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RTC: Extract RTC api file
This commit is contained in:
忘篱
parent
d603b1580e
commit
acc53ded89
8 changed files with 13 additions and 812 deletions
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@ -47,10 +47,6 @@ using namespace std;
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#include <srs_protocol_amf0.hpp>
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#include <srs_protocol_utility.hpp>
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#include <srs_app_coworkers.hpp>
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#ifdef SRS_RTC
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#include <srs_app_rtc_conn.hpp>
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#include <srs_app_rtc_server.hpp>
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#endif
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srs_error_t srs_api_response_jsonp(ISrsHttpResponseWriter* w, string callback, string data)
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{
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@ -799,760 +795,6 @@ srs_error_t SrsGoApiStreams::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
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return srs_api_response(w, r, obj->dumps());
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}
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#ifdef SRS_RTC
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uint32_t SrsGoApiRtcPlay::ssrc_num = 0;
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SrsGoApiRtcPlay::SrsGoApiRtcPlay(SrsRtcServer* server)
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{
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server_ = server;
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}
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SrsGoApiRtcPlay::~SrsGoApiRtcPlay()
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{
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}
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// Request:
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// POST /rtc/v1/play/
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// {
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// "sdp":"offer...", "streamurl":"webrtc://r.ossrs.net/live/livestream",
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// "api":'http...", "clientip":"..."
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// }
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// Response:
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// {"sdp":"answer...", "sid":"..."}
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// @see https://github.com/rtcdn/rtcdn-draft
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srs_error_t SrsGoApiRtcPlay::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
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{
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srs_error_t err = srs_success;
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SrsJsonObject* res = SrsJsonAny::object();
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SrsAutoFree(SrsJsonObject, res);
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if ((err = do_serve_http(w, r, res)) != srs_success) {
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srs_warn("RTC error %s", srs_error_desc(err).c_str()); srs_freep(err);
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return srs_api_response_code(w, r, SRS_CONSTS_HTTP_BadRequest);
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}
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return srs_api_response(w, r, res->dumps());
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}
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srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res)
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{
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srs_error_t err = srs_success;
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// For each RTC session, we use short-term HTTP connection.
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SrsHttpHeader* hdr = w->header();
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hdr->set("Connection", "Close");
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// Parse req, the request json object, from body.
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SrsJsonObject* req = NULL;
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SrsAutoFree(SrsJsonObject, req);
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if (true) {
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string req_json;
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if ((err = r->body_read_all(req_json)) != srs_success) {
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return srs_error_wrap(err, "read body");
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}
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SrsJsonAny* json = SrsJsonAny::loads(req_json);
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if (!json || !json->is_object()) {
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return srs_error_new(ERROR_RTC_API_BODY, "invalid body %s", req_json.c_str());
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}
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req = json->to_object();
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}
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// Fetch params from req object.
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SrsJsonAny* prop = NULL;
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if ((prop = req->ensure_property_string("sdp")) == NULL) {
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return srs_error_wrap(err, "not sdp");
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}
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string remote_sdp_str = prop->to_str();
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if ((prop = req->ensure_property_string("streamurl")) == NULL) {
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return srs_error_wrap(err, "not streamurl");
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}
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string streamurl = prop->to_str();
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string clientip;
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if ((prop = req->ensure_property_string("clientip")) != NULL) {
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clientip = prop->to_str();
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}
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string api;
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if ((prop = req->ensure_property_string("api")) != NULL) {
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api = prop->to_str();
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}
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// TODO: FIXME: Parse vhost.
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// Parse app and stream from streamurl.
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string app;
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string stream_name;
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if (true) {
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string tcUrl;
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srs_parse_rtmp_url(streamurl, tcUrl, stream_name);
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int port;
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string schema, host, vhost, param;
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srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
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}
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// For client to specifies the EIP of server.
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string eip = r->query_get("eip");
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// For client to specifies whether encrypt by SRTP.
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string encrypt = r->query_get("encrypt");
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srs_trace("RTC play %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, encrypt=%s",
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streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(),
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eip.c_str(), encrypt.c_str());
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// TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
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SrsSdp remote_sdp;
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if ((err = remote_sdp.parse(remote_sdp_str)) != srs_success) {
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return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
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}
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if ((err = check_remote_sdp(remote_sdp)) != srs_success) {
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return srs_error_wrap(err, "remote sdp check failed");
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}
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SrsRequest request;
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request.app = app;
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request.stream = stream_name;
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// TODO: FIXME: Parse vhost.
