ast_rtp_codecs_payloads_destroy(&codecs);

Since:: 11

Definition at line 446 of file rtp_engine.c.

References ao2_cleanup, and ast_rtp_codecs::payloads.

Referenced by ast_rtp_codecs_payloads_clear(), instance_destructor(), jingle_interpret_description(), and process_sdp().

{
   ao2_cleanup(codecs->payloads);
}

int ast_rtp_codecs_payloads_initialize ( struct ast_rtp_codecs * codecs )

Initialize an RTP codecs structure.

Parameters:

codecs The codecs structure to initialize

Return values:

0	success
-1	failure

Example usage:

 struct ast_rtp_codecs codecs;
 ast_rtp_codecs_payloads_initialize(&codecs);

Since:: 11

Definition at line 437 of file rtp_engine.c.

References ao2_container_alloc, AST_RTP_MAX_PT, ast_rtp_codecs::payloads, rtp_payload_type_cmp(), and rtp_payload_type_hash().

Referenced by ast_rtp_codecs_payloads_clear(), ast_rtp_instance_new(), jingle_interpret_description(), and process_sdp().

{
   if (!(codecs->payloads = ao2_container_alloc(AST_RTP_MAX_PT, rtp_payload_type_hash, rtp_payload_type_cmp))) {
      return -1;
   }

   return 0;
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 532 of file rtp_engine.c.

References ao2_alloc, ao2_find, ao2_link_flags, ao2_ref, ast_debug, ast_format_copy(), AST_RTP_MAX_PT, ast_rwlock_rdlock, ast_rwlock_unlock, ast_rtp_payload_type::asterisk_format, ast_rtp_instance::engine, ast_rtp_payload_type::format, OBJ_KEY, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_engine::payload_set, ast_rtp_codecs::payloads, ast_rtp_payload_type::rtp_code, static_RTP_PT, and type.

Referenced by gtalk_is_answered(), gtalk_newcall(), jingle_interpret_description(), jingle_newcall(), and process_sdp().

{
   struct ast_rtp_payload_type *type;

   ast_rwlock_rdlock(&static_RTP_PT_lock);

   if (payload < 0 || payload >= AST_RTP_MAX_PT) {
      ast_rwlock_unlock(&static_RTP_PT_lock);
      return;
   }

   if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
      if (!(type = ao2_alloc(sizeof(*type), NULL))) {
         ast_rwlock_unlock(&static_RTP_PT_lock);
         return;
      }
      type->payload = payload;
      ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
   }

   type->asterisk_format = static_RTP_PT[payload].asterisk_format;
   type->rtp_code = static_RTP_PT[payload].rtp_code;
   type->payload = payload;
   ast_format_copy(&type->format, &static_RTP_PT[payload].format);

   ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);

   if (instance && instance->engine && instance->engine->payload_set) {
      instance->engine->payload_set(instance, payload, type->asterisk_format, &type->format, type->rtp_code);
   }

   ao2_ref(type, -1);

   ast_rwlock_unlock(&static_RTP_PT_lock);
}

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.

Parameters:

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

Return values:

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.

Since:: 1.8

Definition at line 630 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_interpret_description(), jingle_newcall(), process_sdp(), set_dtmf_payload(), and setup_rtp_connection().

{
   return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
}

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.

Parameters:

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.

Return values:

0	on success
-1	if the payload type is out of range
-2	if the mimeType/mimeSubtype combination was not found

Since:: 1.8

Definition at line 568 of file rtp_engine.c.

References ao2_alloc, ao2_find, ao2_link_flags, ao2_ref, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, ast_format_set(), AST_RTP_MAX_PT, ast_rtp_mime_types, AST_RTP_OPT_G726_NONSTANDARD, ast_rwlock_rdlock, ast_rwlock_unlock, ast_rtp_payload_type::asterisk_format, ast_rtp_instance::engine, ast_rtp_payload_type::format, ast_format::id, mime_types_len, OBJ_KEY, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_engine::payload_set, ast_rtp_mime_type::payload_type, ast_rtp_codecs::payloads, ast_rtp_payload_type::rtp_code, ast_rtp_mime_type::sample_rate, ast_rtp_mime_type::subtype, type, and ast_rtp_mime_type::type.

Referenced by ast_rtp_codecs_payloads_set_rtpmap_type(), jingle_interpret_description(), process_sdp_a_audio(), process_sdp_a_text(), and process_sdp_a_video().

{
   unsigned int i;
   int found = 0;

   if (pt < 0 || pt >= AST_RTP_MAX_PT)
      return -1; /* bogus payload type */

   ast_rwlock_rdlock(&mime_types_lock);
   for (i = 0; i < mime_types_len; ++i) {
      const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
      struct ast_rtp_payload_type *type;

      if (strcasecmp(mimesubtype, t->subtype)) {
         continue;
      }

      if (strcasecmp(mimetype, t->type)) {
         continue;
      }

      /* if both sample rates have been supplied, and they don't match,
       * then this not a match; if one has not been supplied, then the
       * rates are not compared */
      if (sample_rate && t->sample_rate &&
          (sample_rate != t->sample_rate)) {
         continue;
      }

      found = 1;

      if (!(type = ao2_find(codecs->payloads, &pt, OBJ_KEY | OBJ_NOLOCK))) {
         if (!(type = ao2_alloc(sizeof(*type), NULL))) {
            continue;
         }
         type->payload = pt;
         ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
      }

      *type = t->payload_type;
      type->payload = pt;

      if ((t->payload_type.format.id == AST_FORMAT_G726) && t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
         ast_format_set(&type->format, AST_FORMAT_G726_AAL2, 0);
      }

      if (instance && instance->engine && instance->engine->payload_set) {
         instance->engine->payload_set(instance, pt, type->asterisk_format, &type->format, type->rtp_code);
      }

      ao2_ref(type, -1);

      break;
   }
   ast_rwlock_unlock(&mime_types_lock);

   return (found ? 0 : -2);
}

void ast_rtp_codecs_payloads_unset	(	struct ast_rtp_codecs *	codecs,
		struct ast_rtp_instance *	instance,
		int	payload
	)

Remove payload information.

Parameters:

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.

Since:: 1.8

Definition at line 635 of file rtp_engine.c.

References ao2_find, ast_debug, AST_RTP_MAX_PT, ast_rtp_instance::engine, OBJ_KEY, OBJ_NODATA, OBJ_NOLOCK, OBJ_UNLINK, ast_rtp_engine::payload_set, and ast_rtp_codecs::payloads.

Referenced by process_sdp_a_audio(), and process_sdp_a_video().

{
   if (payload < 0 || payload >= AST_RTP_MAX_PT) {
      return;
   }

   ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);

   ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK | OBJ_NODATA | OBJ_UNLINK);

   if (instance && instance->engine && instance->engine->payload_set) {
      instance->engine->payload_set(instance, payload, 0, NULL, 0);
   }
}

void ast_rtp_dtls_cfg_copy	(	const struct ast_rtp_dtls_cfg *	src_cfg,
		struct ast_rtp_dtls_cfg *	dst_cfg
	)

Copy contents of a DTLS configuration structure.

Parameters:

src_cfg	source DTLS configuration structure
dst_cfg	destination DTLS configuration structure

Definition at line 2147 of file rtp_engine.c.

References ast_strdup, ast_rtp_dtls_cfg::cafile, ast_rtp_dtls_cfg::capath, ast_rtp_dtls_cfg::certfile, ast_rtp_dtls_cfg::cipher, ast_rtp_dtls_cfg::default_setup, ast_rtp_dtls_cfg::enabled, ast_rtp_dtls_cfg::pvtfile, ast_rtp_dtls_cfg::rekey, ast_rtp_dtls_cfg::suite, and ast_rtp_dtls_cfg::verify.

Referenced by check_peer_ok(), and create_addr_from_peer().

{
   dst_cfg->enabled = src_cfg->enabled;
   dst_cfg->verify = src_cfg->verify;
   dst_cfg->rekey = src_cfg->rekey;
   dst_cfg->suite = src_cfg->suite;
   dst_cfg->certfile = ast_strdup(src_cfg->certfile);
   dst_cfg->pvtfile = ast_strdup(src_cfg->pvtfile);
   dst_cfg->cipher = ast_strdup(src_cfg->cipher);
   dst_cfg->cafile = ast_strdup(src_cfg->cafile);
   dst_cfg->capath = ast_strdup(src_cfg->capath);
   dst_cfg->default_setup = src_cfg->default_setup;
}

void ast_rtp_dtls_cfg_free ( struct ast_rtp_dtls_cfg * dtls_cfg )

Free contents of a DTLS configuration structure.

Parameters:

dtls_cfg a DTLS configuration structure

Definition at line 2161 of file rtp_engine.c.

References ast_free, ast_rtp_dtls_cfg::cafile, ast_rtp_dtls_cfg::capath, ast_rtp_dtls_cfg::certfile, ast_rtp_dtls_cfg::cipher, and ast_rtp_dtls_cfg::pvtfile.

Referenced by __sip_destroy(), and sip_destroy_peer().

{
   ast_free(dtls_cfg->certfile);
   ast_free(dtls_cfg->pvtfile);
   ast_free(dtls_cfg->cipher);
   ast_free(dtls_cfg->cafile);
   ast_free(dtls_cfg->capath);
}

int ast_rtp_dtls_cfg_parse	(	struct ast_rtp_dtls_cfg *	dtls_cfg,
		const char *	name,
		const char *	value
	)

Parse DTLS related configuration options.

Parameters:

dtls_cfg	a DTLS configuration structure
name	name of the configuration option
value	value of the configuration option

Return values:

0	if handled
-1	if not handled

Definition at line 2107 of file rtp_engine.c.

References ast_free, AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_PASSIVE, ast_strdup, ast_true(), ast_rtp_dtls_cfg::cafile, ast_rtp_dtls_cfg::capath, ast_rtp_dtls_cfg::certfile, ast_rtp_dtls_cfg::cipher, ast_rtp_dtls_cfg::default_setup, ast_rtp_dtls_cfg::enabled, ast_rtp_dtls_cfg::pvtfile, ast_rtp_dtls_cfg::rekey, and ast_rtp_dtls_cfg::verify.

Referenced by build_peer().

{
   if (!strcasecmp(name, "dtlsenable")) {
      dtls_cfg->enabled = ast_true(value) ? 1 : 0;
   } else if (!strcasecmp(name, "dtlsverify")) {
      dtls_cfg->verify = ast_true(value) ? 1 : 0;
   } else if (!strcasecmp(name, "dtlsrekey")) {
      if (sscanf(value, "%30u", &dtls_cfg->rekey) != 1) {
         return -1;
      }
   } else if (!strcasecmp(name, "dtlscertfile")) {
      ast_free(dtls_cfg->certfile);
      dtls_cfg->certfile = ast_strdup(value);
   } else if (!strcasecmp(name, "dtlsprivatekey")) {
      ast_free(dtls_cfg->pvtfile);
      dtls_cfg->pvtfile = ast_strdup(value);
   } else if (!strcasecmp(name, "dtlscipher")) {
      ast_free(dtls_cfg->cipher);
      dtls_cfg->cipher = ast_strdup(value);
   } else if (!strcasecmp(name, "dtlscafile")) {
      ast_free(dtls_cfg->cafile);
      dtls_cfg->cafile = ast_strdup(value);
   } else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
      ast_free(dtls_cfg->capath);
      dtls_cfg->capath = ast_strdup(value);
   } else if (!strcasecmp(name, "dtlssetup")) {
      if (!strcasecmp(value, "active")) {
         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTIVE;
      } else if (!strcasecmp(value, "passive")) {
         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_PASSIVE;
      } else if (!strcasecmp(value, "actpass")) {
         dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTPASS;
      }
   } else {
      return -1;
   }

   return 0;
}

int ast_rtp_engine_init ( void )

initializes the rtp engine arrays

Definition at line 2267 of file rtp_engine.c.

