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Older/MediaServer/Rtsp/RtspSession.cpp

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add ZLMediaKit code for learning.
2024-09-28 23:55:00 +08:00
/*
* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
*
* Use of this source code is governed by MIT-like 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.
*/
#include <atomic>
#include <iomanip>
#include "Common/config.h"
#include "UDPServer.h"
#include "RtspSession.h"
#include "Util/MD5.h"
#include "Util/base64.h"
#include "RtpMultiCaster.h"
#include "Rtcp/RtcpContext.h"
using namespace std;
using namespace toolkit;
namespace mediakit {
/**
* rtsp协议有多种方式传输rtp数据包，目前已支持包括以下4种
* 1: rtp over udp ,这种方式是rtp通过单独的udp端口传输
* 2: rtp over udp_multicast,这种方式是rtp通过共享udp组播端口传输
* 3: rtp over tcp,这种方式是通过rtsp信令tcp通道完成传输
* 4: rtp over http，下面着重讲解：rtp over http
*
* rtp over http 是把rtsp协议伪装成http协议以达到穿透防火墙的目的，
* 此时播放器会发送两次http请求至rtsp服务器，第一次是http get请求，
* 第二次是http post请求。
*
* 这两次请求通过http请求头中的x-sessioncookie键完成绑定
*
* 第一次http get请求用于接收rtp、rtcp和rtsp回复，后续该链接不再发送其他请求
* 第二次http post请求用于发送rtsp请求，rtsp握手结束后可能会断开连接，此时我们还要维持rtp发送
* 需要指出的是http post请求中的content负载就是base64编码后的rtsp请求包，
* 播放器会把rtsp请求伪装成http content负载发送至rtsp服务器，然后rtsp服务器又把回复发送给第一次http get请求的tcp链接
* 这样，对防火墙而言，本次rtsp会话就是两次http请求，防火墙就会放行数据
*
* zlmediakit在处理rtsp over http的请求时，会把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;
RtspSession::RtspSession(const Socket::Ptr &sock) : Session(sock) {
GET_CONFIG(uint32_t,keep_alive_sec,Rtsp::kKeepAliveSecond);
sock->setSendTimeOutSecond(keep_alive_sec);
}
void RtspSession::onError(const SockException &err) {
bool is_player = !_push_src_ownership;
uint64_t duration = _alive_ticker.createdTime() / 1000;
WarnP(this) << (is_player ? "RTSP播放器(" : "RTSP推流器(")
<< _media_info.shortUrl()
<< ")断开:" << err.what()
<< ",耗时(s):" << duration;
if (_rtp_type == Rtsp::RTP_MULTICAST) {
//取消UDP端口监听
UDPServer::Instance().stopListenPeer(get_peer_ip().data(), this);
}
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
GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
if (_bytes_usage >= iFlowThreshold * 1024) {
NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, is_player, *this);
}
//如果是主动关闭的，那么不延迟注销
if (_push_src && _continue_push_ms && err.getErrCode() != Err_shutdown) {
//取消所有权
_push_src_ownership = nullptr;
//延时10秒注销流
auto push_src = std::move(_push_src);
getPoller()->doDelayTask(_continue_push_ms, [push_src]() { return 0; });
}
}
void RtspSession::onManager() {
GET_CONFIG(uint32_t, handshake_sec, Rtsp::kHandshakeSecond);
GET_CONFIG(uint32_t, keep_alive_sec, Rtsp::kKeepAliveSecond);
if (_alive_ticker.createdTime() > handshake_sec * 1000) {
if (_sessionid.size() == 0) {
shutdown(SockException(Err_timeout,"illegal connection"));
return;
}
}
if (_push_src && _alive_ticker.elapsedTime() > keep_alive_sec * 1000) {
//推流超时
shutdown(SockException(Err_timeout, "pusher session timeout"));
return;
}
if (!_push_src && _rtp_type == Rtsp::RTP_UDP && _alive_ticker.elapsedTime() > keep_alive_sec * 4000) {
//rtp over udp播放器超时
shutdown(SockException(Err_timeout, "rtp over udp player timeout"));
}
}
void RtspSession::onRecv(const Buffer::Ptr &buf) {
_alive_ticker.resetTime();
_bytes_usage += buf->size();
if (_on_recv) {
//http poster的请求数据转发给http getter处理
_on_recv(buf);
} else {
input(buf->data(), buf->size());
}
}
void RtspSession::onWholeRtspPacket(Parser &parser) {
string method = parser.method(); //提取出请求命令字
_cseq = atoi(parser["CSeq"].data());
if (_content_base.empty() && method != "GET" && method != "POST" ) {
RtspUrl rtsp;
rtsp.parse(parser.url());
_content_base = rtsp._url;
_media_info.parse(parser.fullUrl());
_media_info.schema = RTSP_SCHEMA;
}
using rtsp_request_handler = void (RtspSession::*)(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);
});
auto it = s_cmd_functions.find(method);
if (it == s_cmd_functions.end()) {
sendRtspResponse("403 Forbidden");
