mirror of
https://github.com/ossrs/srs.git
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2374 lines
74 KiB
C++
2374 lines
74 KiB
C++
/**
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* The MIT License (MIT)
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*
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* Copyright (c) 2013-2020 John
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
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* the Software, and to permit persons to whom the Software is furnished to do so,
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* subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
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* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
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* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
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* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
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* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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*/
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#include <srs_app_rtc_conn.hpp>
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using namespace std;
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#include <sys/socket.h>
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#include <netinet/in.h>
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#include <arpa/inet.h>
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#include <stdlib.h>
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#include <fcntl.h>
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#include <unistd.h>
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#include <sstream>
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#include <srs_core_autofree.hpp>
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#include <srs_kernel_buffer.hpp>
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#include <srs_kernel_rtc_rtp.hpp>
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#include <srs_kernel_error.hpp>
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#include <srs_kernel_log.hpp>
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#include <srs_rtc_stun_stack.hpp>
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#include <srs_rtmp_stack.hpp>
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#include <srs_rtmp_msg_array.hpp>
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#include <srs_app_rtc_dtls.hpp>
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#include <srs_app_utility.hpp>
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#include <srs_app_config.hpp>
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#include <srs_app_rtc_queue.hpp>
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#include <srs_app_source.hpp>
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#include <srs_app_server.hpp>
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#include <srs_service_utility.hpp>
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#include <srs_http_stack.hpp>
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#include <srs_app_http_api.hpp>
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#include <srs_app_statistic.hpp>
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#include <srs_app_pithy_print.hpp>
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#include <srs_service_st.hpp>
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#include <srs_app_rtc_server.hpp>
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#include <srs_app_rtc_source.hpp>
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// TODO: FIXME: Move to utility.
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string gen_random_str(int len)
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{
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static string random_table = "0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ";
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string ret;
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ret.reserve(len);
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for (int i = 0; i < len; ++i) {
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ret.append(1, random_table[random() % random_table.size()]);
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}
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return ret;
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}
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const int SRTP_MASTER_KEY_KEY_LEN = 16;
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const int SRTP_MASTER_KEY_SALT_LEN = 14;
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uint64_t SrsNtp::kMagicNtpFractionalUnit = 1ULL << 32;
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SrsNtp::SrsNtp()
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{
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system_ms_ = 0;
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ntp_ = 0;
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ntp_second_ = 0;
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ntp_fractions_ = 0;
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}
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SrsNtp::~SrsNtp()
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{
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}
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SrsNtp SrsNtp::from_time_ms(uint64_t ms)
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{
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SrsNtp srs_ntp;
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srs_ntp.system_ms_ = ms;
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srs_ntp.ntp_second_ = ms / 1000;
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srs_ntp.ntp_fractions_ = (static_cast<double>(ms % 1000 / 1000.0)) * kMagicNtpFractionalUnit;
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srs_ntp.ntp_ = (static_cast<uint64_t>(srs_ntp.ntp_second_) << 32) | srs_ntp.ntp_fractions_;
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return srs_ntp;
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}
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SrsNtp SrsNtp::to_time_ms(uint64_t ntp)
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{
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SrsNtp srs_ntp;
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srs_ntp.ntp_ = ntp;
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srs_ntp.ntp_second_ = (ntp & 0xFFFFFFFF00000000ULL) >> 32;
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srs_ntp.ntp_fractions_ = (ntp & 0x00000000FFFFFFFFULL);
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srs_ntp.system_ms_ = (static_cast<uint64_t>(srs_ntp.ntp_second_) * 1000) +
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(static_cast<double>(static_cast<uint64_t>(srs_ntp.ntp_fractions_) * 1000.0) / kMagicNtpFractionalUnit);
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return srs_ntp;
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}
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SrsRtcDtls::SrsRtcDtls(SrsRtcSession* s)
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{
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session_ = s;
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dtls = NULL;
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bio_in = NULL;
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bio_out = NULL;
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client_key = "";
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server_key = "";
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srtp_send = NULL;
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srtp_recv = NULL;
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handshake_done = false;
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}
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SrsRtcDtls::~SrsRtcDtls()
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{
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if (dtls) {
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// this function will free bio_in and bio_out
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SSL_free(dtls);
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dtls = NULL;
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}
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if (srtp_send) {
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srtp_dealloc(srtp_send);
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}
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if (srtp_recv) {
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srtp_dealloc(srtp_recv);
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}
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}
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srs_error_t SrsRtcDtls::initialize(SrsRequest* r)
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{
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srs_error_t err = srs_success;
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if ((err = SrsDtls::instance()->init(r)) != srs_success) {
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return srs_error_wrap(err, "DTLS init");
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}
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// TODO: FIXME: Support config by vhost to use RSA or ECDSA certificate.