References add_static_payload(), AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G719, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_H261, AST_FORMAT_H263, AST_FORMAT_H263_PLUS, AST_FORMAT_H264, AST_FORMAT_ILBC, AST_FORMAT_JPEG, AST_FORMAT_LPC10, AST_FORMAT_MP4_VIDEO, AST_FORMAT_PNG, ast_format_set(), AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_SPEEX16, AST_FORMAT_SPEEX32, AST_FORMAT_T140, AST_FORMAT_T140RED, AST_FORMAT_ULAW, AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rwlock_init, and set_next_mime_type().

Referenced by main().

{
   struct ast_format tmpfmt;

   ast_rwlock_init(&mime_types_lock);
   ast_rwlock_init(&static_RTP_PT_lock);

   /* Define all the RTP mime types available */
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0, "audio", "G723", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0, "audio", "GSM", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "PCMU", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "G711U", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "PCMA", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "G711A", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0, "audio", "G726-32", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0, "audio", "DVI4", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0, "audio", "L16", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16", 16000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16-256", 16000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0, "audio", "LPC", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729A", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G.729", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0, "audio", "speex", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0,  "audio", "speex", 16000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0,  "audio", "speex", 32000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0, "audio", "iLBC", 8000);
   /* this is the sample rate listed in the RTP profile for the G.722 codec, *NOT* the actual sample rate of the media stream */
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0, "audio", "G722", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0, "audio", "AAL2-G726-32", 8000);
   set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
   set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
   set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0, "video", "JPEG", 90000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_PNG, 0), 0, "video", "PNG", 90000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0, "video", "H261", 90000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0, "video", "H263", 90000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0, "video", "h263-1998", 90000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0, "video", "H264", 90000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0, "video", "MP4V-ES", 90000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0, "text", "RED", 1000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0, "text", "T140", 1000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
   set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);

   /* Define the static rtp payload mappings */
   add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
   #ifdef USE_DEPRECATED_G726
   add_static_payload(2, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
   #endif
   add_static_payload(3, ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0);
   add_static_payload(4, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0);
   add_static_payload(5, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0);/* 8 kHz */
   add_static_payload(6, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 16 kHz */
   add_static_payload(7, ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0);
   add_static_payload(8, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0);
   add_static_payload(9, ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0);
   add_static_payload(10, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 2 channels */
   add_static_payload(11, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 1 channel */
   add_static_payload(13, NULL, AST_RTP_CN);
   add_static_payload(16, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 11.025 kHz */
   add_static_payload(17, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 22.050 kHz */
   add_static_payload(18, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0);
   add_static_payload(19, NULL, AST_RTP_CN);         /* Also used for CN */
   add_static_payload(26, ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0);
   add_static_payload(31, ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0);
   add_static_payload(34, ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0);
   add_static_payload(97, ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0);
   add_static_payload(98, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
   add_static_payload(99, ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0);
   add_static_payload(101, NULL, AST_RTP_DTMF);
   add_static_payload(102, ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0);
   add_static_payload(103, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
   add_static_payload(104, ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0);
   add_static_payload(105, ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0);   /* Real time text chat (with redundancy encoding) */
   add_static_payload(106, ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0);     /* Real time text chat */
   add_static_payload(110, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0);
   add_static_payload(111, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);
   add_static_payload(112, ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0);
   add_static_payload(115, ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0);
   add_static_payload(116, ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0);
   add_static_payload(117, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0);
   add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
   add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
   add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */

   return 0;
}

int ast_rtp_engine_load_format ( const struct ast_format * format )

Custom formats declared in codecs.conf at startup must be communicated to the rtp_engine so their mime type can payload number can be initialized.

Definition at line 2220 of file rtp_engine.c.

References add_static_payload(), AST_FORMAT_CELT, ast_format_rate(), AST_FORMAT_SILK, ast_format::id, and set_next_mime_type().

Referenced by ast_format_attr_reg_interface().

{
   switch (format->id) {
   case AST_FORMAT_SILK:
      set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
      add_static_payload(-1, format, 0);
      break;
   case AST_FORMAT_CELT:
      set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
      add_static_payload(-1, format, 0);
      break;
   default:
      break;
   }

   return 0;
}

int ast_rtp_engine_register2	(	struct ast_rtp_engine *	engine,
		struct ast_module *	module
	)

Register an RTP engine.

Parameters:

engine	Structure of the RTP engine to register
module	Module that the RTP engine is part of

Return values:

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.

Note:: It is recommended that you use the ast_rtp_engine_register macro so that the module is associated with the RTP engine and use counting is performed.

Since:: 1.8

Definition at line 114 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, LOG_WARNING, ast_rtp_engine::mod, ast_rtp_engine::name, ast_rtp_engine::new, ast_rtp_engine::read, and ast_rtp_engine::write.

{
   struct ast_rtp_engine *current_engine;

   /* Perform a sanity check on the engine structure to make sure it has the basics */
   if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
      ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
      return -1;
   }

   /* Link owner module to the RTP engine for reference counting purposes */
   engine->mod = module;

   AST_RWLIST_WRLOCK(&engines);

   /* Ensure that no two modules with the same name are registered at the same time */
   AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
      if (!strcmp(current_engine->name, engine->name)) {
         ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
         AST_RWLIST_UNLOCK(&engines);
         return -1;
      }
   }

   /* The engine survived our critique. Off to the list it goes to be used */
   AST_RWLIST_INSERT_TAIL(&engines, engine, entry);

   AST_RWLIST_UNLOCK(&engines);

   ast_verb(2, "Registered RTP engine '%s'\n", engine->name);

   return 0;
}

int ast_rtp_engine_register_srtp	(	struct ast_srtp_res *	srtp_res,
		struct ast_srtp_policy_res *	policy_res
	)

Definition at line 2037 of file rtp_engine.c.

References policy_res, and srtp_res.

Referenced by res_srtp_init().

{
   if (res_srtp || res_srtp_policy) {
      return -1;
   }
   if (!srtp_res || !policy_res) {
      return -1;
   }

   res_srtp = srtp_res;
   res_srtp_policy = policy_res;

   return 0;
}

int ast_rtp_engine_srtp_is_registered ( void )

Definition at line 2058 of file rtp_engine.c.

References res_srtp_policy.

Referenced by ast_rtp_dtls_set_configuration(), dialog_initialize_dtls_srtp(), sdp_crypto_activate(), sdp_crypto_process(), sdp_crypto_setup(), set_crypto_policy(), and setup_srtp().

{
   return res_srtp && res_srtp_policy;
}

int ast_rtp_engine_unload_format ( const struct ast_format * format )

Formats requiring the use of a format attribute interface must have that interface registered in order for the rtp engine to handle it correctly. If an attribute interface is unloaded, this function must be called to notify the rtp_engine.

Definition at line 2238 of file rtp_engine.c.

References ast_format_cmp(), AST_FORMAT_CMP_EQUAL, AST_RTP_MAX_PT, ast_rtp_mime_types, ast_rwlock_unlock, ast_rwlock_wrlock, mime_types_len, and static_RTP_PT.

Referenced by ast_format_attr_unreg_interface().

{
   int x;
   int y = 0;

   ast_rwlock_wrlock(&static_RTP_PT_lock);
   /* remove everything pertaining to this format id from the lists */
   for (x = 0; x < AST_RTP_MAX_PT; x++) {
      if (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
         memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
      }
   }
   ast_rwlock_unlock(&static_RTP_PT_lock);


   ast_rwlock_wrlock(&mime_types_lock);
   /* rebuild the list skipping the items matching this id */
   for (x = 0; x < mime_types_len; x++) {
      if (ast_format_cmp(&ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
         continue;
      }
      ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
      y++;
   }
   mime_types_len = y;
   ast_rwlock_unlock(&mime_types_lock);
   return 0;
}

int ast_rtp_engine_unregister ( struct ast_rtp_engine * engine )

Unregister an RTP engine.

Parameters:

engine Structure of the RTP engine to unregister

Return values:

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.

Since:: 1.8

Definition at line 148 of file rtp_engine.c.

References AST_RWLIST_REMOVE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_verb, and ast_rtp_engine::name.

Referenced by load_module(), and unload_module().

{
   struct ast_rtp_engine *current_engine = NULL;

   AST_RWLIST_WRLOCK(&engines);

   if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
      ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
   }

   AST_RWLIST_UNLOCK(&engines);

   return current_engine ? 0 : -1;
}

void ast_rtp_engine_unregister_srtp ( void )

Definition at line 2052 of file rtp_engine.c.

Referenced by res_srtp_shutdown().

{
   res_srtp = NULL;
   res_srtp_policy = NULL;
}

int ast_rtp_glue_register2	(	struct ast_rtp_glue *	glue,
		struct ast_module *	module
	)

Register RTP glue.

Parameters:

glue	The glue to register
module	Module that the RTP glue is part of

Return values:

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.

Note:: It is recommended that you use the ast_rtp_glue_register macro so that the module is associated with the RTP glue and use counting is performed.

Since:: 1.8

Definition at line 163 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, LOG_WARNING, ast_rtp_glue::mod, and ast_rtp_glue::type.

{
   struct ast_rtp_glue *current_glue = NULL;

   if (ast_strlen_zero(glue->type)) {
      return -1;
   }

   glue->mod = module;

   AST_RWLIST_WRLOCK(&glues);

   AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
      if (!strcasecmp(current_glue->type, glue->type)) {
         ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
         AST_RWLIST_UNLOCK(&glues);
         return -1;
      }
   }

   AST_RWLIST_INSERT_TAIL(&glues, glue, entry);

   AST_RWLIST_UNLOCK(&glues);

   ast_verb(2, "Registered RTP glue '%s'\n", glue->type);

   return 0;
}

int ast_rtp_glue_unregister ( struct ast_rtp_glue * glue )

Unregister RTP glue.

Parameters:

glue	The glue to unregister

Return values:

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.

Since:: 1.8

Definition at line 192 of file rtp_engine.c.

References AST_RWLIST_REMOVE, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_verb, and ast_rtp_glue::type.

Referenced by load_module(), and unload_module().

{
   struct ast_rtp_glue *current_glue = NULL;

   AST_RWLIST_WRLOCK(&glues);

   if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
      ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
   }

   AST_RWLIST_UNLOCK(&glues);

   return current_glue ? 0 : -1;
}

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.

Parameters:

instance The RTP instance

Return values:

0	success
-1	failure

Example usage:

 ast_rtp_instance_activate(instance);

This tells the underlying RTP engine of instance that packets will now flow.

Since:: 1.8

Definition at line 1978 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().

{
   return instance->engine->activate ? instance->engine->activate(instance) : 0;
}

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.

Parameters:

instance	the RTP instance
remote_policy	the remote endpoint's policy
local_policy	our policy for this RTP instance's remote endpoint

Return values:

0	Success
non-zero	Failure

Definition at line 2063 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 dtls_srtp_setup(), and sdp_crypto_activate().

{
   int res = 0;

   if (!res_srtp) {
      return -1;
   }

   if (!instance->srtp) {
      res = res_srtp->create(&instance->srtp, instance, remote_policy);
   } else {
      res = res_srtp->replace(&instance->srtp, instance, remote_policy);
   }
   if (!res) {
      res = res_srtp->add_stream(instance->srtp, local_policy);
   }

   return res;
}

void ast_rtp_instance_available_formats	(	struct ast_rtp_instance *	instance,
		struct ast_format_cap *	to_endpoint,
		struct ast_format_cap *	to_asterisk,
		struct ast_format_cap *	result
	)

Request the formats that can be transcoded.

Parameters:

instance	The RTP instance
to_endpoint	Formats being sent/received towards the endpoint
to_asterisk	Formats being sent/received towards Asterisk
result	capabilities structure to store and return supported formats in.

Example usage:

 ast_rtp_instance_available_formats(instance, to_capabilities, from_capabilities, result_capabilities);

This sees if it is possible to have ulaw communicated to the endpoint but signed linear received into Asterisk.

Since:: 1.8

Definition at line 1966 of file rtp_engine.c.

References ast_format_cap_is_empty(), ast_translate_available_formats(), ast_rtp_engine::available_formats, and ast_rtp_instance::engine.

Referenced by sip_call().

{
   if (instance->engine->available_formats) {
      instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
      if (!ast_format_cap_is_empty(result)) {
         return;
      }
   }

   ast_translate_available_formats(to_endpoint, to_asterisk, result);
}

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.

