From 0be1c388e47ebb82d9b97eff45224c242ba4718e Mon Sep 17 00:00:00 2001
From: Ben Ford <bford@digium.com>
Date: Thu, 22 Feb 2018 14:27:26 -0600
Subject: [PATCH] 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
---
include/asterisk/rtp_engine.h | 4 ++++
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 c77be4584b..3812cb159e 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 ce8ed82dfe..9f0cdd3007 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));
--
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