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if ((dtls = SSL_new(SrsDtls::instance()->get_dtls_ctx())) == NULL) {
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return srs_error_new(ERROR_OpenSslCreateSSL, "SSL_new dtls");
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}
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// Dtls setup passive, as server role.
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SSL_set_accept_state(dtls);
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if ((bio_in = BIO_new(BIO_s_mem())) == NULL) {
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return srs_error_new(ERROR_OpenSslBIONew, "BIO_new in");
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}
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if ((bio_out = BIO_new(BIO_s_mem())) == NULL) {
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BIO_free(bio_in);
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return srs_error_new(ERROR_OpenSslBIONew, "BIO_new out");
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}
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SSL_set_bio(dtls, bio_in, bio_out);
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return err;
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}
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srs_error_t SrsRtcDtls::handshake()
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{
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srs_error_t err = srs_success;
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int ret = SSL_do_handshake(dtls);
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unsigned char *out_bio_data;
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int out_bio_len = BIO_get_mem_data(bio_out, &out_bio_data);
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int ssl_err = SSL_get_error(dtls, ret);
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switch(ssl_err) {
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case SSL_ERROR_NONE: {
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if ((err = on_dtls_handshake_done()) != srs_success) {
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return srs_error_wrap(err, "dtls handshake done handle");
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}
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break;
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}
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case SSL_ERROR_WANT_READ: {
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break;
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}
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case SSL_ERROR_WANT_WRITE: {
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break;
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}
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default: {
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break;
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}
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}
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if (out_bio_len) {
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if ((err = session_->sendonly_skt->sendto(out_bio_data, out_bio_len, 0)) != srs_success) {
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return srs_error_wrap(err, "send dtls packet");
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}
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}
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if (session_->blackhole && session_->blackhole_addr && session_->blackhole_stfd) {
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// Ignore any error for black-hole.
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void* p = out_bio_data; int len = out_bio_len; SrsRtcSession* s = session_;
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srs_sendto(s->blackhole_stfd, p, len, (sockaddr*)s->blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
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}
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return err;
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}
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srs_error_t SrsRtcDtls::on_dtls(char* data, int nb_data)
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{
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srs_error_t err = srs_success;
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if (BIO_reset(bio_in) != 1) {
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return srs_error_new(ERROR_OpenSslBIOReset, "BIO_reset");
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}
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if (BIO_reset(bio_out) != 1) {
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return srs_error_new(ERROR_OpenSslBIOReset, "BIO_reset");
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}
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if (BIO_write(bio_in, data, nb_data) <= 0) {
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// TODO: 0 or -1 maybe block, use BIO_should_retry to check.
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return srs_error_new(ERROR_OpenSslBIOWrite, "BIO_write");
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}
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if (session_->blackhole && session_->blackhole_addr && session_->blackhole_stfd) {
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// Ignore any error for black-hole.
