diff options
Diffstat (limited to 'include/asterisk')
| -rw-r--r-- | include/asterisk/res_pjsip.h | 2 | ||||
| -rw-r--r-- | include/asterisk/res_pjsip_session.h | 19 | ||||
| -rw-r--r-- | include/asterisk/rtp_engine.h | 52 |
3 files changed, 73 insertions, 0 deletions
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h index 2cd27d37f..d499d5514 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -688,6 +688,8 @@ 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; }; /*! diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h index e298e1f32..eae29de04 100644 --- a/include/asterisk/res_pjsip_session.h +++ b/include/asterisk/res_pjsip_session.h @@ -99,6 +99,12 @@ 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; }; /*! @@ -833,6 +839,19 @@ 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 5f439163f..20ae959e9 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 */ @@ -1507,6 +1513,20 @@ 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 * @@ -2266,6 +2286,8 @@ 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); @@ -2411,6 +2433,36 @@ 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 * @{ */ |