diff options
| author | Joshua Colp <jcolp@digium.com> | 2017-06-30 18:55:57 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2017-07-13 14:47:50 +0000 |
| commit | 065c3005ad920f5fe2cedcf062e38b8e28eeb015 (patch) | |
| tree | 6f10effde768ae19faa7342b1eeef1004cf2fddc /include/asterisk/res_pjsip_session.h | |
| parent | e1c0e14fac7cdca276a0fd61fa891ea54f4ab5d8 (diff) | |
res_rtp_asterisk / res_pjsip: Add support for BUNDLE.
BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.
This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.
For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.
ASTERISK-27118
Change-Id: I96c0920b9f9aca7382256484765a239017973c11
Diffstat (limited to 'include/asterisk/res_pjsip_session.h')
| -rw-r--r-- | include/asterisk/res_pjsip_session.h | 19 |
1 files changed, 19 insertions, 0 deletions
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 { \ |