rtp_engine.h File Reference

00365                              {
00366         AST_RTP_DTLS_CONNECTION_NEW,      /*!< Endpoint wants to use a new connection */
00367    AST_RTP_DTLS_CONNECTION_EXISTING, /*!< Endpoint wishes to use existing connection */
00368 };

enum ast_rtp_dtls_hash

DTLS fingerprint hashes.

Enumerator:

AST_RTP_DTLS_HASH_SHA1

SHA-1 fingerprint hash

Definition at line 371 of file rtp_engine.h.

00371                        {
00372    AST_RTP_DTLS_HASH_SHA1, /*!< SHA-1 fingerprint hash */
00373 };

enum ast_rtp_dtls_setup

DTLS setup types.

Enumerator:

AST_RTP_DTLS_SETUP_ACTIVE	Endpoint is willing to inititate connections
AST_RTP_DTLS_SETUP_PASSIVE	Endpoint is willing to accept connections
AST_RTP_DTLS_SETUP_ACTPASS	Endpoint is willing to both accept and initiate connections
AST_RTP_DTLS_SETUP_HOLDCONN	Endpoint does not want the connection to be established right now

Definition at line 357 of file rtp_engine.h.

00357                         {
00358    AST_RTP_DTLS_SETUP_ACTIVE,   /*!< Endpoint is willing to inititate connections */
00359    AST_RTP_DTLS_SETUP_PASSIVE,  /*!< Endpoint is willing to accept connections */
00360    AST_RTP_DTLS_SETUP_ACTPASS,  /*!< Endpoint is willing to both accept and initiate connections */
00361    AST_RTP_DTLS_SETUP_HOLDCONN, /*!< Endpoint does not want the connection to be established right now */
00362 };

enum ast_rtp_dtmf_mode

RTP DTMF Modes

Enumerator:

AST_RTP_DTMF_MODE_NONE	No DTMF is being carried over the RTP stream
AST_RTP_DTMF_MODE_RFC2833	DTMF is being carried out of band using RFC2833
AST_RTP_DTMF_MODE_INBAND	DTMF is being carried inband over the RTP stream

Definition at line 119 of file rtp_engine.h.

00119                        {
00120    /*! No DTMF is being carried over the RTP stream */
00121    AST_RTP_DTMF_MODE_NONE = 0,
00122    /*! DTMF is being carried out of band using RFC2833 */
00123    AST_RTP_DTMF_MODE_RFC2833,
00124    /*! DTMF is being carried inband over the RTP stream */
00125    AST_RTP_DTMF_MODE_INBAND,
00126 };

enum ast_rtp_glue_result

Result codes when RTP glue is queried for information

Enumerator:

AST_RTP_GLUE_RESULT_FORBID	No remote or local bridging is permitted
AST_RTP_GLUE_RESULT_REMOTE	Move RTP stream to be remote between devices directly
AST_RTP_GLUE_RESULT_LOCAL	Perform RTP engine level bridging if possible

Definition at line 129 of file rtp_engine.h.

00129                          {
00130    /*! No remote or local bridging is permitted */
00131    AST_RTP_GLUE_RESULT_FORBID = 0,
00132    /*! Move RTP stream to be remote between devices directly */
00133    AST_RTP_GLUE_RESULT_REMOTE,
00134    /*! Perform RTP engine level bridging if possible */
00135    AST_RTP_GLUE_RESULT_LOCAL,
00136 };

enum ast_rtp_ice_candidate_type

ICE candidate types.

Enumerator:

AST_RTP_ICE_CANDIDATE_TYPE_HOST	ICE host candidate. A host candidate represents the actual local transport address in the host.
AST_RTP_ICE_CANDIDATE_TYPE_SRFLX	ICE server reflexive candidate, which represents the public mapped address of the local address.
AST_RTP_ICE_CANDIDATE_TYPE_RELAYED	ICE relayed candidate, which represents the address allocated in TURN server.

Definition at line 319 of file rtp_engine.h.

00319                                 {
00320    AST_RTP_ICE_CANDIDATE_TYPE_HOST,    /*!< ICE host candidate. A host candidate represents the actual local transport address in the host. */
00321    AST_RTP_ICE_CANDIDATE_TYPE_SRFLX,   /*!< ICE server reflexive candidate, which represents the public mapped address of the local address. */
00322    AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, /*!< ICE relayed candidate, which represents the address allocated in TURN server. */
00323 };

enum ast_rtp_instance_stat

Statistics that can be retrieved from an RTP instance

Enumerator:

AST_RTP_INSTANCE_STAT_ALL	Retrieve all statistics
AST_RTP_INSTANCE_STAT_TXCOUNT	Retrieve number of packets transmitted
AST_RTP_INSTANCE_STAT_RXCOUNT	Retrieve number of packets received
AST_RTP_INSTANCE_STAT_COMBINED_LOSS	Retrieve ALL statistics relating to packet loss
AST_RTP_INSTANCE_STAT_TXPLOSS	Retrieve number of packets lost for transmitting
AST_RTP_INSTANCE_STAT_RXPLOSS	Retrieve number of packets lost for receiving
AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS	Retrieve maximum number of packets lost on remote side
AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS	Retrieve minimum number of packets lost on remote side
AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS	Retrieve average number of packets lost on remote side
AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS	Retrieve standard deviation of packets lost on remote side
AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS	Retrieve maximum number of packets lost on local side
AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS	Retrieve minimum number of packets lost on local side
AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS	Retrieve average number of packets lost on local side
AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS	Retrieve standard deviation of packets lost on local side
AST_RTP_INSTANCE_STAT_COMBINED_JITTER	Retrieve ALL statistics relating to jitter
AST_RTP_INSTANCE_STAT_TXJITTER	Retrieve jitter on transmitted packets
AST_RTP_INSTANCE_STAT_RXJITTER	Retrieve jitter on received packets
AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER	Retrieve maximum jitter on remote side
AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER	Retrieve minimum jitter on remote side
AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER	Retrieve average jitter on remote side
AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER	Retrieve standard deviation jitter on remote side
AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER	Retrieve maximum jitter on local side
AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER	Retrieve minimum jitter on local side
AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER	Retrieve average jitter on local side
AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER	Retrieve standard deviation jitter on local side
AST_RTP_INSTANCE_STAT_COMBINED_RTT	Retrieve ALL statistics relating to round trip time
AST_RTP_INSTANCE_STAT_RTT	Retrieve round trip time
AST_RTP_INSTANCE_STAT_MAX_RTT	Retrieve maximum round trip time
AST_RTP_INSTANCE_STAT_MIN_RTT	Retrieve minimum round trip time
AST_RTP_INSTANCE_STAT_NORMDEVRTT	Retrieve average round trip time
AST_RTP_INSTANCE_STAT_STDEVRTT	Retrieve standard deviation round trip time
AST_RTP_INSTANCE_STAT_LOCAL_SSRC	Retrieve local SSRC
AST_RTP_INSTANCE_STAT_REMOTE_SSRC	Retrieve remote SSRC

Definition at line 151 of file rtp_engine.h.

