// // Copyright (c) 2013-2021 Winlin // // SPDX-License-Identifier: MIT // #include #include #include #include #include #include #include #include #include #include #include using namespace std; SrsGoApiRtcPlay::SrsGoApiRtcPlay(SrsRtcServer* server) { server_ = server; } SrsGoApiRtcPlay::~SrsGoApiRtcPlay() { } // Request: // POST /rtc/v1/play/ // { // "sdp":"offer...", "streamurl":"webrtc://r.ossrs.net/live/livestream", // "api":'http...", "clientip":"..." // } // Response: // {"sdp":"answer...", "sid":"..."} // @see https://github.com/rtcdn/rtcdn-draft srs_error_t SrsGoApiRtcPlay::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r) { srs_error_t err = srs_success; SrsJsonObject* res = SrsJsonAny::object(); SrsAutoFree(SrsJsonObject, res); if ((err = do_serve_http(w, r, res)) != srs_success) { srs_warn("RTC error %s", srs_error_desc(err).c_str()); srs_freep(err); return srs_api_response_code(w, r, SRS_CONSTS_HTTP_BadRequest); } return srs_api_response(w, r, res->dumps()); } srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res) { srs_error_t err = srs_success; // 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; SrsAutoFree(SrsJsonObject, req); 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(); } string tid; if ((prop = req->ensure_property_string("tid")) != NULL) { tid = prop->to_str(); } // TODO: FIXME: Parse vhost. // 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"); string codec = r->query_get("codec"); // For client to specifies whether encrypt by SRTP. string srtp = r->query_get("encrypt"); string dtls = r->query_get("dtls"); srs_trace("RTC play %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s, srtp=%s, dtls=%s", streamurl.c_str(), api.c_str(), tid.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(), codec.c_str(), srtp.c_str(), dtls.c_str() ); // The RTC user config object. SrsRtcUserConfig ruc; ruc.eip_ = eip; ruc.codec_ = codec; ruc.publish_ = false; ruc.dtls_ = (dtls != "false"); if (srtp.empty()) { ruc.srtp_ = _srs_config->get_rtc_server_encrypt(); } else { ruc.srtp_ = (srtp != "false"); } // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information. if ((err = ruc.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(ruc.remote_sdp_)) != srs_success) { return srs_error_wrap(err, "remote sdp check failed"); } ruc.req_->app = app; ruc.req_->stream = stream_name; // TODO: FIXME: Parse vhost. // discovery vhost, resolve the vhost from config SrsConfDirective* parsed_vhost = _srs_config->get_vhost(""); if (parsed_vhost) { ruc.req_->vhost = parsed_vhost->arg0(); } SrsSdp local_sdp; // Config for SDP and session. local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost); local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost); // Whether enabled. bool server_enabled = _srs_config->get_rtc_server_enabled(); bool rtc_enabled = _srs_config->get_rtc_enabled(ruc.req_->vhost); if (server_enabled && !rtc_enabled) { srs_warn("RTC disabled in vhost %s", ruc.req_->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, ruc.req_->vhost.c_str()); } // TODO: FIXME: When server enabled, but vhost disabled, should report error. SrsRtcConnection* session = NULL; if ((err = server_->create_session(&ruc, local_sdp, &session)) != srs_success) { return srs_error_wrap(err, "create session, dtls=%u, srtp=%u, eip=%s", ruc.dtls_, ruc.srtp_, eip.c_str()); } ostringstream os; if ((err = local_sdp.encode(os)) != srs_success) { return srs_error_wrap(err, "encode sdp"); } string local_sdp_str = os.str(); // Filter the \r\n to \\r\\n for JSON. local_sdp_str = srs_string_replace(local_sdp_str.c_str(), "\r\n", "\\r\\n"); res->set("code", SrsJsonAny::integer(ERROR_SUCCESS)); res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str())); // 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, dtls=%u, srtp=%u, offer=%dB, answer=%dB", session->username().c_str(), ruc.dtls_, ruc.srtp_, remote_sdp_str.length(), local_sdp_str.length()); srs_trace("RTC remote offer: %s", srs_string_replace(remote_sdp_str.c_str(), "\r\n", "\\r\\n").c_str()); srs_trace("RTC