支持根据情况选择是否发生rtp

webrtc/DtlsTransport.hpp

@@ -228,7 +228,7 @@ namespace RTC
 		bool handshakeDoneNow{ false };
 		std::string remoteCert;
 		//最大不超过mtu
-		static constexpr int SslReadBufferSize{ 1600 };
+		static constexpr int SslReadBufferSize{ 2000 };
         uint8_t sslReadBuffer[SslReadBufferSize];
 };
 } // namespace RTC

webrtc/SrtpSession.hpp

@@ -59,7 +59,7 @@ namespace RTC
 		// Allocated by this.
 		srtp_t session{ nullptr };
 		//rtp包最大1600
-        static constexpr size_t EncryptBufferSize{ 1600 };
+        static constexpr size_t EncryptBufferSize{ 2000 };
         uint8_t EncryptBuffer[EncryptBufferSize];
         DepLibSRTP::Ptr _env;
 	};

webrtc/WebRtcTransport.cpp

@@ -226,23 +226,17 @@ void WebRtcTransportImp::onStartWebRTC() {
     });
 }
 
-uint8_t WebRtcTransportImp::getSendPayloadType(TrackType type) {
-    for (auto &m : getSdp(SdpType::answer).media) {
-        if (m.type == type) {
-            return m.plan[0].pt;
-        }
-    }
-    return 0;
-}
-
 void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
-    //需要修改pt
-    if (rtp->type == TrackVideo) {
-        rtp->getHeader()->pt = getSendPayloadType(rtp->type);
-        sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush);
-    } else {
-
+    if (!_send_rtp_pt[rtp->type]) {
+        //忽略，对方不支持该编码类型
+        return;
     }
+    auto tmp = rtp->getHeader()->pt;
+    //设置pt
+    rtp->getHeader()->pt = _send_rtp_pt[rtp->type];
+    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush);
+    //还原pt
+    rtp->getHeader()->pt = tmp;
 }
 
 bool WebRtcTransportImp::canSendRtp() const{
@@ -250,7 +244,6 @@ bool WebRtcTransportImp::canSendRtp() const{
     return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly;
 }
 
-
 void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
     WebRtcTransport::onCheckSdp(type, sdp);
     if (type != SdpType::answer || !canSendRtp()) {
@@ -263,19 +256,21 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
         }
         m.rtp_ssrc.ssrc = _src->getSsrc(m.type);
         m.rtp_ssrc.cname = "zlmediakit-rtc";
+        auto rtsp_media = _rtsp_send_sdp.getMedia(m.type);
+        if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
+            _send_rtp_pt[m.type] = m.plan[0].pt;
+        }
     }
 }
 
 void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
     WebRtcTransport::onRtcConfigure(configure);
-
-    RtcSession sdp;
-    sdp.loadFrom(_src->getSdp(), false);
+    _rtsp_send_sdp.loadFrom(_src->getSdp(), false);
 
     configure.audio.enable = false;
     configure.video.enable = false;
 
-    for (auto &m : sdp.media) {
+    for (auto &m : _rtsp_send_sdp.media) {
         switch (m.type) {
             case TrackVideo: {
                 configure.video.enable = true;

webrtc/WebRtcTransport.h

@@ -127,7 +127,6 @@ private:
     void onDestory() override;
     void onSendRtp(const RtpPacket::Ptr &rtp, bool flush);
     SdpAttrCandidate::Ptr getIceCandidate() const;
-    uint8_t getSendPayloadType(TrackType type);
     bool canSendRtp() const;
 
 private:
@@ -135,6 +134,8 @@ private:
     RtspMediaSource::Ptr _src;
     RtspMediaSource::RingType::RingReader::Ptr _reader;
     RtcSession _answer_sdp;
+    mutable RtcSession _rtsp_send_sdp;
+    mutable uint8_t _send_rtp_pt[2] = {0, 0};
 };