<p>Benjamin Keith Ford has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/8383">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">Add extended properties to rtp_engine for RTP retransmission support.<br><br>A couple of additional properties are needed in rtp_engine to enable<br>support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and<br>AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically<br>if an endpoint has the webrtc option enabled. While this adds no<br>functionality currently, it will serve as a building block for future<br>changes for RTP retransmission support.<br><br>For more information, refer to the wiki page:<br>https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements<br><br>Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc<br>---<br>M include/asterisk/rtp_engine.h<br>M res/res_pjsip_sdp_rtp.c<br>2 files changed, 11 insertions(+), 4 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/83/8383/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h<br>index c77be45..b1272c1 100644<br>--- a/include/asterisk/rtp_engine.h<br>+++ b/include/asterisk/rtp_engine.h<br>@@ -122,6 +122,10 @@<br> AST_RTP_PROPERTY_RTCP,<br> /*! Enable Asymmetric RTP Codecs */<br> AST_RTP_PROPERTY_ASYMMETRIC_CODEC,<br>+ /*! Enable packet retransmission for received packets */<br>+ AST_RTP_PROPERTY_RETRANS_RECV,<br>+ /*! Enable packet retransmission for sent packets */<br>+ AST_RTP_PROPERTY_RETRANS_SEND,<br> <br> /*!<br> * \brief Maximum number of RTP properties supported<br>diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c<br>index 9e04119..25ffd16 100644<br>--- a/res/res_pjsip_sdp_rtp.c<br>+++ b/res/res_pjsip_sdp_rtp.c<br>@@ -219,10 +219,13 @@<br> (session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) {<br> ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,<br> session->endpoint->media.cos_audio, "SIP RTP Audio");<br>- } else if (session_media->type == AST_MEDIA_TYPE_VIDEO &&<br>- (session->endpoint->media.tos_video || session->endpoint->media.cos_video)) {<br>- ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,<br>- session->endpoint->media.cos_video, "SIP RTP Video");<br>+ } else if (session_media->type == AST_MEDIA_TYPE_VIDEO) {<br>+ ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_RECV, session->endpoint->media.webrtc);<br>+ ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_SEND, session->endpoint->media.webrtc);<br>+ if (session->endpoint->media.tos_video || session->endpoint->media.cos_video) {<br>+ ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,<br>+ session->endpoint->media.cos_video, "SIP RTP Video");<br>+ }<br> }<br> <br> ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));<br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/8383">change 8383</a>. To unsubscribe, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/8383"/><meta itemprop="name" content="View Change"/></div></div>
<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: 15 </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>
<div style="display:none"> Gerrit-Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc </div>
<div style="display:none"> Gerrit-Change-Number: 8383 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Benjamin Keith Ford <bford@digium.com> </div>