throw SockException(Err_shutdown, StrPrinter << "403 Forbidden:" << method);
}
(this->*(it->second))(parser);
parser.clear();
}
void RtspSession::onRtpPacket(const char *data, size_t len) {
uint8_t interleaved = data[1];
if (interleaved % 2 == 0) {
auto track_idx = getTrackIndexByInterleaved(interleaved);
handleOneRtp(track_idx, _sdp_track[track_idx]->_type, _sdp_track[track_idx]->_samplerate, (uint8_t *) data + RtpPacket::kRtpTcpHeaderSize, len - RtpPacket::kRtpTcpHeaderSize);
} else {
auto track_idx = getTrackIndexByInterleaved(interleaved - 1);
onRtcpPacket(track_idx, _sdp_track[track_idx], data + RtpPacket::kRtpTcpHeaderSize, len - RtpPacket::kRtpTcpHeaderSize);
}
}
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);
if ((RtcpType) rtcp->pt == RtcpType::RTCP_SR) {
auto sr = (RtcpSR *) (rtcp);
//设置rtp时间戳与ntp时间戳的对应关系
setNtpStamp(track_idx, sr->rtpts, sr->getNtpUnixStampMS());
}
}
}
ssize_t RtspSession::getContentLength(Parser &parser) {
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);
}
void RtspSession::handleReq_Options(const Parser &parser) {
//支持这些命令
sendRtspResponse("200 OK",{"Public" , "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, ANNOUNCE, RECORD, SET_PARAMETER, GET_PARAMETER"});
}
void RtspSession::handleReq_ANNOUNCE(const Parser &parser) {
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);
_media_info.parse(full_url);
}
if (_media_info.app.empty() || _media_info.stream.empty()) {
//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
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);
}
auto onRes = [this, parser, full_url](const string &err, const ProtocolOption &option) {
if (!err.empty()) {
sendRtspResponse("401 Unauthorized", { "Content-Type", "text/plain" }, err);
shutdown(SockException(Err_shutdown, StrPrinter << "401 Unauthorized:" << err));
return;
}
assert(!_push_src);
auto src = MediaSource::find(RTSP_SCHEMA, _media_info.vhost, _media_info.app, _media_info.stream);
auto push_failed = (bool)src;
while (src) {
//尝试断连后继续推流
auto rtsp_src = dynamic_pointer_cast<RtspMediaSourceImp>(src);
if (!rtsp_src) {
//源不是rtsp推流产生的
break;
}
auto ownership = rtsp_src->getOwnership();
if (!ownership) {
//获取推流源所有权失败
break;
}
_push_src = std::move(rtsp_src);
_push_src_ownership = std::move(ownership);
push_failed = false;
break;
}
if (push_failed) {
sendRtspResponse("406 Not Acceptable", { "Content-Type", "text/plain" }, "Already publishing.");
string err = StrPrinter << "ANNOUNCE: Already publishing:" << _media_info.shortUrl() << endl;
throw SockException(Err_shutdown, err);
}
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;
}
_rtcp_context.clear();
for (auto &track : _sdp_track) {
_rtcp_context.emplace_back(std::make_shared<RtcpContextForRecv>());
}
if (!_push_src) {
_push_src = std::make_shared<RtspMediaSourceImp>(_media_info);
//获取所有权
_push_src_ownership = _push_src->getOwnership();
_push_src->setProtocolOption(option);
_push_src->setSdp(parser.content());
}
_push_src->setListener(static_pointer_cast<RtspSession>(shared_from_this()));
_continue_push_ms = option.continue_push_ms;
sendRtspResponse("200 OK");
};
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
Broadcast::PublishAuthInvoker invoker = [weak_self, onRes](const string &err, const ProtocolOption &option) {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
strong_self->async([weak_self, onRes, err, option]() {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
onRes(err, option);
});
};
//rtsp推流需要鉴权
auto flag = NOTICE_EMIT(BroadcastMediaPublishArgs, Broadcast::kBroadcastMediaPublish, MediaOriginType::rtsp_push, _media_info, invoker, *this);
if (!flag) {
//该事件无人监听,默认不鉴权
onRes("", ProtocolOption());
}
}
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 available 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();
}
}
void RtspSession::emitOnPlay(){
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
//url鉴权回调