Parameters:

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

Return values:

Bridge result

Note:: This should only be used by channel drivers in their technology declaration.

Since:: 1.8

Definition at line 1465 of file rtp_engine.c.

References ast_rtp_glue::allow_rtp_remote, ast_rtp_glue::allow_vrtp_remote, AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, ast_channel_lock, ast_channel_name(), ast_channel_rawreadformat(), ast_channel_rawwriteformat(), ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_trylock, ast_channel_unlock, ast_check_hangup(), ast_codec_pref_getsize(), ast_debug, ast_format_cap_alloc_nolock(), ast_format_cap_destroy(), ast_format_cap_has_joint(), ast_format_cap_is_empty(), ast_getformatname_multiple(), 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, remote_bridge_loop(), ast_sockaddr::ss, type, and unref_instance_cond().

{
   struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
         *vinstance0 = NULL, *vinstance1 = NULL,
         *tinstance0 = NULL, *tinstance1 = NULL;
   struct ast_rtp_glue *glue0, *glue1;
   struct ast_sockaddr addr1 = { {0, }, }, addr2 = { {0, }, };
   enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
   enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
   enum ast_bridge_result res = AST_BRIDGE_FAILED;
   enum ast_rtp_dtmf_mode dmode;
   struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
   struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
   int unlock_chans = 1;
   int read_ptime0, read_ptime1, write_ptime0, write_ptime1;

   if (!cap0 || !cap1) {
      unlock_chans = 0;
      goto done;
   }

   /* Lock both channels so we can look for the glue that binds them together */
   ast_channel_lock(c0);
   while (ast_channel_trylock(c1)) {
      ast_channel_unlock(c0);
      usleep(1);
      ast_channel_lock(c0);
   }

   /* Ensure neither channel got hungup during lock avoidance */
   if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
      ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
      goto done;
   }

   /* Grab glue that binds each channel to something using the RTP engine */
   if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
      ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
      goto done;
   }

   audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
   video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;

   audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
   video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;

   /* If the channels are of the same technology, they might have limitations on remote bridging */
   if (ast_channel_tech(c0) == ast_channel_tech(c1)) {
      if (audio_glue0_res == audio_glue1_res && audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
         if (glue0->allow_rtp_remote && !(glue0->allow_rtp_remote(c0, c1))) {
            /* If the allow_rtp_remote indicates that remote isn't allowed, revert to local bridge */
            audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
         }
      }
      if (video_glue0_res == video_glue1_res && video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
         if (glue0->allow_vrtp_remote && !(glue0->allow_vrtp_remote(c0, c1))) {
            /* if the allow_vrtp_remote indicates that remote isn't allowed, revert to local bridge */
            video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
         }
      }
   }

   /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
   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)) {
      audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
   }
   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)) {
      audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
   }

   /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
   if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }


   /* If address families differ, force a local bridge */
   ast_rtp_instance_get_remote_address(instance0, &addr1);
   ast_rtp_instance_get_remote_address(instance1, &addr2);

   if (addr1.ss.ss_family != addr2.ss.ss_family ||
      (ast_sockaddr_is_ipv4_mapped(&addr1) != ast_sockaddr_is_ipv4_mapped(&addr2))) {
      audio_glue0_res = AST_RTP_GLUE_RESULT_LOCAL;
      audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
   }

   /* If we need to get DTMF see if we can do it outside of the RTP stream itself */
   dmode = ast_rtp_instance_dtmf_mode_get(instance0);
   if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && dmode) {
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }
   dmode = ast_rtp_instance_dtmf_mode_get(instance1);
   if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && dmode) {
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }

   /* 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 */
   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)))) {
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }

   /* Make sure that codecs match */
   if (glue0->get_codec){
      glue0->get_codec(c0, cap0);
   }
   if (glue1->get_codec) {
      glue1->get_codec(c1, cap1);
   }
   if (!ast_format_cap_is_empty(cap0) && !ast_format_cap_is_empty(cap1) && !ast_format_cap_has_joint(cap0, cap1)) {
      char tmp0[256] = { 0, };
      char tmp1[256] = { 0, };
      ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n",
         ast_getformatname_multiple(tmp0, sizeof(tmp0), cap0),
         ast_getformatname_multiple(tmp1, sizeof(tmp1), cap1));
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }

   read_ptime0 = (ast_codec_pref_getsize(&instance0->codecs.pref, ast_channel_rawreadformat(c0))).cur_ms;
   read_ptime1 = (ast_codec_pref_getsize(&instance1->codecs.pref, ast_channel_rawreadformat(c1))).cur_ms;
   write_ptime0 = (ast_codec_pref_getsize(&instance0->codecs.pref, ast_channel_rawwriteformat(c0))).cur_ms;
   write_ptime1 = (ast_codec_pref_getsize(&instance1->codecs.pref, ast_channel_rawwriteformat(c1))).cur_ms;

   if (read_ptime0 != write_ptime1 || read_ptime1 != write_ptime0) {
      ast_debug(1, "Packetization differs between RTP streams (%d != %d or %d != %d). Cannot native bridge in RTP\n",
            read_ptime0, write_ptime1, read_ptime1, write_ptime0);
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }

   instance0->glue = glue0;
   instance1->glue = glue1;
   instance0->chan = c0;
   instance1->chan = c1;

   /* Depending on the end result for bridging either do a local bridge or remote bridge */
   if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
      ast_verb(3, "Locally bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
      res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
   } else {
      ast_verb(3, "Remotely bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
      res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
            tinstance0, tinstance1, glue0, glue1, cap0, cap1, timeoutms, flags,
            fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
   }

   instance0->glue = NULL;
   instance1->glue = NULL;
   instance0->chan = NULL;
   instance1->chan = NULL;

   unlock_chans = 0;

done:
   if (unlock_chans) {
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
   }
   ast_format_cap_destroy(cap1);
   ast_format_cap_destroy(cap0);

   unref_instance_cond(&instance0);
   unref_instance_cond(&instance1);
   unref_instance_cond(&vinstance0);
   unref_instance_cond(&vinstance1);
   unref_instance_cond(&tinstance0);
   unref_instance_cond(&tinstance1);

   return res;
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 914 of file rtp_engine.c.

References ast_rtp_engine::change_source, and ast_rtp_instance::engine.

Referenced by jingle_indicate(), mgcp_indicate(), oh323_indicate(), sip_indicate(), skinny_indicate(), and unistim_indicate().

{
   if (instance->engine->change_source) {
      instance->engine->change_source(instance);
   }
}

int ast_rtp_instance_destroy ( struct ast_rtp_instance * instance )

Destroy an RTP instance.

Parameters:

instance The RTP instance to destroy

Return values:

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.

Since:: 1.8

Definition at line 229 of file rtp_engine.c.

References ao2_ref.

Referenced by __oh323_destroy(), __sip_destroy(), cleanup_connection(), destroy_endpoint(), destroy_rtp(), gtalk_free_pvt(), jingle_free_pvt(), jingle_session_destructor(), mgcp_hangup(), multicast_rtp_hangup(), multicast_rtp_request(), oh323_alloc(), start_rtp(), unalloc_sub(), and unistim_hangup_clean().

{
   ao2_ref(instance, -1);

   return 0;
}

int ast_rtp_instance_dtmf_begin	(	struct ast_rtp_instance *	instance,
		char	digit
	)

Begin sending a DTMF digit.

Parameters:

instance	The RTP instance to send the DTMF on
digit	What DTMF digit to send

Return values:

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.

Since:: 1.8

Definition at line 883 of file rtp_engine.c.

References ast_rtp_engine::dtmf_begin, and ast_rtp_instance::engine.

Referenced by gtalk_digit_begin(), jingle_digit_begin(), mgcp_senddigit_begin(), oh323_digit_begin(), and sip_senddigit_begin().

{
   return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
}

int ast_rtp_instance_dtmf_end	(	struct ast_rtp_instance *	instance,
		char	digit
	)

Stop sending a DTMF digit.

Parameters:

instance	The RTP instance to stop the DTMF on
digit	What DTMF digit to stop

Return values:

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.

Since:: 1.8

Definition at line 888 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().

{
   return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
}

int ast_rtp_instance_dtmf_end_with_duration	(	struct ast_rtp_instance *	instance,
		char	digit,
		unsigned int	duration
	)

Definition at line 892 of file rtp_engine.c.

References ast_rtp_engine::dtmf_end_with_duration, and ast_rtp_instance::engine.

Referenced by gtalk_digit_end(), jingle_digit_end(), and sip_senddigit_end().

{
   return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
}

enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get ( struct ast_rtp_instance * instance )

Get the DTMF mode of an RTP instance.

Parameters:

instance The RTP instance to get the DTMF mode of

Return values:

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'.

Since:: 1.8

Definition at line 902 of file rtp_engine.c.

References ast_rtp_engine::dtmf_mode_get, and ast_rtp_instance::engine.

Referenced by ast_rtp_instance_bridge().

{
   return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
}

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.

Parameters:

instance	the RTP instance to set DTMF mode on
dtmf_mode	The DTMF mode that is in use

Return values:

0	success
-1	failure

Example usage:

 ast_rtp_instance_dtmf_mode_set(instance, AST_RTP_DTMF_MODE_RFC2833);

This sets the RTP instance to use RFC2833 for DTMF transmission and receiving.

Since:: 1.8

Definition at line 897 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().

{
   return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
}

int ast_rtp_instance_early_bridge	(	struct ast_channel *	c0,
		struct ast_channel *	c1
	)

Early bridge two channels that use RTP instances.

Parameters:

c0	First channel part of the bridge
c1	Second channel part of the bridge

Return values:

0	success
-1	failure

Note:: This should only be used by channel drivers in their technology declaration.

Since:: 1.8

Definition at line 1732 of file rtp_engine.c.

References ast_channel_lock, ast_channel_name(), ast_channel_tech(), ast_channel_trylock, ast_channel_unlock, ast_debug, ast_format_cap_alloc_nolock(), ast_format_cap_destroy(), ast_format_cap_has_joint(), 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, type, unref_instance_cond(), and ast_rtp_glue::update_peer.

{
   struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
         *vinstance0 = NULL, *vinstance1 = NULL,
         *tinstance0 = NULL, *tinstance1 = NULL;
   struct ast_rtp_glue *glue0, *glue1;
   enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
   enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
   struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
   struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
   int res = 0;

   /* If there is no second channel just immediately bail out, we are of no use in that scenario */
   if (!c1) {
      ast_format_cap_destroy(cap0);
      ast_format_cap_destroy(cap1);
      return -1;
   }

   /* Lock both channels so we can look for the glue that binds them together */
   ast_channel_lock(c0);
   while (ast_channel_trylock(c1)) {
      ast_channel_unlock(c0);
      usleep(1);
      ast_channel_lock(c0);
   }

   if (!cap1 || !cap0) {
      goto done;
   }

   /* Grab glue that binds each channel to something using the RTP engine */
   if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
      ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
      goto done;
   }

   audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
   video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;

   audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
   video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;

   /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
   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)) {
      audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
   }
   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)) {
      audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
   }
   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) {
      glue0->get_codec(c0, cap0);
   }
   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) {
      glue1->get_codec(c1, cap1);
   }

   /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
   if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
      goto done;
   }

   /* Make sure we have matching codecs */
   if (!ast_format_cap_has_joint(cap0, cap1)) {
      goto done;
   }

   /* Bridge media early */
   if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
      ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
   }

   res = 0;

done:
   ast_channel_unlock(c0);
   ast_channel_unlock(c1);

   ast_format_cap_destroy(cap0);
   ast_format_cap_destroy(cap1);

   unref_instance_cond(&instance0);
   unref_instance_cond(&instance1);
   unref_instance_cond(&vinstance0);
   unref_instance_cond(&vinstance1);
   unref_instance_cond(&tinstance0);
   unref_instance_cond(&tinstance1);

   if (!res) {
      ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
   }

   return res;
}

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.

Parameters:

c_dst	Destination channel to copy to
c_src	Source channel to copy from

Since:: 1.8

Definition at line 1646 of file rtp_engine.c.

