diff options
| author | Kevin Harwell <kharwell@digium.com> | 2017-07-10 18:17:44 -0500 |
|---|---|---|
| committer | Kevin Harwell <kharwell@digium.com> | 2017-07-13 18:19:35 -0500 |
| commit | 7da6ddda30ab9291ec810fa88d4219145616bae8 (patch) | |
| tree | 89ad7fa5ae53b18a0a6412e85903ff7d8cd9d58b /include/asterisk/res_pjsip_session.h | |
| parent | 0f45c979a3de00b320e05ba93309cf412e9e2702 (diff) | |
res_pjsip: Add "webrtc" configuration option
This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:
rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.
Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.
ASTERISK-27119 #close
Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
Diffstat (limited to 'include/asterisk/res_pjsip_session.h')
| -rw-r--r-- | include/asterisk/res_pjsip_session.h | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h index eae29de04..eae11af43 100644 --- a/include/asterisk/res_pjsip_session.h +++ b/include/asterisk/res_pjsip_session.h @@ -105,6 +105,8 @@ struct ast_sip_session_media { int bundle_group; /*! \brief Whether this stream is currently bundled or not */ unsigned int bundled; + /*! \brief RTP/Media streams association identifier */ + char *msid; }; /*! |