auto onRes = [weak_self](const string &err) {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
if (!err.empty()) {
//播放url鉴权失败
strong_self->sendRtspResponse("401 Unauthorized", {"Content-Type", "text/plain"}, err);
strong_self->shutdown(SockException(Err_shutdown, StrPrinter << "401 Unauthorized:" << err));
return;
}
strong_self->onAuthSuccess();
};
Broadcast::AuthInvoker invoker = [weak_self, onRes](const string &err) {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
strong_self->async([onRes, err, weak_self]() {
onRes(err);
});
};
//广播通用播放url鉴权事件
auto flag = _emit_on_play ? false : NOTICE_EMIT(BroadcastMediaPlayedArgs, Broadcast::kBroadcastMediaPlayed, _media_info, invoker, *this);
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"];
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
//rtsp专属鉴权是否开启事件回调
onGetRealm invoker = [weak_self, authorization](const string &realm) {
auto strong_self = weak_self.lock();
if (!strong_self) {
//本对象已经销毁
return;
}
//切换到自己的线程然后执行
strong_self->async([weak_self, realm, authorization]() {
auto strong_self = weak_self.lock();
if (!strong_self) {
//本对象已经销毁
return;
}
if (realm.empty()) {
//无需rtsp专属认证, 那么继续url通用鉴权认证(on_play)
strong_self->emitOnPlay();
return;
}
//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)
strong_self->_rtsp_realm = realm;
strong_self->onAuthUser(realm, authorization);
});
};
if(_rtsp_realm.empty()){
//广播是否需要rtsp专属认证事件
if (!NOTICE_EMIT(BroadcastOnGetRtspRealmArgs, Broadcast::kBroadcastOnGetRtspRealm, _media_info, invoker, *this)) {
//无人监听此事件，说明无需认证
invoker("");
}
}else{
invoker(_rtsp_realm);
}
}
void RtspSession::onAuthSuccess() {
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
MediaSource::findAsync(_media_info, weak_self.lock(), [weak_self](const MediaSource::Ptr &src){
auto strong_self = weak_self.lock();
if(!strong_self){
return;
}
auto rtsp_src = dynamic_pointer_cast<RtspMediaSource>(src);
if (!rtsp_src) {
//未找到相应的MediaSource
string err = StrPrinter << "no such stream:" << strong_self->_media_info.shortUrl();
strong_self->send_StreamNotFound();
strong_self->shutdown(SockException(Err_shutdown,err));
return;
}
//找到了相应的rtsp流
strong_self->_sdp_track = SdpParser(rtsp_src->getSdp()).getAvailableTrack();
if (strong_self->_sdp_track.empty()) {
//该流无效
WarnL << "sdp中无有效track，该流无效:" << rtsp_src->getSdp();
strong_self->send_StreamNotFound();
strong_self->shutdown(SockException(Err_shutdown,"can not find any available track in sdp"));
return;
}
strong_self->_rtcp_context.clear();
for (auto &track : strong_self->_sdp_track) {
strong_self->_rtcp_context.emplace_back(std::make_shared<RtcpContextForSend>());
}
strong_self->_sessionid = makeRandStr(12);
strong_self->_play_src = rtsp_src;
for(auto &track : strong_self->_sdp_track){
track->_ssrc = rtsp_src->getSsrc(track->_type);
track->_seq = rtsp_src->getSeqence(track->_type);
track->_time_stamp = rtsp_src->getTimeStamp(track->_type);
}
strong_self->sendRtspResponse("200 OK",
{"Content-Base", strong_self->_content_base + "/",
"x-Accept-Retransmit","our-retransmit",
"x-Accept-Dynamic-Rate","1"
},rtsp_src->getSdp());
});
}
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
_auth_nonce = makeRandStr(32);
sendRtspResponse("401 Unauthorized", { "WWW-Authenticate", StrPrinter << "Digest realm=\"" << realm << "\",nonce=\"" << _auth_nonce << "\"" });
} 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));
}
}
void RtspSession::onAuthBasic(const string &realm, const string &auth_base64) {
//base64认证
auto user_passwd = decodeBase64(auth_base64);
auto user_pwd_vec = split(user_passwd, ":");
if (user_pwd_vec.size() < 2) {
// 认证信息格式不合法，回复401 Unauthorized
onAuthFailed(realm, "can not find user and passwd when basic64 auth");
return;
}
auto user = user_pwd_vec[0];
auto pwd = user_pwd_vec[1];
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
onAuth invoker = [pwd, realm, weak_self](bool encrypted, const string &good_pwd) {
auto strong_self = weak_self.lock();