References ast_channel_lock_both, ast_channel_name(), ast_channel_tech(), ast_channel_unlock, ast_debug, ast_format_cap_alloc_nolock(), ast_format_cap_destroy(), ast_format_cap_has_joint(), 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, type, unref_instance_cond(), and ast_rtp_glue::update_peer.

Referenced by dial_exec_full(), and do_forward().

{
   struct ast_rtp_instance *instance_dst = NULL, *instance_src = NULL,
      *vinstance_dst = NULL, *vinstance_src = NULL,
      *tinstance_dst = NULL, *tinstance_src = NULL;
   struct ast_rtp_glue *glue_dst, *glue_src;
   enum ast_rtp_glue_result audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
   enum ast_rtp_glue_result audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
   struct ast_format_cap *cap_dst = ast_format_cap_alloc_nolock();
   struct ast_format_cap *cap_src = ast_format_cap_alloc_nolock();

   /* Lock both channels so we can look for the glue that binds them together */
   ast_channel_lock_both(c_dst, c_src);

   if (!cap_src || !cap_dst) {
      goto done;
   }

   /* Grab glue that binds each channel to something using the RTP engine */
   if (!(glue_dst = ast_rtp_instance_get_glue(ast_channel_tech(c_dst)->type)) || !(glue_src = ast_rtp_instance_get_glue(ast_channel_tech(c_src)->type))) {
      ast_debug(1, "Can't find native functions for channel '%s'\n", glue_dst ? ast_channel_name(c_src) : ast_channel_name(c_dst));
      goto done;
   }

   audio_glue_dst_res = glue_dst->get_rtp_info(c_dst, &instance_dst);
   video_glue_dst_res = glue_dst->get_vrtp_info ? glue_dst->get_vrtp_info(c_dst, &vinstance_dst) : AST_RTP_GLUE_RESULT_FORBID;

   audio_glue_src_res = glue_src->get_rtp_info(c_src, &instance_src);
   video_glue_src_res = glue_src->get_vrtp_info ? glue_src->get_vrtp_info(c_src, &vinstance_src) : AST_RTP_GLUE_RESULT_FORBID;

   /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
   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)) {
      audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
   }
   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)) {
      audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
   }
   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) {
      glue_dst->get_codec(c_dst, cap_dst);
   }
   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) {
      glue_src->get_codec(c_src, cap_src);
   }

   /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
   if (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE) {
      goto done;
   }

   /* Make sure we have matching codecs */
   if (!ast_format_cap_has_joint(cap_dst, cap_src)) {
      goto done;
   }

   ast_rtp_codecs_payloads_copy(&instance_src->codecs, &instance_dst->codecs, instance_dst);

   if (vinstance_dst && vinstance_src) {
      ast_rtp_codecs_payloads_copy(&vinstance_src->codecs, &vinstance_dst->codecs, vinstance_dst);
   }
   if (tinstance_dst && tinstance_src) {
      ast_rtp_codecs_payloads_copy(&tinstance_src->codecs, &tinstance_dst->codecs, tinstance_dst);
   }

   if (glue_dst->update_peer(c_dst, instance_src, vinstance_src, tinstance_src, cap_src, 0)) {
      ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
         ast_channel_name(c_dst), ast_channel_name(c_src));
   } else {
      ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
         ast_channel_name(c_dst), ast_channel_name(c_src));
   }

done:
   ast_channel_unlock(c_dst);
   ast_channel_unlock(c_src);

   ast_format_cap_destroy(cap_dst);
   ast_format_cap_destroy(cap_src);

   unref_instance_cond(&instance_dst);
   unref_instance_cond(&instance_src);
   unref_instance_cond(&vinstance_dst);
   unref_instance_cond(&vinstance_src);
   unref_instance_cond(&tinstance_dst);
   unref_instance_cond(&tinstance_src);
}

int ast_rtp_instance_fd	(	struct ast_rtp_instance *	instance,
		int	rtcp
	)

Get the file descriptor for an RTP session (or RTCP)

Parameters:

instance	Instance to get the file descriptor for
rtcp	Whether to retrieve the file descriptor for RTCP or not

Return values:

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.

Since:: 1.8

Definition at line 933 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_enable_video(), jingle_new(), mgcp_new(), process_sdp(), sip_new(), sip_set_rtp_peer(), skinny_new(), start_rtp(), and unistim_new().

{
   return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
}

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.

Parameters:

instance The RTP instance

Return values:

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'.

Since:: 1.8

Definition at line 2027 of file rtp_engine.c.

References ast_rtp_instance::glue.

{
   return instance->glue;
}

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.

Parameters:

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

Return values:

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.

Since:: 1.8

Definition at line 352 of file rtp_engine.c.

References ast_sockaddr_cmp(), ast_sockaddr_copy(), and ast_rtp_instance::local_address.

{
   if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
      ast_sockaddr_copy(address, &instance->local_address);
      return 1;
   }

   return 0;
}

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.

Parameters:

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

Return values:

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.

Since:: 1.8

Definition at line 369 of file rtp_engine.c.

References ast_sockaddr_cmp(), ast_sockaddr_copy(), and ast_rtp_instance::remote_address.

Referenced by sip_set_rtp_peer().

{
   if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) {
      ast_sockaddr_copy(address, &instance->remote_address);
      return 1;
   }

   return 0;
}

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.

Parameters:

instance The RTP instance that we want

Return values:

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.

Since:: 1.8

Definition at line 1641 of file rtp_engine.c.

References ast_rtp_instance::bridged.

Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and dialog_needdestroy().

{
   return instance->bridged;
}

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.

Parameters:

instance The RTP instance

Return values:

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'.

Note:: This will only return a channel while in a local or remote bridge.

Since:: 1.8

Definition at line 2032 of file rtp_engine.c.

References ast_rtp_instance::chan.

{
   return instance->chan;
}

struct ast_rtp_codecs* ast_rtp_instance_get_codecs ( struct ast_rtp_instance * instance ) [read]

Get the codecs structure of an RTP instance.

Parameters:

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'.

Since:: 1.8

Definition at line 416 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_add_payloads_to_description(), jingle_enable_video(), jingle_interpret_description(), jingle_new(), jingle_newcall(), multicast_rtp_write(), process_sdp(), process_sdp_a_audio(), send_start_rtp(), set_dtmf_payload(), set_peer_capabilities(), setup_rtp_connection(), start_rtp(), and transmit_response_with_sdp().

{
   return &instance->codecs;
}

void* ast_rtp_instance_get_data ( struct ast_rtp_instance * instance )

Get the data portion of an RTP instance.

Parameters:

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'.

Since:: 1.8

Definition at line 302 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_activate(), ast_rtp_alt_remote_address_set(), ast_rtp_change_source(), ast_rtp_destroy(), ast_rtp_dtls_active(), ast_rtp_dtls_get_connection(), ast_rtp_dtls_get_fingerprint(), ast_rtp_dtls_get_setup(), ast_rtp_dtls_reset(), ast_rtp_dtls_set_configuration(), ast_rtp_dtls_set_fingerprint(), ast_rtp_dtls_set_setup(), ast_rtp_dtls_stop(), 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(), dtls_srtp_renegotiate(), 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().

{
   return instance->data;
}

struct ast_rtp_engine_dtls* ast_rtp_instance_get_dtls ( struct ast_rtp_instance * instance ) [read]

Obtain a pointer to the DTLS support present on an RTP instance.

Parameters:

instance the RTP instance

Return values:

DTLS	support if present
NULL	if no DTLS support available

Definition at line 2102 of file rtp_engine.c.

References ast_rtp_engine::dtls, and ast_rtp_instance::engine.

Referenced by add_dtls_to_sdp(), dialog_initialize_dtls_srtp(), get_sdp_rtp_profile(), process_crypto(), process_sdp(), and process_sdp_a_dtls().

{
   return instance->engine->dtls;
}

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.

Parameters:

instance The RTP instance

Return values:

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'.

Since:: 1.8

Definition at line 2022 of file rtp_engine.c.

References ast_rtp_instance::engine.

{
   return instance->engine;
}

void* ast_rtp_instance_get_extended_prop	(	struct ast_rtp_instance *	instance,
		int	property
	)

Get the value of an RTP instance extended property.

Parameters:

instance	The RTP instance to get the extended property on
property	The extended property to get

Since:: 1.8

Definition at line 393 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::extended_prop_get.

{
   if (instance->engine->extended_prop_get) {
      return instance->engine->extended_prop_get(instance, property);
   }

   return NULL;
}

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.

Parameters:

type	Name of the glue we want

Return values:

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'.

Since:: 1.8

Definition at line 938 of file rtp_engine.c.

References AST_RWLIST_RDLOCK, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, 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().

{
   struct ast_rtp_glue *glue = NULL;

   AST_RWLIST_RDLOCK(&glues);

   AST_RWLIST_TRAVERSE(&glues, glue, entry) {
      if (!strcasecmp(glue->type, type)) {
         break;
      }
   }

   AST_RWLIST_UNLOCK(&glues);

   return glue;
}

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.

Parameters:

instance The RTP instance

Return values:

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'.

Since:: 1.8

Definition at line 2012 of file rtp_engine.c.

References ast_rtp_instance::holdtimeout.

Referenced by check_rtp_timeout().

{
   return instance->holdtimeout;
}

struct ast_rtp_engine_ice* ast_rtp_instance_get_ice ( struct ast_rtp_instance * instance ) [read]

Obtain a pointer to the ICE support present on an RTP instance.

Parameters:

instance the RTP instance

Return values:

ICE	support if present
NULL	if no ICE support available

Definition at line 2097 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::ice.

Referenced by add_ice_to_sdp(), dialog_initialize_rtp(), jingle_add_google_candidates_to_transport(), jingle_add_ice_udp_candidates_to_transport(), jingle_enable_video(), jingle_interpret_google_transport(), jingle_interpret_ice_udp_transport(), jingle_new(), jingle_outgoing_hook(), jingle_request(), process_sdp_a_ice(), and start_ice().

{
   return instance->engine->ice;
}

int ast_rtp_instance_get_keepalive ( struct ast_rtp_instance * instance )

Get the RTP keepalive interval.

Parameters:

instance The RTP instance

Return values:

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'.

Since:: 1.8

Definition at line 2017 of file rtp_engine.c.

References ast_rtp_instance::keepalive.

Referenced by check_rtp_timeout().

{
   return instance->keepalive;
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 363 of file rtp_engine.c.

References ast_sockaddr_copy(), and ast_rtp_instance::local_address.

Referenced by add_sdp(), apply_directmedia_acl(), 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(), send_start_rtp(), sip_acf_channel_read(), skinny_set_rtp_peer(), and unistim_set_rtp_peer().

{
   ast_sockaddr_copy(address, &instance->local_address);
}

int ast_rtp_instance_get_prop	(	struct ast_rtp_instance *	instance,
		enum ast_rtp_property	property
	)

Get the value of an RTP instance property.

Parameters:

instance	The RTP instance to get the property from
property	The property to get

Return values:

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.

Since:: 1.8

Definition at line 411 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().

{
   return instance->properties[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.

Parameters:

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

Return values:

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.

Since:: 1.8

Definition at line 1842 of file rtp_engine.c.

Referenced by ast_rtp_instance_set_stats_vars(), handle_request_bye(), sip_acf_channel_read(), and sip_hangup().

{
   struct ast_rtp_instance_stats stats = { 0, };
   enum ast_rtp_instance_stat stat;

   /* Determine what statistics we will need to retrieve based on field passed in */
   if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
      stat = AST_RTP_INSTANCE_STAT_ALL;
   } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
      stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
   } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
      stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
   } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
      stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
   } else {
      return NULL;
   }

   /* Attempt to actually retrieve the statistics we need to generate the quality string */
   if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
      return NULL;
   }

   /* Now actually fill the buffer with the good information */
   if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
      snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
          stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.rxjitter, stats.rxcount, stats.txjitter, stats.txcount, stats.txploss, stats.rtt);
   } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
      snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
          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));
   } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
      snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
          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));
   } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
      snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
   }

   return buf;
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 380 of file rtp_engine.c.

References ast_sockaddr_copy(), and ast_rtp_instance::remote_address.

