src/Rtcp/RtcpContext.cpp

@@ -107,6 +107,11 @@ uint32_t RtcpContextForSend::getRtt(uint32_t ssrc) const {
 }
 
 Buffer::Ptr RtcpContextForSend::createRtcpSR(uint32_t rtcp_ssrc) {
+    uint64_t now = getCurrentMillisecond();
+    if (now - _last_sr_tsp < 5000) {
+        return nullptr;
+    }
+    _last_sr_tsp = now;
     auto rtcp = RtcpSR::create(0);
     rtcp->setNtpStamp(_last_ntp_stamp_ms);
     rtcp->rtpts = htonl(_last_rtp_stamp);
@@ -117,7 +122,7 @@ Buffer::Ptr RtcpContextForSend::createRtcpSR(uint32_t rtcp_ssrc) {
     // 记录上次发送的sender report信息，用于后续统计rtt  [AUTO-TRANSLATED:1d22d2c8]
     // Record the last sent sender report information for subsequent RTT statistics
     auto last_sr_lsr = ((ntohl(rtcp->ntpmsw) & 0xFFFF) << 16) | ((ntohl(rtcp->ntplsw) >> 16) & 0xFFFF);
-    _sender_report_ntp[last_sr_lsr] = getCurrentMillisecond();
+    _sender_report_ntp[last_sr_lsr] = now;
     if (_sender_report_ntp.size() >= 5) {
         // 删除最早的sr rtcp  [AUTO-TRANSLATED:2457e08d]
         // Delete the earliest sr rtcp

src/Rtcp/RtcpContext.h

@@ -154,6 +154,7 @@ public:
     uint32_t getRtt(uint32_t ssrc) const;
 
 private:
+    uint64_t _last_sr_tsp = 0;
     std::map<uint32_t /*ssrc*/, uint32_t /*rtt*/> _rtt;
     std::map<uint32_t /*last_sr_lsr*/, uint64_t /*ntp stamp*/> _sender_report_ntp;
 

srt/SrtCaller.cpp

@@ -35,10 +35,6 @@ void SrtUrl::parse(const string &strUrl) {
     auto ip = findSubString(url.data(), "://", "?");
     splitUrl(ip, _host, _port);
 
-    if (!SockUtil::getDomainIP(_host.c_str(), _port, _addr, AF_INET, SOCK_DGRAM, IPPROTO_UDP)) {
-        throw std::invalid_argument("invalid host: " + _host);
-    }
-
     auto _params = findSubString(url.data(), "?" , NULL);
 
     auto kv = Parser::parseArgs(_params);
@@ -84,9 +80,10 @@ SrtCaller::~SrtCaller(void) {
 void SrtCaller::onConnect() {
     //DebugL;
 
+    auto peer_addr = SockUtil::make_sockaddr(_url._host.c_str(), (_url._port));
     _socket = Socket::createSocket(_poller, false);
-    _socket->bindUdpSock(0, _url._addr.ss_family == AF_INET ? "0.0.0.0" : "::");
-    _socket->bindPeerAddr((struct sockaddr *)&_url._addr, 0, true);
+    _socket->bindUdpSock(0, SockUtil::is_ipv4(_url._host.data()) ? "0.0.0.0" : "::");
+    _socket->bindPeerAddr((struct sockaddr *)&peer_addr, 0, true);
 
     weak_ptr<SrtCaller> weak_self = shared_from_this();
     _socket->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
@@ -283,7 +280,8 @@ void SrtCaller::sendHandshakeInduction() {
     req->srt_socket_id                  = _socket_id;
     req->syn_cookie                     = 0;
 
-    req->assignPeerIPBE(&_url._addr);
+    auto dataSenderAddr = SockUtil::make_sockaddr(_url._host.c_str(), _url._port);
+    req->assignPeerIPBE(&dataSenderAddr);
     req->storeToData();
     _handleshake_req = req;
     sendControlPacket(req, true);
@@ -328,7 +326,8 @@ void SrtCaller::sendHandshakeConclusion() {
     req->srt_socket_id                   = _socket_id;
     req->syn_cookie                      = _sync_cookie;
 
-    req->assignPeerIPBE(&_url._addr);
+    auto addr = SockUtil::make_sockaddr(_url._host.c_str(), _url._port);
+    req->assignPeerIPBE(&addr);
 
     HSExtMessage::Ptr ext = std::make_shared<HSExtMessage>();
     ext->extension_type = HSExt::SRT_CMD_HSREQ;

srt/SrtCaller.h

@@ -33,18 +33,15 @@ namespace mediakit {
 
 // 解析srt 信令url的工具类
 class SrtUrl {
-public:
-    void parse(const std::string &url);
-
 public:
     std::string _full_url;
     std::string _params;
-    std::string _streamid;
-    sockaddr_storage _addr;
-
-private:
-    uint16_t _port;
     std::string _host;
+    uint16_t _port;
+    std::string _streamid;
+
+public:
+    void parse(const std::string &url);
 };
 
 // 实现了webrtc代理拉流功能

webrtc/WebRtcTransport.cpp

@@ -1497,13 +1497,10 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
     sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);
     _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
 
-    if (_rtcp_sr_send_ticker.elapsedTime() > 5000) {
-        _rtcp_sr_send_ticker.resetTime();
-        if (track->rtcp_context_send) {
-            auto sr = track->rtcp_context_send->createRtcpSR(track->answer_ssrc_rtp);
-            if (sr && sr->size() > 0) {
-                sendRtcpPacket(sr->data(), sr->size(), true);
-            }
+    if (track->rtcp_context_send) {
+        auto sr = track->rtcp_context_send->createRtcpSR(track->answer_ssrc_rtp);
+        if (sr && sr->size() > 0) {
+            sendRtcpPacket(sr->data(), sr->size(), true);
         }
     }
 }

webrtc/WebRtcTransport.h

@@ -380,7 +380,6 @@ private:
     // pli rtcp timer
     toolkit::Ticker _pli_ticker;
 
-    toolkit::Ticker _rtcp_sr_send_ticker;
     toolkit::Ticker _rtcp_rr_send_ticker;
 
     // twcc rtcp发送上下文对象  [AUTO-TRANSLATED:aef6476a]