if (!strong_self) {
//本对象已经销毁
return;
}
//切换到自己的线程执行
strong_self->async([weak_self, good_pwd, pwd, realm]() {
auto strong_self = weak_self.lock();
if (!strong_self) {
//本对象已经销毁
return;
}
//base64忽略encrypted参数，上层必须传入明文密码
if (pwd == good_pwd) {
//提供的密码且匹配正确
strong_self->onAuthSuccess();
return;
}
//密码错误
strong_self->onAuthFailed(realm, StrPrinter << "password mismatch when base64 auth:" << pwd << " != " << good_pwd);
});
};
//此时必须提供明文密码
if (!NOTICE_EMIT(BroadcastOnRtspAuthArgs, Broadcast::kBroadcastOnRtspAuth, _media_info, realm, user, true, invoker, *this)) {
//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
WarnP(this) << "请监听kBroadcastOnRtspAuth事件！";
//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
invoker(false, pwd);
}
}
void RtspSession::onAuthDigest(const string &realm,const string &auth_md5){
DebugP(this) << auth_md5;
auto mapTmp = Parser::parseArgs(auth_md5, ",", "=");
decltype(mapTmp) map;
for(auto &pr : mapTmp){
map[trim(string(pr.first)," \"")] = trim(pr.second," \"");
}
//check realm
if(realm != map["realm"]){
onAuthFailed(realm,StrPrinter << "realm not mached:" << realm << " != " << map["realm"]);
return ;
}
//check nonce
auto nonce = map["nonce"];
if(_auth_nonce != nonce){
onAuthFailed(realm,StrPrinter << "nonce not mached:" << nonce << " != " << _auth_nonce);
return ;
}
//check username and uri
auto username = map["username"];
auto uri = map["uri"];
auto response = map["response"];
if(username.empty() || uri.empty() || response.empty()){
onAuthFailed(realm,StrPrinter << "username/uri/response empty:" << username << "," << uri << "," << response);
return ;
}
auto realInvoker = [this,realm,nonce,uri,username,response](bool ignoreAuth,bool encrypted,const string &good_pwd){
if(ignoreAuth){
//忽略认证
TraceP(this) << "auth ignored";
onAuthSuccess();
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)当password为MD5编码,则
response = md5( password:nonce:md5(public_method:url) );
(2)当password为ANSI字符串,则
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();
if(strcasecmp(good_response.data(),response.data()) == 0){
//认证成功！md5不区分大小写
onAuthSuccess();
}else{
//认证失败！
onAuthFailed(realm, StrPrinter << "password mismatch when md5 auth:" << good_response << " != " << response );
}
};
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
onAuth invoker = [realInvoker,weak_self](bool encrypted,const string &good_pwd){
auto strong_self = weak_self.lock();
if(!strong_self){
return;
}
//切换到自己的线程确保realInvoker执行时，this指针有效
strong_self->async([realInvoker,weak_self,encrypted,good_pwd](){
auto strong_self = weak_self.lock();
if(!strong_self){
return;
}
realInvoker(false,encrypted,good_pwd);
});
};
//此时可以提供明文或md5加密的密码
if(!NOTICE_EMIT(BroadcastOnRtspAuthArgs, Broadcast::kBroadcastOnRtspAuth, _media_info, realm, username, false, invoker, *this)){
//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
WarnP(this) << "请监听kBroadcastOnRtspAuth事件！";
//但是我们还是忽略认证以便完成播放
realInvoker(true,true,"");
}
}
void RtspSession::onAuthUser(const string &realm,const string &authorization){
if(authorization.empty()){
onAuthFailed(realm,"", false);
return;
}
//请求中包含认证信息
auto authType = findSubString(authorization.data(), NULL, " ");
auto authStr = findSubString(authorization.data(), " ", NULL);
if(authType.empty() || authStr.empty()){
//认证信息格式不合法，回复401 Unauthorized
onAuthFailed(realm,"can not find auth type or auth string");
return;
}
if(authType == "Basic"){
//base64认证，需要明文密码
onAuthBasic(realm,authStr);
}else if(authType == "Digest"){
//md5认证
onAuthDigest(realm,authStr);
}else{
//其他认证方式？不支持！
onAuthFailed(realm,StrPrinter << "unsupported auth type:" << authType);
}
}
void RtspSession::send_StreamNotFound() {
sendRtspResponse("404 Stream Not Found",{"Connection","Close"});
}
void RtspSession::send_UnsupportedTransport() {
sendRtspResponse("461 Unsupported Transport",{"Connection","Close"});
}
void RtspSession::send_SessionNotFound() {
sendRtspResponse("454 Session Not Found",{"Connection","Close"});