Referenced by add_sdp(), apply_directmedia_acl(), 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(), dtls_srtp_check_pending(), find_rtp_port(), gtalk_update_stun(), handle_response_invite(), jingle_interpret_ice_udp_transport(), multicast_rtp_write(), multicast_send_control_packet(), oh323_set_rtp_peer(), process_cn_rfc3389(), process_dtmf_rfc2833(), process_sdp(), remote_bridge_loop(), send_start_rtp(), sip_acf_channel_read(), skinny_set_rtp_peer(), transmit_modify_with_sdp(), and unistim_set_rtp_peer().

{
   ast_sockaddr_copy(address, &instance->remote_address);
}

struct ast_srtp* ast_rtp_instance_get_srtp ( struct ast_rtp_instance * instance ) [read]

Obtain the SRTP instance associated with an RTP instance.

Parameters:

instance the RTP instance

Return values:

the	SRTP instance on success
NULL	if no SRTP instance exists

Definition at line 2083 of file rtp_engine.c.

References ast_rtp_instance::srtp.

Referenced by __rtp_recvfrom(), __rtp_sendto(), and ast_rtp_change_source().

{
   return instance->srtp;
}

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.

Parameters:

instance	Instance to get statistics on
stats	Structure to put results into
stat	What statistic(s) to retrieve

Return values:

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.

Since:: 1.8

Definition at line 1837 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(), dtls_srtp_setup(), sdp_crypto_activate(), show_chanstats_cb(), and sip_acf_channel_read().

{
   return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
}

int ast_rtp_instance_get_timeout ( struct ast_rtp_instance * instance )

Get the RTP timeout value.

Parameters:

instance The RTP instance

Return values:

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'.

Since:: 1.8

Definition at line 2007 of file rtp_engine.c.

References ast_rtp_instance::timeout.

Referenced by check_rtp_timeout().

{
   return instance->timeout;
}

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.

Parameters:

chan	Our own Asterisk channel
instance	The first RTP instance
peer	The peer Asterisk channel

Return values:

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.

Since:: 1.8

Definition at line 1926 of file rtp_engine.c.

References ao2_ref, ast_channel_lock, ast_channel_tech(), 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, type, and ast_rtp_glue::type.

Referenced by sip_setoption().

{
   struct ast_rtp_glue *glue;
   struct ast_rtp_instance *peer_instance = NULL;
   int res = -1;

   if (!instance->engine->make_compatible) {
      return -1;
   }

   ast_channel_lock(peer);

   if (!(glue = ast_rtp_instance_get_glue(ast_channel_tech(peer)->type))) {
      ast_channel_unlock(peer);
      return -1;
   }

   glue->get_rtp_info(peer, &peer_instance);
   if (!peer_instance) {
      ast_log(LOG_ERROR, "Unable to get_rtp_info for peer type %s\n", glue->type);
      ast_channel_unlock(peer);
      return -1;
   }
   if (peer_instance->engine != instance->engine) {
      ast_log(LOG_ERROR, "Peer engine mismatch for type %s\n", glue->type);
      ast_channel_unlock(peer);
      ao2_ref(peer_instance, -1);
      return -1;
   }

   res = instance->engine->make_compatible(chan, instance, peer, peer_instance);

   ast_channel_unlock(peer);

   ao2_ref(peer_instance, -1);
   peer_instance = NULL;

   return res;
}

struct ast_rtp_instance* ast_rtp_instance_new	(	const char *	engine_name,
		struct ast_sched_context *	sched,
		const struct ast_sockaddr *	sa,
		void *	data
	)		`[read]`

Create a new RTP instance.

Parameters:

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

Return values:

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.

Note:: The RTP engine does not have to use the address provided when creating an RTP instance. It may choose to use another depending on it's own configuration.

Since:: 1.8

Definition at line 236 of file rtp_engine.c.

References ao2_alloc, ao2_ref, ast_debug, ast_log(), ast_module_ref(), ast_module_unref(), ast_rtp_codecs_payloads_initialize(), AST_RWLIST_FIRST, AST_RWLIST_RDLOCK, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, ast_sockaddr_copy(), ast_strlen_zero(), ast_rtp_instance::codecs, ast_rtp_instance::engine, 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(), jingle_enable_video(), multicast_rtp_request(), and start_rtp().

{
   struct ast_sockaddr address = {{0,}};
   struct ast_rtp_instance *instance = NULL;
   struct ast_rtp_engine *engine = NULL;

   AST_RWLIST_RDLOCK(&engines);

   /* If an engine name was specified try to use it or otherwise use the first one registered */
   if (!ast_strlen_zero(engine_name)) {
      AST_RWLIST_TRAVERSE(&engines, engine, entry) {
         if (!strcmp(engine->name, engine_name)) {
            break;
         }
      }
   } else {
      engine = AST_RWLIST_FIRST(&engines);
   }

   /* If no engine was actually found bail out now */
   if (!engine) {
      ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
      AST_RWLIST_UNLOCK(&engines);
      return NULL;
   }

   /* Bump up the reference count before we return so the module can not be unloaded */
   ast_module_ref(engine->mod);

   AST_RWLIST_UNLOCK(&engines);

   /* Allocate a new RTP instance */
   if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
      ast_module_unref(engine->mod);
      return NULL;
   }
   instance->engine = engine;
   ast_sockaddr_copy(&instance->local_address, sa);
   ast_sockaddr_copy(&address, sa);

   if (ast_rtp_codecs_payloads_initialize(&instance->codecs)) {
      ao2_ref(instance, -1);
      return NULL;
   }

   ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);

   /* And pass it off to the engine to setup */
   if (instance->engine->new(instance, sched, &address, data)) {
      ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
      ao2_ref(instance, -1);
      return NULL;
   }

   ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);

   return instance;
}

struct ast_frame* ast_rtp_instance_read	(	struct ast_rtp_instance *	instance,
		int	rtcp
	)		`[read]`

Receive a frame over RTP.

Parameters:

instance	The RTP instance to receive frame on
rtcp	Whether to read in RTCP or not

Return values:

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.

Since:: 1.8

Definition at line 312 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::read.

Referenced by gtalk_rtp_read(), jingle_read(), jingle_rtp_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), sip_rtp_read(), skinny_rtp_read(), and unistim_rtp_read().

{
   return instance->engine->read(instance, rtcp);
}

int ast_rtp_instance_sendcng	(	struct ast_rtp_instance *	instance,
		int	level
	)

Send a comfort noise packet to the RTP instance.

Parameters:

instance	The RTP instance
level	Magnitude of the noise level

Return values:

0	Success
non-zero	Failure

Definition at line 2088 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::sendcng.

Referenced by check_rtp_timeout().

{
   if (instance->engine->sendcng) {
      return instance->engine->sendcng(instance, level);
   }

   return -1;
}

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.

Parameters:

instance	The RTP instance to change the address on
address	Address to set it to

Return values:

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.

Since:: 1.8

Definition at line 338 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().

{
   ast_sockaddr_copy(&instance->alt_remote_address, address);

   /* oink */

   if (instance->engine->alt_remote_address_set) {
      instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address);
   }

   return 0;
}

void ast_rtp_instance_set_data	(	struct ast_rtp_instance *	instance,
		void *	data
	)

Set the data portion of an RTP instance.

Parameters:

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.

Since:: 1.8

Definition at line 297 of file rtp_engine.c.

References ast_rtp_instance::data.

Referenced by ast_rtp_new(), and multicast_rtp_new().

{
   instance->data = data;
}

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.

Parameters:

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

Since:: 1.8

Definition at line 386 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::extended_prop_set.

{
   if (instance->engine->extended_prop_set) {
      instance->engine->extended_prop_set(instance, property, value);
   }
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 1997 of file rtp_engine.c.

References ast_rtp_instance::holdtimeout, and ast_rtp_instance::timeout.

Referenced by check_rtp_timeout(), and dialog_initialize_rtp().

{
   instance->holdtimeout = timeout;
}

void ast_rtp_instance_set_keepalive	(	struct ast_rtp_instance *	instance,
		int	timeout
	)

Set the RTP keepalive interval.

Parameters:

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.

Since:: 1.8

Definition at line 2002 of file rtp_engine.c.

References ast_rtp_instance::keepalive.

Referenced by dialog_initialize_rtp().

{
   instance->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.

Parameters:

instance	The RTP instance to change the address on
address	Address to set it to

Return values:

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.

Since:: 1.8

Definition at line 317 of file rtp_engine.c.

References ast_sockaddr_copy(), and ast_rtp_instance::local_address.

Referenced by ast_rtp_new().

{
   ast_sockaddr_copy(&instance->local_address, address);
   return 0;
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 402 of file rtp_engine.c.

References ast_rtp_instance::engine, ast_rtp_engine::prop_set, ast_rtp_instance::properties, and value.

Referenced by __oh323_rtp_create(), create_addr_from_peer(), dialog_initialize_rtp(), do_setnat(), gtalk_alloc(), handle_request_invite(), jingle_alloc(), jingle_enable_video(), oh323_rtp_read(), process_sdp(), sip_dtmfmode(), sip_set_rtp_peer(), and start_rtp().

{
   instance->properties[property] = value;

   if (instance->engine->prop_set) {
      instance->engine->prop_set(instance, property, value);
   }
}

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.

Parameters:

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

Return 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.

Since:: 1.8

Definition at line 921 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().

{
   return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
}

int ast_rtp_instance_set_read_format	(	struct ast_rtp_instance *	instance,
		struct ast_format *	format
	)

Request that the underlying RTP engine provide audio frames in a specific format.

Parameters:

instance	The RTP instance to change read format on
format	Format that frames are wanted in

Return values:

0	success
-1	failure

Example usage:

 struct ast_format tmp_fmt;
 ast_rtp_instance_set_read_format(instance, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0));

This requests that the RTP engine provide audio frames in the ULAW format.

Since:: 1.8

Definition at line 1916 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::set_read_format.

Referenced by sip_new(), and sip_setoption().

{
   return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
}

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.

Parameters:

instance	The RTP instance to change the address on
address	Address to set it to

Return values:

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.

Since:: 1.8

Definition at line 324 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(), jingle_interpret_ice_udp_transport(), multicast_rtp_request(), process_sdp(), setup_rtp_connection(), and start_rtp().

{
   ast_sockaddr_copy(&instance->remote_address, address);

   /* moo */

   if (instance->engine->remote_address_set) {
      instance->engine->remote_address_set(instance, &instance->remote_address);
   }

   return 0;
}

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.

Parameters:

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.

Since:: 1.8

Definition at line 1882 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().

{
   char quality_buf[AST_MAX_USER_FIELD], *quality;
   struct ast_channel *bridge = ast_bridged_channel(chan);

   if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
      pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
      if (bridge) {
         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
      }
   }

   if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
      pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
      if (bridge) {
         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
      }
   }

   if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
      pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
      if (bridge) {
         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
      }
   }

   if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
      pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
      if (bridge) {
         pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
      }
   }
}

void ast_rtp_instance_set_timeout	(	struct ast_rtp_instance *	instance,
		int	timeout
	)

Set the RTP timeout value.

Parameters:

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.

Since:: 1.8

Definition at line 1992 of file rtp_engine.c.

References ast_rtp_instance::timeout.

Referenced by check_rtp_timeout(), and dialog_initialize_rtp().

{
   instance->timeout = timeout;
}

int ast_rtp_instance_set_write_format	(	struct ast_rtp_instance *	instance,
		struct ast_format *	format
	)

Tell underlying RTP engine that audio frames will be provided in a specific format.

Parameters:

instance	The RTP instance to change write format on
format	Format that frames will be provided in

Return values:

0	success
-1	failure

Example usage:

 struct ast_format tmp_fmt;
 ast_rtp_instance_set_write_format(instance, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0));

This tells the underlying RTP engine that audio frames will be provided to it in ULAW format.

Since:: 1.8

Definition at line 1921 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::set_write_format.

Referenced by sip_new(), and sip_setoption().

{
   return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
}

void ast_rtp_instance_stop ( struct ast_rtp_instance * instance )

Stop an RTP instance.

Parameters:

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.

Since:: 1.8

Definition at line 926 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::stop.

