ZLMediaKit/src/Rtsp/RtspSession.cpp

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/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
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*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
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*
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* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
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*/
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#include <atomic>
#include <iomanip>
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#include "Common/config.h"
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#include "UDPServer.h"
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#include "RtspSession.h"
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#include "Util/MD5.h"
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#include "Util/base64.h"
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using namespace std;
using namespace toolkit;
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namespace mediakit {
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/**
* rtsp协议有多种方式传输rtp数据包4
* 1: rtp over udp ,rtp通过单独的udp端口传输
* 2: rtp over udp_multicast,rtp通过共享udp组播端口传输
* 3: rtp over tcp,rtsp信令tcp通道完成传输
* 4: rtp over httprtp over http
*
* rtp over http rtsp协议伪装成http协议以达到穿透防火墙的目的
* http请求至rtsp服务器http get请求
* http post请求
*
* http请求头中的x-sessioncookie键完成绑定
*
* http get请求用于接收rtprtcp和rtsp回复
* http post请求用于发送rtsp请求rtsp握手结束后可能会断开连接rtp发送
* http post请求中的content负载就是base64编码后的rtsp请求包
* rtsp请求伪装成http content负载发送至rtsp服务器rtsp服务器又把回复发送给第一次http get请求的tcp链接
* rtsp会话就是两次http请求
*
* zlmediakit在处理rtsp over http的请求时http poster中的content数据base64解码后转发给http getter处理
*/
//rtsp over http 情况下get请求实例在请求实例用于接收rtp数据包
static unordered_map<string, weak_ptr<RtspSession> > g_mapGetter;
//对g_mapGetter上锁保护
static recursive_mutex g_mtxGetter;
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RtspSession::RtspSession(const Socket::Ptr &sock) : TcpSession(sock) {
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DebugP(this);
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GET_CONFIG(uint32_t,keep_alive_sec,Rtsp::kKeepAliveSecond);
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sock->setSendTimeOutSecond(keep_alive_sec);
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}
RtspSession::~RtspSession() {
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DebugP(this);
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}
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void RtspSession::onError(const SockException &err) {
bool isPlayer = !_push_src;
uint64_t duration = _alive_ticker.createdTime() / 1000;
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WarnP(this) << (isPlayer ? "RTSP播放器(" : "RTSP推流器(")
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<< _media_info._vhost << "/"
<< _media_info._app << "/"
<< _media_info._streamid
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<< ")断开:" << err.what()
<< ",耗时(s):" << duration;
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if (_rtp_type == Rtsp::RTP_MULTICAST) {
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//取消UDP端口监听
UDPServer::Instance().stopListenPeer(get_peer_ip().data(), this);
}
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if (_http_x_sessioncookie.size() != 0) {
//移除http getter的弱引用记录
lock_guard<recursive_mutex> lock(g_mtxGetter);
g_mapGetter.erase(_http_x_sessioncookie);
}
//流量统计事件广播
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GET_CONFIG(uint32_t,iFlowThreshold,General::kFlowThreshold);
if(_bytes_usage >= iFlowThreshold * 1024){
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NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, isPlayer, static_cast<SockInfo &>(*this));
}
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}
void RtspSession::onManager() {
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GET_CONFIG(uint32_t,handshake_sec,Rtsp::kHandshakeSecond);
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GET_CONFIG(uint32_t,keep_alive_sec,Rtsp::kKeepAliveSecond);
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if (_alive_ticker.createdTime() > handshake_sec * 1000) {
if (_sessionid.size() == 0) {
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shutdown(SockException(Err_timeout,"illegal connection"));
return;
}
}
if ((_rtp_type == Rtsp::RTP_UDP || _push_src ) && _alive_ticker.elapsedTime() > keep_alive_sec * 1000 && _enable_send_rtp) {
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//如果是推流端或者rtp over udp类型的播放端那么就做超时检测
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shutdown(SockException(Err_timeout,"rtp over udp session timeouted"));
return;
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}
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}
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void RtspSession::onRecv(const Buffer::Ptr &buf) {
_alive_ticker.resetTime();
_bytes_usage += buf->size();
if (_on_recv) {
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//http poster的请求数据转发给http getter处理
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_on_recv(buf);
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} else {
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input(buf->data(), buf->size());
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}
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}
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void RtspSession::onWholeRtspPacket(Parser &parser) {
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string method = parser.Method(); //提取出请求命令字
_cseq = atoi(parser["CSeq"].data());
if(_content_base.empty() && method != "GET"){
_content_base = parser.Url();
_media_info.parse(parser.FullUrl());
_media_info._schema = RTSP_SCHEMA;
}
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typedef void (RtspSession::*rtsp_request_handler)(const Parser &parser);
static unordered_map<string, rtsp_request_handler> s_cmd_functions;
static onceToken token( []() {
s_cmd_functions.emplace("OPTIONS",&RtspSession::handleReq_Options);
s_cmd_functions.emplace("DESCRIBE",&RtspSession::handleReq_Describe);
s_cmd_functions.emplace("ANNOUNCE",&RtspSession::handleReq_ANNOUNCE);
s_cmd_functions.emplace("RECORD",&RtspSession::handleReq_RECORD);
s_cmd_functions.emplace("SETUP",&RtspSession::handleReq_Setup);
s_cmd_functions.emplace("PLAY",&RtspSession::handleReq_Play);
s_cmd_functions.emplace("PAUSE",&RtspSession::handleReq_Pause);
s_cmd_functions.emplace("TEARDOWN",&RtspSession::handleReq_Teardown);
s_cmd_functions.emplace("GET",&RtspSession::handleReq_Get);
s_cmd_functions.emplace("POST",&RtspSession::handleReq_Post);
s_cmd_functions.emplace("SET_PARAMETER",&RtspSession::handleReq_SET_PARAMETER);
s_cmd_functions.emplace("GET_PARAMETER",&RtspSession::handleReq_SET_PARAMETER);
}, []() {});
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auto it = s_cmd_functions.find(method);
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if (it == s_cmd_functions.end()) {
