上一篇 http请求播放流程中
SrsGoApiRtcPlay::do_serve_http
// For each RTC session, we use short-term HTTP connection.
1、对于RTC会话，使用HTTP短连接。
// Parse req, the request json object, from body.
2、从body中解析请求的json对象
// Fetch params from req object.
3、从请求对象中获取参数，包括远程对端SDP信息、流地址、客户端IP、api、tid信息。
// Parse app and stream from streamurl.
4、解析出application和流名称。和RTMP中的类似。
// For client to specifies the EIP of server.
5、客户端指定信息
// DTLS/SRTP need reconfig after negotiating sdp.
6、在sdp协商结束后需要重新配置DTLS/SRTP
// It seems remote_sdp doesn't represents the full SDP information.
7、设置remote_sdp, 创建RTC用户配置对象，并设置相应的参数值，生成local_sdp；
创建RTC会话
srs_error_t SrsRtcServer::create_session(SrsRtcUserConfig* conf, SrsSdp& local_sdp, SrsRtcConnection** psession)
{
srs_error_t err = srs_success;
SrsContextId cid = _srs_context->get_id();
SrsRequest* req = conf->req;
if (conf->publish) {
SrsRtcStream* source = NULL;
// 根据req获取或创建source，用于_srs_rtc_sources管理。
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
return srs_error_wrap(err, "create source");
}
if (!source->can_publish()) {
return srs_error_new(ERROR_RTC_SOURCE_BUSY, "stream %s busy", req->get_stream_url().c_str());
}
}
// TODO: FIXME: add do_create_session to error process.
// 创建会话
SrsRtcConnection* session = new SrsRtcConnection(this, cid);
if ((err = do_create_session(conf, local_sdp, session)) != srs_success) {
srs_freep(session);
return srs_error_wrap(err, "create session");
}
*psession = session;
return err;
}
srs_error_t SrsRtcServer::do_create_session(SrsRtcUserConfig* conf, SrsSdp& local_sdp, SrsRtcConnection* session)
{
srs_error_t err = srs_success;
SrsRequest* req = conf->req;
//首先根据conf->publish类型为sdp媒体信息添加发布者/播放者
// first add publisher/player for negotiate sdp media info
if (conf->publish) {
if ((err = session->add_publisher(conf, local_sdp)) != srs_success) {
return srs_error_wrap(err, "add publisher");
}
} else {
if ((err = session->add_player(conf, local_sdp)) != srs_success) {
return srs_error_wrap(err, "add player");
}
}
// All tracks default as inactive, so we must enable them.
session->set_all_tracks_status(req->get_stream_url(), conf->publish, true);
std::string local_pwd = srs_random_str(32);
std::string local_ufrag = "";
// TODO: FIXME: Rename for a better name, it's not an username.
std::string username = "";
while (true) {
local_ufrag = srs_random_str(8);
username = local_ufrag + ":" + conf->remote_sdp.get_ice_ufrag();
if (!_srs_rtc_manager->find_by_name(username)) {
break;
}
}
local_sdp.set_ice_ufrag(local_ufrag);
local_sdp.set_ice_pwd(local_pwd);
local_sdp.set_fingerprint_algo("sha-256");
local_sdp.set_fingerprint(_srs_rtc_dtls_certificate->get_fingerprint());
// We allows to mock the eip of server.
if (!conf->eip.empty()) {
string host;
int port = _srs_config->get_rtc_server_listen();
srs_parse_hostport(conf->eip, host, port);
local_sdp.add_candidate(host, port, "host");
srs_trace("RTC: Use candidate mock_eip %s as %s:%d", conf->eip.c_str(), host.c_str(), port);
} else {
std::vector<string> candidate_ips = get_candidate_ips();
for (int i = 0; i < (int)candidate_ips.size(); ++i) {
local_sdp.add_candidate(candidate_ips[i], _srs_config->get_rtc_server_listen(), "host");
}
srs_trace("RTC: Use candidates %s", srs_join_vector_string(candidate_ips, ", ").c_str());
}
// Setup the negotiate DTLS by config.