Referenced by destroy_rtp(), jingle_session_destructor(), process_sdp(), setup_rtp_connection(), and stop_media_flows().

{
   if (instance->engine->stop) {
      instance->engine->stop(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.

Parameters:

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'.

Since:: 1.8

Definition at line 1983 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::stun_request.

Referenced by gtalk_update_stun(), jingle_interpret_google_transport(), and jingle_update_stun().

{
   if (instance->engine->stun_request) {
      instance->engine->stun_request(instance, suggestion, username);
   }
}

void ast_rtp_instance_update_source ( struct ast_rtp_instance * instance )

Indicate that the RTP marker bit should be set on an RTP stream.

Parameters:

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.

Since:: 1.8

Definition at line 907 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::update_source.

Referenced by jingle_indicate(), mgcp_indicate(), oh323_indicate(), sip_answer(), sip_indicate(), sip_write(), and skinny_indicate().

{
   if (instance->engine->update_source) {
      instance->engine->update_source(instance);
   }
}

int ast_rtp_instance_write	(	struct ast_rtp_instance *	instance,
		struct ast_frame *	frame
	)

Send a frame out over RTP.

Parameters:

instance	The RTP instance to send frame out on
frame	the frame to send out

Return values:

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.

Since:: 1.8

Definition at line 307 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().

{
   return instance->engine->write(instance, frame);
}

char* ast_rtp_lookup_mime_multiple2	(	struct ast_str *	buf,
		struct ast_format_cap *	ast_format_capability,
		int	rtp_capability,
		const int	asterisk_format,
		enum ast_rtp_options	options
	)

Convert formats into a string and put them into a buffer.

Parameters:

buf	Buffer to put the mime output into
ast_format_capability	Asterisk Formats we are looking up.
rtp_capability	RTP codes that we are looking up
asterisk_format	Non-zero if the ast_format_capability structure is to be used, 0 if rtp_capability is to be used
options	Additional options that may change the result

Return values:

non-NULL	success
NULL	failure

Example usage:

 char buf[256] = "";
 struct ast_format tmp_fmt;
 struct ast_format_cap *cap = ast_format_cap_alloc_nolock();
 ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0));
 ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_GSM, 0));
 char *mime = ast_rtp_lookup_mime_multiple2(&buf, sizeof(buf), cap, 0, 1, 0);
 ast_format_cap_destroy(cap);

This returns the mime values for ULAW and ALAW in the buffer pointed to by buf.

Since:: 1.8

Definition at line 838 of file rtp_engine.c.

References ast_format_cap_iter_end(), ast_format_cap_iter_next(), ast_format_cap_iter_start(), ast_rtp_lookup_mime_subtype2(), AST_RTP_MAX, ast_str_append(), ast_str_buffer(), and name.

Referenced by process_sdp().

{
   int found = 0;
   const char *name;
   if (!buf) {
      return NULL;
   }


   if (asterisk_format) {
      struct ast_format tmp_fmt;
      ast_format_cap_iter_start(ast_format_capability);
      while (!ast_format_cap_iter_next(ast_format_capability, &tmp_fmt)) {
         name = ast_rtp_lookup_mime_subtype2(asterisk_format, &tmp_fmt, 0, options);
         ast_str_append(&buf, 0, "%s|", name);
         found = 1;
      }
      ast_format_cap_iter_end(ast_format_capability);

   } else {
      int x;
      ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
      for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
         if (rtp_capability & x) {
            name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
            ast_str_append(&buf, 0, "%s|", name);
            found = 1;
         }
      }
   }

   ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");

   return ast_str_buffer(buf);
}

const char* ast_rtp_lookup_mime_subtype2	(	const int	asterisk_format,
		struct ast_format *	format,
		int	code,
		enum ast_rtp_options	options
	)

Retrieve mime subtype information on a payload.

Parameters:

asterisk_format	Non-zero to look up using Asterisk format
format	Asterisk format to look up
code	RTP code to look up
options	Additional options that may change the result

Return values:

Mime	subtype success
NULL	failure

Example usage:

 const char *subtype = ast_rtp_lookup_mime_subtype2(1, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0), 0, 0);

This looks up the mime subtype for the ULAW format.

Since:: 1.8

Definition at line 788 of file rtp_engine.c.

References ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, AST_FORMAT_G726_AAL2, ast_rtp_mime_types, AST_RTP_OPT_G726_NONSTANDARD, ast_rwlock_rdlock, ast_rwlock_unlock, ast_format::id, mime_types_len, ast_rtp_mime_type::payload_type, ast_rtp_payload_type::rtp_code, 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(), jingle_add_payloads_to_description(), transmit_connect(), transmit_connect_with_sdp(), transmit_modify_request(), and transmit_modify_with_sdp().

{
   int i;
   const char *res = "";

   ast_rwlock_rdlock(&mime_types_lock);
   for (i = 0; i < mime_types_len; i++) {
      if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
         (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
         if ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
            res = "G726-32";
            break;
         } else {
            res = ast_rtp_mime_types[i].subtype;
            break;
         }
      } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
         ast_rtp_mime_types[i].payload_type.rtp_code == code) {

         res = ast_rtp_mime_types[i].subtype;
         break;
      }
   }
   ast_rwlock_unlock(&mime_types_lock);

   return res;
}

unsigned int ast_rtp_lookup_sample_rate2	(	int	asterisk_format,
		struct ast_format *	format,
		int	code
	)

Get the sample rate associated with known RTP payload types.

Parameters:

asterisk_format	True if the value in format is to be used.
An	asterisk format
code	from AST_RTP list

Returns:: the sample rate if the format was found, zero if it was not found

Since:: 1.8

Definition at line 816 of file rtp_engine.c.

References ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, ast_rtp_mime_types, ast_rwlock_rdlock, ast_rwlock_unlock, mime_types_len, ast_rtp_mime_type::payload_type, ast_rtp_payload_type::rtp_code, and ast_rtp_mime_type::sample_rate.

Referenced by add_codec_to_sdp(), add_noncodec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), and jingle_add_payloads_to_description().

{
   unsigned int i;
   unsigned int res = 0;

   ast_rwlock_rdlock(&mime_types_lock);
   for (i = 0; i < mime_types_len; ++i) {
      if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
         (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
         res = ast_rtp_mime_types[i].sample_rate;
         break;
      } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
         ast_rtp_mime_types[i].payload_type.rtp_code == code) {
         res = ast_rtp_mime_types[i].sample_rate;
         break;
      }
   }
   ast_rwlock_unlock(&mime_types_lock);

   return res;
}

int ast_rtp_red_buffer	(	struct ast_rtp_instance *	instance,
		struct ast_frame *	frame
	)

Buffer a frame in an RTP instance for RED.

Parameters:

instance	The instance to buffer the frame on
frame	Frame that we want to buffer

Return values:

0	success
-1	failure

Since:: 1.8

Definition at line 1832 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::red_buffer.

Referenced by sip_write().

{
   return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
}

int ast_rtp_red_init	(	struct ast_rtp_instance *	instance,
		int	buffer_time,
		int *	payloads,
		int	generations
	)

Initialize RED support on an RTP instance.

Parameters:

instance	The instance to initialize RED support on
buffer_time	How long to buffer before sending
payloads	Payload values
generations	Number of generations

Return values:

0	success
-1	failure

Since:: 1.8

Definition at line 1827 of file rtp_engine.c.

References ast_rtp_instance::engine, and ast_rtp_engine::red_init.

Referenced by process_sdp().

{
   return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
}

static void instance_destructor ( void * obj ) [static]

Definition at line 207 of file rtp_engine.c.

References ast_debug, ast_module_unref(), ast_rtp_codecs_payloads_destroy(), ast_rtp_instance::codecs, ast_rtp_instance::data, ast_srtp_res::destroy, ast_rtp_engine::destroy, ast_rtp_instance::engine, ast_rtp_engine::mod, ast_rtp_engine::name, and ast_rtp_instance::srtp.

Referenced by ast_rtp_instance_new().

{
   struct ast_rtp_instance *instance = obj;

   /* Pass us off to the engine to destroy */
   if (instance->data && instance->engine->destroy(instance)) {
      ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
      return;
   }

   if (instance->srtp) {
      res_srtp->destroy(instance->srtp);
   }

   ast_rtp_codecs_payloads_destroy(&instance->codecs);

   /* Drop our engine reference */
   ast_module_unref(instance->engine->mod);

   ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
}

static enum ast_bridge_result local_bridge_loop	(	struct ast_channel *	c0,
		struct ast_channel *	c1,
		struct ast_rtp_instance *	instance0,
		struct ast_rtp_instance *	instance1,
		int	timeoutms,
		int	flags,
		struct ast_frame **	fo,
		struct ast_channel **	rc,
		void *	pvt0,
		void *	pvt1
	)		`[static]`

Definition at line 955 of file rtp_engine.c.

References AST_BRIDGE_COMPLETE, AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_BRIDGE_FAILED, AST_BRIDGE_FAILED_NOWARN, AST_BRIDGE_IGNORE_SIGS, AST_BRIDGE_RETRY, ast_channel_audiohooks(), ast_channel_connected_line_macro(), ast_channel_connected_line_sub(), ast_channel_framehooks(), ast_channel_hangupcause_hash_set(), ast_channel_masq(), ast_channel_masqr(), ast_channel_monitor(), ast_channel_name(), ast_channel_rawreadformat(), ast_channel_rawwriteformat(), ast_channel_redirecting_macro(), ast_channel_redirecting_sub(), ast_channel_tech_pvt(), ast_channel_unlock, ast_check_hangup(), AST_CONTROL_CONNECTED_LINE, AST_CONTROL_HOLD, AST_CONTROL_PVT_CAUSE_CODE, AST_CONTROL_REDIRECTING, AST_CONTROL_SRCUPDATE, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_UNHOLD, AST_CONTROL_UPDATE_RTP_PEER, AST_CONTROL_VIDUPDATE, ast_debug, ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_framehook_list_is_empty(), ast_frfree, ast_indicate_data(), ast_poll_channel_add(), ast_poll_channel_del(), ast_read(), ast_remaining_ms(), ast_tvnow(), ast_waitfor_n(), ast_write(), ast_rtp_instance::bridged, ast_frame::data, ast_frame::datalen, ast_rtp_instance::engine, ast_frame::frametype, ast_frame_subclass::integer, ast_rtp_engine::local_bridge, ast_frame::ptr, and ast_frame::subclass.

Referenced by ast_rtp_instance_bridge().