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// discovery vhost, resolve the vhost from config
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SrsConfDirective* parsed_vhost = _srs_config->get_vhost("");
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if (parsed_vhost) {
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request.vhost = parsed_vhost->arg0();
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}
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SrsSdp local_sdp;
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if ((err = exchange_sdp(&request, remote_sdp, local_sdp)) != srs_success) {
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return srs_error_wrap(err, "remote sdp have error or unsupport attributes");
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}
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// Whether enabled.
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bool server_enabled = _srs_config->get_rtc_server_enabled();
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bool rtc_enabled = _srs_config->get_rtc_enabled(request.vhost);
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if (server_enabled && !rtc_enabled) {
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srs_warn("RTC disabled in vhost %s", request.vhost.c_str());
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}
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if (!server_enabled || !rtc_enabled) {
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return srs_error_new(ERROR_RTC_DISABLED, "Disabled server=%d, rtc=%d, vhost=%s",
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server_enabled, rtc_enabled, request.vhost.c_str());
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}
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// TODO: FIXME: When server enabled, but vhost disabled, should report error.
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SrsRtcSession* session = NULL;
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if ((err = server_->create_session(&request, remote_sdp, local_sdp, eip, false, &session)) != srs_success) {
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return srs_error_wrap(err, "create session");
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}
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if (encrypt.empty()) {
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session->set_encrypt(_srs_config->get_rtc_server_encrypt());
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} else {
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session->set_encrypt(encrypt != "false");
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}
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ostringstream os;
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if ((err = local_sdp.encode(os)) != srs_success) {
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return srs_error_wrap(err, "encode sdp");
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}
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string local_sdp_str = os.str();
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srs_verbose("local_sdp=%s", local_sdp_str.c_str());
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res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
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res->set("server", SrsJsonAny::integer(SrsStatistic::instance()->server_id()));
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// TODO: add candidates in response json?
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res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
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res->set("sessionid", SrsJsonAny::str(session->username().c_str()));
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srs_trace("RTC username=%s, offer=%dB, answer=%dB", session->username().c_str(),
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remote_sdp_str.length(), local_sdp_str.length());
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return err;
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}
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srs_error_t SrsGoApiRtcPlay::check_remote_sdp(const SrsSdp& remote_sdp)
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{
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srs_error_t err = srs_success;
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if (remote_sdp.group_policy_ != "BUNDLE") {
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return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "now only support BUNDLE, group policy=%s", remote_sdp.group_policy_.c_str());
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}
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if (remote_sdp.media_descs_.empty()) {
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return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no media descriptions");
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}
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for (std::vector<SrsMediaDesc>::const_iterator iter = remote_sdp.media_descs_.begin(); iter != remote_sdp.media_descs_.end(); ++iter) {
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if (iter->type_ != "audio" && iter->type_ != "video") {
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return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "unsupport media type=%s", iter->type_.c_str());
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}
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if (! iter->rtcp_mux_) {
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return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "now only suppor rtcp-mux");
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}
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for (std::vector<SrsMediaPayloadType>::const_iterator iter_media = iter->payload_types_.begin(); iter_media != iter->payload_types_.end(); ++iter_media) {
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if (iter->sendonly_) {
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return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "play API only support sendrecv/recvonly");
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}
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}
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}
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return err;
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}
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srs_error_t SrsGoApiRtcPlay::exchange_sdp(SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp)
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{
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srs_error_t err = srs_success;
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local_sdp.version_ = "0";
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local_sdp.username_ = RTMP_SIG_SRS_SERVER;
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local_sdp.session_id_ = srs_int2str((int64_t)this);
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local_sdp.session_version_ = "2";
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local_sdp.nettype_ = "IN";
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local_sdp.addrtype_ = "IP4";
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local_sdp.unicast_address_ = "0.0.0.0";
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local_sdp.session_name_ = "SRSPlaySession";
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local_sdp.msid_semantic_ = "WMS";
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local_sdp.msids_.push_back(req->app + "/" + req->stream);
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local_sdp.group_policy_ = "BUNDLE";
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bool nack_enabled = _srs_config->get_rtc_nack_enabled(req->vhost);
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for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
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const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_[i];
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if (remote_media_desc.is_audio()) {
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local_sdp.media_descs_.push_back(SrsMediaDesc("audio"));
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} else if (remote_media_desc.is_video()) {
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local_sdp.media_descs_.push_back(SrsMediaDesc("video"));
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}
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SrsMediaDesc& local_media_desc = local_sdp.media_descs_.back();
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if (remote_media_desc.is_audio()) {
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// TODO: check opus format specific param
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std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("opus");
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for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
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local_media_desc.payload_types_.push_back(*iter);
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SrsMediaPayloadType& payload_type = local_media_desc.payload_types_.back();
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// TODO: FIXME: Only support some transport algorithms.