}
void RtspSession::handleReq_Setup(const Parser &parser) {
//处理setup命令，该函数可能进入多次
int trackIdx = getTrackIndexByControlUrl(parser.fullUrl());
SdpTrack::Ptr &trackRef = _sdp_track[trackIdx];
if (trackRef->_inited) {
//已经初始化过该Track
throw SockException(Err_shutdown, "can not setup one track twice");
}
static auto getRtpTypeStr = [](const int type) {
switch (type)
{
case Rtsp::RTP_TCP:
return "TCP";
case Rtsp::RTP_UDP:
return "UDP";
case Rtsp::RTP_MULTICAST:
return "MULTICAST";
default:
return "Invalid";
}
};
if (_rtp_type == Rtsp::RTP_Invalid) {
auto &strTransport = parser["Transport"];
auto rtpType = Rtsp::RTP_Invalid;
if (strTransport.find("TCP") != string::npos) {
rtpType = Rtsp::RTP_TCP;
} else if (strTransport.find("multicast") != string::npos) {
rtpType = Rtsp::RTP_MULTICAST;
} else {
rtpType = Rtsp::RTP_UDP;
}
//检查RTP传输类型限制
GET_CONFIG(int, transport, Rtsp::kRtpTransportType);
if (transport != Rtsp::RTP_Invalid && transport != rtpType) {
WarnL << "rtsp client setup transport " << getRtpTypeStr(rtpType) << " but config force transport " << getRtpTypeStr(transport);
//配置限定RTSP传输方式，但是客户端握手方式不一致，返回461
sendRtspResponse("461 Unsupported transport");
return;
}
_rtp_type = rtpType;
}
trackRef->_inited = true; //现在初始化
//允许接收rtp、rtcp包
RtspSplitter::enableRecvRtp(_rtp_type == Rtsp::RTP_TCP);
switch (_rtp_type) {
case Rtsp::RTP_TCP: {
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;
}
sendRtspResponse("200 OK",
{"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"
});
}
break;
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) {
//分配端口失败
send_NotAcceptable();
throw SockException(Err_shutdown, ex.what());
}
_rtp_socks[trackIdx] = pr.first;
_rtcp_socks[trackIdx] = pr.second;
//设置客户端内网端口信息
string strClientPort = findSubString(parser["Transport"].data(), "client_port=", NULL);
uint16_t ui16RtpPort = atoi(findSubString(strClientPort.data(), NULL, "-").data());
uint16_t ui16RtcpPort = atoi(findSubString(strClientPort.data(), "-", NULL).data());
auto peerAddr = SockUtil::make_sockaddr(get_peer_ip().data(), ui16RtpPort);
//设置rtp发送目标地址
pr.first->bindPeerAddr((struct sockaddr *) (&peerAddr), 0, true);
//设置rtcp发送目标地址
peerAddr = SockUtil::make_sockaddr(get_peer_ip().data(), ui16RtcpPort);
pr.second->bindPeerAddr((struct sockaddr *) (&peerAddr), 0, true);
//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",
{"Transport", StrPrinter << "RTP/AVP/UDP;unicast;"
<< "client_port=" << strClientPort << ";"
<< "server_port=" << pr.first->get_local_port() << "-"
<< pr.second->get_local_port() << ";"
<< "ssrc=" << printSSRC(trackRef->_ssrc)
});
}
break;
case Rtsp::RTP_MULTICAST: {
if(!_multicaster){
_multicaster = RtpMultiCaster::get(*this, get_local_ip(), _media_info, _multicast_ip, _multicast_video_port, _multicast_audio_port);
if (!_multicaster) {
send_NotAcceptable();
throw SockException(Err_shutdown, "can not get a available udp multicast socket");
}
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
_multicaster->setDetachCB(this, [weak_self]() {
auto strong_self = weak_self.lock();
if(!strong_self) {
return;
}
strong_self->safeShutdown(SockException(Err_shutdown,"ring buffer detached"));
});
}
int iSrvPort = _multicaster->getMultiCasterPort(trackRef->_type);
//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);
if (!pSockRtcp) {
//分配端口失败
send_NotAcceptable();
throw SockException(Err_shutdown, "open shared rtcp socket failed");
}
startListenPeerUdpData(trackIdx);
GET_CONFIG(uint32_t,udpTTL,MultiCast::kUdpTTL);
sendRtspResponse("200 OK",
{"Transport", StrPrinter << "RTP/AVP;multicast;"
<< "destination=" << _multicaster->getMultiCasterIP() << ";"
<< "source=" << get_local_ip() << ";"
<< "port=" << iSrvPort << "-" << pSockRtcp->get_local_port() << ";"
<< "ttl=" << udpTTL << ";"
<< "ssrc=" << printSSRC(trackRef->_ssrc)
});
}
break;
default:
break;
}
}
void RtspSession::handleReq_Play(const Parser &parser) {