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sendRtspResponse("403 Forbidden");
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shutdown(SockException(Err_shutdown,StrPrinter << "403 Forbidden:" << method));
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return;
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}
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auto &fun = it->second;
try {
(this->*fun)(parser);
}catch (SockException &ex){
if(ex){
shutdown(ex);
}
}catch (exception &ex){
shutdown(SockException(Err_shutdown,ex.what()));
}
parser.Clear();
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}
void RtspSession::onRtpPacket(const char *data, size_t len) {
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uint8_t interleaved = data[1];
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if (interleaved % 2 == 0) {
if (!_push_src) {
return;
}
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auto track_idx = getTrackIndexByInterleaved(interleaved);
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handleOneRtp(track_idx, _sdp_track[track_idx]->_type, _sdp_track[track_idx]->_samplerate, (uint8_t *) data + RtpPacket::kRtpTcpHeaderSize, len - RtpPacket::kRtpTcpHeaderSize);
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} else {
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auto track_idx = getTrackIndexByInterleaved(interleaved - 1);
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onRtcpPacket(track_idx, _sdp_track[track_idx], data + RtpPacket::kRtpTcpHeaderSize, len - RtpPacket::kRtpTcpHeaderSize);
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}
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}
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void RtspSession::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, const char *data, size_t len){
auto rtcp_arr = RtcpHeader::loadFromBytes((char *) data, len);
for (auto &rtcp : rtcp_arr) {
_rtcp_context[track_idx]->onRtcp(rtcp);
}
}
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ssize_t RtspSession::getContentLength(Parser &parser) {
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if(parser.Method() == "POST"){
//http post请求的content数据部分是base64编码后的rtsp请求信令包
return remainDataSize();
}
return RtspSplitter::getContentLength(parser);
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}
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void RtspSession::handleReq_Options(const Parser &parser) {
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//支持这些命令
sendRtspResponse("200 OK",{"Public" , "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, ANNOUNCE, RECORD, SET_PARAMETER, GET_PARAMETER"});
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}
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void RtspSession::handleReq_ANNOUNCE(const Parser &parser) {
auto src = dynamic_pointer_cast<RtspMediaSource>(MediaSource::find(RTSP_SCHEMA,
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_media_info._vhost,
_media_info._app,
_media_info._streamid));
if (src) {
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sendRtspResponse("406 Not Acceptable", {"Content-Type", "text/plain"}, "Already publishing.");
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string err = StrPrinter << "ANNOUNCE:"
<< "Already publishing:"
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<< _media_info._vhost << " "
<< _media_info._app << " "
<< _media_info._streamid << endl;
throw SockException(Err_shutdown, err);
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}
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auto full_url = parser.FullUrl();
_content_base = full_url;
if (end_with(full_url, ".sdp")) {
//去除.sdp后缀防止EasyDarwin推流器强制添加.sdp后缀
full_url = full_url.substr(0, full_url.length() - 4);
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_media_info.parse(full_url);
}
if (_media_info._app.empty() || _media_info._streamid.empty()) {
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//推流rtsp url必须最少两级(rtsp://host/app/stream_id)不允许莫名其妙的推流url
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static constexpr auto err = "rtsp推流url非法,最少确保两级rtsp url";
sendRtspResponse("403 Forbidden", {"Content-Type", "text/plain"}, err);
throw SockException(Err_shutdown, StrPrinter << err << ":" << full_url);
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}
auto onRes = [this, parser, full_url](const string &err, bool enableHls, bool enableMP4){
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bool authSuccess = err.empty();
if (!authSuccess) {
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sendRtspResponse("401 Unauthorized", {"Content-Type", "text/plain"}, err);
shutdown(SockException(Err_shutdown, StrPrinter << "401 Unauthorized:" << err));
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return;
}
SdpParser sdpParser(parser.Content());
_sessionid = makeRandStr(12);
_sdp_track = sdpParser.getAvailableTrack();
if (_sdp_track.empty()) {
//sdp无效
static constexpr auto err = "sdp中无有效track";
sendRtspResponse("403 Forbidden", {"Content-Type", "text/plain"}, err);
shutdown(SockException(Err_shutdown, StrPrinter << err << ":" << full_url));
return;
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}
_rtcp_context.clear();
for (auto &track : _sdp_track) {
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, true));
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}
_push_src = std::make_shared<RtspMediaSourceImp>(_media_info._vhost, _media_info._app, _media_info._streamid);
_push_src->setListener(dynamic_pointer_cast<MediaSourceEvent>(shared_from_this()));
_push_src->setProtocolTranslation(enableHls, enableMP4);
_push_src->setSdp(sdpParser.toString());
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sendRtspResponse("200 OK");
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};
weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
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Broadcast::PublishAuthInvoker invoker = [weakSelf, onRes](const string &err, bool enableHls, bool enableMP4) {
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auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
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return;
}
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strongSelf->async([weakSelf, onRes, err, enableHls, enableMP4]() {
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auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
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return;
}
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onRes(err, enableHls, enableMP4);
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});
};
//rtsp推流需要鉴权
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auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish, _media_info, invoker, static_cast<SockInfo &>(*this));
if (!flag) {
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//该事件无人监听,默认不鉴权
GET_CONFIG(bool, toHls, General::kPublishToHls);
GET_CONFIG(bool, toMP4, General::kPublishToMP4);
onRes("", toHls, toMP4);
}
}