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vector<string> rtcp_fb;
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payload_type.rtcp_fb_.swap(rtcp_fb);
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for (int j = 0; j < (int)rtcp_fb.size(); j++) {
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if (nack_enabled) {
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if (rtcp_fb.at(j) == "nack" || rtcp_fb.at(j) == "nack pli") {
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payload_type.rtcp_fb_.push_back(rtcp_fb.at(j));
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}
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}
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}
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// Only choose one match opus codec.
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break;
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}
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if (local_media_desc.payload_types_.empty()) {
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return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid opus payload type");
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}
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} else if (remote_media_desc.is_video()) {
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std::deque<SrsMediaPayloadType> backup_payloads;
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std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");
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for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
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if (iter->format_specific_param_.empty()) {
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backup_payloads.push_front(*iter);
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continue;
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}
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H264SpecificParam h264_param;
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if ((err = parse_h264_fmtp(iter->format_specific_param_, h264_param)) != srs_success) {
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srs_error_reset(err); continue;
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}
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// Try to pick the "best match" H.264 payload type.
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if (h264_param.packetization_mode == "1" && h264_param.level_asymmerty_allow == "1") {
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local_media_desc.payload_types_.push_back(*iter);
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SrsMediaPayloadType& payload_type = local_media_desc.payload_types_.back();
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// TODO: FIXME: Only support some transport algorithms.
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vector<string> rtcp_fb;
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payload_type.rtcp_fb_.swap(rtcp_fb);
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for (int j = 0; j < (int)rtcp_fb.size(); j++) {
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if (nack_enabled) {
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if (rtcp_fb.at(j) == "nack" || rtcp_fb.at(j) == "nack pli") {
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payload_type.rtcp_fb_.push_back(rtcp_fb.at(j));
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}
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}
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}
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// Only choose first match H.264 payload type.
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break;
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}
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backup_payloads.push_back(*iter);
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}
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// Try my best to pick at least one media payload type.
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if (local_media_desc.payload_types_.empty() && ! backup_payloads.empty()) {
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srs_warn("choose backup H.264 payload type=%d", backup_payloads.front().payload_type_);
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local_media_desc.payload_types_.push_back(backup_payloads.front());
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}
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if (local_media_desc.payload_types_.empty()) {
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return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid H.264 payload type");
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}
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}
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local_media_desc.mid_ = remote_media_desc.mid_;
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local_sdp.groups_.push_back(local_media_desc.mid_);
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local_media_desc.port_ = 9;
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local_media_desc.protos_ = "UDP/TLS/RTP/SAVPF";
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if (remote_media_desc.session_info_.setup_ == "active") {
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local_media_desc.session_info_.setup_ = "passive";
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} else if (remote_media_desc.session_info_.setup_ == "passive") {
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local_media_desc.session_info_.setup_ = "active";
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} else if (remote_media_desc.session_info_.setup_ == "actpass") {
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local_media_desc.session_info_.setup_ = "passive";
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} else {
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// @see: https://tools.ietf.org/html/rfc4145#section-4.1
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// The default value of the setup attribute in an offer/answer exchange
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// is 'active' in the offer and 'passive' in the answer.
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local_media_desc.session_info_.setup_ = "passive";
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}
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if (remote_media_desc.sendonly_) {
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local_media_desc.recvonly_ = true;
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} else if (remote_media_desc.recvonly_) {
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local_media_desc.sendonly_ = true;
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} else if (remote_media_desc.sendrecv_) {
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local_media_desc.sendrecv_ = true;
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}
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local_media_desc.rtcp_mux_ = true;
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local_media_desc.rtcp_rsize_ = true;
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// TODO: FIXME: Avoid SSRC collision.