if (_sdp_track.empty() || parser["Session"] != _sessionid) {
send_SessionNotFound();
throw SockException(Err_shutdown, _sdp_track.empty() ? "can not find any available track when play" : "session not found when play");
}
auto play_src = _play_src.lock();
if(!play_src){
send_StreamNotFound();
shutdown(SockException(Err_shutdown,"rtsp stream released"));
return;
}
bool use_gop = true;
auto &strScale = parser["Scale"];
auto &strRange = parser["Range"];
StrCaseMap res_header;
if (!strScale.empty()) {
//这是设置播放速度
res_header.emplace("Scale", strScale);
auto speed = atof(strScale.data());
play_src->speed(speed);
InfoP(this) << "rtsp set play speed:" << speed;
}
if (!strRange.empty()) {
//这是seek操作
res_header.emplace("Range", strRange);
auto strStart = findSubString(strRange.data(), "npt=", "-");
if (strStart == "now") {
strStart = "0";
}
auto iStartTime = 1000 * (float) atof(strStart.data());
use_gop = !play_src->seekTo((uint32_t) iStartTime);
InfoP(this) << "rtsp seekTo(ms):" << iStartTime;
}
vector<TrackType> inited_tracks;
_StrPrinter rtp_info;
for (auto &track : _sdp_track) {
if (track->_inited == false) {
//为支持播放器播放单一track, 不校验没有发setup的track
continue;
}
inited_tracks.emplace_back(track->_type);
track->_ssrc = play_src->getSsrc(track->_type);
track->_seq = play_src->getSeqence(track->_type);
track->_time_stamp = play_src->getTimeStamp(track->_type);
rtp_info << "url=" << track->getControlUrl(_content_base) << ";"
<< "seq=" << track->_seq << ";"
<< "rtptime=" << (int64_t)(track->_time_stamp) * (int64_t)(track->_samplerate/ 1000) << ",";
}
rtp_info.pop_back();
res_header.emplace("RTP-Info", rtp_info);
//已存在Range时不覆盖
res_header.emplace("Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << play_src->getTimeStamp(TrackInvalid) / 1000.0);
sendRtspResponse("200 OK", res_header);
//设置播放track
if (inited_tracks.size() == 1) {
_target_play_track = inited_tracks[0];
InfoP(this) << "指定播放track:" << _target_play_track;
}
//在回复rtsp信令后再恢复播放
play_src->pause(false);
setSocketFlags();
if (!_play_reader && _rtp_type != Rtsp::RTP_MULTICAST) {
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
_play_reader = play_src->getRing()->attach(getPoller(), use_gop);
_play_reader->setGetInfoCB([weak_self]() {
Any ret;
ret.set(static_pointer_cast<SockInfo>(weak_self.lock()));
return ret;
});
_play_reader->setDetachCB([weak_self]() {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
strong_self->shutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
});
_play_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pack) {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
strong_self->sendRtpPacket(pack);
});
}
}
void RtspSession::handleReq_Pause(const Parser &parser) {
if (parser["Session"] != _sessionid) {
send_SessionNotFound();
throw SockException(Err_shutdown, "session not found when pause");
}
sendRtspResponse("200 OK");
auto play_src = _play_src.lock();
if (play_src) {
play_src->pause(true);
}
}
void RtspSession::handleReq_Teardown(const Parser &parser) {
_push_src = nullptr;
//此时回复可能触发broken pipe事件，从而直接触发onError回调；所以需要先把_push_src置空，防止触发断流续推功能
sendRtspResponse("200 OK");
throw SockException(Err_shutdown,"recv teardown request");
}
void RtspSession::handleReq_Get(const Parser &parser) {
_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] = static_pointer_cast<RtspSession>(shared_from_this());
}
void RtspSession::handleReq_Post(const Parser &parser) {
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()) {
throw SockException(Err_shutdown,"can not find http getter by x-sessioncookie");
}
//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
_on_recv = [this,httpGetterWeak](const Buffer::Ptr &buf){
auto httpGetterStrong = httpGetterWeak.lock();
if(!httpGetterStrong){
shutdown(SockException(Err_shutdown,"http getter released"));
return;
}
//切换到http getter的线程
httpGetterStrong->async([buf,httpGetterWeak](){
auto httpGetterStrong = httpGetterWeak.lock();
if(!httpGetterStrong){