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void* p = data; int len = nb_data; SrsRtcSession* s = session_;
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srs_sendto(s->blackhole_stfd, p, len, (sockaddr*)s->blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
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}
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if (!handshake_done) {
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err = handshake();
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} else {
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while (BIO_ctrl_pending(bio_in) > 0) {
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char dtls_read_buf[8092];
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int nb = SSL_read(dtls, dtls_read_buf, sizeof(dtls_read_buf));
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if (nb > 0) {
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if ((err =on_dtls_application_data(dtls_read_buf, nb)) != srs_success) {
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return srs_error_wrap(err, "dtls application data process");
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}
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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 SrsRtcDtls::on_dtls_handshake_done()
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{
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srs_error_t err = srs_success;
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srs_trace("rtc session=%s, DTLS handshake done.", session_->id().c_str());
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handshake_done = true;
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if ((err = srtp_initialize()) != srs_success) {
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return srs_error_wrap(err, "srtp init failed");
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}
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return session_->on_connection_established();
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}
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srs_error_t SrsRtcDtls::on_dtls_application_data(const char* buf, const int nb_buf)
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{
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srs_error_t err = srs_success;
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// TODO: process SCTP protocol(WebRTC datachannel support)
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return err;
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}
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srs_error_t SrsRtcDtls::srtp_initialize()
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{
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srs_error_t err = srs_success;
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unsigned char material[SRTP_MASTER_KEY_LEN * 2] = {0}; // client(SRTP_MASTER_KEY_KEY_LEN + SRTP_MASTER_KEY_SALT_LEN) + server
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static const string dtls_srtp_lable = "EXTRACTOR-dtls_srtp";
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if (!SSL_export_keying_material(dtls, material, sizeof(material), dtls_srtp_lable.c_str(), dtls_srtp_lable.size(), NULL, 0, 0)) {
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return srs_error_new(ERROR_RTC_SRTP_INIT, "SSL_export_keying_material failed");
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}
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size_t offset = 0;
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std::string client_master_key(reinterpret_cast<char*>(material), SRTP_MASTER_KEY_KEY_LEN);
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offset += SRTP_MASTER_KEY_KEY_LEN;
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std::string server_master_key(reinterpret_cast<char*>(material + offset), SRTP_MASTER_KEY_KEY_LEN);
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offset += SRTP_MASTER_KEY_KEY_LEN;
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std::string client_master_salt(reinterpret_cast<char*>(material + offset), SRTP_MASTER_KEY_SALT_LEN);
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offset += SRTP_MASTER_KEY_SALT_LEN;
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std::string server_master_salt(reinterpret_cast<char*>(material + offset), SRTP_MASTER_KEY_SALT_LEN);
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client_key = client_master_key + client_master_salt;
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server_key = server_master_key + server_master_salt;
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if ((err = srtp_send_init()) != srs_success) {
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return srs_error_wrap(err, "srtp send init failed");
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}
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if ((err = srtp_recv_init()) != srs_success) {
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return srs_error_wrap(err, "srtp recv init failed");
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}
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return err;
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}
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srs_error_t SrsRtcDtls::srtp_send_init()
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{
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srs_error_t err = srs_success;
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srtp_policy_t policy;
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bzero(&policy, sizeof(policy));
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// TODO: Maybe we can use SRTP-GCM in future.
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// @see https://bugs.chromium.org/p/chromium/issues/detail?id=713701
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// @see https://groups.google.com/forum/#!topic/discuss-webrtc/PvCbWSetVAQ
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srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
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srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
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policy.ssrc.type = ssrc_any_outbound;
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policy.ssrc.value = 0;
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// TODO: adjust window_size
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policy.window_size = 8192;
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policy.allow_repeat_tx = 1;
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policy.next = NULL;
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uint8_t *key = new uint8_t[server_key.size()];
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memcpy(key, server_key.data(), server_key.size());
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policy.key = key;
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if (srtp_create(&srtp_send, &policy) != srtp_err_status_ok) {
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srs_freepa(key);
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return srs_error_new(ERROR_RTC_SRTP_INIT, "srtp_create failed");
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}
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srs_freepa(key);
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return err;
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}
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srs_error_t SrsRtcDtls::srtp_recv_init()
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{
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srs_error_t err = srs_success;
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srtp_policy_t policy;
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bzero(&policy, sizeof(policy));
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srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
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srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
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policy.ssrc.type = ssrc_any_inbound;
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policy.ssrc.value = 0;
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// TODO: adjust window_size
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policy.window_size = 8192;
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policy.allow_repeat_tx = 1;
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policy.next = NULL;
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uint8_t *key = new uint8_t[client_key.size()];
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memcpy(key, client_key.data(), client_key.size());
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policy.key = key;
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// TODO: FIXME: Wrap error code.
