From ffdc13bfb9f25dff1277bdd70a20f0b4aee3b309 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?=E5=BC=A0=E4=BC=A0=E5=B3=B0?=
 <59160162+zhang-chuanfeng@users.noreply.github.com>
Date: Tue, 5 Mar 2024 10:42:22 +0800
Subject: [PATCH] =?UTF-8?q?RTP=20proxy=E9=80=9A=E8=BF=87UDP=E6=94=B6?=
 =?UTF-8?q?=E6=B5=81=EF=BC=8C=E8=B0=83=E6=95=B4udp=20recv=20socket=20buffe?=
 =?UTF-8?q?r=20size=E6=88=90=E9=85=8D=E7=BD=AE=20(#3336)?=
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国标推流有些情况需要UDP方式接收流，端口复用同一个UDP端口可能需要根据服务器性能
---
 conf/config.ini        |  3 +++
 src/Common/config.cpp  |  2 ++
 src/Common/config.h    |  2 ++
 src/Rtp/RtpServer.cpp  |  4 +++-
 src/Rtp/RtpSession.cpp | 11 +++++++----
 src/Rtp/RtpSession.h   |  1 +
 6 files changed, 18 insertions(+), 5 deletions(-)

diff --git a/conf/config.ini b/conf/config.ini
index a66de19f..21e268fd 100644
--- a/conf/config.ini
+++ b/conf/config.ini
@@ -332,6 +332,9 @@ gop_cache=1
 #国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
 #最好为20 的倍数，程序自动向20的倍数取整
 rtp_g711_dur_ms = 100
+#udp接收数据socket buffer大小配置
+#4*1024*1024=4196304
+udp_recv_socket_buffer=4194304
 
 [rtc]
 #rtc播放推流、播放超时时间
diff --git a/src/Common/config.cpp b/src/Common/config.cpp
index f31546fe..a72ea8da 100644
--- a/src/Common/config.cpp
+++ b/src/Common/config.cpp
@@ -345,6 +345,7 @@ const string kPSPT = RTP_PROXY_FIELD "ps_pt";
 const string kOpusPT = RTP_PROXY_FIELD "opus_pt";
 const string kGopCache = RTP_PROXY_FIELD "gop_cache";
 const string kRtpG711DurMs = RTP_PROXY_FIELD "rtp_g711_dur_ms";
+const string kUdpRecvSocketBuffer = RTP_PROXY_FIELD "udp_recv_socket_buffer";
 
 static onceToken token([]() {
     mINI::Instance()[kDumpDir] = "";
@@ -356,6 +357,7 @@ static onceToken token([]() {
     mINI::Instance()[kOpusPT] = 100;
     mINI::Instance()[kGopCache] = 1;
     mINI::Instance()[kRtpG711DurMs] = 100;
+    mINI::Instance()[kUdpRecvSocketBuffer] = 4 * 1024 * 1024;
 });
 } // namespace RtpProxy
 
diff --git a/src/Common/config.h b/src/Common/config.h
index 3196654b..16ef6303 100644
--- a/src/Common/config.h
+++ b/src/Common/config.h
@@ -400,6 +400,8 @@ extern const std::string kGopCache;
 //国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
 //最好为20 的倍数，程序自动向20的倍数取整
 extern const std::string kRtpG711DurMs;
+// udp recv socket buffer size
+extern const std::string kUdpRecvSocketBuffer;
 } // namespace RtpProxy
 
 /**
diff --git a/src/Rtp/RtpServer.cpp b/src/Rtp/RtpServer.cpp
index 51bb5555..560792b3 100644
--- a/src/Rtp/RtpServer.cpp
+++ b/src/Rtp/RtpServer.cpp
@@ -174,7 +174,8 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
     }
 
     //设置udp socket读缓存
-    SockUtil::setRecvBuf(rtp_socket->rawFD(), 4 * 1024 * 1024);
+    GET_CONFIG(int, udpRecvSocketBuffer, RtpProxy::kUdpRecvSocketBuffer);
+    SockUtil::setRecvBuf(rtp_socket->rawFD(), udpRecvSocketBuffer);
 
     TcpServer::Ptr tcp_server;
     _tcp_mode = tcp_mode;
@@ -223,6 +224,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
         //单端口多线程接收多个流，根据ssrc区分流
         udp_server = std::make_shared<UdpServer>(rtp_socket->getPoller());
         (*udp_server)[RtpSession::kOnlyAudio] = only_audio;
+        (*udp_server)[RtpSession::kUdpRecvBuffer] = udpRecvSocketBuffer;
         udp_server->start<RtpSession>(local_port, local_ip);
         rtp_socket = nullptr;
     }
diff --git a/src/Rtp/RtpSession.cpp b/src/Rtp/RtpSession.cpp
index 006b1c82..41c76287 100644
--- a/src/Rtp/RtpSession.cpp
+++ b/src/Rtp/RtpSession.cpp
@@ -24,6 +24,7 @@ namespace mediakit{
 const string RtpSession::kStreamID = "stream_id";
 const string RtpSession::kSSRC = "ssrc";
 const string RtpSession::kOnlyAudio = "only_audio";
+const string RtpSession::kUdpRecvBuffer = "udp_recv_socket_buffer";
 
 void RtpSession::attachServer(const Server &server) {
     setParams(const_cast<Server &>(server));
@@ -33,6 +34,12 @@ void RtpSession::setParams(mINI &ini) {
     _stream_id = ini[kStreamID];
     _ssrc = ini[kSSRC];
     _only_audio = ini[kOnlyAudio];
+    int udp_socket_buffer = ini[kUdpRecvBuffer];
+    if (_is_udp) {
+        // 设置udp socket读缓存
+        SockUtil::setRecvBuf(getSock()->rawFD(),
+            (udp_socket_buffer > 0) ? udp_socket_buffer : (4 * 1024 * 1024));
+    }
 }
 
 RtpSession::RtpSession(const Socket::Ptr &sock)
@@ -40,10 +47,6 @@ RtpSession::RtpSession(const Socket::Ptr &sock)
     socklen_t addr_len = sizeof(_addr);
     getpeername(sock->rawFD(), (struct sockaddr *)&_addr, &addr_len);
     _is_udp = sock->sockType() == SockNum::Sock_UDP;
-    if (_is_udp) {
-        // 设置udp socket读缓存
-        SockUtil::setRecvBuf(getSock()->rawFD(), 4 * 1024 * 1024);
-    }
 }
 
 RtpSession::~RtpSession() = default;
diff --git a/src/Rtp/RtpSession.h b/src/Rtp/RtpSession.h
index 5b8d7e1f..de2fa75f 100644
--- a/src/Rtp/RtpSession.h
+++ b/src/Rtp/RtpSession.h
@@ -25,6 +25,7 @@ public:
     static const std::string kStreamID;
     static const std::string kSSRC;
     static const std::string kOnlyAudio;
+    static const std::string kUdpRecvBuffer;
 
     RtpSession(const toolkit::Socket::Ptr &sock);
     ~RtpSession() override;