return;
}
httpGetterStrong->onRecv(std::make_shared<BufferString>(decodeBase64(string(buf->data(), buf->size()))));
});
};
if(!parser.content().empty()){
//http poster后面的粘包
_on_recv(std::make_shared<BufferString>(parser.content()));
}
sendRtspResponse("200 OK",
{"Cache-Control","no-store",
"Pragma","no-store",
"Content-Type","application/x-rtsp-tunnelled",
},"","HTTP/1.0");
}
void RtspSession::handleReq_SET_PARAMETER(const Parser &parser) {
//TraceP(this) <<endl;
sendRtspResponse("200 OK");
}
void RtspSession::send_NotAcceptable() {
sendRtspResponse("406 Not Acceptable",{"Connection","Close"});
}
void RtspSession::onRtpSorted(RtpPacket::Ptr rtp, int track_idx) {
if (_push_src) {
_push_src->onWrite(std::move(rtp), false);
} else {
WarnL << "Not a rtsp push!";
}
}
void RtspSession::onRcvPeerUdpData(int interleaved, const Buffer::Ptr &buf, const struct sockaddr_storage &addr) {
//这是rtcp心跳包，说明播放器还存活
_alive_ticker.resetTime();
if (interleaved % 2 == 0) {
if (_push_src) {
//这是rtsp推流上来的rtp包
auto &ref = _sdp_track[interleaved / 2];
handleOneRtp(interleaved / 2, ref->_type, ref->_samplerate, (uint8_t *) buf->data(), buf->size());
} else if (!_udp_connected_flags.count(interleaved)) {
//这是rtsp播放器的rtp打洞包
_udp_connected_flags.emplace(interleaved);
if (_rtp_socks[interleaved / 2]) {
_rtp_socks[interleaved / 2]->bindPeerAddr((struct sockaddr *)&addr);
}
}
} else {
//rtcp包
if (!_udp_connected_flags.count(interleaved)) {
_udp_connected_flags.emplace(interleaved);
if (_rtcp_socks[(interleaved - 1) / 2]) {
_rtcp_socks[(interleaved - 1) / 2]->bindPeerAddr((struct sockaddr *)&addr);
}
}
onRtcpPacket((interleaved - 1) / 2, _sdp_track[(interleaved - 1) / 2], buf->data(), buf->size());
}
}
void RtspSession::startListenPeerUdpData(int track_idx) {
weak_ptr<RtspSession> weak_self = static_pointer_cast<RtspSession>(shared_from_this());
auto peer_ip = get_peer_ip();
auto onUdpData = [weak_self,peer_ip](const Buffer::Ptr &buf, struct sockaddr *peer_addr, int interleaved){
auto strong_self = weak_self.lock();
if (!strong_self) {
return false;
}
if (SockUtil::inet_ntoa(peer_addr) != peer_ip) {
WarnP(strong_self.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(peer_addr);
return true;
}
struct sockaddr_storage addr = *((struct sockaddr_storage *)peer_addr);
strong_self->async([weak_self, buf, addr, interleaved]() {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
try {
strong_self->onRcvPeerUdpData(interleaved, buf, addr);
} catch (SockException &ex) {
strong_self->shutdown(ex);
} catch (std::exception &ex) {
strong_self->shutdown(SockException(Err_other, ex.what()));
}
});
return true;
};
switch (_rtp_type){
case Rtsp::RTP_MULTICAST:{
//组播使用的共享rtcp端口
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);
});
}
break;
case Rtsp::RTP_UDP:{
auto setEvent = [&](Socket::Ptr &sock,int interleaved){
if(!sock){
WarnP(this) << "udp端口为空:" << interleaved;
return;
}
sock->setOnRead([onUdpData,interleaved](const Buffer::Ptr &pBuf, struct sockaddr *pPeerAddr , int addr_len){
onUdpData(pBuf, pPeerAddr, interleaved);
});
};
setEvent(_rtp_socks[track_idx], 2 * track_idx );
setEvent(_rtcp_socks[track_idx], 2 * track_idx + 1 );
}
break;
default:
break;
}
}
static string dateStr(){
char buf[64];
time_t tt = time(NULL);
strftime(buf, sizeof buf, "%a, %b %d %Y %H:%M:%S GMT", gmtime(&tt));
return buf;
}
bool RtspSession::sendRtspResponse(const string &res_code, const StrCaseMap &header_const, const string &sdp, const char *protocol){
auto header = header_const;
header.emplace("CSeq",StrPrinter << _cseq);
if(!_sessionid.empty()){
header.emplace("Session", _sessionid);
}
header.emplace("Server",kServerName);
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;
}
// DebugP(this) << printer;
return send(std::make_shared<BufferString>(std::move(printer))) > 0 ;
}
ssize_t RtspSession::send(Buffer::Ptr pkt){
// if(!_enableSendRtp){
// DebugP(this) << pkt->data();
// }
_bytes_usage += pkt->size();
