<p>Joshua Colp has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/8572">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">rtp: Add REMB RTP property and set it on PJSIP video RTP.<br><br>This change adds a property to RTP instances to indicate that<br>REMB support is enabled and that sending/receiving should be<br>passed through.<br><br>This also enables it on video RTP instances in PJSIP if<br>WebRTC support is enabled.<br><br>Change-Id: I1902dda1c0882bd1a0d71b2f120684b44b97e789<br>---<br>M include/asterisk/rtp_engine.h<br>M res/res_pjsip_sdp_rtp.c<br>2 files changed, 3 insertions(+), 0 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/72/8572/1</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 3812cb1..4e32d6b 100644<br>--- a/include/asterisk/rtp_engine.h<br>+++ b/include/asterisk/rtp_engine.h<br>@@ -126,6 +126,8 @@<br>  AST_RTP_PROPERTY_RETRANS_RECV,<br>        /*! Enable packet retransmission for sent packets */<br>  AST_RTP_PROPERTY_RETRANS_SEND,<br>+       /*! Enable REMB sending and receiving passthrough support */<br>+ AST_RTP_PROPERTY_REMB,<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 25ffd16..102c3b4 100644<br>--- a/res/res_pjsip_sdp_rtp.c<br>+++ b/res/res_pjsip_sdp_rtp.c<br>@@ -222,6 +222,7 @@<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>+              ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_REMB, 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></pre><p>To view, visit <a href="https://gerrit.asterisk.org/8572">change 8572</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/8572"/><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: I1902dda1c0882bd1a0d71b2f120684b44b97e789 </div>
<div style="display:none"> Gerrit-Change-Number: 8572 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Joshua Colp <jcolp@digium.com> </div>