00151                            {
00152    /*! Retrieve all statistics */
00153    AST_RTP_INSTANCE_STAT_ALL = 0,
00154    /*! Retrieve number of packets transmitted */
00155    AST_RTP_INSTANCE_STAT_TXCOUNT,
00156    /*! Retrieve number of packets received */
00157    AST_RTP_INSTANCE_STAT_RXCOUNT,
00158    /*! Retrieve ALL statistics relating to packet loss */
00159    AST_RTP_INSTANCE_STAT_COMBINED_LOSS,
00160    /*! Retrieve number of packets lost for transmitting */
00161    AST_RTP_INSTANCE_STAT_TXPLOSS,
00162    /*! Retrieve number of packets lost for receiving */
00163    AST_RTP_INSTANCE_STAT_RXPLOSS,
00164    /*! Retrieve maximum number of packets lost on remote side */
00165    AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS,
00166    /*! Retrieve minimum number of packets lost on remote side */
00167    AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS,
00168    /*! Retrieve average number of packets lost on remote side */
00169    AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS,
00170    /*! Retrieve standard deviation of packets lost on remote side */
00171    AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS,
00172    /*! Retrieve maximum number of packets lost on local side */
00173    AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS,
00174    /*! Retrieve minimum number of packets lost on local side */
00175    AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS,
00176    /*! Retrieve average number of packets lost on local side */
00177    AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS,
00178    /*! Retrieve standard deviation of packets lost on local side */
00179    AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS,
00180    /*! Retrieve ALL statistics relating to jitter */
00181    AST_RTP_INSTANCE_STAT_COMBINED_JITTER,
00182    /*! Retrieve jitter on transmitted packets */
00183    AST_RTP_INSTANCE_STAT_TXJITTER,
00184    /*! Retrieve jitter on received packets */
00185    AST_RTP_INSTANCE_STAT_RXJITTER,
00186    /*! Retrieve maximum jitter on remote side */
00187    AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER,
00188    /*! Retrieve minimum jitter on remote side */
00189    AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER,
00190    /*! Retrieve average jitter on remote side */
00191    AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER,
00192    /*! Retrieve standard deviation jitter on remote side */
00193    AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER,
00194    /*! Retrieve maximum jitter on local side */
00195    AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER,
00196    /*! Retrieve minimum jitter on local side */
00197    AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER,
00198    /*! Retrieve average jitter on local side */
00199    AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER,
00200    /*! Retrieve standard deviation jitter on local side */
00201    AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER,
00202    /*! Retrieve ALL statistics relating to round trip time */
00203    AST_RTP_INSTANCE_STAT_COMBINED_RTT,
00204    /*! Retrieve round trip time */
00205    AST_RTP_INSTANCE_STAT_RTT,
00206    /*! Retrieve maximum round trip time */
00207    AST_RTP_INSTANCE_STAT_MAX_RTT,
00208    /*! Retrieve minimum round trip time */
00209    AST_RTP_INSTANCE_STAT_MIN_RTT,
00210    /*! Retrieve average round trip time */
00211    AST_RTP_INSTANCE_STAT_NORMDEVRTT,
00212    /*! Retrieve standard deviation round trip time */
00213    AST_RTP_INSTANCE_STAT_STDEVRTT,
00214    /*! Retrieve local SSRC */
00215    AST_RTP_INSTANCE_STAT_LOCAL_SSRC,
00216    /*! Retrieve remote SSRC */
00217    AST_RTP_INSTANCE_STAT_REMOTE_SSRC,
00218 };

enum ast_rtp_instance_stat_field

Field statistics that can be retrieved from an RTP instance

Enumerator:

AST_RTP_INSTANCE_STAT_FIELD_QUALITY	Retrieve quality information
AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER	Retrieve quality information about jitter
AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS	Retrieve quality information about packet loss
AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT	Retrieve quality information about round trip time

Definition at line 139 of file rtp_engine.h.

00139                                  {
00140    /*! Retrieve quality information */
00141    AST_RTP_INSTANCE_STAT_FIELD_QUALITY = 0,
00142    /*! Retrieve quality information about jitter */
00143    AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER,
00144    /*! Retrieve quality information about packet loss */
00145    AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS,
00146    /*! Retrieve quality information about round trip time */
00147    AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT,
00148 };

enum ast_rtp_options

Additional RTP options

Enumerator:

AST_RTP_OPT_G726_NONSTANDARD

Remote side is using non-standard G.726

Definition at line 113 of file rtp_engine.h.

00113                      {
00114    /*! Remote side is using non-standard G.726 */
00115    AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
00116 };

enum ast_rtp_property

RTP Properties that can be set on an RTP instance

Enumerator:

AST_RTP_PROPERTY_NAT	Enable symmetric RTP support
AST_RTP_PROPERTY_DTMF	RTP instance will be carrying DTMF (using RFC2833)
AST_RTP_PROPERTY_DTMF_COMPENSATE	Expect unreliable DTMF from remote party
AST_RTP_PROPERTY_STUN	Enable STUN support
AST_RTP_PROPERTY_RTCP	Enable RTCP support
AST_RTP_PROPERTY_MAX	Maximum number of RTP properties supported. Note: THIS MUST BE THE LAST ENTRY IN THIS ENUM.

Definition at line 92 of file rtp_engine.h.

00092                       {
00093    /*! Enable symmetric RTP support */
00094    AST_RTP_PROPERTY_NAT = 0,
00095    /*! RTP instance will be carrying DTMF (using RFC2833) */
00096    AST_RTP_PROPERTY_DTMF,
00097    /*! Expect unreliable DTMF from remote party */
00098    AST_RTP_PROPERTY_DTMF_COMPENSATE,
00099    /*! Enable STUN support */
00100    AST_RTP_PROPERTY_STUN,
00101    /*! Enable RTCP support */
00102    AST_RTP_PROPERTY_RTCP,
00103 
00104    /*!
00105     * \brief Maximum number of RTP properties supported
00106     *
00107     * \note THIS MUST BE THE LAST ENTRY IN THIS ENUM.
00108     */
00109    AST_RTP_PROPERTY_MAX,
00110 };

Function Documentation

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 765 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().

00766 {
00767    struct ast_rtp_payload_type *type;
00768    int res = -1;
00769 
00770    /* Search the payload type in the codecs passed */
00771    if ((type = ao2_find(codecs->payloads, &code, OBJ_NOLOCK | OBJ_KEY)))
00772    {
00773       res = type->payload;
00774       ao2_ref(type, -1);
00775       return res;
00776    }
00777 
00778    return res;
00779 }

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, ast_rtp_payload_type::format, format, OBJ_KEY, OBJ_NOLOCK, ast_rtp_codecs::payloads, and type.

Referenced by process_sdp_a_audio(), and process_sdp_a_video().

00674 {
00675    struct ast_rtp_payload_type *type;
00676    struct ast_format *format;
00677 
00678    if (payload < 0 || payload >= AST_RTP_MAX_PT) {
00679       return NULL;
00680    }
00681 
00682    if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
00683       return NULL;
00684    }
00685 
00686    format = type->asterisk_format ? &type->format : NULL;
00687 
00688    ao2_ref(type, -1);
00689 
00690    return format;
00691 }

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 866 of file rtp_engine.c.

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

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

00867 {
00868    codecs->pref = *prefs;
00869 
00870    if (instance && instance->engine->packetization_set) {
00871       instance->engine->packetization_set(instance, &instance->codecs.pref);
00872    }
00873 }

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 734 of file rtp_engine.c.

References ao2_callback, ao2_find, ao2_ref, ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, AST_RTP_MAX_PT, ast_rwlock_rdlock, ast_rwlock_unlock, OBJ_KEY, OBJ_NOLOCK, ast_rtp_payload_type::payload, ast_rtp_codecs::payloads, ast_rtp_payload_type::rtp_code, rtp_payload_type_find_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().

00735 {
00736    struct ast_rtp_payload_type *type;
00737    int i, res = -1;
00738 
00739    if (asterisk_format && format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_format, (void*)format))) {
00740       res = type->payload;
00741       ao2_ref(type, -1);
00742       return res;
00743    } else if (!asterisk_format && (type = ao2_find(codecs->payloads, &code, OBJ_NOLOCK | OBJ_KEY))) {
00744       res = type->payload;
00745       ao2_ref(type, -1);
00746       return res;
00747    }
00748 
00749    ast_rwlock_rdlock(&static_RTP_PT_lock);
00750    for (i = 0; i < AST_RTP_MAX_PT; i++) {
00751       if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
00752          (ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
00753          res = i;
00754          break;
00755       } else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
00756          (static_RTP_PT[i].rtp_code == code)) {
00757          res = i;
00758          break;
00759       }
00760    }
00761    ast_rwlock_unlock(&static_RTP_PT_lock);
00762 
00763    return res;
00764 }

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().

00718 {
00719    ast_format_cap_remove_all(astformats);
00720    *nonastformats = 0;
00721 
00722    ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_ast, astformats);
00723    ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_nonast, nonastformats);
00724 }

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().