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if (!ssrc_num) {
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ssrc_num = ::getpid() * 10000 + ::getpid() * 100 + ::getpid();
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}
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if (local_media_desc.sendonly_ || local_media_desc.sendrecv_) {
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SrsSSRCInfo ssrc_info;
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ssrc_info.ssrc_ = ++ssrc_num;
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// TODO:use formated cname
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ssrc_info.cname_ = "test_sdp_cname";
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local_media_desc.ssrc_infos_.push_back(ssrc_info);
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}
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}
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return err;
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}
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uint32_t SrsGoApiRtcPublish::ssrc_num = 0;
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SrsGoApiRtcPublish::SrsGoApiRtcPublish(SrsRtcServer* server)
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{
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server_ = server;
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}
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SrsGoApiRtcPublish::~SrsGoApiRtcPublish()
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{
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}
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// Request:
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// POST /rtc/v1/publish/
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// {
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// "sdp":"offer...", "streamurl":"webrtc://r.ossrs.net/live/livestream",
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// "api":'http...", "clientip":"..."
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// }
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// Response:
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// {"sdp":"answer...", "sid":"..."}
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// @see https://github.com/rtcdn/rtcdn-draft
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srs_error_t SrsGoApiRtcPublish::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
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{
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srs_error_t err = srs_success;
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SrsJsonObject* res = SrsJsonAny::object();
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SrsAutoFree(SrsJsonObject, res);
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if ((err = do_serve_http(w, r, res)) != srs_success) {
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srs_warn("RTC error %s", srs_error_desc(err).c_str()); srs_freep(err);
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return srs_api_response_code(w, r, SRS_CONSTS_HTTP_BadRequest);
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}
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return srs_api_response(w, r, res->dumps());
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}
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srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res)
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{
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srs_error_t err = srs_success;
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// For each RTC session, we use short-term HTTP connection.
|
||||
SrsHttpHeader* hdr = w->header();
|
||||
hdr->set("Connection", "Close");
|
||||
|
||||
// Parse req, the request json object, from body.
|
||||
SrsJsonObject* req = NULL;
|
||||
if (true) {
|
||||
string req_json;
|
||||
if ((err = r->body_read_all(req_json)) != srs_success) {
|
||||
return srs_error_wrap(err, "read body");
|
||||
}
|
||||
|
||||
SrsJsonAny* json = SrsJsonAny::loads(req_json);
|
||||
if (!json || !json->is_object()) {
|
||||
return srs_error_new(ERROR_RTC_API_BODY, "invalid body %s", req_json.c_str());
|
||||
}
|
||||
|
||||
req = json->to_object();
|
||||
}
|
||||
|
||||
// Fetch params from req object.
|
||||
SrsJsonAny* prop = NULL;
|
||||
if ((prop = req->ensure_property_string("sdp")) == NULL) {
|
||||
return srs_error_wrap(err, "not sdp");
|
||||
}
|
||||
string remote_sdp_str = prop->to_str();
|
||||
|
||||
if ((prop = req->ensure_property_string("streamurl")) == NULL) {
|
||||
return srs_error_wrap(err, "not streamurl");
|
||||
}
|
||||
string streamurl = prop->to_str();
|
||||
|
||||
string clientip;
|
||||
if ((prop = req->ensure_property_string("clientip")) != NULL) {
|
||||
clientip = prop->to_str();
|
||||
}
|
||||
|
||||
string api;
|
||||
if ((prop = req->ensure_property_string("api")) != NULL) {
|
||||
api = prop->to_str();
|
||||
}
|
||||
|
||||
// Parse app and stream from streamurl.
|
||||
string app;
|
||||
string stream_name;
|
||||
if (true) {
|
||||
string tcUrl;
|
||||
srs_parse_rtmp_url(streamurl, tcUrl, stream_name);
|
||||
|
||||
int port;
|
||||
string schema, host, vhost, param;
|
||||
srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
|
||||
}
|
||||
|
||||
// For client to specifies the EIP of server.