void RtspSession::handleReq_RECORD(const Parser &parser){
if (_sdp_track.empty() || parser["Session"] != _sessionid) {
send_SessionNotFound();
throw SockException(Err_shutdown, _sdp_track.empty() ? "can not find any availabe track when record" : "session not found when record");
}
_StrPrinter rtp_info;
for (auto &track : _sdp_track) {
if (track->_inited == false) {
//还有track没有setup
shutdown(SockException(Err_shutdown, "track not setuped"));
return;
}
rtp_info << "url=" << track->getControlUrl(_content_base) << ",";
}
rtp_info.pop_back();
sendRtspResponse("200 OK", {"RTP-Info", rtp_info});
if (_rtp_type == Rtsp::RTP_TCP) {
//如果是rtsp推流服务器并且是TCP推流设置socket flags,,这样能提升接收性能
setSocketFlags();
}
}
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void RtspSession::emitOnPlay(){
weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
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//url鉴权回调
auto onRes = [weakSelf](const string &err) {
auto strongSelf = weakSelf.lock();
if (!strongSelf) {
return;
}
if (!err.empty()) {
//播放url鉴权失败
strongSelf->sendRtspResponse("401 Unauthorized", {"Content-Type", "text/plain"}, err);
strongSelf->shutdown(SockException(Err_shutdown, StrPrinter << "401 Unauthorized:" << err));
return;
}
strongSelf->onAuthSuccess();
};
Broadcast::AuthInvoker invoker = [weakSelf, onRes](const string &err) {
auto strongSelf = weakSelf.lock();
if (!strongSelf) {
return;
}
strongSelf->async([onRes, err, weakSelf]() {
onRes(err);
});
};
//广播通用播放url鉴权事件
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auto flag = _emit_on_play ? false : NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPlayed, _media_info, invoker, static_cast<SockInfo &>(*this));
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if (!flag) {
//该事件无人监听,默认不鉴权
onRes("");
}
//已经鉴权过了
_emit_on_play = true;
}
void RtspSession::handleReq_Describe(const Parser &parser) {
//该请求中的认证信息
auto authorization = parser["Authorization"];
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weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
//rtsp专属鉴权是否开启事件回调
onGetRealm invoker = [weakSelf, authorization](const string &realm) {
auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
//本对象已经销毁
return;
}
//切换到自己的线程然后执行
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strongSelf->async([weakSelf, realm, authorization]() {
auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
//本对象已经销毁
return;
}
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if (realm.empty()) {
//无需rtsp专属认证, 那么继续url通用鉴权认证(on_play)
strongSelf->emitOnPlay();
return;
}
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//该流需要rtsp专属认证开启rtsp专属认证后将不再触发url通用鉴权认证(on_play)
strongSelf->_rtsp_realm = realm;
strongSelf->onAuthUser(realm, authorization);
});
};
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if(_rtsp_realm.empty()){
//广播是否需要rtsp专属认证事件
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if (!NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastOnGetRtspRealm, _media_info, invoker, static_cast<SockInfo &>(*this))) {
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//无人监听此事件,说明无需认证
invoker("");
}
}else{
invoker(_rtsp_realm);
}
}
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void RtspSession::onAuthSuccess() {
TraceP(this);
weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
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MediaSource::findAsync(_media_info, weakSelf.lock(), [weakSelf](const MediaSource::Ptr &src){
auto strongSelf = weakSelf.lock();
if(!strongSelf){
return;
}
auto rtsp_src = dynamic_pointer_cast<RtspMediaSource>(src);
if (!rtsp_src) {
//未找到相应的MediaSource
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string err = StrPrinter << "no such stream:" << strongSelf->_media_info._vhost << " " << strongSelf->_media_info._app << " " << strongSelf->_media_info._streamid;
strongSelf->send_StreamNotFound();
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strongSelf->shutdown(SockException(Err_shutdown,err));
return;
}
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//找到了相应的rtsp流
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strongSelf->_sdp_track = SdpParser(rtsp_src->getSdp()).getAvailableTrack();
if (strongSelf->_sdp_track.empty()) {
//该流无效
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WarnL << "sdp中无有效track该流无效:" << rtsp_src->getSdp();
strongSelf->send_StreamNotFound();
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strongSelf->shutdown(SockException(Err_shutdown,"can not find any available track in sdp"));
return;
}
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strongSelf->_rtcp_context.clear();
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for (auto &track : strongSelf->_sdp_track) {
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strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, false));
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}
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strongSelf->_sessionid = makeRandStr(12);
strongSelf->_play_src = rtsp_src;
for(auto &track : strongSelf->_sdp_track){
track->_ssrc = rtsp_src->getSsrc(track->_type);
track->_seq = rtsp_src->getSeqence(track->_type);
track->_time_stamp = rtsp_src->getTimeStamp(track->_type);
}
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strongSelf->sendRtspResponse("200 OK",
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{"Content-Base", strongSelf->_content_base + "/",
"x-Accept-Retransmit","our-retransmit",
"x-Accept-Dynamic-Rate","1"
},rtsp_src->getSdp());
});
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}
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void RtspSession::onAuthFailed(const string &realm,const string &why,bool close) {
GET_CONFIG(bool,authBasic,Rtsp::kAuthBasic);
if (!authBasic) {
//我们需要客户端优先以md5方式认证
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_auth_nonce = makeRandStr(32);
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sendRtspResponse("401 Unauthorized",
{"WWW-Authenticate",
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StrPrinter << "Digest realm=\"" << realm << "\",nonce=\"" << _auth_nonce << "\"" });
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}else {
//当然我们也支持base64认证,但是我们不建议这样做
sendRtspResponse("401 Unauthorized",
{"WWW-Authenticate",
StrPrinter << "Basic realm=\"" << realm << "\"" });
}
if(close){
shutdown(SockException(Err_shutdown,StrPrinter << "401 Unauthorized:" << why));
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}
}
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void RtspSession::onAuthBasic(const string &realm,const string &auth_base64){
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//base64认证
char user_pwd_buf[512];
av_base64_decode((uint8_t *) user_pwd_buf, auth_base64.data(), (int)auth_base64.size());
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auto user_pwd_vec = split(user_pwd_buf, ":");
if (user_pwd_vec.size() < 2) {
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//认证信息格式不合法回复401 Unauthorized
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onAuthFailed(realm, "can not find user and passwd when basic64 auth");