00651 {
00652    struct ast_rtp_payload_type result = { .asterisk_format = 0, }, *type;
00653 
00654    if (payload < 0 || payload >= AST_RTP_MAX_PT) {
00655       return result;
00656    }
00657 
00658    if ((type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
00659       result = *type;
00660       ao2_ref(type, -1);
00661    }
00662 
00663    if (!result.rtp_code && !result.asterisk_format) {
00664       ast_rwlock_rdlock(&static_RTP_PT_lock);
00665       result = static_RTP_PT[payload];
00666       ast_rwlock_unlock(&static_RTP_PT_lock);
00667    }
00668 
00669    return result;
00670 }

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().

00452 {
00453    ast_rtp_codecs_payloads_destroy(codecs);
00454 
00455    if (instance && instance->engine && instance->engine->payload_set) {
00456       int i;
00457       for (i = 0; i < AST_RTP_MAX_PT; i++) {
00458          instance->engine->payload_set(instance, i, 0, NULL, 0);
00459       }
00460    }
00461 
00462    ast_rtp_codecs_payloads_initialize(codecs);
00463 }

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().

00500 {
00501    int i;
00502    struct ast_rtp_payload_type *type;
00503 
00504    for (i = 0; i < AST_RTP_MAX_PT; i++) {
00505       struct ast_rtp_payload_type *new_type;
00506 
00507       if (!(type = ao2_find(src->payloads, &i, OBJ_KEY | OBJ_NOLOCK))) {
00508          continue;
00509       }
00510 
00511       if (!(new_type = ao2_alloc(sizeof(*new_type), NULL))) {
00512          continue;
00513       }
00514 
00515       ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
00516 
00517       new_type->payload = i;
00518       *new_type = *type;
00519 
00520       ao2_link_flags(dest->payloads, new_type, OBJ_NOLOCK);
00521 
00522       ao2_ref(new_type, -1);
00523 
00524       if (instance && instance->engine && instance->engine->payload_set) {
00525          instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
00526       }
00527 
00528       ao2_ref(type, -1);
00529    }
00530 }

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

Set payload information on an RTP instance to the default.

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.

00466 {
00467    int i;
00468 
00469    ast_rwlock_rdlock(&static_RTP_PT_lock);
00470    for (i = 0; i < AST_RTP_MAX_PT; i++) {
00471       if (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
00472          struct ast_rtp_payload_type *type;
00473 
00474          if (!(type = ao2_alloc(sizeof(*type), NULL))) {
00475             /* Unfortunately if this occurs the payloads container will not contain all possible default payloads
00476              * but we err on the side of doing what we can in the hopes that the extreme memory conditions which
00477              * caused this to occur will go away.
00478              */
00479             continue;
00480          }
00481 
00482          type->payload = i;
00483          type->asterisk_format = static_RTP_PT[i].asterisk_format;
00484          type->rtp_code = static_RTP_PT[i].rtp_code;
00485          ast_format_copy(&type->format, &static_RTP_PT[i].format);
00486 
00487          ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
00488 
00489          if (instance && instance->engine && instance->engine->payload_set) {
00490             instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
00491          }
00492 
00493          ao2_ref(type, -1);
00494       }
00495    }
00496    ast_rwlock_unlock(&static_RTP_PT_lock);
00497 }

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().

00447 {
00448    ao2_cleanup(codecs->payloads);
00449 }

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().

00438 {
00439    if (!(codecs->payloads = ao2_container_alloc(AST_RTP_MAX_PT, rtp_payload_type_hash, rtp_payload_type_cmp))) {
00440       return -1;
00441    }
00442 
00443    return 0;
00444 }

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().

00533 {
00534    struct ast_rtp_payload_type *type;
00535 
00536    ast_rwlock_rdlock(&static_RTP_PT_lock);
00537 
00538    if (payload < 0 || payload >= AST_RTP_MAX_PT) {
00539       ast_rwlock_unlock(&static_RTP_PT_lock);
00540       return;
00541    }
00542 
00543    if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
00544       if (!(type = ao2_alloc(sizeof(*type), NULL))) {
00545          ast_rwlock_unlock(&static_RTP_PT_lock);
00546          return;
00547       }
00548       type->payload = payload;
00549       ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
00550    }
00551 
00552    type->asterisk_format = static_RTP_PT[payload].asterisk_format;
00553    type->rtp_code = static_RTP_PT[payload].rtp_code;
00554    type->payload = payload;
00555    ast_format_copy(&type->format, &static_RTP_PT[payload].format);
00556 
00557    ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
00558 
00559    if (instance && instance->engine && instance->engine->payload_set) {
00560       instance->engine->payload_set(instance, payload, type->asterisk_format, &type->format, type->rtp_code);
00561    }
00562 
00563    ao2_ref(type, -1);
00564 
00565    ast_rwlock_unlock(&static_RTP_PT_lock);
00566 }

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().

00631 {
00632    return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
00633 }

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, 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, ast_rtp_mime_type::type, and 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().

00572 {
00573    unsigned int i;
00574    int found = 0;
00575 
00576    if (pt < 0 || pt >= AST_RTP_MAX_PT)
00577       return -1; /* bogus payload type */
00578 
00579    ast_rwlock_rdlock(&mime_types_lock);
00580    for (i = 0; i < mime_types_len; ++i) {
00581       const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
00582       struct ast_rtp_payload_type *type;
00583 
00584       if (strcasecmp(mimesubtype, t->subtype)) {
00585          continue;
00586       }
00587 
00588       if (strcasecmp(mimetype, t->type)) {
00589          continue;
00590       }
00591 
00592       /* if both sample rates have been supplied, and they don't match,
00593        * then this not a match; if one has not been supplied, then the
00594        * rates are not compared */
00595       if (sample_rate && t->sample_rate &&
00596           (sample_rate != t->sample_rate)) {
00597          continue;
00598       }
00599 
00600       found = 1;
00601 
00602       if (!(type = ao2_find(codecs->payloads, &pt, OBJ_KEY | OBJ_NOLOCK))) {
00603          if (!(type = ao2_alloc(sizeof(*type), NULL))) {
00604             continue;
00605          }
00606          type->payload = pt;
00607          ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
00608       }
00609 
00610       *type = t->payload_type;
00611       type->payload = pt;
00612 
00613       if ((t->payload_type.format.id == AST_FORMAT_G726) && t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
00614          ast_format_set(&type->format, AST_FORMAT_G726_AAL2, 0);
00615       }
00616 
00617       if (instance && instance->engine && instance->engine->payload_set) {
00618          instance->engine->payload_set(instance, pt, type->asterisk_format, &type->format, type->rtp_code);
00619       }
00620 
00621       ao2_ref(type, -1);
00622 
00623       break;
00624    }
00625    ast_rwlock_unlock(&mime_types_lock);
00626 
00627    return (found ? 0 : -2);
00628 }

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().

00636 {
00637    if (payload < 0 || payload >= AST_RTP_MAX_PT) {
00638       return;
00639    }
00640 
00641    ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
00642 
00643    ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK | OBJ_NODATA | OBJ_UNLINK);
00644 
00645    if (instance && instance->engine && instance->engine->payload_set) {
00646       instance->engine->payload_set(instance, payload, 0, NULL, 0);
00647    }
00648 }

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 2108 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().

02109 {
02110    dst_cfg->enabled = src_cfg->enabled;
02111    dst_cfg->verify = src_cfg->verify;
02112    dst_cfg->rekey = src_cfg->rekey;
02113    dst_cfg->suite = src_cfg->suite;
02114    dst_cfg->certfile = ast_strdup(src_cfg->certfile);
02115    dst_cfg->pvtfile = ast_strdup(src_cfg->pvtfile);
02116    dst_cfg->cipher = ast_strdup(src_cfg->cipher);
02117    dst_cfg->cafile = ast_strdup(src_cfg->cafile);
02118    dst_cfg->capath = ast_strdup(src_cfg->capath);
02119    dst_cfg->default_setup = src_cfg->default_setup;
02120 }

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 2122 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().

02123 {
02124    ast_free(dtls_cfg->certfile);
02125    ast_free(dtls_cfg->pvtfile);
02126    ast_free(dtls_cfg->cipher);
02127    ast_free(dtls_cfg->cafile);
02128    ast_free(dtls_cfg->capath);
02129 }

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 2068 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().