local answer: %s", local_sdp_str.c_str()); return err; } srs_error_t SrsGoApiRtcPlay::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::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::const_iterator iter_media = iter->payload_types_.begin(); iter_media != iter->payload_types_.end(); ++iter_media) { if (iter->sendonly_) { return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "play API only support sendrecv/recvonly"); } } } return err; } SrsGoApiRtcPublish::SrsGoApiRtcPublish(SrsRtcServer* server) { server_ = server; } SrsGoApiRtcPublish::~SrsGoApiRtcPublish() { } // Request: // POST /rtc/v1/publish/ // { // "sdp":"offer...", "streamurl":"webrtc://r.ossrs.net/live/livestream", // "api":'http...", "clientip":"..." // } // Response: // {"sdp":"answer...", "sid":"..."} // @see https://github.com/rtcdn/rtcdn-draft srs_error_t SrsGoApiRtcPublish::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r) { srs_error_t err = srs_success; SrsJsonObject* res = SrsJsonAny::object(); SrsAutoFree(SrsJsonObject, res); if ((err = do_serve_http(w, r, res)) != srs_success) { srs_warn("RTC error %s", srs_error_desc(err).c_str()); srs_freep(err); return srs_api_response_code(w, r, SRS_CONSTS_HTTP_BadRequest); } return srs_api_response(w, r, res->dumps()); } srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res) { srs_error_t err = srs_success; // 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(); } string tid; if ((prop = req->ensure_property_string("tid")) != NULL) { tid = 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"); string codec = r->query_get("codec"); srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s", streamurl.c_str(), api.c_str(), tid.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(), codec.c_str() ); // The RTC user config object. SrsRtcUserConfig ruc; ruc.eip_ = eip; ruc.codec_ = codec; ruc.publish_ = true; ruc.dtls_ = ruc.srtp_ = true; // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information. if ((err = ruc.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(ruc.remote_sdp_)) != srs_success) { return srs_error_wrap(err, "remote sdp check failed"); } ruc.req_->app = app; ruc.req_->stream = stream_name; // TODO: FIXME: Parse vhost. // discovery vhost, resolve the vhost from config SrsConfDirective* parsed_vhost = _srs_config->get_vhost(""); if (parsed_vhost) { ruc.req_->vhost = parsed_vhost->arg0(); } SrsSdp local_sdp; // TODO: FIXME: move to create_session. // Config for SDP and session. local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost); local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost); // Whether enabled. bool server_enabled = _srs_config->get_rtc_server_enabled(); bool rtc_enabled = _srs_config->get_rtc_enabled(ruc.req_->vhost); if (server_enabled && !rtc_enabled) { srs_warn("RTC disabled in vhost %s", ruc.req_->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, ruc.req_->vhost.c_str()); } // TODO: FIXME: When server enabled, but vhost disabled, should report error. SrsRtcConnection* session = NULL; if ((err = server_->create_session(&ruc, local_sdp, &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(); // Filter the \r\n to \\r\\n for JSON. local_sdp_str = srs_string_replace(local_sdp_str.c_str(), "\r\n", "\\r\\n"); res->set("code", SrsJsonAny::integer(ERROR_SUCCESS)); res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str())); // 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()); srs_trace("RTC remote offer: %s", srs_string_replace(remote_sdp_str.c_str(), "\r\n", "\\r\\n").c_str()); srs_trace("RTC local answer: %s", local_sdp_str.c_str()); 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::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::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; } 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); } SrsRtcConnection* 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 username=%s, drop=%s/%d", username.c_str(), dropv.c_str(), drop); return srs_success; }