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return;
}
auto user = user_pwd_vec[0];
auto pwd = user_pwd_vec[1];
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weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
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onAuth invoker = [pwd, realm, weakSelf](bool encrypted, const string &good_pwd) {
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auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
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//本对象已经销毁
return;
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}
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//切换到自己的线程执行
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strongSelf->async([weakSelf, good_pwd, pwd, realm]() {
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auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
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//本对象已经销毁
return;
}
//base64忽略encrypted参数上层必须传入明文密码
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if (pwd == good_pwd) {
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//提供的密码且匹配正确
strongSelf->onAuthSuccess();
return;
}
//密码错误
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strongSelf->onAuthFailed(realm, StrPrinter << "password mismatch when base64 auth:" << pwd << " != " << good_pwd);
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});
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};
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//此时必须提供明文密码
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if (!NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastOnRtspAuth, _media_info, realm, user, true, invoker, static_cast<SockInfo &>(*this))) {
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//表明该流需要认证却没监听请求密码事件,这一般是大意的程序所为,警告之
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WarnP(this) << "请监听kBroadcastOnRtspAuth事件";
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//但是我们还是忽略认证以便完成播放
//我们输入的密码是明文
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invoker(false, pwd);
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}
}
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void RtspSession::onAuthDigest(const string &realm,const string &auth_md5){
DebugP(this) << auth_md5;
auto mapTmp = Parser::parseArgs(auth_md5, ",", "=");
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decltype(mapTmp) map;
for(auto &pr : mapTmp){
map[trim(string(pr.first)," \"")] = trim(pr.second," \"");
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}
//check realm
if(realm != map["realm"]){
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onAuthFailed(realm,StrPrinter << "realm not mached:" << realm << " != " << map["realm"]);
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return ;
}
//check nonce
auto nonce = map["nonce"];
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if(_auth_nonce != nonce){
onAuthFailed(realm,StrPrinter << "nonce not mached:" << nonce << " != " << _auth_nonce);
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return ;
}
//check username and uri
auto username = map["username"];
auto uri = map["uri"];
auto response = map["response"];
if(username.empty() || uri.empty() || response.empty()){
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onAuthFailed(realm,StrPrinter << "username/uri/response empty:" << username << "," << uri << "," << response);
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return ;
}
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auto realInvoker = [this,realm,nonce,uri,username,response](bool ignoreAuth,bool encrypted,const string &good_pwd){
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if(ignoreAuth){
//忽略认证
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TraceP(this) << "auth ignored";
onAuthSuccess();
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return;
}
/*
response计算方法如下
RTSP客户端应该使用username + password并计算response如下:
(1)password为MD5编码,
response = md5( password:nonce:md5(public_method:url) );
(2)password为ANSI字符串,
response= md5( md5(username:realm:password):nonce:md5(public_method:url) );
*/
auto encrypted_pwd = good_pwd;
if(!encrypted){
//提供的是明文密码
encrypted_pwd = MD5(username+ ":" + realm + ":" + good_pwd).hexdigest();
}
auto good_response = MD5( encrypted_pwd + ":" + nonce + ":" + MD5(string("DESCRIBE") + ":" + uri).hexdigest()).hexdigest();
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if(strcasecmp(good_response.data(),response.data()) == 0){
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//认证成功md5不区分大小写
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onAuthSuccess();
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}else{
//认证失败!
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onAuthFailed(realm, StrPrinter << "password mismatch when md5 auth:" << good_response << " != " << response );
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}
};
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weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
onAuth invoker = [realInvoker,weakSelf](bool encrypted,const string &good_pwd){
auto strongSelf = weakSelf.lock();
if(!strongSelf){
return;
}
//切换到自己的线程确保realInvoker执行时this指针有效
strongSelf->async([realInvoker,weakSelf,encrypted,good_pwd](){
auto strongSelf = weakSelf.lock();
if(!strongSelf){
return;
}
realInvoker(false,encrypted,good_pwd);
});
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};
//此时可以提供明文或md5加密的密码
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if(!NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastOnRtspAuth, _media_info, realm, username, false, invoker, static_cast<SockInfo &>(*this))){
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//表明该流需要认证却没监听请求密码事件,这一般是大意的程序所为,警告之
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WarnP(this) << "请监听kBroadcastOnRtspAuth事件";
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//但是我们还是忽略认证以便完成播放
realInvoker(true,true,"");
}
}
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void RtspSession::onAuthUser(const string &realm,const string &authorization){
if(authorization.empty()){
onAuthFailed(realm,"", false);
return;
}
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//请求中包含认证信息
auto authType = FindField(authorization.data(),NULL," ");
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auto authStr = FindField(authorization.data()," ",NULL);
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if(authType.empty() || authStr.empty()){
//认证信息格式不合法回复401 Unauthorized
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onAuthFailed(realm,"can not find auth type or auth string");
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return;
}
if(authType == "Basic"){
//base64认证需要明文密码
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onAuthBasic(realm,authStr);
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}else if(authType == "Digest"){
//md5认证
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onAuthDigest(realm,authStr);
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}else{
//其他认证方式?不支持!