02069 {
02070    if (!strcasecmp(name, "dtlsenable")) {
02071       dtls_cfg->enabled = ast_true(value) ? 1 : 0;
02072    } else if (!strcasecmp(name, "dtlsverify")) {
02073       dtls_cfg->verify = ast_true(value) ? 1 : 0;
02074    } else if (!strcasecmp(name, "dtlsrekey")) {
02075       if (sscanf(value, "%30u", &dtls_cfg->rekey) != 1) {
02076          return -1;
02077       }
02078    } else if (!strcasecmp(name, "dtlscertfile")) {
02079       ast_free(dtls_cfg->certfile);
02080       dtls_cfg->certfile = ast_strdup(value);
02081    } else if (!strcasecmp(name, "dtlsprivatekey")) {
02082       ast_free(dtls_cfg->pvtfile);
02083       dtls_cfg->pvtfile = ast_strdup(value);
02084    } else if (!strcasecmp(name, "dtlscipher")) {
02085       ast_free(dtls_cfg->cipher);
02086       dtls_cfg->cipher = ast_strdup(value);
02087    } else if (!strcasecmp(name, "dtlscafile")) {
02088       ast_free(dtls_cfg->cafile);
02089       dtls_cfg->cafile = ast_strdup(value);
02090    } else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
02091       ast_free(dtls_cfg->capath);
02092       dtls_cfg->capath = ast_strdup(value);
02093    } else if (!strcasecmp(name, "dtlssetup")) {
02094       if (!strcasecmp(value, "active")) {
02095          dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTIVE;
02096       } else if (!strcasecmp(value, "passive")) {
02097          dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_PASSIVE;
02098       } else if (!strcasecmp(value, "actpass")) {
02099          dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTPASS;
02100       }
02101    } else {
02102       return -1;
02103    }
02104 
02105    return 0;
02106 }

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 2181 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().

02182 {
02183    switch (format->id) {
02184    case AST_FORMAT_SILK:
02185       set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
02186       add_static_payload(-1, format, 0);
02187       break;
02188    case AST_FORMAT_CELT:
02189       set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
02190       add_static_payload(-1, format, 0);
02191       break;
02192    default:
02193       break;
02194    }
02195 
02196    return 0;
02197 }

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, ast_rtp_engine::entry, LOG_WARNING, ast_rtp_engine::mod, ast_rtp_engine::name, ast_rtp_engine::new, ast_rtp_engine::read, and ast_rtp_engine::write.

00115 {
00116    struct ast_rtp_engine *current_engine;
00117 
00118    /* Perform a sanity check on the engine structure to make sure it has the basics */
00119    if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
00120       ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
00121       return -1;
00122    }
00123 
00124    /* Link owner module to the RTP engine for reference counting purposes */
00125    engine->mod = module;
00126 
00127    AST_RWLIST_WRLOCK(&engines);
00128 
00129    /* Ensure that no two modules with the same name are registered at the same time */
00130    AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
00131       if (!strcmp(current_engine->name, engine->name)) {
00132          ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
00133          AST_RWLIST_UNLOCK(&engines);
00134          return -1;
00135       }
00136    }
00137 
00138    /* The engine survived our critique. Off to the list it goes to be used */
00139    AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
00140 
00141    AST_RWLIST_UNLOCK(&engines);
00142 
00143    ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
00144 
00145    return 0;
00146 }

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

Definition at line 1998 of file rtp_engine.c.

Referenced by res_srtp_init().

01999 {
02000    if (res_srtp || res_srtp_policy) {
02001       return -1;
02002    }
02003    if (!srtp_res || !policy_res) {
02004       return -1;
02005    }
02006 
02007    res_srtp = srtp_res;
02008    res_srtp_policy = policy_res;
02009 
02010    return 0;
02011 }

int ast_rtp_engine_srtp_is_registered ( void )

Definition at line 2019 of file rtp_engine.c.

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

02020 {
02021    return res_srtp && res_srtp_policy;
02022 }

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 2199 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, and static_RTP_PT.

Referenced by ast_format_attr_unreg_interface().

02200 {
02201    int x;
02202    int y = 0;
02203 
02204    ast_rwlock_wrlock(&static_RTP_PT_lock);
02205    /* remove everything pertaining to this format id from the lists */
02206    for (x = 0; x < AST_RTP_MAX_PT; x++) {
02207       if (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
02208          memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
02209       }
02210    }
02211    ast_rwlock_unlock(&static_RTP_PT_lock);
02212 
02213 
02214    ast_rwlock_wrlock(&mime_types_lock);
02215    /* rebuild the list skipping the items matching this id */
02216    for (x = 0; x < mime_types_len; x++) {
02217       if (ast_format_cmp(&ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
02218          continue;
02219       }
02220       ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
02221       y++;
02222    }
02223    mime_types_len = y;
02224    ast_rwlock_unlock(&mime_types_lock);
02225    return 0;
02226 }

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, ast_rtp_engine::entry, and ast_rtp_engine::name.

Referenced by load_module(), and unload_module().

00149 {
00150    struct ast_rtp_engine *current_engine = NULL;
00151 
00152    AST_RWLIST_WRLOCK(&engines);
00153 
00154    if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
00155       ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
00156    }
00157 
00158    AST_RWLIST_UNLOCK(&engines);
00159 
00160    return current_engine ? 0 : -1;
00161 }

void ast_rtp_engine_unregister_srtp ( void )

Definition at line 2013 of file rtp_engine.c.

Referenced by res_srtp_shutdown().

02014 {
02015    res_srtp = NULL;
02016    res_srtp_policy = NULL;
02017 }

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, ast_rtp_glue::entry, LOG_WARNING, ast_rtp_glue::mod, and ast_rtp_glue::type.

00164 {
00165    struct ast_rtp_glue *current_glue = NULL;
00166 
00167    if (ast_strlen_zero(glue->type)) {
00168       return -1;
00169    }
00170 
00171    glue->mod = module;
00172 
00173    AST_RWLIST_WRLOCK(&glues);
00174 
00175    AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
00176       if (!strcasecmp(current_glue->type, glue->type)) {
00177          ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
00178          AST_RWLIST_UNLOCK(&glues);
00179          return -1;
00180       }
00181    }
00182 
00183    AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
00184 
00185    AST_RWLIST_UNLOCK(&glues);
00186 
00187    ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
00188 
00189    return 0;
00190 }

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, ast_rtp_glue::entry, and ast_rtp_glue::type.

Referenced by load_module(), and unload_module().

00193 {
00194    struct ast_rtp_glue *current_glue = NULL;
00195 
00196    AST_RWLIST_WRLOCK(&glues);
00197 
00198    if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
00199       ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
00200    }
00201 
00202    AST_RWLIST_UNLOCK(&glues);
00203 
00204    return current_glue ? 0 : -1;
00205 }

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 1939 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().

01940 {
01941    return instance->engine->activate ? instance->engine->activate(instance) : 0;
01942 }

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 2024 of file rtp_engine.c.

References ast_srtp_res::add_stream, ast_srtp_res::create, ast_srtp_res::replace, and ast_rtp_instance::srtp.

Referenced by sdp_crypto_activate().

02025 {
02026    int res = 0;
02027 
02028    if (!res_srtp) {
02029       return -1;
02030    }
02031 
02032    if (!instance->srtp) {
02033       res = res_srtp->create(&instance->srtp, instance, remote_policy);
02034    } else {
02035       res = res_srtp->replace(&instance->srtp, instance, remote_policy);
02036    }
02037    if (!res) {
02038       res = res_srtp->add_stream(instance->srtp, local_policy);
02039    }
02040 
02041    return res;
02042 }

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 1927 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().

01928 {
01929    if (instance->engine->available_formats) {
01930       instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
01931       if (!ast_format_cap_is_empty(result)) {
01932          return;
01933       }
01934    }
01935 
01936    ast_translate_available_formats(to_endpoint, to_asterisk, result);
01937 }

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 1441 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_tech(), ast_channel_tech_pvt(), ast_channel_trylock, ast_channel_unlock, ast_check_hangup(), 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_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, remote_bridge_loop(), ast_sockaddr::ss, type, and unref_instance_cond().