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if (srtp_create(&srtp_recv, &policy) != srtp_err_status_ok) {
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srs_freepa(key);
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return srs_error_new(ERROR_RTC_SRTP_INIT, "srtp_create failed");
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}
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srs_freepa(key);
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return err;
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}
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srs_error_t SrsRtcDtls::protect_rtp(char* out_buf, const char* in_buf, int& nb_out_buf)
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{
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srs_error_t err = srs_success;
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if (srtp_send) {
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memcpy(out_buf, in_buf, nb_out_buf);
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// TODO: FIXME: Wrap error code.
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if (srtp_protect(srtp_send, out_buf, &nb_out_buf) != 0) {
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return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect failed");
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}
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return err;
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}
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return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect failed");
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}
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// TODO: FIXME: Merge with protect_rtp.
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srs_error_t SrsRtcDtls::protect_rtp2(void* rtp_hdr, int* len_ptr)
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{
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srs_error_t err = srs_success;
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if (!srtp_send) {
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return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect");
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}
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// TODO: FIXME: Wrap error code.
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if (srtp_protect(srtp_send, rtp_hdr, len_ptr) != 0) {
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return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect");
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}
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return err;
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}
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srs_error_t SrsRtcDtls::unprotect_rtp(char* out_buf, const char* in_buf, int& nb_out_buf)
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{
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srs_error_t err = srs_success;
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if (srtp_recv) {
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memcpy(out_buf, in_buf, nb_out_buf);
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srtp_err_status_t r0 = srtp_unprotect(srtp_recv, out_buf, &nb_out_buf);
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if (r0 != srtp_err_status_ok) {
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return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "unprotect r0=%u", r0);
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}
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return err;
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}
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return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtp unprotect failed");
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}
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srs_error_t SrsRtcDtls::protect_rtcp(char* out_buf, const char* in_buf, int& nb_out_buf)
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{
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srs_error_t err = srs_success;
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if (srtp_send) {
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memcpy(out_buf, in_buf, nb_out_buf);
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// TODO: FIXME: Wrap error code.
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if (srtp_protect_rtcp(srtp_send, out_buf, &nb_out_buf) != 0) {
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return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtcp protect failed");
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}
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return err;
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}
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return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtcp protect failed");
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}
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srs_error_t SrsRtcDtls::unprotect_rtcp(char* out_buf, const char* in_buf, int& nb_out_buf)
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{
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srs_error_t err = srs_success;
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if (srtp_recv) {
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memcpy(out_buf, in_buf, nb_out_buf);
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// TODO: FIXME: Wrap error code.
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if (srtp_unprotect_rtcp(srtp_recv, out_buf, &nb_out_buf) != srtp_err_status_ok) {
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return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtcp unprotect failed");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtcp unprotect failed");
|
|
}
|
|
|
|
SrsRtcOutgoingInfo::SrsRtcOutgoingInfo()
|
|
{
|
|
#if defined(SRS_DEBUG)
|
|
debug_id = 0;
|
|
#endif
|
|
|
|
nn_rtp_pkts = 0;
|
|
nn_audios = nn_extras = 0;
|
|
nn_videos = nn_samples = 0;
|
|
nn_bytes = nn_rtp_bytes = 0;
|
|
nn_padding_bytes = nn_paddings = 0;
|
|
}
|
|
|
|
SrsRtcOutgoingInfo::~SrsRtcOutgoingInfo()
|
|
{
|
|
}
|
|
|
|
SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, int parent_cid)
|
|
{
|
|
_parent_cid = parent_cid;
|
|
trd = new SrsDummyCoroutine();
|
|
|
|
session_ = s;
|
|
|
|
audio_sequence = 0;
|
|
video_sequence = 0;
|
|
mw_msgs = 0;
|
|
realtime = true;
|
|
|
|
// TODO: FIXME: Config the capacity?
