From a42e5f6470000c346cf091b2517202abb39aa08c Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Mon, 10 May 2021 18:11:12 +0800 Subject: [PATCH] =?UTF-8?q?=E5=AE=8C=E5=96=84ssrc=E5=A4=84=E7=90=86?= =?UTF-8?q?=E9=80=BB=E8=BE=91?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 43 ++++++++++++++++++++++++++++---------- webrtc/WebRtcTransport.h | 11 +++++----- 2 files changed, 38 insertions(+), 16 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index d7aa1a49..bec317f5 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -185,6 +185,9 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){ if (sdp.group.mids.empty()) { throw std::invalid_argument("只支持group BUNDLE模式"); } + if (type == SdpType::offer) { + sdp.checkValidSSRC(); + } } void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const { @@ -390,6 +393,16 @@ bool WebRtcTransportImp::canRecvRtp() const{ return _push_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly); } +const RtcSession& WebRtcTransportImp::getSdpWithSSRC() const{ + auto &offer = getSdp(SdpType::offer); + if (offer.haveSSRC()) { + return offer; + } + auto &answer = getSdp(SdpType::answer); + CHECK(answer.haveSSRC()); + return answer; +} + void WebRtcTransportImp::onStartWebRTC() { //获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息 for (auto &m : getSdp(SdpType::offer).media) { @@ -398,11 +411,12 @@ void WebRtcTransportImp::onStartWebRTC() { if (!hit_pan) { continue; } + auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type); //获取offer端rtp的ssrc和pt相关信息 auto &ref = _rtp_info_pt[plan.pt]; - _rtp_info_ssrc[m.rtp_rtx_ssrc[0].ssrc] = &ref; + _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref; ref.plan = &plan; - ref.media = &m; + ref.media = m_with_ssrc; ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); @@ -441,14 +455,16 @@ void WebRtcTransportImp::onStartWebRTC() { RtcSession rtsp_send_sdp; rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); - for (auto &m : getSdp(SdpType::answer).media) { + for (auto &m : getSdp(SdpType::answer).media) { if (m.type == TrackApplication) { continue; } auto rtsp_media = rtsp_send_sdp.getMedia(m.type); if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { - //记录发送rtp时约定的pt，届时发送rtp时需要修改pt - _send_rtp_pt[m.type] = m.plan[0].pt; + auto it = _rtp_info_pt.find(m.plan[0].pt); + CHECK(it != _rtp_info_pt.end()); + //记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc + _send_rtp_info[m.type] = &it->second; } } } @@ -457,10 +473,11 @@ void WebRtcTransportImp::onStartWebRTC() { void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ WebRtcTransport::onCheckSdp(type, sdp); if (type != SdpType::answer) { + //我们只修改answer sdp return; } - //修改sdp的ip、端口信息 + //修改answer sdp的ip、端口信息 GET_CONFIG(string, extern_ip, RTC::kExternIP); for (auto &m : sdp.media) { m.addr.reset(); @@ -472,7 +489,8 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ sdp.origin.address = m.addr.address; } - if (!canSendRtp()) { + if (!canSendRtp() || getSdp(SdpType::offer).haveSSRC()) { + //offer sdp未包含ssrc相关信息，那么我们才在answer sdp中回复ssrc相关信息 return; } @@ -686,20 +704,23 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp } void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ - auto pt = _send_rtp_pt[rtp->type]; - if (pt == 0xFF) { + auto info = _send_rtp_info[rtp->type]; + if (!info) { //忽略，对方不支持该编码类型 return; } _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type); //统计rtp发送情况，好做sr汇报 - _rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); + _rtp_info_pt[info->plan->pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) { auto header = (RtpHeader *)buf; - header->pt = _send_rtp_pt[type]; + auto info = _send_rtp_info[type]; + //修改目标pt和ssrc + header->pt = info->plan->pt; + header->ssrc = htons(info->media->rtp_rtx_ssrc[0].ssrc); changeRtpExtId(header, _rtp_ext_type_to_id); } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 44fdf973..9e6116c1 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -188,6 +188,7 @@ private: SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; bool canRecvRtp() const; + const RtcSession& getSdpWithSSRC() const; class RtpPayloadInfo { public: @@ -215,8 +216,8 @@ private: Ticker _alive_ticker; //pli rtcp计时器 Ticker _pli_ticker; - //记录协商的rtp的pt类型 - uint8_t _send_rtp_pt[2] = {0xFF, 0xFF}; + //记录协商的发送rtp的pt和ssrc + RtpPayloadInfo* _send_rtp_info[2] = {nullptr, nullptr}; //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; //推流的rtsp源 @@ -226,9 +227,9 @@ private: //播放rtsp源的reader对象 RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 - unordered_map _rtp_info_pt; - //根据推流端rtcp的ssrc获取相关信息 - unordered_map _rtp_info_ssrc; + unordered_map _rtp_info_pt; + //根据rtcp的ssrc获取相关信息 + unordered_map _rtp_info_ssrc; //发送rtp时需要修改rtp ext id map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id