01442 {
01443    struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
01444          *vinstance0 = NULL, *vinstance1 = NULL,
01445          *tinstance0 = NULL, *tinstance1 = NULL;
01446    struct ast_rtp_glue *glue0, *glue1;
01447    struct ast_sockaddr addr1 = { {0, }, }, addr2 = { {0, }, };
01448    enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
01449    enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
01450    enum ast_bridge_result res = AST_BRIDGE_FAILED;
01451    enum ast_rtp_dtmf_mode dmode;
01452    struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
01453    struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
01454    int unlock_chans = 1;
01455 
01456    if (!cap0 || !cap1) {
01457       unlock_chans = 0;
01458       goto done;
01459    }
01460 
01461    /* Lock both channels so we can look for the glue that binds them together */
01462    ast_channel_lock(c0);
01463    while (ast_channel_trylock(c1)) {
01464       ast_channel_unlock(c0);
01465       usleep(1);
01466       ast_channel_lock(c0);
01467    }
01468 
01469    /* Ensure neither channel got hungup during lock avoidance */
01470    if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
01471       ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
01472       goto done;
01473    }
01474 
01475    /* Grab glue that binds each channel to something using the RTP engine */
01476    if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
01477       ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
01478       goto done;
01479    }
01480 
01481    audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
01482    video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
01483 
01484    audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
01485    video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
01486 
01487    /* Apply any limitations on direct media bridging that may be present */
01488    if (audio_glue0_res == audio_glue1_res && audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
01489       if (glue0->allow_rtp_remote && !(glue0->allow_rtp_remote(c0, instance1))) {
01490          /* If the allow_rtp_remote indicates that remote isn't allowed, revert to local bridge */
01491          audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
01492       } else if (glue1->allow_rtp_remote && !(glue1->allow_rtp_remote(c1, instance0))) {
01493          audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
01494       }
01495    }
01496    if (video_glue0_res == video_glue1_res && video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
01497       if (glue0->allow_vrtp_remote && !(glue0->allow_vrtp_remote(c0, instance1))) {
01498          /* if the allow_vrtp_remote indicates that remote isn't allowed, revert to local bridge */
01499          video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
01500       } else if (glue1->allow_vrtp_remote && !(glue1->allow_vrtp_remote(c1, instance0))) {
01501          video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
01502       }
01503    }
01504 
01505    /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
01506    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)) {
01507       audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
01508    }
01509    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)) {
01510       audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
01511    }
01512 
01513    /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
01514    if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
01515       res = AST_BRIDGE_FAILED_NOWARN;
01516       goto done;
01517    }
01518 
01519 
01520    /* If address families differ, force a local bridge */
01521    ast_rtp_instance_get_remote_address(instance0, &addr1);
01522    ast_rtp_instance_get_remote_address(instance1, &addr2);
01523 
01524    if (addr1.ss.ss_family != addr2.ss.ss_family ||
01525       (ast_sockaddr_is_ipv4_mapped(&addr1) != ast_sockaddr_is_ipv4_mapped(&addr2))) {
01526       audio_glue0_res = AST_RTP_GLUE_RESULT_LOCAL;
01527       audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
01528    }
01529 
01530    /* If we need to get DTMF see if we can do it outside of the RTP stream itself */
01531    dmode = ast_rtp_instance_dtmf_mode_get(instance0);
01532    if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && dmode) {
01533       res = AST_BRIDGE_FAILED_NOWARN;
01534       goto done;
01535    }
01536    dmode = ast_rtp_instance_dtmf_mode_get(instance1);
01537    if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && dmode) {
01538       res = AST_BRIDGE_FAILED_NOWARN;
01539       goto done;
01540    }
01541 
01542    /* 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 */
01543    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)))) {
01544       res = AST_BRIDGE_FAILED_NOWARN;
01545       goto done;
01546    }
01547 
01548    /* Make sure that codecs match */
01549    if (glue0->get_codec){
01550       glue0->get_codec(c0, cap0);
01551    }
01552    if (glue1->get_codec) {
01553       glue1->get_codec(c1, cap1);
01554    }
01555    if (!ast_format_cap_is_empty(cap0) && !ast_format_cap_is_empty(cap1) && !ast_format_cap_has_joint(cap0, cap1)) {
01556       char tmp0[256] = { 0, };
01557       char tmp1[256] = { 0, };
01558       ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n",
01559          ast_getformatname_multiple(tmp0, sizeof(tmp0), cap0),
01560          ast_getformatname_multiple(tmp1, sizeof(tmp1), cap1));
01561       res = AST_BRIDGE_FAILED_NOWARN;
01562       goto done;
01563    }
01564 
01565    instance0->glue = glue0;
01566    instance1->glue = glue1;
01567    instance0->chan = c0;
01568    instance1->chan = c1;
01569 
01570    /* Depending on the end result for bridging either do a local bridge or remote bridge */
01571    if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
01572       ast_verb(3, "Locally bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
01573       res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
01574    } else {
01575       ast_verb(3, "Remotely bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
01576       res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
01577             tinstance0, tinstance1, glue0, glue1, cap0, cap1, timeoutms, flags,
01578             fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
01579    }
01580 
01581    instance0->glue = NULL;
01582    instance1->glue = NULL;
01583    instance0->chan = NULL;
01584    instance1->chan = NULL;
01585 
01586    unlock_chans = 0;
01587 
01588 done:
01589    if (unlock_chans) {
01590       ast_channel_unlock(c0);
01591       ast_channel_unlock(c1);
01592    }
01593    ast_format_cap_destroy(cap1);
01594    ast_format_cap_destroy(cap0);
01595 
01596    unref_instance_cond(&instance0);
01597    unref_instance_cond(&instance1);
01598    unref_instance_cond(&vinstance0);
01599    unref_instance_cond(&vinstance1);
01600    unref_instance_cond(&tinstance0);
01601    unref_instance_cond(&tinstance1);
01602 
01603    return res;
01604 }

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 906 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().

00907 {
00908    if (instance->engine->change_source) {
00909       instance->engine->change_source(instance);
00910    }
00911 }

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(), gtalk_free_pvt(), jingle_free_pvt(), jingle_session_destructor(), mgcp_hangup(), multicast_rtp_hangup(), multicast_rtp_request(), oh323_alloc(), setsubstate(), skinny_hangup(), start_rtp(), unalloc_sub(), and unistim_hangup_clean().

00230 {
00231    ao2_ref(instance, -1);
00232 
00233    return 0;
00234 }

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 875 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().

00876 {
00877    return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
00878 }

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 880 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().

00881 {
00882    return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
00883 }

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

Definition at line 884 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().

00885 {
00886    return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
00887 }

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 894 of file rtp_engine.c.

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

Referenced by ast_rtp_instance_bridge().

00895 {
00896    return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
00897 }

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 889 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().

00890 {
00891    return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
00892 }

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 1697 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.

01698 {
01699    struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
01700          *vinstance0 = NULL, *vinstance1 = NULL,
01701          *tinstance0 = NULL, *tinstance1 = NULL;
01702    struct ast_rtp_glue *glue0, *glue1;
01703    enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
01704    enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
01705    struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
01706    struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
01707    int res = 0;
01708 
01709    /* If there is no second channel just immediately bail out, we are of no use in that scenario */
01710    if (!c1) {
01711       ast_format_cap_destroy(cap0);
01712       ast_format_cap_destroy(cap1);
01713       return -1;
01714    }
01715 
01716    /* Lock both channels so we can look for the glue that binds them together */
01717    ast_channel_lock(c0);
01718    while (ast_channel_trylock(c1)) {
01719       ast_channel_unlock(c0);
01720       usleep(1);
01721       ast_channel_lock(c0);
01722    }
01723 
01724    if (!cap1 || !cap0) {
01725       goto done;
01726    }
01727 
01728    /* Grab glue that binds each channel to something using the RTP engine */
01729    if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
01730       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
01731       goto done;
01732    }
01733 
01734    audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
01735    video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
01736 
01737    audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
01738    video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
01739 
01740    /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
01741    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)) {
01742       audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
01743    }
01744    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)) {
01745       audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
01746    }
01747    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) {
01748       glue0->get_codec(c0, cap0);
01749    }
01750    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) {
01751       glue1->get_codec(c1, cap1);
01752    }
01753 
01754    /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
01755    if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
01756       goto done;
01757    }
01758 
01759    /* Make sure we have matching codecs */
01760    if (!ast_format_cap_has_joint(cap0, cap1)) {
01761       goto done;
01762    }
01763 
01764    /* Bridge media early */
01765    if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
01766       ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
01767    }
01768 
01769    res = 0;
01770 
01771 done:
01772    ast_channel_unlock(c0);
01773    ast_channel_unlock(c1);
01774 
01775    ast_format_cap_destroy(cap0);
01776    ast_format_cap_destroy(cap1);
01777 
01778    unref_instance_cond(&instance0);
01779    unref_instance_cond(&instance1);
01780    unref_instance_cond(&vinstance0);
01781    unref_instance_cond(&vinstance1);
01782    unref_instance_cond(&tinstance0);
01783    unref_instance_cond(&tinstance1);
01784 
01785    if (!res) {
01786       ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
01787    }
01788 
01789    return res;
01790 }

void ast_rtp_instance_early_bridge_make_compatible	(	struct ast_channel *	c0,
		struct ast_channel *	c1
	)

Make two channels compatible for early bridging.

