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完善nack/srtp相关代码

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This commit is contained in:
ziyue 2021-07-28 11:18:09 +08:00
parent 80ab84cbb5
commit 5c52c636a3
4 changed files with 8 additions and 17 deletions
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2
webrtc/Nack.cpp
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@ -192,7 +192,7 @@ uint64_t NackContext::reSendNack() {
it = _nack_send_status.erase(it); it = _nack_send_status.erase(it);
continue; continue;
} }
if (now - it->second.update_stamp < 2 * _rtt) { if (now - it->second.update_stamp < kNackIntervalRatio * _rtt) {
//距离上次nack不足2倍的rtt，不用再发送nack //距离上次nack不足2倍的rtt，不用再发送nack
++it; ++it;
continue; continue;

4
webrtc/Nack.h
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@ -39,11 +39,13 @@ public:
using Ptr = std::shared_ptr<NackContext>; using Ptr = std::shared_ptr<NackContext>;
using onNack = function<void(const FCI_NACK &nack)>; using onNack = function<void(const FCI_NACK &nack)>;
//最大保留的rtp丢包状态个数 //最大保留的rtp丢包状态个数
static constexpr auto kNackMaxSize = 100; static constexpr auto kNackMaxSize = 1024;
//rtp丢包状态最长保留时间 //rtp丢包状态最长保留时间
static constexpr auto kNackMaxMS = 3 * 1000; static constexpr auto kNackMaxMS = 3 * 1000;
//nack最多请求重传10次 //nack最多请求重传10次
static constexpr auto kNackMaxCount = 10; static constexpr auto kNackMaxCount = 10;
//nack重传频率，rtt的倍数
static constexpr auto kNackIntervalRatio = 2.0f;
NackContext() = default; NackContext() = default;
~NackContext() = default; ~NackContext() = default;

16
webrtc/WebRtcTransport.cpp
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@ -97,12 +97,7 @@ void WebRtcTransport::OnDtlsTransportConnected(
std::string &remoteCert) { std::string &remoteCert) {
InfoL; InfoL;
_srtp_session_send = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen); _srtp_session_send = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen);
_srtp_session_recv = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
string srtpRemoteKey_str((char *) srtpRemoteKey, srtpRemoteKeyLen);
_srtp_session_recv_alloc = [srtpCryptoSuite, srtpRemoteKey_str]() {
return std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite,
(uint8_t *) srtpRemoteKey_str.data(), srtpRemoteKey_str.size());
};
onStartWebRTC(); onStartWebRTC();
} }
@ -255,19 +250,14 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
_dtls_transport->ProcessDtlsData((uint8_t *) buf, len); _dtls_transport->ProcessDtlsData((uint8_t *) buf, len);
return; return;
} }
RtpHeader *rtp = (RtpHeader *) buf;
auto it = _srtp_session_recv.find(rtp->pt);
if (it == _srtp_session_recv.end()) {
it = _srtp_session_recv.emplace((uint8_t) rtp->pt, _srtp_session_recv_alloc()).first;
}
if (is_rtp(buf)) { if (is_rtp(buf)) {
if (it->second->DecryptSrtp((uint8_t *) buf, &len)) { if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
onRtp(buf, len); onRtp(buf, len);
} }
return; return;
} }
if (is_rtcp(buf)) { if (is_rtcp(buf)) {
if (it->second->DecryptSrtcp((uint8_t *) buf, &len)) { if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
onRtcp(buf, len); onRtcp(buf, len);
} }
return; return;

3
webrtc/WebRtcTransport.h
View File

@ -122,10 +122,9 @@ private:
std::shared_ptr<RTC::IceServer> _ice_server; std::shared_ptr<RTC::IceServer> _ice_server;
std::shared_ptr<RTC::DtlsTransport> _dtls_transport; std::shared_ptr<RTC::DtlsTransport> _dtls_transport;
std::shared_ptr<RTC::SrtpSession> _srtp_session_send; std::shared_ptr<RTC::SrtpSession> _srtp_session_send;
std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
RtcSession::Ptr _offer_sdp; RtcSession::Ptr _offer_sdp;
RtcSession::Ptr _answer_sdp; RtcSession::Ptr _answer_sdp;
function<std::shared_ptr<RTC::SrtpSession>() > _srtp_session_recv_alloc;
std::unordered_map<uint8_t /*pt*/, std::shared_ptr<RTC::SrtpSession> > _srtp_session_recv;
}; };
class RtpChannel; class RtpChannel;