|
||||
string eip = r->query_get("eip");
|
||||
|
||||
srs_trace("RTC publish %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s",
|
||||
streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str());
|
||||
|
||||
// TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
|
||||
SrsSdp remote_sdp;
|
||||
if ((err = remote_sdp.parse(remote_sdp_str)) != srs_success) {
|
||||
return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
|
||||
}
|
||||
|
||||
if ((err = check_remote_sdp(remote_sdp)) != srs_success) {
|
||||
return srs_error_wrap(err, "remote sdp check failed");
|
||||
}
|
||||
|
||||
SrsRequest request;
|
||||
request.app = app;
|
||||
request.stream = stream_name;
|
||||
|
||||
// TODO: FIXME: Parse vhost.
|
||||
// discovery vhost, resolve the vhost from config
|
||||
SrsConfDirective* parsed_vhost = _srs_config->get_vhost("");
|
||||
if (parsed_vhost) {
|
||||
request.vhost = parsed_vhost->arg0();
|
||||
}
|
||||
|
||||
SrsSdp local_sdp;
|
||||
if ((err = exchange_sdp(&request, remote_sdp, local_sdp)) != srs_success) {
|
||||
return srs_error_wrap(err, "remote sdp have error or unsupport attributes");
|
||||
}
|
||||
|
||||
// Whether enabled.
|
||||
bool server_enabled = _srs_config->get_rtc_server_enabled();
|
||||
bool rtc_enabled = _srs_config->get_rtc_enabled(request.vhost);
|
||||
if (server_enabled && !rtc_enabled) {
|
||||
srs_warn("RTC disabled in vhost %s", request.vhost.c_str());
|
||||
}
|
||||
if (!server_enabled || !rtc_enabled) {
|
||||
return srs_error_new(ERROR_RTC_DISABLED, "Disabled server=%d, rtc=%d, vhost=%s",
|
||||
server_enabled, rtc_enabled, request.vhost.c_str());
|
||||
}
|
||||
|
||||
// TODO: FIXME: When server enabled, but vhost disabled, should report error.
|
||||
SrsRtcSession* session = NULL;
|
||||
if ((err = server_->create_session(&request, remote_sdp, local_sdp, eip, true, &session)) != srs_success) {
|
||||
return srs_error_wrap(err, "create session");
|
||||
}
|
||||
|
||||
ostringstream os;
|
||||
if ((err = local_sdp.encode(os)) != srs_success) {
|
||||
return srs_error_wrap(err, "encode sdp");
|
||||
}
|
||||
|
||||
string local_sdp_str = os.str();
|
||||
|
||||
srs_verbose("local_sdp=%s", local_sdp_str.c_str());
|
||||
|
||||
res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
|
||||
res->set("server", SrsJsonAny::integer(SrsStatistic::instance()->server_id()));
|
||||
|
||||
// TODO: add candidates in response json?
|
||||
|
||||
res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
|
||||
res->set("sessionid", SrsJsonAny::str(session->username().c_str()));
|
||||
|
||||
srs_trace("RTC username=%s, offer=%dB, answer=%dB", session->username().c_str(),
|
||||
remote_sdp_str.length(), local_sdp_str.length());
|
||||
|
||||
return err;
|
||||
}
|
||||
|
||||
srs_error_t SrsGoApiRtcPublish::check_remote_sdp(const SrsSdp& remote_sdp)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
if (remote_sdp.group_policy_ != "BUNDLE") {
|
||||
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "now only support BUNDLE, group policy=%s", remote_sdp.group_policy_.c_str());
|
||||
}
|
||||
|
||||
if (remote_sdp.media_descs_.empty()) {
|
||||
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no media descriptions");
|
||||
}
|
||||
|
||||
for (std::vector<SrsMediaDesc>::const_iterator iter = remote_sdp.media_descs_.begin(); iter != remote_sdp.media_descs_.end(); ++iter) {
|
||||
if (iter->type_ != "audio" && iter->type_ != "video") {
|
||||
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "unsupport media type=%s", iter->type_.c_str());
|
||||
}
|
||||
|
||||
if (! iter->rtcp_mux_) {
|
||||
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "now only suppor rtcp-mux");
|
||||
}
|
||||
|
||||
for (std::vector<SrsMediaPayloadType>::const_iterator iter_media = iter->payload_types_.begin(); iter_media != iter->payload_types_.end(); ++iter_media) {