{
   enum ast_bridge_result res = AST_BRIDGE_FAILED;
   struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
   struct ast_frame *fr = NULL;
   struct timeval start;

   /* Start locally bridging both instances */
   if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
      ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c0), ast_channel_name(c1));
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED_NOWARN;
   }
   if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
      ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c1), ast_channel_name(c0));
      if (instance0->engine->local_bridge) {
         instance0->engine->local_bridge(instance0, NULL);
      }
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      return AST_BRIDGE_FAILED_NOWARN;
   }

   ast_channel_unlock(c0);
   ast_channel_unlock(c1);

   instance0->bridged = instance1;
   instance1->bridged = instance0;

   ast_poll_channel_add(c0, c1);

   /* Hop into a loop waiting for a frame from either channel */
   cs[0] = c0;
   cs[1] = c1;
   cs[2] = NULL;
   start = ast_tvnow();
   for (;;) {
      int ms;
      /* If the underlying formats have changed force this bridge to break */
      if ((ast_format_cmp(ast_channel_rawreadformat(c0), ast_channel_rawwriteformat(c1)) == AST_FORMAT_CMP_NOT_EQUAL) ||
         (ast_format_cmp(ast_channel_rawreadformat(c1), ast_channel_rawwriteformat(c0)) == AST_FORMAT_CMP_NOT_EQUAL)) {
         ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
         res = AST_BRIDGE_FAILED_NOWARN;
         break;
      }
      /* Check if anything changed */
      if ((ast_channel_tech_pvt(c0) != pvt0) ||
          (ast_channel_tech_pvt(c1) != pvt1) ||
          (ast_channel_masq(c0) || ast_channel_masqr(c0) || ast_channel_masq(c1) || ast_channel_masqr(c1)) ||
          (ast_channel_monitor(c0) || ast_channel_audiohooks(c0) || ast_channel_monitor(c1) || ast_channel_audiohooks(c1)) ||
          (!ast_framehook_list_is_empty(ast_channel_framehooks(c0)) || !ast_framehook_list_is_empty(ast_channel_framehooks(c1)))) {
         ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
         /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
         if ((ast_channel_masq(c0) || ast_channel_masqr(c0)) && (fr = ast_read(c0))) {
            ast_frfree(fr);
         }
         if ((ast_channel_masq(c1) || ast_channel_masqr(c1)) && (fr = ast_read(c1))) {
            ast_frfree(fr);
         }
         res = AST_BRIDGE_RETRY;
         break;
      }
      /* Wait on a channel to feed us a frame */
      ms = ast_remaining_ms(start, timeoutms);
      if (!(who = ast_waitfor_n(cs, 2, &ms))) {
         if (!ms) {
            res = AST_BRIDGE_RETRY;
            break;
         }
         ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
         if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
            break;
         }
         continue;
      }
      /* Read in frame from channel */
      fr = ast_read(who);
      other = (who == c0) ? c1 : c0;
      /* Depending on the frame we may need to break out of our bridge */
      if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
             ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
             ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
         /* Record received frame and who */
         *fo = fr;
         *rc = who;
         ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
         res = AST_BRIDGE_COMPLETE;
         break;
      } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
         if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
             (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
             (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) ||
             (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
            /* If we are going on hold, then break callback mode and P2P bridging */
            if (fr->subclass.integer == AST_CONTROL_HOLD) {
               if (instance0->engine->local_bridge) {
                  instance0->engine->local_bridge(instance0, NULL);
               }
               if (instance1->engine->local_bridge) {
                  instance1->engine->local_bridge(instance1, NULL);
               }
               instance0->bridged = NULL;
               instance1->bridged = NULL;
            } else if (fr->subclass.integer == AST_CONTROL_UNHOLD) {
               if (instance0->engine->local_bridge) {
                  instance0->engine->local_bridge(instance0, instance1);
               }
               if (instance1->engine->local_bridge) {
                  instance1->engine->local_bridge(instance1, instance0);
               }
               instance0->bridged = instance1;
               instance1->bridged = instance0;
            }
            /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
            if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
            if (ast_channel_connected_line_sub(who, other, fr, 1) &&
               ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
            if (ast_channel_redirecting_sub(who, other, fr, 1) &&
               ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
            ast_channel_hangupcause_hash_set(other, fr->data.ptr, fr->datalen);
            ast_frfree(fr);
         } else {
            *fo = fr;
            *rc = who;
            ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, ast_channel_name(who));
            res = AST_BRIDGE_COMPLETE;
            break;
         }
      } else {
         if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
             (fr->frametype == AST_FRAME_DTMF_END) ||
             (fr->frametype == AST_FRAME_VOICE) ||
             (fr->frametype == AST_FRAME_VIDEO) ||
             (fr->frametype == AST_FRAME_IMAGE) ||
             (fr->frametype == AST_FRAME_HTML) ||
             (fr->frametype == AST_FRAME_MODEM) ||
             (fr->frametype == AST_FRAME_TEXT)) {
            ast_write(other, fr);
         }

         ast_frfree(fr);
      }
      /* Swap priority */
      cs[2] = cs[0];
      cs[0] = cs[1];
      cs[1] = cs[2];
   }

   /* Stop locally bridging both instances */
   if (instance0->engine->local_bridge) {
      instance0->engine->local_bridge(instance0, NULL);
   }
   if (instance1->engine->local_bridge) {
      instance1->engine->local_bridge(instance1, NULL);
   }

   instance0->bridged = NULL;
   instance1->bridged = NULL;

   ast_poll_channel_del(c0, c1);

   return res;
}

static enum ast_bridge_result remote_bridge_loop	(	struct ast_channel *	c0,
		struct ast_channel *	c1,
		struct ast_rtp_instance *	instance0,
		struct ast_rtp_instance *	instance1,
		struct ast_rtp_instance *	vinstance0,
		struct ast_rtp_instance *	vinstance1,
		struct ast_rtp_instance *	tinstance0,
		struct ast_rtp_instance *	tinstance1,
		struct ast_rtp_glue *	glue0,
		struct ast_rtp_glue *	glue1,
		struct ast_format_cap *	cap0,
		struct ast_format_cap *	cap1,
		int	timeoutms,
		int	flags,
		struct ast_frame **	fo,
		struct ast_channel **	rc,
		void *	pvt0,
		void *	pvt1
	)		`[static]`

Definition at line 1134 of file rtp_engine.c.

References AST_BRIDGE_COMPLETE, AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, AST_BRIDGE_FAILED, AST_BRIDGE_IGNORE_SIGS, AST_BRIDGE_RETRY, ast_channel_audiohooks(), ast_channel_connected_line_macro(), ast_channel_connected_line_sub(), ast_channel_flags(), ast_channel_framehooks(), ast_channel_hangupcause_hash_set(), ast_channel_lock, ast_channel_masq(), ast_channel_masqr(), ast_channel_monitor(), ast_channel_name(), ast_channel_redirecting_macro(), ast_channel_redirecting_sub(), ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_unlock, ast_check_hangup(), AST_CONTROL_CONNECTED_LINE, AST_CONTROL_HOLD, AST_CONTROL_PVT_CAUSE_CODE, AST_CONTROL_REDIRECTING, AST_CONTROL_SRCUPDATE, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_UNHOLD, AST_CONTROL_UPDATE_RTP_PEER, AST_CONTROL_VIDUPDATE, ast_debug, AST_FLAG_ZOMBIE, ast_format_cap_append(), ast_format_cap_copy(), ast_format_cap_destroy(), ast_format_cap_dup(), ast_format_cap_identical(), ast_format_cap_remove_all(), AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_framehook_list_is_empty(), ast_frfree, ast_getformatname_multiple(), ast_indicate_data(), ast_log(), ast_poll_channel_add(), ast_poll_channel_del(), ast_read(), ast_remaining_ms(), ast_rtp_instance_get_glue(), ast_rtp_instance_get_remote_address(), ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_test_flag, ast_tvnow(), ast_waitfor_n(), ast_write(), ast_rtp_instance::bridged, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_rtp_glue::get_codec, ast_frame_subclass::integer, LOG_WARNING, ast_frame::ptr, ast_frame::subclass, tmp_buf, type, and ast_rtp_glue::update_peer.

Referenced by ast_rtp_instance_bridge().

{
   enum ast_bridge_result res = AST_BRIDGE_FAILED;
   struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
   struct ast_format_cap *oldcap0 = ast_format_cap_dup(cap0);
   struct ast_format_cap *oldcap1 = ast_format_cap_dup(cap1);
   struct ast_sockaddr ac1 = {{0,}}, vac1 = {{0,}}, tac1 = {{0,}}, ac0 = {{0,}}, vac0 = {{0,}}, tac0 = {{0,}};
   struct ast_sockaddr t1 = {{0,}}, vt1 = {{0,}}, tt1 = {{0,}}, t0 = {{0,}}, vt0 = {{0,}}, tt0 = {{0,}};
   struct ast_frame *fr = NULL;
   struct timeval start;

   if (!oldcap0 || !oldcap1) {
      ast_channel_unlock(c0);
      ast_channel_unlock(c1);
      goto remote_bridge_cleanup;
   }
   /* Test the first channel */
   if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0))) {
      ast_rtp_instance_get_remote_address(instance1, &ac1);
      if (vinstance1) {
         ast_rtp_instance_get_remote_address(vinstance1, &vac1);
      }
      if (tinstance1) {
         ast_rtp_instance_get_remote_address(tinstance1, &tac1);
      }
   } else {
      ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
   }

   /* Test the second channel */
   if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0))) {
      ast_rtp_instance_get_remote_address(instance0, &ac0);
      if (vinstance0) {
         ast_rtp_instance_get_remote_address(instance0, &vac0);
      }
      if (tinstance0) {
         ast_rtp_instance_get_remote_address(instance0, &tac0);
      }
   } else {
      ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c1), ast_channel_name(c0));
   }

   ast_channel_unlock(c0);
   ast_channel_unlock(c1);

   instance0->bridged = instance1;
   instance1->bridged = instance0;

   ast_poll_channel_add(c0, c1);

   /* Go into a loop handling any stray frames that may come in */
   cs[0] = c0;
   cs[1] = c1;
   cs[2] = NULL;
   start = ast_tvnow();
   for (;;) {
      int ms;
      /* Check if anything changed */
      if ((ast_channel_tech_pvt(c0) != pvt0) ||
          (ast_channel_tech_pvt(c1) != pvt1) ||
          (ast_channel_masq(c0) || ast_channel_masqr(c0) || ast_channel_masq(c1) || ast_channel_masqr(c1)) ||
          (ast_channel_monitor(c0) || ast_channel_audiohooks(c0) || ast_channel_monitor(c1) || ast_channel_audiohooks(c1)) ||
          (!ast_framehook_list_is_empty(ast_channel_framehooks(c0)) || !ast_framehook_list_is_empty(ast_channel_framehooks(c1)))) {
         ast_debug(1, "Oooh, something is weird, backing out\n");
         res = AST_BRIDGE_RETRY;
         break;
      }

      /* Check if they have changed their address */
      ast_rtp_instance_get_remote_address(instance1, &t1);
      if (vinstance1) {
         ast_rtp_instance_get_remote_address(vinstance1, &vt1);
      }
      if (tinstance1) {
         ast_rtp_instance_get_remote_address(tinstance1, &tt1);
      }
      ast_channel_lock(c1);
      if (glue1->get_codec && ast_channel_tech_pvt(c1)) {
         ast_format_cap_remove_all(cap1);
         glue1->get_codec(c1, cap1);
      }
      ast_channel_unlock(c1);

      ast_rtp_instance_get_remote_address(instance0, &t0);
      if (vinstance0) {
         ast_rtp_instance_get_remote_address(vinstance0, &vt0);
      }
      if (tinstance0) {
         ast_rtp_instance_get_remote_address(tinstance0, &tt0);
      }
      ast_channel_lock(c0);
      if (glue0->get_codec && ast_channel_tech_pvt(c0)) {
         ast_format_cap_remove_all(cap0);
         glue0->get_codec(c0, cap0);
      }
      ast_channel_unlock(c0);

      if ((ast_sockaddr_cmp(&t1, &ac1)) ||
          (vinstance1 && ast_sockaddr_cmp(&vt1, &vac1)) ||
          (tinstance1 && ast_sockaddr_cmp(&tt1, &tac1)) ||
          (!ast_format_cap_identical(cap1, oldcap1))) {
         char tmp_buf[512] = { 0, };
         ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
              ast_channel_name(c1), ast_sockaddr_stringify(&t1),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
         ast_debug(1, "Oooh, '%s' changed end vaddress to %s (format %s)\n",
              ast_channel_name(c1), ast_sockaddr_stringify(&vt1),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
         ast_debug(1, "Oooh, '%s' changed end taddress to %s (format %s)\n",
              ast_channel_name(c1), ast_sockaddr_stringify(&tt1),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              ast_channel_name(c1), ast_sockaddr_stringify(&ac1),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              ast_channel_name(c1), ast_sockaddr_stringify(&vac1),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              ast_channel_name(c1), ast_sockaddr_stringify(&tac1),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
         if (glue0->update_peer(c0,
                      ast_sockaddr_isnull(&t1)  ? NULL : instance1,
                      ast_sockaddr_isnull(&vt1) ? NULL : vinstance1,
                      ast_sockaddr_isnull(&tt1) ? NULL : tinstance1,
                      cap1, 0)) {
            ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
         }
         ast_sockaddr_copy(&ac1, &t1);
         ast_sockaddr_copy(&vac1, &vt1);
         ast_sockaddr_copy(&tac1, &tt1);
         ast_format_cap_copy(oldcap1, cap1);
      }
      if ((ast_sockaddr_cmp(&t0, &ac0)) ||
          (vinstance0 && ast_sockaddr_cmp(&vt0, &vac0)) ||
          (tinstance0 && ast_sockaddr_cmp(&tt0, &tac0)) ||
          (!ast_format_cap_identical(cap0, oldcap0))) {
         char tmp_buf[512] = { 0, };
         ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
              ast_channel_name(c0), ast_sockaddr_stringify(&t0),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap0));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              ast_channel_name(c0), ast_sockaddr_stringify(&ac0),
              ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap0));
         if (glue1->update_peer(c1, t0.len ? instance0 : NULL,
                  vt0.len ? vinstance0 : NULL,
                  tt0.len ? tinstance0 : NULL,
                  cap0, 0)) {
            ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", ast_channel_name(c1), ast_channel_name(c0));
         }
         ast_sockaddr_copy(&ac0, &t0);
         ast_sockaddr_copy(&vac0, &vt0);
         ast_sockaddr_copy(&tac0, &tt0);
         ast_format_cap_copy(oldcap0, cap0);
      }

      ms = ast_remaining_ms(start, timeoutms);
      /* Wait for frame to come in on the channels */
      if (!(who = ast_waitfor_n(cs, 2, &ms))) {
         if (!ms) {
            res = AST_BRIDGE_RETRY;
            break;
         }
         ast_debug(1, "Ooh, empty read...\n");
         if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
            break;
         }
         continue;
      }
      fr = ast_read(who);
      other = (who == c0) ? c1 : c0;
      if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
             (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
              ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
         /* Break out of bridge */
         *fo = fr;
         *rc = who;
         ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
         res = AST_BRIDGE_COMPLETE;
         break;
      } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
         if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
             (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
             (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) ||
            (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
            if (fr->subclass.integer == AST_CONTROL_HOLD) {
               /* If we someone went on hold we want the other side to reinvite back to us */
               if (who == c0) {
                  glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
               } else {
                  glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
               }
            } else if (fr->subclass.integer == AST_CONTROL_UNHOLD ||
               fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER) {
               /* If they went off hold they should go back to being direct, or if we have
                * been told to force a peer update, go ahead and do it. */
               if (who == c0) {
                  glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0);
               } else {
                  glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0);
               }
            }
            /* Update local address information */
            ast_rtp_instance_get_remote_address(instance0, &t0);
            ast_sockaddr_copy(&ac0, &t0);
            ast_rtp_instance_get_remote_address(instance1, &t1);
            ast_sockaddr_copy(&ac1, &t1);
            /* Update codec information */
            ast_channel_lock(c0);
            if (glue0->get_codec && ast_channel_tech_pvt(c0)) {
               ast_format_cap_remove_all(cap0);
               ast_format_cap_remove_all(oldcap0);
               glue0->get_codec(c0, cap0);
               ast_format_cap_append(oldcap0, cap0);

            }
            ast_channel_unlock(c0);
            ast_channel_lock(c1);
            if (glue1->get_codec && ast_channel_tech_pvt(c1)) {
               ast_format_cap_remove_all(cap1);
               ast_format_cap_remove_all(oldcap1);
               glue1->get_codec(c1, cap1);
               ast_format_cap_append(oldcap1, cap1);
            }
            ast_channel_unlock(c1);
            /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
            if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
            if (ast_channel_connected_line_sub(who, other, fr, 1) &&
               ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
            if (ast_channel_redirecting_sub(who, other, fr, 1) &&
               ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
            ast_channel_hangupcause_hash_set(other, fr->data.ptr, fr->datalen);
            ast_frfree(fr);
         } else {
            *fo = fr;
            *rc = who;
            ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, ast_channel_name(who));
            res = AST_BRIDGE_COMPLETE;
            break;
         }
      } else {
         if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
             (fr->frametype == AST_FRAME_DTMF_END) ||
             (fr->frametype == AST_FRAME_VOICE) ||
             (fr->frametype == AST_FRAME_VIDEO) ||
             (fr->frametype == AST_FRAME_IMAGE) ||
             (fr->frametype == AST_FRAME_HTML) ||
             (fr->frametype == AST_FRAME_MODEM) ||
             (fr->frametype == AST_FRAME_TEXT)) {
            ast_write(other, fr);
         }
         ast_frfree(fr);
      }
      /* Swap priority */
      cs[2] = cs[0];
      cs[0] = cs[1];
      cs[1] = cs[2];
   }

   if (ast_test_flag(ast_channel_flags(c0), AST_FLAG_ZOMBIE)) {
      ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", ast_channel_name(c0));
   } else if (ast_channel_tech_pvt(c0) != pvt0) {
      ast_debug(1, "Channel c0->'%s' pvt changed, in bridge with c1->'%s'\n", ast_channel_name(c0), ast_channel_name(c1));
   } else if (glue0 != ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) {
      ast_debug(1, "Channel c0->'%s' technology changed, in bridge with c1->'%s'\n", ast_channel_name(c0), ast_channel_name(c1));
   } else if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
      ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", ast_channel_name(c0));
   }
   if (ast_test_flag(ast_channel_flags(c1), AST_FLAG_ZOMBIE)) {
      ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", ast_channel_name(c1));
   } else if (ast_channel_tech_pvt(c1) != pvt1) {
      ast_debug(1, "Channel c1->'%s' pvt changed, in bridge with c0->'%s'\n", ast_channel_name(c1), ast_channel_name(c0));
   } else if (glue1 != ast_rtp_instance_get_glue(ast_channel_tech(c1)->type)) {
      ast_debug(1, "Channel c1->'%s' technology changed, in bridge with c0->'%s'\n", ast_channel_name(c1), ast_channel_name(c0));
   } else if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
      ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", ast_channel_name(c1));
   }

   instance0->bridged = NULL;
   instance1->bridged = NULL;

   ast_poll_channel_del(c0, c1);

remote_bridge_cleanup:
   ast_format_cap_destroy(oldcap0);
   ast_format_cap_destroy(oldcap1);

   return res;
}

static int rtp_payload_type_add_ast	(	void *	obj,
		void *	arg,
		int	flags
	)		`[static]`

Definition at line 693 of file rtp_engine.c.

References ast_format_cap_add(), ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::format, and type.

Referenced by ast_rtp_codecs_payload_formats().

{
   struct ast_rtp_payload_type *type = obj;
   struct ast_format_cap *astformats = arg;

   if (type->asterisk_format) {
      ast_format_cap_add(astformats, &type->format);
   }

   return 0;
}

static int rtp_payload_type_add_nonast	(	void *	obj,
		void *	arg,
		int	flags
	)		`[static]`

Definition at line 705 of file rtp_engine.c.

References ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::rtp_code, and type.

Referenced by ast_rtp_codecs_payload_formats().

{
   struct ast_rtp_payload_type *type = obj;
   int *nonastformats = arg;

   if (!type->asterisk_format) {
      *nonastformats |= type->rtp_code;
   }

   return 0;
}

static int rtp_payload_type_cmp	(	void *	obj,
		void *	arg,
		int	flags
	)		`[static]`

Definition at line 429 of file rtp_engine.c.

References CMP_MATCH, CMP_STOP, OBJ_KEY, and ast_rtp_payload_type::payload.

Referenced by ast_rtp_codecs_payloads_initialize().

{
   struct ast_rtp_payload_type *type1 = obj, *type2 = arg;
   const int *payload = arg;

   return (type1->payload == (OBJ_KEY ? *payload : type2->payload)) ? CMP_MATCH | CMP_STOP : 0;
}

static int rtp_payload_type_find_format	(	void *	obj,
		void *	arg,
		int	flags
	)		`[static]`

Definition at line 726 of file rtp_engine.c.

References ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, ast_rtp_payload_type::asterisk_format, CMP_MATCH, CMP_STOP, format, ast_rtp_payload_type::format, and type.

Referenced by ast_rtp_codecs_payload_code().

{
   struct ast_rtp_payload_type *type = obj;
   struct ast_format *format = arg;

   return (type->asterisk_format && (ast_format_cmp(&type->format, format) != AST_FORMAT_CMP_NOT_EQUAL)) ? CMP_MATCH | CMP_STOP : 0;
}

static int rtp_payload_type_find_nonast_format	(	void *	obj,
		void *	arg,
		int	flags
	)		`[static]`

Definition at line 734 of file rtp_engine.c.

References ast_rtp_payload_type::asterisk_format, CMP_MATCH, CMP_STOP, ast_rtp_payload_type::rtp_code, and type.

Referenced by ast_rtp_codecs_payload_code().

{
   struct ast_rtp_payload_type *type = obj;
   int *rtp_code = arg;

   return ((!type->asterisk_format && (type->rtp_code == *rtp_code)) ? CMP_MATCH | CMP_STOP : 0);
}

static int rtp_payload_type_hash	(	const void *	obj,
		const int	flags
	)		`[static]`

Definition at line 421 of file rtp_engine.c.

References OBJ_KEY, ast_rtp_payload_type::payload, and type.

Referenced by ast_rtp_codecs_payloads_initialize().

{
   const struct ast_rtp_payload_type *type = obj;
   const int *payload = obj;

   return (flags & OBJ_KEY) ? *payload : type->payload;
}

static void set_next_mime_type	(	const struct ast_format *	format,
		int	rtp_code,
		char *	type,
		char *	subtype,
		unsigned int	sample_rate
	)		`[static]`

Definition at line 2170 of file rtp_engine.c.

References ARRAY_LEN, ast_format_copy(), ast_rtp_mime_types, ast_rwlock_unlock, ast_rwlock_wrlock, ast_rtp_payload_type::asterisk_format, mime_types_len, ast_rtp_mime_type::payload_type, ast_rtp_payload_type::rtp_code, ast_rtp_mime_type::sample_rate, ast_rtp_mime_type::subtype, type, and ast_rtp_mime_type::type.

Referenced by ast_rtp_engine_init(), and ast_rtp_engine_load_format().

{
   int x = mime_types_len;
   if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
      return;
   }

   ast_rwlock_wrlock(&mime_types_lock);
   if (format) {
      ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
      ast_format_copy(&ast_rtp_mime_types[x].payload_type.format, format);
   } else {
      ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
   }
   ast_rtp_mime_types[x].type = type;
   ast_rtp_mime_types[x].subtype = subtype;
   ast_rtp_mime_types[x].sample_rate = sample_rate;
   mime_types_len++;
   ast_rwlock_unlock(&mime_types_lock);
}

static void unref_instance_cond ( struct ast_rtp_instance ** instance ) [static]

Conditionally unref an rtp instance.

Definition at line 1457 of file rtp_engine.c.

References ao2_ref.

Referenced by ast_rtp_instance_bridge(), ast_rtp_instance_early_bridge(), and ast_rtp_instance_early_bridge_make_compatible().

{
   if (*instance) {
      ao2_ref(*instance, -1);
      *instance = NULL;
   }
}

Variable Documentation

struct ast_rtp_mime_type ast_rtp_mime_types[128] [static]

Referenced by ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_engine_unload_format(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), and set_next_mime_type().

struct engines engines [static]

struct glues glues [static]

int mime_types_len = 0 [static]

Definition at line 99 of file rtp_engine.c.

Referenced by ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_engine_unload_format(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), and set_next_mime_type().

ast_rwlock_t mime_types_lock [static]

Definition at line 98 of file rtp_engine.c.

struct ast_srtp_res* res_srtp = NULL

Definition at line 49 of file rtp_engine.c.

struct ast_srtp_policy_res* res_srtp_policy = NULL

Definition at line 50 of file rtp_engine.c.

Referenced by ast_rtp_engine_srtp_is_registered().

struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] [static]

Mapping between Asterisk codecs and rtp payload types.

Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s: also, our own choices for dynamic payload types. This is our master table for transmission

See http://www.iana.org/assignments/rtp-parameters for a list of assigned values

Definition at line 111 of file rtp_engine.c.

Referenced by add_static_payload(), ast_rtp_codecs_payload_code(), ast_rtp_codecs_payload_lookup(), ast_rtp_codecs_payloads_default(), ast_rtp_codecs_payloads_set_m_type(), and ast_rtp_engine_unload_format().

ast_rwlock_t static_RTP_PT_lock [static]

Definition at line 112 of file rtp_engine.c.

Asterisk developer's documentation

Data Structures

Functions

Variables

Detailed Description

Function Documentation

Variable Documentation