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RTC: Support drop for specified PT
This commit is contained in:
parent
b9bd12192b
commit
3510c74cd4
5 changed files with 46 additions and 13 deletions
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@ -497,6 +497,9 @@ vhost rtc.vhost.srs.com {
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# The version of dtls, support dtls1.0, dtls1.2, and auto
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# default: auto
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dtls_version auto;
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# Drop the packet with the pt(payload type), 0 never drop.
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# default: 0
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drop_for_pt 0;
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}
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# whether enable min delay mode for vhost.
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# default: on, for RTC.
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@ -3935,7 +3935,7 @@ srs_error_t SrsConfig::check_normal_config()
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for (int j = 0; j < (int)conf->directives.size(); j++) {
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string m = conf->at(j)->name;
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if (m != "enabled" && m != "bframe" && m != "aac" && m != "stun_timeout" && m != "stun_strict_check"
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&& m != "dtls_role" && m != "dtls_version") {
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&& m != "dtls_role" && m != "dtls_version" && m != "drop_for_pt") {
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return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
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}
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}
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@ -5056,6 +5056,23 @@ std::string SrsConfig::get_rtc_dtls_version(string vhost)
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return conf->arg0();
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}
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int SrsConfig::get_rtc_drop_for_pt(string vhost)
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{
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static int DEFAULT = 0;
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SrsConfDirective* conf = get_vhost(vhost);
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if (!conf) {
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return DEFAULT;
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}
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conf = conf->get("drop_for_pt");
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if (!conf || conf->arg0().empty()) {
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return DEFAULT;
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}
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return ::atoi(conf->arg0().c_str());
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}
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bool SrsConfig::get_rtc_nack_enabled(string vhost)
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{
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static bool DEFAULT = true;
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@ -546,6 +546,7 @@ public:
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bool get_rtc_stun_strict_check(std::string vhost);
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std::string get_rtc_dtls_role(std::string vhost);
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std::string get_rtc_dtls_version(std::string vhost);
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int get_rtc_drop_for_pt(std::string vhost);
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bool get_rtc_nack_enabled(std::string vhost);
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bool get_rtc_twcc_enabled(std::string vhost);
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@ -855,6 +855,7 @@ SrsRtcPublisher::SrsRtcPublisher(SrsRtcSession* session)
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source = NULL;
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nn_simulate_nack_drop = 0;
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nack_enabled_ = false;
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pt_to_drop_ = 0;
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nn_audio_frames = 0;
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twcc_ext_id_ = 0;
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@ -892,9 +893,10 @@ srs_error_t SrsRtcPublisher::initialize(uint32_t vssrc, uint32_t assrc, uint8_t
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}
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// TODO: FIXME: Support reload.
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nack_enabled_ = _srs_config->get_rtc_nack_enabled(session_->req->vhost);
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pt_to_drop_ = (uint16_t)_srs_config->get_rtc_drop_for_pt(session_->req->vhost);
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srs_trace("RTC publisher video(ssrc=%u), audio(ssrc=%u), nack=%d",
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video_ssrc, audio_ssrc, nack_enabled_);
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srs_trace("RTC publisher video(ssrc=%u), audio(ssrc=%u), nack=%d, pt-drop=%u",
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video_ssrc, audio_ssrc, nack_enabled_, pt_to_drop_);
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if ((err = report_timer->tick(0 * SRS_UTIME_MILLISECONDS)) != srs_success) {
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return srs_error_wrap(err, "hourglass tick");
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@ -1133,6 +1135,7 @@ srs_error_t SrsRtcPublisher::on_twcc(uint16_t sn) {
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srs_utime_t now = srs_get_system_time();
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return rtcp_twcc_.recv_packet(sn, now);
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}
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srs_error_t SrsRtcPublisher::on_rtp(char* data, int nb_data)
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{
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srs_error_t err = srs_success;
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@ -1144,18 +1147,21 @@ srs_error_t SrsRtcPublisher::on_rtp(char* data, int nb_data)
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return err;
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}
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// Decode the header first.
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SrsRtpHeader h;
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if (pt_to_drop_ && twcc_ext_id_) {
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SrsBuffer b(data, nb_data);
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h.ignore_padding(true); h.set_extensions(&extension_types_);
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if ((err = h.decode(&b)) != srs_success) {
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return srs_error_wrap(err, "twcc decode header");
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}
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}
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// We must parse the TWCC from RTP header before SRTP unprotect, because:
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// 1. Client may send some padding packets with invalid SequenceNumber, which causes the SRTP fail.
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// 2. Server may send multiple duplicated NACK to client, and got more than one ARQ packet, which also fail SRTP.
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// so, we must parse the header before SRTP unprotect(which may fail and drop packet).
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if (0 != twcc_ext_id_) {
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SrsBuffer b(data, nb_data); SrsRtpHeader h;
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h.ignore_padding(true);
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h.set_extensions(&extension_types_);
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if ((err = h.decode(&b)) != srs_success) {
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return srs_error_wrap(err, "twcc decode header");
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}
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if (twcc_ext_id_) {
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uint16_t twcc_sn = 0;
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if ((err = h.get_twcc_sequence_number(twcc_sn)) == srs_success) {
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if((err = on_twcc(twcc_sn)) != srs_success) {
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@ -1166,6 +1172,11 @@ srs_error_t SrsRtcPublisher::on_rtp(char* data, int nb_data)
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}
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}
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// If payload type is configed to drop, ignore this packet.
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if (pt_to_drop_ && pt_to_drop_ == h.get_payload_type()) {
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return err;
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}
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// Decrypt the cipher to plaintext RTP data.
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int nb_unprotected_buf = nb_data;
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char* unprotected_buf = new char[kRtpPacketSize];
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@ -251,6 +251,9 @@ private:
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SrsRtcSession* session_;
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uint32_t video_ssrc;
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uint32_t audio_ssrc;
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uint16_t pt_to_drop_;
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// Whether enabled nack.
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bool nack_enabled_;
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private:
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bool request_keyframe_;
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SrsRtpRingBuffer* video_queue_;
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@ -260,8 +263,6 @@ private:
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private:
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SrsRequest* req;
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SrsRtcSource* source;
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// Whether enabled nack.
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bool nack_enabled_;
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// Simulators.
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int nn_simulate_nack_drop;
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private:
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