return Session::send(std::move(pkt));
}
bool RtspSession::sendRtspResponse(const string &res_code, const std::initializer_list<string> &header, const string &sdp, const char *protocol) {
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);
}
int RtspSession::getTrackIndexByTrackType(TrackType type) {
for (size_t i = 0; i < _sdp_track.size(); ++i) {
if (type == _sdp_track[i]->_type) {
return i;
}
}
if (_sdp_track.size() == 1) {
return 0;
}
throw SockException(Err_shutdown, StrPrinter << "no such track with type:" << getTrackString(type));
}
int RtspSession::getTrackIndexByControlUrl(const string &control_url) {
for (size_t i = 0; i < _sdp_track.size(); ++i) {
if (control_url == _sdp_track[i]->getControlUrl(_content_base)) {
return i;
}
}
if (_sdp_track.size() == 1) {
return 0;
}
throw SockException(Err_shutdown, StrPrinter << "no such track with control url:" << control_url);
}
int RtspSession::getTrackIndexByInterleaved(int interleaved) {
for (size_t i = 0; i < _sdp_track.size(); ++i) {
if (_sdp_track[i]->_interleaved == interleaved) {
return i;
}
}
if (_sdp_track.size() == 1) {
return 0;
}
throw SockException(Err_shutdown, StrPrinter << "no such track with interleaved:" << interleaved);
}
bool RtspSession::close(MediaSource &sender) {
//此回调在其他线程触发
string err = StrPrinter << "close media: " << sender.getUrl();
safeShutdown(SockException(Err_shutdown,err));
return true;
}
int RtspSession::totalReaderCount(MediaSource &sender) {
return _push_src ? _push_src->totalReaderCount() : sender.readerCount();
}
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();
}
toolkit::EventPoller::Ptr RtspSession::getOwnerPoller(MediaSource &sender) {
return getPoller();
}
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];
rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
if (!rtp->ntp_stamp && !rtp->getStamp()) {
// 忽略时间戳都为0的rtp
return;
}
auto &ticker = _rtcp_send_tickers[track_index];
//send rtcp every 5 second
if (ticker.elapsedTime() > 5 * 1000 || (_send_sr_rtcp[track_index] && !_push_src)) {
//确保在发送rtp前，先发送一次sender report rtcp(用于播放器同步音视频)
ticker.resetTime();
_send_sr_rtcp[track_index] = false;
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));
}
};
auto ssrc = rtp->getSSRC();
auto rtcp = _push_src ? rtcp_ctx->createRtcpRR(ssrc + 1, ssrc) : rtcp_ctx->createRtcpSR(ssrc);
auto rtcp_sdes = RtcpSdes::create({kServerName});
rtcp_sdes->chunks.type = (uint8_t)SdesType::RTCP_SDES_CNAME;
rtcp_sdes->chunks.ssrc = htonl(ssrc);
send_rtcp(this, track_index, std::move(rtcp));
send_rtcp(this, track_index, RtcpHeader::toBuffer(rtcp_sdes));
}
}
void RtspSession::sendRtpPacket(const RtspMediaSource::RingDataType &pkt) {
switch (_rtp_type) {
case Rtsp::RTP_TCP: {
setSendFlushFlag(false);
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
if (_target_play_track == TrackInvalid || _target_play_track == rtp->type) {
updateRtcpContext(rtp);
send(rtp);
}
});
flushAll();
setSendFlushFlag(true);
}
break;
case Rtsp::RTP_UDP: {
//下标0表示视频，1表示音频
Socket::Ptr rtp_socks[2];
rtp_socks[TrackVideo] = _rtp_socks[getTrackIndexByTrackType(TrackVideo)];
rtp_socks[TrackAudio] = _rtp_socks[getTrackIndexByTrackType(TrackAudio)];
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
if (_target_play_track == TrackInvalid || _target_play_track == rtp->type) {
updateRtcpContext(rtp);
auto &sock = rtp_socks[rtp->type];
if (!sock) {
shutdown(SockException(Err_shutdown, "udp sock not opened yet"));
return;
}
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
sock->send(std::make_shared<BufferRtp>(rtp, RtpPacket::kRtpTcpHeaderSize), nullptr, 0, false);
}
});
for (auto &sock : rtp_socks) {
if (sock) {
sock->flushAll();
}
}
}
break;
default:
break;
}
}
void RtspSession::setSocketFlags(){
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
setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
}
}
}
/* namespace mediakit */
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