|
||||
if (iter->recvonly_) {
|
||||
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "publish API only support sendrecv/sendonly");
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
return err;
|
||||
}
|
||||
|
||||
srs_error_t SrsGoApiRtcPublish::exchange_sdp(SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
local_sdp.version_ = "0";
|
||||
|
||||
local_sdp.username_ = RTMP_SIG_SRS_SERVER;
|
||||
local_sdp.session_id_ = srs_int2str((int64_t)this);
|
||||
local_sdp.session_version_ = "2";
|
||||
local_sdp.nettype_ = "IN";
|
||||
local_sdp.addrtype_ = "IP4";
|
||||
local_sdp.unicast_address_ = "0.0.0.0";
|
||||
|
||||
local_sdp.session_name_ = "SRSPublishSession";
|
||||
|
||||
local_sdp.msid_semantic_ = "WMS";
|
||||
local_sdp.msids_.push_back(req->app + "/" + req->stream);
|
||||
|
||||
local_sdp.group_policy_ = "BUNDLE";
|
||||
|
||||
bool nack_enabled = _srs_config->get_rtc_nack_enabled(req->vhost);
|
||||
|
||||
for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
|
||||
const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_[i];
|
||||
|
||||
if (remote_media_desc.is_audio()) {
|
||||
local_sdp.media_descs_.push_back(SrsMediaDesc("audio"));
|
||||
} else if (remote_media_desc.is_video()) {
|
||||
local_sdp.media_descs_.push_back(SrsMediaDesc("video"));
|
||||
}
|
||||
|
||||
SrsMediaDesc& local_media_desc = local_sdp.media_descs_.back();
|
||||
|
||||
if (remote_media_desc.is_audio()) {
|
||||
// TODO: check opus format specific param
|
||||
std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("opus");
|
||||
for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
|
||||
local_media_desc.payload_types_.push_back(*iter);
|
||||
SrsMediaPayloadType& payload_type = local_media_desc.payload_types_.back();
|
||||
|
||||
// TODO: FIXME: Only support some transport algorithms.
|
||||
vector<string> rtcp_fb;
|
||||
payload_type.rtcp_fb_.swap(rtcp_fb);
|
||||
for (int j = 0; j < (int)rtcp_fb.size(); j++) {
|
||||
if (nack_enabled) {
|
||||
if (rtcp_fb.at(j) == "nack" || rtcp_fb.at(j) == "nack pli") {
|
||||
payload_type.rtcp_fb_.push_back(rtcp_fb.at(j));
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// Only choose one match opus codec.
|
||||
break;
|
||||
}
|
||||
|
||||
if (local_media_desc.payload_types_.empty()) {
|
||||
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found opus payload type");
|
||||
}
|
||||
|
||||
} else if (remote_media_desc.is_video()) {
|
||||
std::deque<SrsMediaPayloadType> backup_payloads;
|
||||
std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");
|
||||
for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
|
||||
if (iter->format_specific_param_.empty()) {
|
||||
backup_payloads.push_front(*iter);
|
||||
continue;
|
||||
}
|
||||
H264SpecificParam h264_param;
|
||||
if ((err = parse_h264_fmtp(iter->format_specific_param_, h264_param)) != srs_success) {
|
||||
srs_error_reset(err); continue;
|
||||
}
|
||||
|
||||
// Try to pick the "best match" H.264 payload type.
|
||||
if (h264_param.packetization_mode == "1" && h264_param.level_asymmerty_allow == "1") {
|
||||
local_media_desc.payload_types_.push_back(*iter);
|
||||
SrsMediaPayloadType& payload_type = local_media_desc.payload_types_.back();
|
||||
|
||||
// TODO: FIXME: Only support some transport algorithms.
|
||||
vector<string> rtcp_fb;
|
||||
payload_type.rtcp_fb_.swap(rtcp_fb);
|
||||
for (int j = 0; j < (int)rtcp_fb.size(); j++) {
|
||||
if (nack_enabled) {
|
||||
if (rtcp_fb.at(j) == "nack" || rtcp_fb.at(j) == "nack pli") {
|
||||
payload_type.rtcp_fb_.push_back(rtcp_fb.at(j));
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// Only choose first match H.264 payload type.
|
||||
break;
|
||||
}
|
||||
|
||||
backup_payloads.push_back(*iter);
|
||||
}
|
||||
|
||||
// Try my best to pick at least one media payload type.