Parameters:

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

Since:: 1.8

Definition at line 1611 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().

01612 {
01613    struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
01614       *vinstance0 = NULL, *vinstance1 = NULL,
01615       *tinstance0 = NULL, *tinstance1 = NULL;
01616    struct ast_rtp_glue *glue0, *glue1;
01617    enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
01618    enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
01619    struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
01620    struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
01621 
01622    /* Lock both channels so we can look for the glue that binds them together */
01623    ast_channel_lock_both(c0, c1);
01624 
01625    if (!cap1 || !cap0) {
01626       goto done;
01627    }
01628 
01629    /* Grab glue that binds each channel to something using the RTP engine */
01630    if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
01631       ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
01632       goto done;
01633    }
01634 
01635    audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
01636    video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
01637 
01638    audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
01639    video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
01640 
01641    /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
01642    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)) {
01643       audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
01644    }
01645    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)) {
01646       audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
01647    }
01648    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) {
01649       glue0->get_codec(c0, cap0);
01650    }
01651    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) {
01652       glue1->get_codec(c1, cap1);
01653    }
01654 
01655    /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
01656    if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
01657       goto done;
01658    }
01659 
01660    /* Make sure we have matching codecs */
01661    if (!ast_format_cap_has_joint(cap0, cap1)) {
01662       goto done;
01663    }
01664 
01665    ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
01666 
01667    if (vinstance0 && vinstance1) {
01668       ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
01669    }
01670    if (tinstance0 && tinstance1) {
01671       ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
01672    }
01673 
01674    if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
01675       ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
01676          ast_channel_name(c0), ast_channel_name(c1));
01677    } else {
01678       ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
01679          ast_channel_name(c0), ast_channel_name(c1));
01680    }
01681 
01682 done:
01683    ast_channel_unlock(c0);
01684    ast_channel_unlock(c1);
01685 
01686    ast_format_cap_destroy(cap0);
01687    ast_format_cap_destroy(cap1);
01688 
01689    unref_instance_cond(&instance0);
01690    unref_instance_cond(&instance1);
01691    unref_instance_cond(&vinstance0);
01692    unref_instance_cond(&vinstance1);
01693    unref_instance_cond(&tinstance0);
01694    unref_instance_cond(&tinstance1);
01695 }

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 925 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().

00926 {
00927    return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
00928 }

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 1988 of file rtp_engine.c.

References ast_rtp_instance::glue.

01989 {
01990    return instance->glue;
01991 }

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.

00354 {
00355    if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
00356       ast_sockaddr_copy(address, &instance->local_address);
00357       return 1;
00358    }
00359 
00360    return 0;
00361 }

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().

00371 {
00372    if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) {
00373       ast_sockaddr_copy(address, &instance->remote_address);
00374       return 1;
00375    }
00376 
00377    return 0;
00378 }

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 1606 of file rtp_engine.c.

References ast_rtp_instance::bridged.

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

01607 {
01608    return instance->bridged;
01609 }

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 1993 of file rtp_engine.c.

References ast_rtp_instance::chan.

01994 {
01995    return instance->chan;
01996 }

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().

00417 {
00418    return &instance->codecs;
00419 }

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_alt_remote_address_set(), ast_rtp_change_source(), ast_rtp_destroy(), ast_rtp_dtmf_begin(), ast_rtp_dtmf_continuation(), ast_rtp_dtmf_end_with_duration(), ast_rtp_dtmf_mode_get(), ast_rtp_dtmf_mode_set(), ast_rtp_fd(), ast_rtp_get_stat(), ast_rtp_ice_add_remote_candidate(), ast_rtp_ice_get_local_candidates(), ast_rtp_ice_get_password(), ast_rtp_ice_get_ufrag(), ast_rtp_ice_lite(), ast_rtp_ice_set_authentication(), ast_rtp_ice_start(), ast_rtp_ice_stop(), ast_rtp_local_bridge(), ast_rtp_on_turn_rtcp_state(), ast_rtp_on_turn_rtp_state(), ast_rtp_on_turn_rx_rtcp_data(), ast_rtp_on_turn_rx_rtp_data(), ast_rtp_prop_set(), ast_rtp_qos_set(), ast_rtp_raw_write(), ast_rtp_read(), ast_rtp_remote_address_set(), ast_rtp_sendcng(), ast_rtp_stop(), ast_rtp_stun_request(), ast_rtp_update_source(), ast_rtp_write(), bridge_p2p_rtp_write(), create_dtmf_frame(), multicast_rtp_activate(), multicast_rtp_destroy(), multicast_rtp_write(), process_cn_rfc3389(), process_dtmf_cisco(), process_dtmf_rfc2833(), red_write(), rtp_red_buffer(), and rtp_red_init().

00303 {
00304    return instance->data;
00305 }

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 2063 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().

02064 {
02065    return instance->engine->dtls;
02066 }

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 1983 of file rtp_engine.c.

References ast_rtp_instance::engine.

01984 {
01985    return instance->engine;
01986 }

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.

00394 {
00395    if (instance->engine->extended_prop_get) {
00396       return instance->engine->extended_prop_get(instance, property);
00397    }
00398 
00399    return NULL;
00400 }

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 930 of file rtp_engine.c.

References AST_RWLIST_RDLOCK, AST_RWLIST_TRAVERSE, AST_RWLIST_UNLOCK, ast_rtp_glue::entry, and ast_rtp_glue::type.

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

00931 {
00932    struct ast_rtp_glue *glue = NULL;
00933 
00934    AST_RWLIST_RDLOCK(&glues);
00935 
00936    AST_RWLIST_TRAVERSE(&glues, glue, entry) {
00937       if (!strcasecmp(glue->type, type)) {
00938          break;
00939       }
00940    }
00941 
00942    AST_RWLIST_UNLOCK(&glues);
00943 
00944    return glue;
00945 }

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 1973 of file rtp_engine.c.

References ast_rtp_instance::holdtimeout.

Referenced by check_rtp_timeout().

01974 {
01975    return instance->holdtimeout;
01976 }

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 2058 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(), process_sdp_a_ice(), and start_ice().

02059 {
02060    return instance->engine->ice;
02061 }

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 1978 of file rtp_engine.c.

References ast_rtp_instance::keepalive.

Referenced by check_rtp_timeout().

01979 {
01980    return instance->keepalive;
01981 }

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(), ast_rtp_on_turn_rx_rtp_data(), 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(), sip_allow_anyrtp_remote(), and skinny_set_rtp_peer().

00365 {
00366    ast_sockaddr_copy(address, &instance->local_address);
00367 }

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().

00412 {
00413    return instance->properties[property];
00414 }

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 1807 of file rtp_engine.c.

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

01808 {
01809    struct ast_rtp_instance_stats stats = { 0, };
01810    enum ast_rtp_instance_stat stat;
01811 
01812    /* Determine what statistics we will need to retrieve based on field passed in */
01813    if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
01814       stat = AST_RTP_INSTANCE_STAT_ALL;
01815    } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
01816       stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
01817    } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
01818       stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
01819    } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
01820       stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
01821    } else {
01822       return NULL;
01823    }
01824 
01825    /* Attempt to actually retrieve the statistics we need to generate the quality string */
01826    if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
01827       return NULL;
01828    }
01829 
01830    /* Now actually fill the buffer with the good information */
01831    if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
01832       snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
01833           stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
01834    } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
01835       snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
01836           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));
01837    } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
01838       snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
01839           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));
01840    } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
01841       snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
01842    }
01843 
01844    return buf;
01845 }

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(), 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(), find_rtp_port(), gtalk_update_stun(), 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(), sip_allow_anyrtp_remote(), skinny_set_rtp_peer(), and transmit_modify_with_sdp().

00382 {
00383    ast_sockaddr_copy(address, &instance->remote_address);
00384 }

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 2044 of file rtp_engine.c.

References ast_rtp_instance::srtp.

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

02045 {
02046    return instance->srtp;
02047 }

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 1802 of file rtp_engine.c.

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

Referenced by ast_rtp_instance_get_quality(), ast_srtp_unprotect(), sdp_crypto_activate(), show_chanstats_cb(), and sip_acf_channel_read().

01803 {
01804    return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
01805 }

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 1968 of file rtp_engine.c.

References ast_rtp_instance::timeout.

Referenced by check_rtp_timeout().

01969 {
01970    return instance->timeout;
01971 }

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 1891 of file rtp_engine.c.

References ao2_ref, ast_channel_lock, ast_channel_tech(), ast_channel_unlock, ast_rtp_instance_get_glue(), ast_rtp_instance::engine, ast_rtp_glue::get_rtp_info, ast_rtp_engine::make_compatible, and type.

Referenced by sip_setoption().

01892 {
01893    struct ast_rtp_glue *glue;
01894    struct ast_rtp_instance *peer_instance = NULL;
01895    int res = -1;
01896 
01897    if (!instance->engine->make_compatible) {
01898       return -1;
01899    }
01900 
01901    ast_channel_lock(peer);
01902 
01903    if (!(glue = ast_rtp_instance_get_glue(ast_channel_tech(peer)->type))) {
01904       ast_channel_unlock(peer);
01905       return -1;
01906    }
01907 
01908    glue->get_rtp_info(peer, &peer_instance);
01909 
01910    if (!peer_instance || peer_instance->engine != instance->engine) {
01911       ast_channel_unlock(peer);
01912       ao2_ref(peer_instance, -1);
01913       peer_instance = NULL;
01914       return -1;
01915    }
01916 
01917    res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
01918 
01919    ast_channel_unlock(peer);
01920 
01921    ao2_ref(peer_instance, -1);
01922    peer_instance = NULL;
01923 
01924    return res;
01925 }

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, ast_rtp_engine::entry, instance_destructor(), ast_rtp_instance::local_address, LOG_ERROR, ast_rtp_engine::mod, ast_rtp_engine::name, and ast_rtp_engine::new.

Referenced by __oh323_rtp_create(), dialog_initialize_rtp(), gtalk_alloc(), jingle_alloc(), jingle_enable_video(), multicast_rtp_request(), and start_rtp().

00239 {
00240    struct ast_sockaddr address = {{0,}};
00241    struct ast_rtp_instance *instance = NULL;
00242    struct ast_rtp_engine *engine = NULL;
00243 
00244    AST_RWLIST_RDLOCK(&engines);
00245 
00246    /* If an engine name was specified try to use it or otherwise use the first one registered */
00247    if (!ast_strlen_zero(engine_name)) {
00248       AST_RWLIST_TRAVERSE(&engines, engine, entry) {
00249          if (!strcmp(engine->name, engine_name)) {
00250             break;
00251          }
00252       }
00253    } else {
00254       engine = AST_RWLIST_FIRST(&engines);
00255    }
00256 
00257    /* If no engine was actually found bail out now */
00258    if (!engine) {
00259       ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
00260       AST_RWLIST_UNLOCK(&engines);
00261       return NULL;
00262    }
00263 
00264    /* Bump up the reference count before we return so the module can not be unloaded */
00265    ast_module_ref(engine->mod);
00266 
00267    AST_RWLIST_UNLOCK(&engines);
00268 
00269    /* Allocate a new RTP instance */
00270    if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
00271       ast_module_unref(engine->mod);
00272       return NULL;
00273    }
00274    instance->engine = engine;
00275    ast_sockaddr_copy(&instance->local_address, sa);
00276    ast_sockaddr_copy(&address, sa);
00277 
00278    if (ast_rtp_codecs_payloads_initialize(&instance->codecs)) {
00279       ao2_ref(instance, -1);
00280       return NULL;
00281    }
00282 
00283    ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
00284 
00285    /* And pass it off to the engine to setup */
00286    if (instance->engine->new(instance, sched, &address, data)) {
00287       ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
00288       ao2_ref(instance, -1);
00289       return NULL;
00290    }
00291 
00292    ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
00293 
00294    return instance;
00295 }

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().

00313 {
00314    return instance->engine->read(instance, rtcp);
00315 }

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 2049 of file rtp_engine.c.

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

Referenced by check_rtp_timeout().

02050 {
02051    if (instance->engine->sendcng) {
02052       return instance->engine->sendcng(instance, level);
02053    }
02054 
02055    return -1;
02056 }

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().

00340 {
00341    ast_sockaddr_copy(&instance->alt_remote_address, address);
00342 
00343    /* oink */
00344 
00345    if (instance->engine->alt_remote_address_set) {
00346       instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address);
00347    }
00348 
00349    return 0;
00350 }

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().

00298 {
00299    instance->data = data;
00300 }

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.

00387 {
00388    if (instance->engine->extended_prop_set) {
00389       instance->engine->extended_prop_set(instance, property, value);
00390    }
00391 }

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 1958 of file rtp_engine.c.

References ast_rtp_instance::holdtimeout.

Referenced by check_rtp_timeout(), and dialog_initialize_rtp().

01959 {
01960    instance->holdtimeout = timeout;
01961 }

void ast_rtp_instance_set_keepalive	(	struct ast_rtp_instance *	instance,
		int	timeout
	)

Set the RTP keepalive interval.

Parameters:

	instance	The RTP instance
	timeout	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 1963 of file rtp_engine.c.

References ast_rtp_instance::keepalive.

Referenced by dialog_initialize_rtp().

01964 {
01965    instance->keepalive = interval;
01966 }

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().

00319 {
00320    ast_sockaddr_copy(&instance->local_address, address);
00321    return 0;
00322 }

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, and ast_rtp_instance::properties.

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().

00403 {
00404    instance->properties[property] = value;
00405 
00406    if (instance->engine->prop_set) {
00407       instance->engine->prop_set(instance, property, value);
00408    }
00409 }

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 913 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().

00914 {
00915    return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
00916 }

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 1881 of file rtp_engine.c.

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

Referenced by sip_new(), and sip_setoption().

01882 {
01883    return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
01884 }

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().

00326 {
00327    ast_sockaddr_copy(&instance->remote_address, address);
00328 
00329    /* moo */
00330 
00331    if (instance->engine->remote_address_set) {
00332       instance->engine->remote_address_set(instance, &instance->remote_address);
00333    }
00334 
00335    return 0;
00336 }

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 1847 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().

01848 {
01849    char quality_buf[AST_MAX_USER_FIELD], *quality;
01850    struct ast_channel *bridge = ast_bridged_channel(chan);
01851 
01852    if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
01853       pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
01854       if (bridge) {
01855          pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
01856       }
01857    }
01858 
01859    if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
01860       pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
01861       if (bridge) {
01862          pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
01863       }
01864    }
01865 
01866    if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
01867       pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
01868       if (bridge) {
01869          pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
01870       }
01871    }
01872 
01873    if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
01874       pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
01875       if (bridge) {
01876          pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
01877       }
01878    }
01879 }

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 1953 of file rtp_engine.c.

References ast_rtp_instance::timeout.

Referenced by check_rtp_timeout(), and dialog_initialize_rtp().

01954 {
01955    instance->timeout = timeout;
01956 }

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 1886 of file rtp_engine.c.

