diff options
| author | Ben Ford <bford@digium.com> | 2018-02-22 14:27:26 -0600 |
|---|---|---|
| committer | Benjamin Keith Ford <bford@digium.com> | 2018-02-23 12:56:00 -0600 |
| commit | 0be1c388e47ebb82d9b97eff45224c242ba4718e (patch) | |
| tree | 98a7783dfa2e913b5fe35c77c6bd207a114d3f59 | |
| parent | d6d520a0406f7097346edf02a5bc6749779aafd6 (diff) | |
Add extended properties to rtp_engine for RTP retransmission support.
A couple of additional properties are needed in rtp_engine to enable
support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and
AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically
if an endpoint has the webrtc option enabled. While this adds no
functionality currently, it will serve as a building block for future
changes for RTP retransmission support.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc
| -rw-r--r-- | include/asterisk/rtp_engine.h | 4 | ||||
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 11 |
2 files changed, 11 insertions, 4 deletions
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h index c77be4584..3812cb159 100644 --- a/include/asterisk/rtp_engine.h +++ b/include/asterisk/rtp_engine.h @@ -122,6 +122,10 @@ enum ast_rtp_property { AST_RTP_PROPERTY_RTCP, /*! Enable Asymmetric RTP Codecs */ AST_RTP_PROPERTY_ASYMMETRIC_CODEC, + /*! Enable packet retransmission for received packets */ + AST_RTP_PROPERTY_RETRANS_RECV, + /*! Enable packet retransmission for sent packets */ + AST_RTP_PROPERTY_RETRANS_SEND, /*! * \brief Maximum number of RTP properties supported diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index ce8ed82df..9f0cdd300 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -219,10 +219,13 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me (session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) { ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio, session->endpoint->media.cos_audio, "SIP RTP Audio"); - } else if (session_media->type == AST_MEDIA_TYPE_VIDEO && - (session->endpoint->media.tos_video || session->endpoint->media.cos_video)) { - ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video, - session->endpoint->media.cos_video, "SIP RTP Video"); + } else if (session_media->type == AST_MEDIA_TYPE_VIDEO) { + ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_RECV, session->endpoint->media.webrtc); + ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_SEND, session->endpoint->media.webrtc); + if (session->endpoint->media.tos_video || session->endpoint->media.cos_video) { + ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video, + session->endpoint->media.cos_video, "SIP RTP Video"); + } } ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL)); |