|
||||
if (local_media_desc.payload_types_.empty() && ! backup_payloads.empty()) {
|
||||
srs_warn("choose backup H.264 payload type=%d", backup_payloads.front().payload_type_);
|
||||
local_media_desc.payload_types_.push_back(backup_payloads.front());
|
||||
}
|
||||
|
||||
if (local_media_desc.payload_types_.empty()) {
|
||||
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid H.264 payload type");
|
||||
}
|
||||
|
||||
// TODO: FIXME: Support RRTR?
|
||||
//local_media_desc.payload_types_.back().rtcp_fb_.push_back("rrtr");
|
||||
}
|
||||
|
||||
local_media_desc.mid_ = remote_media_desc.mid_;
|
||||
local_sdp.groups_.push_back(local_media_desc.mid_);
|
||||
|
||||
local_media_desc.port_ = 9;
|
||||
local_media_desc.protos_ = "UDP/TLS/RTP/SAVPF";
|
||||
|
||||
if (remote_media_desc.session_info_.setup_ == "active") {
|
||||
local_media_desc.session_info_.setup_ = "passive";
|
||||
} else if (remote_media_desc.session_info_.setup_ == "passive") {
|
||||
local_media_desc.session_info_.setup_ = "active";
|
||||
} else if (remote_media_desc.session_info_.setup_ == "actpass") {
|
||||
local_media_desc.session_info_.setup_ = "passive";
|
||||
} 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_media_desc.session_info_.setup_ = "passive";
|
||||
}
|
||||
|
||||
local_media_desc.rtcp_mux_ = true;
|
||||
|
||||
// For publisher, we are always sendonly.
|
||||
local_media_desc.sendonly_ = false;
|
||||
local_media_desc.recvonly_ = true;
|
||||
local_media_desc.sendrecv_ = false;
|
||||
}
|
||||
|
||||
return err;
|
||||
}
|
||||
|
||||
#ifdef SRS_SIMULATOR
|
||||
SrsGoApiRtcNACK::SrsGoApiRtcNACK(SrsRtcServer* server)
|
||||
{
|
||||
server_ = server;
|
||||
}
|
||||
|
||||
SrsGoApiRtcNACK::~SrsGoApiRtcNACK()
|
||||
{
|
||||
}
|
||||
|
||||
srs_error_t SrsGoApiRtcNACK::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
SrsJsonObject* res = SrsJsonAny::object();
|
||||
SrsAutoFree(SrsJsonObject, res);
|
||||
|
||||
res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
|
||||
|
||||
if ((err = do_serve_http(w, r, res)) != srs_success) {
|
||||
srs_warn("RTC NACK err %s", srs_error_desc(err).c_str());
|
||||
res->set("code", SrsJsonAny::integer(srs_error_code(err)));
|
||||
srs_freep(err);
|
||||
}
|
||||
|
||||
return srs_api_response(w, r, res->dumps());
|
||||
}
|
||||
|
||||
srs_error_t SrsGoApiRtcNACK::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res)
|
||||
{
|
||||
string username = r->query_get("username");
|
||||
string dropv = r->query_get("drop");
|
||||
|
||||
SrsJsonObject* query = SrsJsonAny::object();
|
||||
res->set("query", query);
|
||||
|
||||
query->set("username", SrsJsonAny::str(username.c_str()));
|
||||
query->set("drop", SrsJsonAny::str(dropv.c_str()));
|
||||
query->set("help", SrsJsonAny::str("?username=string&drop=int"));
|
||||
|
||||
int drop = ::atoi(dropv.c_str());
|
||||
if (drop <= 0) {
|
||||
return srs_error_new(ERROR_RTC_INVALID_PARAMS, "invalid drop=%s/%d", dropv.c_str(), drop);
|
||||
}
|
||||
|
||||
SrsRtcSession* session = server_->find_session_by_username(username);
|
||||
if (!session) {
|
||||
return srs_error_new(ERROR_RTC_NO_SESSION, "no session username=%s", username.c_str());
|
||||
}
|
||||
|
||||
session->simulate_nack_drop(drop);
|
||||
|
||||
srs_trace("RTC NACK session peer_id=%s, username=%s, drop=%s/%d", session->peer_id().c_str(),
|
||||
username.c_str(), dropv.c_str(), drop);
|
||||
|
||||
return srs_success;
|
||||
}
|
||||
#endif
|
||||
#endif
|
||||
|
||||
SrsGoApiClients::SrsGoApiClients()
|
||||
{
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue