diff options
Diffstat (limited to 'include')
| -rw-r--r-- | include/asterisk/channel.h | 20 | ||||
| -rw-r--r-- | include/asterisk/config.h | 11 | ||||
| -rw-r--r-- | include/asterisk/config_options.h | 24 | ||||
| -rw-r--r-- | include/asterisk/res_pjsip.h | 22 | ||||
| -rw-r--r-- | include/asterisk/res_pjsip_session.h | 21 | ||||
| -rw-r--r-- | include/asterisk/rtp_engine.h | 52 |
6 files changed, 150 insertions, 0 deletions
diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h index cb2e0e6a4..f0fe5b212 100644 --- a/include/asterisk/channel.h +++ b/include/asterisk/channel.h @@ -2030,6 +2030,26 @@ struct ast_frame *ast_read_stream(struct ast_channel *chan); struct ast_frame *ast_read_noaudio(struct ast_channel *chan); /*! + * \brief Reads a frame, but does not filter to just the default streams, + * returning AST_FRAME_NULL frame if audio. + * + * \param chan channel to read a frame from + * + * \return Returns a frame, or NULL on error. If it returns NULL, you + * best just stop reading frames and assume the channel has been + * disconnected. + * + * \note This function will not perform any filtering and will return + * media frames from all streams on the channel. To determine which + * stream a frame originated from the stream_num on it can be + * examined. + * + * \note Audio is replaced with AST_FRAME_NULL to avoid + * transcode when the resulting audio is not necessary. + */ +struct ast_frame *ast_read_stream_noaudio(struct ast_channel *chan); + +/*! * \brief Write a frame to a channel * This function writes the given frame to the indicated channel. * \param chan destination channel of the frame diff --git a/include/asterisk/config.h b/include/asterisk/config.h index f57966b0b..1addfa317 100644 --- a/include/asterisk/config.h +++ b/include/asterisk/config.h @@ -1086,6 +1086,11 @@ enum ast_parse_flags { PARSE_UINT16 = 0x0005, #endif + /* Returns an int processed by ast_app_parse_timelen. + * The first argument is an enum ast_timelen value (required). + */ + PARSE_TIMELEN = 0x0006, + /* Returns a struct ast_sockaddr, with optional default value * (passed by reference) and port handling (accept, ignore, * require, forbid). The format is 'ipaddress[:port]'. IPv6 address @@ -1152,6 +1157,12 @@ enum ast_parse_flags { * returns 1, b unchanged * ast_parse_arg("12", PARSE_UINT32|PARSE_IN_RANGE|PARSE_RANGE_DEFAULTS, &a, 1, 10); * returns 1, a = 10 + * ast_parse_arg("223", PARSE_TIMELEN|PARSE_IN_RANGE, &a, TIMELEN_SECONDS, -1000, 1000); + * returns 0, a = 1000 + * ast_parse_arg("223", PARSE_TIMELEN|PARSE_IN_RANGE, &a, TIMELEN_SECONDS, -1000, 250000); + * returns 0, a = 223000 + * ast_parse_arg("223", PARSE_TIMELEN|PARSE_IN_RANGE|PARSE_DEFAULT, &a, TIMELEN_SECONDS, 9999, -1000, 250000); + * returns 0, a = 9999 * ast_parse_arg("www.foo.biz:44", PARSE_INADDR, &sa); * returns 0, sa contains address and port * ast_parse_arg("www.foo.biz", PARSE_INADDR|PARSE_PORT_REQUIRE, &sa); diff --git a/include/asterisk/config_options.h b/include/asterisk/config_options.h index f2a457eb5..f4c3db188 100644 --- a/include/asterisk/config_options.h +++ b/include/asterisk/config_options.h @@ -468,6 +468,30 @@ enum aco_option_type { */ OPT_YESNO_T, + /*! \brief Type for default option handler for time length signed integers + * + * \note aco_option_register flags: + * See flags available for use with the PARSE_TIMELEN type for the ast_parse_arg function + * aco_option_register varargs: + * FLDSET macro with the field of type int + * The remaining varargs for should be arguments compatible with the varargs for the + * ast_parse_arg function with the PARSE_TIMELEN type and the flags passed in the + * aco_option_register flags parameter. + * + * \note In most situations, it is preferable to not pass the PARSE_DEFAULT flag. If a config + * contains an invalid value, it is better to let the config loading fail with warnings so that + * the problem is fixed by the administrator. + * + * Example: + * struct test_item { + * int timelen; + * }; + * {code} + * aco_option_register(&cfg_info, "timelen", ACO_EXACT, my_types, "3", OPT_TIMELEN_T, PARSE_IN_RANGE, FLDSET(struct test_item, intopt), TIMELEN_MILLISECONDS, -10, 10); + * {endcode} + */ + OPT_TIMELEN_T, + }; /*! \brief A callback function for handling a particular option diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h index 2cd27d37f..cf366cbab 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -688,6 +688,10 @@ struct ast_sip_endpoint_media_configuration { unsigned int max_audio_streams; /*! Maximum number of video streams to offer/accept */ unsigned int max_video_streams; + /*! Use BUNDLE */ + unsigned int bundle; + /*! Enable webrtc settings and defaults */ + unsigned int webrtc; }; /*! @@ -2059,6 +2063,24 @@ int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text); void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size); /*! + * \brief Create and copy a pj_str_t into a standard character buffer. + * + * pj_str_t is not NULL-terminated. Any place that expects a NULL- + * terminated string needs to have the pj_str_t copied into a separate + * buffer. + * + * Copies the pj_str_t contents into a newly allocated buffer pointed to + * by dest. NULL-terminates the buffer. + * + * \note Caller is responsible for freeing the allocated memory. + * + * \param dest [out] The destination buffer + * \param src The pj_str_t to copy + * \retval Number of characters copied or negative value on error + */ +int ast_copy_pj_str2(char **dest, const pj_str_t *src); + +/*! * \brief Get the looked-up endpoint on an out-of dialog request or response * * The function may ONLY be called on out-of-dialog requests or responses. For diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h index e298e1f32..eae11af43 100644 --- a/include/asterisk/res_pjsip_session.h +++ b/include/asterisk/res_pjsip_session.h @@ -99,6 +99,14 @@ struct ast_sip_session_media { ast_sip_session_media_write_cb write_callback; /*! \brief The stream number to place into any resulting frames */ int stream_num; + /*! \brief Media identifier for this stream (may be shared across multiple streams) */ + char *mid; + /*! \brief The bundle group the stream belongs to */ + int bundle_group; + /*! \brief Whether this stream is currently bundled or not */ + unsigned int bundled; + /*! \brief RTP/Media streams association identifier */ + char *msid; }; /*! @@ -833,6 +841,19 @@ int ast_sip_session_media_add_read_callback(struct ast_sip_session *session, str int ast_sip_session_media_set_write_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media, ast_sip_session_media_write_cb callback); +/*! + * \brief Retrieve the underlying media session that is acting as transport for a media session + * \since 15.0.0 + * + * \param session The session + * \param session_media The media session to retrieve the transport for + * + * \note This operates on the pending media state + * + * \note This function is guaranteed to return non-NULL + */ +struct ast_sip_session_media *ast_sip_session_media_get_transport(struct ast_sip_session *session, struct ast_sip_session_media *session_media); + /*! \brief Determines whether the res_pjsip_session module is loaded */ #define CHECK_PJSIP_SESSION_MODULE_LOADED() \ do { \ diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h index 36f499b50..d030bdb19 100644 --- a/include/asterisk/rtp_engine.h +++ b/include/asterisk/rtp_engine.h @@ -603,6 +603,12 @@ struct ast_rtp_engine { unsigned int (*ssrc_get)(struct ast_rtp_instance *instance); /*! Callback to retrieve RTCP SDES CNAME */ const char *(*cname_get)(struct ast_rtp_instance *instance); + /*! Callback to bundle an RTP instance to another */ + int (*bundle)(struct ast_rtp_instance *child, struct ast_rtp_instance *parent); + /*! Callback to set remote SSRC information */ + void (*set_remote_ssrc)(struct ast_rtp_instance *instance, unsigned int ssrc); + /*! Callback to set the stream identifier */ + void (*set_stream_num)(struct ast_rtp_instance *instance, int stream_num); /*! Callback to pointer for optional ICE support */ struct ast_rtp_engine_ice *ice; /*! Callback to pointer for optional DTLS SRTP support */ @@ -1512,6 +1518,20 @@ void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_fo int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code); /*! + * \brief Set a payload code for use with a specific Asterisk format + * + * \param codecs Codecs structure to manipulate + * \param code The payload code + * \param format Asterisk format + * + * \retval 0 Payload was set to the given format + * \retval -1 Payload was in use or could not be set + * + * \since 15.0.0 + */ +int ast_rtp_codecs_payload_set_rx(struct ast_rtp_codecs *codecs, int code, struct ast_format *format); + +/*! * \brief Retrieve a tx mapped payload type based on whether it is an Asterisk format and the code * \since 14.0.0 * @@ -2271,6 +2291,8 @@ int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level); * * \retval 0 Success * \retval non-zero Failure + * + * \note If no remote policy is provided any existing SRTP policies are left and the new local policy is added */ int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy* remote_policy, struct ast_srtp_policy *local_policy, int rtcp); @@ -2416,6 +2438,36 @@ unsigned int ast_rtp_instance_get_ssrc(struct ast_rtp_instance *rtp); */ const char *ast_rtp_instance_get_cname(struct ast_rtp_instance *rtp); +/*! + * \brief Request that an RTP instance be bundled with another + * \since 15.0.0 + * + * \param child The child RTP instance + * \param parent The parent RTP instance the child should be bundled with + * + * \retval 0 success + * \retval -1 failure + */ +int ast_rtp_instance_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent); + +/*! + * \brief Set the remote SSRC for an RTP instance + * \since 15.0.0 + * + * \param rtp The RTP instance + * \param ssrc The remote SSRC + */ +void ast_rtp_instance_set_remote_ssrc(struct ast_rtp_instance *rtp, unsigned int ssrc); + +/*! + * \brief Set the stream number for an RTP instance + * \since 15.0.0 + * + * \param rtp The RTP instance + * \param stream_num The stream identifier number + */ +void ast_rtp_instance_set_stream_num(struct ast_rtp_instance *instance, int stream_num); + /*! \addtogroup StasisTopicsAndMessages * @{ */ |