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onAuthFailed(realm,StrPrinter << "unsupported auth type:" << authType);
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}
}
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void RtspSession::send_StreamNotFound() {
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sendRtspResponse("404 Stream Not Found",{"Connection","Close"});
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}
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void RtspSession::send_UnsupportedTransport() {
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sendRtspResponse("461 Unsupported Transport",{"Connection","Close"});
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}
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void RtspSession::send_SessionNotFound() {
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sendRtspResponse("454 Session Not Found",{"Connection","Close"});
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}
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void RtspSession::handleReq_Setup(const Parser &parser) {
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//处理setup命令该函数可能进入多次
int trackIdx = getTrackIndexByControlUrl(parser.FullUrl());
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SdpTrack::Ptr &trackRef = _sdp_track[trackIdx];
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if (trackRef->_inited) {
//已经初始化过该Track
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throw SockException(Err_shutdown, "can not setup one track twice");
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}
trackRef->_inited = true; //现在初始化
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if(_rtp_type == Rtsp::RTP_Invalid){
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auto &strTransport = parser["Transport"];
if(strTransport.find("TCP") != string::npos){
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_rtp_type = Rtsp::RTP_TCP;
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}else if(strTransport.find("multicast") != string::npos){
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_rtp_type = Rtsp::RTP_MULTICAST;
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}else{
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_rtp_type = Rtsp::RTP_UDP;
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}
}
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//允许接收rtp、rtcp包
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RtspSplitter::enableRecvRtp(_rtp_type == Rtsp::RTP_TCP);
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switch (_rtp_type) {
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case Rtsp::RTP_TCP: {
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if(_push_src){
//rtsp推流时interleaved由推流者决定
auto key_values = Parser::parseArgs(parser["Transport"],";","=");
int interleaved_rtp = -1 , interleaved_rtcp = -1;
if(2 == sscanf(key_values["interleaved"].data(),"%d-%d",&interleaved_rtp,&interleaved_rtcp)){
trackRef->_interleaved = interleaved_rtp;
}else{
throw SockException(Err_shutdown, "can not find interleaved when setup of rtp over tcp");
}
}else{
//rtsp播放时由于数据共享分发所以interleaved必须由服务器决定
trackRef->_interleaved = 2 * trackRef->_type;
}
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sendRtspResponse("200 OK",
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{"Transport", StrPrinter << "RTP/AVP/TCP;unicast;"
<< "interleaved=" << (int) trackRef->_interleaved << "-"
<< (int) trackRef->_interleaved + 1 << ";"
<< "ssrc=" << printSSRC(trackRef->_ssrc),
"x-Transport-Options", "late-tolerance=1.400000",
"x-Dynamic-Rate", "1"
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});
}
break;
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case Rtsp::RTP_UDP: {
std::pair<Socket::Ptr, Socket::Ptr> pr = std::make_pair(createSocket(),createSocket());
try {
makeSockPair(pr, get_local_ip());
} catch (std::exception &ex) {
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//分配端口失败
send_NotAcceptable();
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throw SockException(Err_shutdown, ex.what());
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}
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_rtp_socks[trackIdx] = pr.first;
_rtcp_socks[trackIdx] = pr.second;
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//设置客户端内网端口信息
string strClientPort = FindField(parser["Transport"].data(), "client_port=", NULL);
uint16_t ui16RtpPort = atoi(FindField(strClientPort.data(), NULL, "-").data());
uint16_t ui16RtcpPort = atoi(FindField(strClientPort.data(), "-", NULL).data());
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struct sockaddr_in peerAddr;
//设置rtp发送目标地址
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peerAddr.sin_family = AF_INET;
peerAddr.sin_port = htons(ui16RtpPort);
peerAddr.sin_addr.s_addr = inet_addr(get_peer_ip().data());
bzero(&(peerAddr.sin_zero), sizeof peerAddr.sin_zero);
pr.first->setSendPeerAddr((struct sockaddr *) (&peerAddr));
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//设置rtcp发送目标地址
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peerAddr.sin_family = AF_INET;
peerAddr.sin_port = htons(ui16RtcpPort);
peerAddr.sin_addr.s_addr = inet_addr(get_peer_ip().data());
bzero(&(peerAddr.sin_zero), sizeof peerAddr.sin_zero);
pr.second->setSendPeerAddr((struct sockaddr *) (&peerAddr));
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//尝试获取客户端nat映射地址
startListenPeerUdpData(trackIdx);
//InfoP(this) << "分配端口:" << srv_port;
sendRtspResponse("200 OK",
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{"Transport", StrPrinter << "RTP/AVP/UDP;unicast;"
<< "client_port=" << strClientPort << ";"
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<< "server_port=" << pr.first->get_local_port() << "-"
<< pr.second->get_local_port() << ";"
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<< "ssrc=" << printSSRC(trackRef->_ssrc)
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});
}
break;
case Rtsp::RTP_MULTICAST: {
if(!_multicaster){
_multicaster = RtpMultiCaster::get(*this, get_local_ip(), _media_info._vhost, _media_info._app, _media_info._streamid);
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if (!_multicaster) {
send_NotAcceptable();
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throw SockException(Err_shutdown, "can not get a available udp multicast socket");
}
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weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
_multicaster->setDetachCB(this, [weakSelf]() {
auto strongSelf = weakSelf.lock();
if(!strongSelf) {
return;
}
strongSelf->safeShutdown(SockException(Err_shutdown,"ring buffer detached"));
});
}
int iSrvPort = _multicaster->getMultiCasterPort(trackRef->_type);
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//我们用trackIdx区分rtp和rtcp包
//由于组播udp端口是共享的而rtcp端口为组播udp端口+1所以rtcp端口需要改成共享端口
auto pSockRtcp = UDPServer::Instance().getSock(*this, get_local_ip().data(), 2 * trackIdx + 1, iSrvPort + 1);
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if (!pSockRtcp) {
//分配端口失败
send_NotAcceptable();
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throw SockException(Err_shutdown, "open shared rtcp socket failed");
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}
startListenPeerUdpData(trackIdx);
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GET_CONFIG(uint32_t,udpTTL,MultiCast::kUdpTTL);
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sendRtspResponse("200 OK",
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{"Transport", StrPrinter << "RTP/AVP;multicast;"
<< "destination=" << _multicaster->getMultiCasterIP() << ";"
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<< "source=" << get_local_ip() << ";"
<< "port=" << iSrvPort << "-" << pSockRtcp->get_local_port() << ";"
<< "ttl=" << udpTTL << ";"
<< "ssrc=" << printSSRC(trackRef->_ssrc)
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});
}
break;
default:
break;
}
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}
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void RtspSession::handleReq_Play(const Parser &parser) {
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if (_sdp_track.empty() || parser["Session"] != _sessionid) {
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send_SessionNotFound();
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throw SockException(Err_shutdown, _sdp_track.empty() ? "can not find any available track when play" : "session not found when play");
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}
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auto play_src = _play_src.lock();
if(!play_src){
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send_StreamNotFound();
shutdown(SockException(Err_shutdown,"rtsp stream released"));
return;
}
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bool useGOP = true;
_enable_send_rtp = false;
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float iStartTime = 0;
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auto strRange = parser["Range"];
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if (strRange.size()) {
//这个是seek操作
auto strStart = FindField(strRange.data(), "npt=", "-");
if (strStart == "now") {
strStart = "0";
}
iStartTime = 1000 * (float)atof(strStart.data());
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InfoP(this) << "rtsp seekTo(ms):" << iStartTime;
useGOP = !play_src->seekTo((uint32_t)iStartTime);
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} else if (play_src->totalReaderCount() == 0) {
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//第一个消费者
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play_src->seekTo(0);
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}
_StrPrinter rtp_info;
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for (auto &track : _sdp_track) {
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if (track->_inited == false) {
//还有track没有setup
shutdown(SockException(Err_shutdown, "track not setuped"));
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return;
}
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track->_ssrc = play_src->getSsrc(track->_type);
track->_seq = play_src->getSeqence(track->_type);
track->_time_stamp = play_src->getTimeStamp(track->_type);
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rtp_info << "url=" << track->getControlUrl(_content_base) << ";"
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<< "seq=" << track->_seq << ";"
<< "rtptime=" << (int) (track->_time_stamp * (track->_samplerate / 1000)) << ",";
}
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rtp_info.pop_back();
sendRtspResponse("200 OK",
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{"Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << (useGOP ? play_src->getTimeStamp(TrackInvalid) / 1000.0 : iStartTime / 1000),
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"RTP-Info",rtp_info
});
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_enable_send_rtp = true;
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setSocketFlags();
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if (!_play_reader && _rtp_type != Rtsp::RTP_MULTICAST) {
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weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
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_play_reader = play_src->getRing()->attach(getPoller(), useGOP);
_play_reader->setDetachCB([weakSelf]() {
auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
return;
}
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strongSelf->shutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
});
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_play_reader->setReadCB([weakSelf](const RtspMediaSource::RingDataType &pack) {
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auto strongSelf = weakSelf.lock();
if (!strongSelf) {
return;
}
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if (strongSelf->_enable_send_rtp) {
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strongSelf->sendRtpPacket(pack);
}
});
}
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}
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void RtspSession::handleReq_Pause(const Parser &parser) {
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if (parser["Session"] != _sessionid) {
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send_SessionNotFound();
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throw SockException(Err_shutdown,"session not found when pause");
}
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sendRtspResponse("200 OK");
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_enable_send_rtp = false;
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}
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void RtspSession::handleReq_Teardown(const Parser &parser) {
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sendRtspResponse("200 OK");
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throw SockException(Err_shutdown,"rtsp player send teardown request");
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}
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void RtspSession::handleReq_Get(const Parser &parser) {
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_http_x_sessioncookie = parser["x-sessioncookie"];
sendRtspResponse("200 OK",
{"Cache-Control","no-store",
"Pragma","no-store",
"Content-Type","application/x-rtsp-tunnelled",
},"","HTTP/1.0");
//注册http getter以便http poster绑定
lock_guard<recursive_mutex> lock(g_mtxGetter);
g_mapGetter[_http_x_sessioncookie] = dynamic_pointer_cast<RtspSession>(shared_from_this());
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}
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void RtspSession::handleReq_Post(const Parser &parser) {
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lock_guard<recursive_mutex> lock(g_mtxGetter);
string sessioncookie = parser["x-sessioncookie"];
//Poster 找到 Getter
auto it = g_mapGetter.find(sessioncookie);
if (it == g_mapGetter.end()) {
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throw SockException(Err_shutdown,"can not find http getter by x-sessioncookie");
}
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//Poster 找到Getter的SOCK
auto httpGetterWeak = it->second;
//移除http getter的弱引用记录
g_mapGetter.erase(sessioncookie);
//http poster收到请求后转发给http getter处理
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_on_recv = [this,httpGetterWeak](const Buffer::Ptr &buf){
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auto httpGetterStrong = httpGetterWeak.lock();
if(!httpGetterStrong){
shutdown(SockException(Err_shutdown,"http getter released"));
return;
}
//切换到http getter的线程
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httpGetterStrong->async([buf,httpGetterWeak](){
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auto httpGetterStrong = httpGetterWeak.lock();
if(!httpGetterStrong){
return;
}
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httpGetterStrong->onRecv(std::make_shared<BufferString>(decodeBase64(string(buf->data(), buf->size()))));
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});