local_sdp.session_negotiate_ = local_sdp.session_config_;
// Setup the negotiate DTLS role.
if (conf->remote_sdp.get_dtls_role() == "active") {
local_sdp.session_negotiate_.dtls_role = "passive";
} else if (conf->remote_sdp.get_dtls_role() == "passive") {
local_sdp.session_negotiate_.dtls_role = "active";
} else if (conf->remote_sdp.get_dtls_role() == "actpass") {
local_sdp.session_negotiate_.dtls_role = local_sdp.session_config_.dtls_role;
} else {
// @see: https://tools.ietf.org/html/rfc4145#section-4.1
// The default value of the setup attribute in an offer/answer exchange
// is 'active' in the offer and 'passive' in the answer.
local_sdp.session_negotiate_.dtls_role = "passive";
}
local_sdp.set_dtls_role(local_sdp.session_negotiate_.dtls_role);
session->set_remote_sdp(conf->remote_sdp);
// We must setup the local SDP, then initialize the session object.
session->set_local_sdp(local_sdp);
session->set_state(WAITING_STUN);
// Before session initialize, we must setup the local SDP.
if ((err = session->initialize(req, conf->dtls, conf->srtp, username)) != srs_success) {
return srs_error_wrap(err, "init");
}
// We allows username is optional, but it never empty here.
_srs_rtc_manager->add_with_name(username, session);
return err;
}
添加播放者
我们本章内容是播放流程，所以就选取session->add_player进行分析
srs_error_t SrsRtcConnection::add_player(SrsRtcUserConfig* conf, SrsSdp& local_sdp)
{
srs_error_t err = srs_success;
SrsRequest* req = conf->req;
if (_srs_rtc_hijacker) {
if ((err = _srs_rtc_hijacker->on_before_play(this, req)) != srs_success) {
return srs_error_wrap(err, "before play");
}
}
if (_srs_config->get_vhost_is_edge(req->vhost)){
SrsSource* rtmp_source = NULL;
if ((err = _srs_sources->fetch_or_create(req, _srs_hybrid->srs()->instance(), &rtmp_source)) != srs_success) {
return srs_error_wrap(err, "create rtmp source");
}
if (rtmp_source && (err = rtmp_source->on_edge_play()) != srs_success){
return srs_error_wrap(err, "rtmp source play");
}
// update traffic state
is_traffic_hit_ = rtmp_source->get_traffic_hit();
}
std::map<uint32_t, SrsRtcTrackDescription*> play_sub_relations;
if ((err = negotiate_play_source_capability(req, conf->remote_sdp, play_sub_relations)) != srs_success) {
return srs_error_wrap(err, "play negotiate");
}
// reconfig dtls/srtp of userconfig after negotiate sdp.
if (conf->req->ext_abilities.is_support_unencrypted()){
conf->dtls = conf->srtp = false;
}
if (!play_sub_relations.size()) {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no play relations");
}
SrsRtcStreamDescription* stream_desc = new SrsRtcStreamDescription();
SrsAutoFree(SrsRtcStreamDescription, stream_desc);
std::map<uint32_t, SrsRtcTrackDescription*>::iterator it = play_sub_relations.begin();
while (it != play_sub_relations.end()) {
SrsRtcTrackDescription* track_desc = it->second;
// TODO: FIXME: we only support one audio track.
if (track_desc->type_ == "audio" && !stream_desc->audio_track_desc_) {
stream_desc->audio_track_desc_ = track_desc->copy();
}
if (track_desc->type_ == "video") {
stream_desc->video_track_descs_.push_back(track_desc->copy());
}
++it;
}
if ((err = generate_play_local_sdp(req, local_sdp, stream_desc, conf->remote_sdp.is_unified())) != srs_success) {
return srs_error_wrap(err, "generate local sdp");
}
if ((err = create_player(req, play_sub_relations)) != srs_success) {
return srs_error_wrap(err, "create player");
}
return err;
}
最主要的是add_player这个函数了，明天再继续分析