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

Referenced by sip_new(), and sip_setoption().

01887 {
01888    return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
01889 }

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 918 of file rtp_engine.c.

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

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

00919 {
00920    if (instance->engine->stop) {
00921       instance->engine->stop(instance);
00922    }
00923 }

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 1944 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().

01947 {
01948    if (instance->engine->stun_request) {
01949       instance->engine->stun_request(instance, suggestion, username);
01950    }
01951 }

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 899 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().

00900 {
00901    if (instance->engine->update_source) {
00902       instance->engine->update_source(instance);
00903    }
00904 }

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().

00308 {
00309    return instance->engine->write(instance, frame);
00310 }

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 830 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().

00831 {
00832    int found = 0;
00833    const char *name;
00834    if (!buf) {
00835       return NULL;
00836    }
00837 
00838 
00839    if (asterisk_format) {
00840       struct ast_format tmp_fmt;
00841       ast_format_cap_iter_start(ast_format_capability);
00842       while (!ast_format_cap_iter_next(ast_format_capability, &tmp_fmt)) {
00843          name = ast_rtp_lookup_mime_subtype2(asterisk_format, &tmp_fmt, 0, options);
00844          ast_str_append(&buf, 0, "%s|", name);
00845          found = 1;
00846       }
00847       ast_format_cap_iter_end(ast_format_capability);
00848 
00849    } else {
00850       int x;
00851       ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
00852       for (x = 1; x < AST_RTP_MAX; x <<= 1) {
00853          if (rtp_capability & x) {
00854             name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
00855             ast_str_append(&buf, 0, "%s|", name);
00856             found = 1;
00857          }
00858       }
00859    }
00860 
00861    ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
00862 
00863    return ast_str_buffer(buf);
00864 }

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 780 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, 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().

00781 {
00782    int i;
00783    const char *res = "";
00784 
00785    ast_rwlock_rdlock(&mime_types_lock);
00786    for (i = 0; i < mime_types_len; i++) {
00787       if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
00788          (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
00789          if ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
00790             res = "G726-32";
00791             break;
00792          } else {
00793             res = ast_rtp_mime_types[i].subtype;
00794             break;
00795          }
00796       } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
00797          ast_rtp_mime_types[i].payload_type.rtp_code == code) {
00798 
00799          res = ast_rtp_mime_types[i].subtype;
00800          break;
00801       }
00802    }
00803    ast_rwlock_unlock(&mime_types_lock);
00804 
00805    return res;
00806 }

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.
	format	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 808 of file rtp_engine.c.

References ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, ast_rtp_mime_types, ast_rwlock_rdlock, ast_rwlock_unlock, 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().

00809 {
00810    unsigned int i;
00811    unsigned int res = 0;
00812 
00813    ast_rwlock_rdlock(&mime_types_lock);
00814    for (i = 0; i < mime_types_len; ++i) {
00815       if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
00816          (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
00817          res = ast_rtp_mime_types[i].sample_rate;
00818          break;
00819       } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
00820          ast_rtp_mime_types[i].payload_type.rtp_code == code) {
00821          res = ast_rtp_mime_types[i].sample_rate;
00822          break;
00823       }
00824    }
00825    ast_rwlock_unlock(&mime_types_lock);
00826 
00827    return res;
00828 }

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 1797 of file rtp_engine.c.

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

Referenced by sip_write().

01798 {
01799    return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
01800 }

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 1792 of file rtp_engine.c.

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

Referenced by process_sdp().

01793 {
01794    return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
01795 }


Data Structures
struct	ast_rtp_codecs
struct	ast_rtp_dtls_cfg
	DTLS configuration structure. More...
struct	ast_rtp_engine
struct	ast_rtp_engine_dtls
	Structure that represents the optional DTLS SRTP support within an RTP engine. More...
struct	ast_rtp_engine_ice
	Structure that represents the optional ICE support within an RTP engine. More...
struct	ast_rtp_engine_ice_candidate
	Structure for an ICE candidate. More...
struct	ast_rtp_glue
struct	ast_rtp_instance_stats
struct	ast_rtp_payload_type
Defines
#define	AST_RED_MAX_GENERATION 5
#define	AST_RTP_CISCO_DTMF (1 << 2)
#define	AST_RTP_CN (1 << 1)
#define	AST_RTP_DTMF (1 << 0)
#define	ast_rtp_engine_register(engine) ast_rtp_engine_register2(engine, ast_module_info->self)
#define	ast_rtp_glue_register(glue) ast_rtp_glue_register2(glue, ast_module_info->self)
#define	AST_RTP_MAX AST_RTP_CISCO_DTMF
#define	AST_RTP_MAX_PT 196
#define	AST_RTP_STAT_SET(current_stat, combined, placement, value)
#define	AST_RTP_STAT_TERMINATOR(combined)
Enumerations
enum	ast_rtp_dtls_connection { AST_RTP_DTLS_CONNECTION_NEW, AST_RTP_DTLS_CONNECTION_EXISTING }
	DTLS connection states. More...
enum	ast_rtp_dtls_hash { AST_RTP_DTLS_HASH_SHA1 }
	DTLS fingerprint hashes. More...
enum	ast_rtp_dtls_setup { AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_PASSIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_HOLDCONN }
	DTLS setup types. More...
enum	ast_rtp_dtmf_mode { AST_RTP_DTMF_MODE_NONE = 0, AST_RTP_DTMF_MODE_RFC2833, AST_RTP_DTMF_MODE_INBAND }
enum	ast_rtp_glue_result { AST_RTP_GLUE_RESULT_FORBID = 0, AST_RTP_GLUE_RESULT_REMOTE, AST_RTP_GLUE_RESULT_LOCAL }
enum	ast_rtp_ice_candidate_type { AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED }
	ICE candidate types. More...
enum	ast_rtp_instance_stat { AST_RTP_INSTANCE_STAT_ALL = 0, AST_RTP_INSTANCE_STAT_TXCOUNT, AST_RTP_INSTANCE_STAT_RXCOUNT, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, AST_RTP_INSTANCE_STAT_TXPLOSS, AST_RTP_INSTANCE_STAT_RXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, AST_RTP_INSTANCE_STAT_TXJITTER, AST_RTP_INSTANCE_STAT_RXJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_RTT, AST_RTP_INSTANCE_STAT_RTT, AST_RTP_INSTANCE_STAT_MAX_RTT, AST_RTP_INSTANCE_STAT_MIN_RTT, AST_RTP_INSTANCE_STAT_NORMDEVRTT, AST_RTP_INSTANCE_STAT_STDEVRTT, AST_RTP_INSTANCE_STAT_LOCAL_SSRC, AST_RTP_INSTANCE_STAT_REMOTE_SSRC }
enum	ast_rtp_instance_stat_field { AST_RTP_INSTANCE_STAT_FIELD_QUALITY = 0, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT }
enum	ast_rtp_options { AST_RTP_OPT_G726_NONSTANDARD = (1 << 0) }
enum	ast_rtp_property { AST_RTP_PROPERTY_NAT = 0, AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, AST_RTP_PROPERTY_STUN, AST_RTP_PROPERTY_RTCP, AST_RTP_PROPERTY_MAX }
Functions
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_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 c0, struct ast_channel c1)
	Make two channels compatible for early bridging.
int	ast_rtp_instance_fd (struct ast_rtp_instance *instance, int rtcp)
	Get the file descriptor for an RTP session (or RTCP).
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 timeout)
	Set the RTP keepalive interval.
int	ast_rtp_instance_set_local_address (struct ast_rtp_instance instance, const struct ast_sockaddr address)
	Set the address that we are expecting to receive RTP on.
void	ast_rtp_instance_set_prop (struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
	Set the value of an RTP instance property.
int	ast_rtp_instance_set_qos (struct ast_rtp_instance instance, int tos, int cos, const char desc)
	Set QoS parameters on an RTP session.
int	ast_rtp_instance_set_read_format (struct ast_rtp_instance instance, 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.

Asterisk developer's documentation

rtp_engine.h File Reference

Data Structures

Defines

Enumerations

Functions

Detailed Description

Define Documentation

Enumeration Type Documentation

Function Documentation