};
if(!parser.Content().empty()){
//http poster后面的粘包
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_on_recv(std::make_shared<BufferString>(parser.Content()));
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}
sendRtspResponse("200 OK",
{"Cache-Control","no-store",
"Pragma","no-store",
"Content-Type","application/x-rtsp-tunnelled",
},"","HTTP/1.0");
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}
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void RtspSession::handleReq_SET_PARAMETER(const Parser &parser) {
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//TraceP(this) <<endl;
sendRtspResponse("200 OK");
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}
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void RtspSession::send_NotAcceptable() {
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sendRtspResponse("406 Not Acceptable",{"Connection","Close"});
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}
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void RtspSession::onRtpSorted(RtpPacket::Ptr rtp, int track_idx) {
_push_src->onWrite(std::move(rtp), false);
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}
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void RtspSession::onRcvPeerUdpData(int interleaved, const Buffer::Ptr &buf, const struct sockaddr &addr) {
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//这是rtcp心跳包说明播放器还存活
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_alive_ticker.resetTime();
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if (interleaved % 2 == 0) {
if (_push_src) {
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//这是rtsp推流上来的rtp包
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auto &ref = _sdp_track[interleaved / 2];
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handleOneRtp(interleaved / 2, ref->_type, ref->_samplerate, (uint8_t *) buf->data(), buf->size());
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} else if (!_udp_connected_flags.count(interleaved)) {
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//这是rtsp播放器的rtp打洞包
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_udp_connected_flags.emplace(interleaved);
_rtp_socks[interleaved / 2]->setSendPeerAddr(&addr);
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}
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} else {
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//rtcp包
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if (!_udp_connected_flags.count(interleaved)) {
_udp_connected_flags.emplace(interleaved);
_rtcp_socks[(interleaved - 1) / 2]->setSendPeerAddr(&addr);
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}
onRtcpPacket((interleaved - 1) / 2, _sdp_track[(interleaved - 1) / 2], buf->data(), buf->size());
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}
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}
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void RtspSession::startListenPeerUdpData(int track_idx) {
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weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
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auto srcIP = inet_addr(get_peer_ip().data());
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auto onUdpData = [weakSelf,srcIP](const Buffer::Ptr &buf, struct sockaddr *peer_addr, int interleaved){
auto strongSelf = weakSelf.lock();
if (!strongSelf) {
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return false;
}
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if (((struct sockaddr_in *) peer_addr)->sin_addr.s_addr != srcIP) {
WarnP(strongSelf.get()) << ((interleaved % 2 == 0) ? "收到其他地址的rtp数据:" : "收到其他地址的rtcp数据:")
<< SockUtil::inet_ntoa(((struct sockaddr_in *) peer_addr)->sin_addr);
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return true;
}
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struct sockaddr addr = *peer_addr;
strongSelf->async([weakSelf, buf, addr, interleaved]() {
auto strongSelf = weakSelf.lock();
if (!strongSelf) {
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return;
}
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strongSelf->onRcvPeerUdpData(interleaved, buf, addr);
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});
return true;
};
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switch (_rtp_type){
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case Rtsp::RTP_MULTICAST:{
//组播使用的共享rtcp端口
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UDPServer::Instance().listenPeer(get_peer_ip().data(), this,
[onUdpData]( int interleaved, const Buffer::Ptr &buf, struct sockaddr *peer_addr) {
return onUdpData(buf, peer_addr, interleaved);
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});
}
break;
case Rtsp::RTP_UDP:{
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auto setEvent = [&](Socket::Ptr &sock,int interleaved){
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if(!sock){
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WarnP(this) << "udp端口为空:" << interleaved;
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return;
}
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sock->setOnRead([onUdpData,interleaved](const Buffer::Ptr &pBuf, struct sockaddr *pPeerAddr , int addr_len){
onUdpData(pBuf, pPeerAddr, interleaved);
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});
};
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setEvent(_rtp_socks[track_idx], 2 * track_idx );
setEvent(_rtcp_socks[track_idx], 2 * track_idx + 1 );
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}
break;
default:
break;
}
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}
static string dateStr(){
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char buf[64];
time_t tt = time(NULL);
strftime(buf, sizeof buf, "%a, %b %d %Y %H:%M:%S GMT", gmtime(&tt));
return buf;
}
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bool RtspSession::sendRtspResponse(const string &res_code, const StrCaseMap &header_const, const string &sdp, const char *protocol){
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auto header = header_const;
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header.emplace("CSeq",StrPrinter << _cseq);
if(!_sessionid.empty()){
header.emplace("Session", _sessionid);
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}
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header.emplace("Server",SERVER_NAME);
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header.emplace("Date",dateStr());
if(!sdp.empty()){
header.emplace("Content-Length",StrPrinter << sdp.size());
header.emplace("Content-Type","application/sdp");
}
_StrPrinter printer;
printer << protocol << " " << res_code << "\r\n";
for (auto &pr : header){
printer << pr.first << ": " << pr.second << "\r\n";
}
printer << "\r\n";
if(!sdp.empty()){
printer << sdp;
}
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// DebugP(this) << printer;
return send(std::make_shared<BufferString>(std::move(printer))) > 0 ;
}
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ssize_t RtspSession::send(Buffer::Ptr pkt){
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// if(!_enableSendRtp){
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// DebugP(this) << pkt->data();
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// }
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_bytes_usage += pkt->size();
return TcpSession::send(std::move(pkt));
}
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bool RtspSession::sendRtspResponse(const string &res_code, const std::initializer_list<string> &header, const string &sdp, const char *protocol) {
