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支持根据情况选择是否发生rtp

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This commit is contained in:
xiongziliang 2021-04-02 21:08:40 +08:00
parent c42b678cab
commit dc485c6211
4 changed files with 19 additions and 23 deletions
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2
webrtc/DtlsTransport.hpp
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@ -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

2
webrtc/SrtpSession.hpp
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@ -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;
};

35
webrtc/WebRtcTransport.cpp
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@ -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;

3
webrtc/WebRtcTransport.h
View File

@ -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};
};