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string key;
StrCaseMap header_map;
int i = 0;
for(auto &val : header){
if(++i % 2 == 0){
header_map.emplace(key,val);
}else{
key = val;
}
}
return sendRtspResponse(res_code,header_map,sdp,protocol);
}
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int RtspSession::getTrackIndexByTrackType(TrackType type) {
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for (unsigned int i = 0; i < _sdp_track.size(); i++) {
if (type == _sdp_track[i]->_type) {
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return i;
}
}
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if(_sdp_track.size() == 1){
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return 0;
}
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throw SockException(Err_shutdown, StrPrinter << "no such track with type:" << (int) type);
}
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int RtspSession::getTrackIndexByControlUrl(const string &control_url) {
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for (unsigned int i = 0; i < _sdp_track.size(); i++) {
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if (control_url == _sdp_track[i]->getControlUrl(_content_base)) {
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return i;
}
}
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if(_sdp_track.size() == 1){
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return 0;
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}
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throw SockException(Err_shutdown, StrPrinter << "no such track with control url:" << control_url);
}
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int RtspSession::getTrackIndexByInterleaved(int interleaved){
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for (unsigned int i = 0; i < _sdp_track.size(); i++) {
if (_sdp_track[i]->_interleaved == interleaved) {
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return i;
}
}
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if(_sdp_track.size() == 1){
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return 0;
}
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throw SockException(Err_shutdown, StrPrinter << "no such track with interleaved:" << interleaved);
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}
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bool RtspSession::close(MediaSource &sender, bool force) {
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//此回调在其他线程触发
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if(!_push_src || (!force && _push_src->totalReaderCount())){
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return false;
}
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string err = StrPrinter << "close media:" << sender.getSchema() << "/" << sender.getVhost() << "/" << sender.getApp() << "/" << sender.getId() << " " << force;
safeShutdown(SockException(Err_shutdown,err));
return true;
}
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int RtspSession::totalReaderCount(MediaSource &sender) {
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return _push_src ? _push_src->totalReaderCount() : sender.readerCount();
}
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MediaOriginType RtspSession::getOriginType(MediaSource &sender) const{
return MediaOriginType::rtsp_push;
}
string RtspSession::getOriginUrl(MediaSource &sender) const {
return _media_info._full_url;
}
std::shared_ptr<SockInfo> RtspSession::getOriginSock(MediaSource &sender) const {
return const_cast<RtspSession *>(this)->shared_from_this();
}
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void RtspSession::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index){
updateRtcpContext(rtp);
}
void RtspSession::updateRtcpContext(const RtpPacket::Ptr &rtp){
int track_index = getTrackIndexByTrackType(rtp->type);
auto &rtcp_ctx = _rtcp_context[track_index];
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rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
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auto &ticker = _rtcp_send_tickers[track_index];
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//send rtcp every 5 second
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if (ticker.elapsedTime() > 5 * 1000) {
ticker.resetTime();
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static auto send_rtcp = [](RtspSession *thiz, int index, Buffer::Ptr ptr) {
if (thiz->_rtp_type == Rtsp::RTP_TCP) {
auto &track = thiz->_sdp_track[index];
thiz->send(makeRtpOverTcpPrefix((uint16_t)(ptr->size()), track->_interleaved + 1));
thiz->send(std::move(ptr));
} else {
thiz->_rtcp_socks[index]->send(std::move(ptr));
}
};
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auto ssrc = rtp->getSSRC();
auto rtcp = _push_src ? rtcp_ctx->createRtcpRR(ssrc + 1, ssrc) : rtcp_ctx->createRtcpSR(ssrc);
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auto rtcp_sdes = RtcpSdes::create({SERVER_NAME});
rtcp_sdes->items.type = (uint8_t)SdesType::RTCP_SDES_CNAME;
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rtcp_sdes->items.ssrc = htonl(ssrc);
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send_rtcp(this, track_index, std::move(rtcp));
send_rtcp(this, track_index, RtcpHeader::toBuffer(rtcp_sdes));
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}
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}
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void RtspSession::sendRtpPacket(const RtspMediaSource::RingDataType &pkt) {
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switch (_rtp_type) {
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case Rtsp::RTP_TCP: {
size_t i = 0;
auto size = pkt->size();
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setSendFlushFlag(false);
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
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updateRtcpContext(rtp);
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if (++i == size) {
setSendFlushFlag(true);
}
send(rtp);
});
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}
break;
case Rtsp::RTP_UDP: {
size_t i = 0;
auto size = pkt->size();
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pkt->for_each([&](const RtpPacket::Ptr &rtp) {
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updateRtcpContext(rtp);
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int track_index = getTrackIndexByTrackType(rtp->type);
auto &pSock = _rtp_socks[track_index];
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if (!pSock) {
shutdown(SockException(Err_shutdown, "udp sock not opened yet"));
return;
}
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_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
pSock->send(std::make_shared<BufferRtp>(rtp, RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
2020-04-07 13:03:53 +08:00
});
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}
break;
default:
break;
}
}
void RtspSession::setSocketFlags(){
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GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
if(mergeWriteMS > 0) {
//推流模式下关闭TCP_NODELAY会增加推流端的延时但是服务器性能将提高
SockUtil::setNoDelay(getSock()->rawFD(), false);
//播放模式下开启MSG_MORE会增加延时但是能提高发送性能
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setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
}
}
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}
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/* namespace mediakit */