|
|
audio_queue_ = new SrsRtpRingBuffer(100);
|
|
video_queue_ = new SrsRtpRingBuffer(1000);
|
|
|
|
nn_simulate_nack_drop = 0;
|
|
nack_enabled_ = false;
|
|
keep_sequence_ = false;
|
|
|
|
_srs_config->subscribe(this);
|
|
}
|
|
|
|
SrsRtcPlayer::~SrsRtcPlayer()
|
|
{
|
|
_srs_config->unsubscribe(this);
|
|
|
|
srs_freep(trd);
|
|
srs_freep(audio_queue_);
|
|
srs_freep(video_queue_);
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::initialize(const uint32_t& vssrc, const uint32_t& assrc, const uint16_t& v_pt, const uint16_t& a_pt)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
video_ssrc = vssrc;
|
|
audio_ssrc = assrc;
|
|
|
|
video_payload_type = v_pt;
|
|
audio_payload_type = a_pt;
|
|
|
|
// TODO: FIXME: Support reload.
|
|
nack_enabled_ = _srs_config->get_rtc_nack_enabled(session_->req->vhost);
|
|
keep_sequence_ = _srs_config->get_rtc_keep_sequence(session_->req->vhost);
|
|
srs_trace("RTC player video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), nack=%d, keep-seq=%d",
|
|
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, nack_enabled_, keep_sequence_);
|
|
|
|
if (_srs_rtc_hijacker) {
|
|
if ((err = _srs_rtc_hijacker->on_start_play(session_, this, session_->req)) != srs_success) {
|
|
return srs_error_wrap(err, "on start play");
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_reload_vhost_play(string vhost)
|
|
{
|
|
SrsRequest* req = session_->req;
|
|
|
|
if (req->vhost != vhost) {
|
|
return srs_success;
|
|
}
|
|
|
|
realtime = _srs_config->get_realtime_enabled(req->vhost, true);
|
|
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime, true);
|
|
|
|
srs_trace("Reload play realtime=%d, mw_msgs=%d", realtime, mw_msgs);
|
|
|
|
return srs_success;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_reload_vhost_realtime(string vhost)
|
|
{
|
|
return on_reload_vhost_play(vhost);
|
|
}
|
|
|
|
int SrsRtcPlayer::cid()
|
|
{
|
|
return trd->cid();
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::start()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
srs_freep(trd);
|
|
trd = new SrsSTCoroutine("rtc_sender", this, _parent_cid);
|
|
|
|
if ((err = trd->start()) != srs_success) {
|
|
return srs_error_wrap(err, "rtc_sender");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtcPlayer::stop()
|
|
{
|
|
trd->stop();
|
|
}
|
|
|
|
void SrsRtcPlayer::stop_loop()
|
|
{
|
|
trd->interrupt();
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::cycle()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRtcSource* source = NULL;
|
|
SrsRequest* req = session_->req;
|
|
|
|
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
|
|
return srs_error_wrap(err, "rtc fetch source failed");
|
|
}
|
|
|
|
SrsRtcConsumer* consumer = NULL;
|
|
SrsAutoFree(SrsRtcConsumer, consumer);
|
|
if ((err = source->create_consumer(consumer)) != srs_success) {
|
|
return srs_error_wrap(err, "rtc create consumer, source url=%s", req->get_stream_url().c_str());
|
|
}
|
|
|
|
// TODO: FIXME: Dumps the SPS/PPS from gop cache, without other frames.
|
|
if ((err = source->consumer_dumps(consumer)) != srs_success) {
|
|
return srs_error_wrap(err, "dumps consumer, source url=%s", req->get_stream_url().c_str());
|
|
}
|
|
|
|
realtime = _srs_config->get_realtime_enabled(req->vhost, true);
|
|
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime, true);
|
|
|
|
srs_trace("RTC source url=%s, source_id=[%d][%d], encrypt=%d, realtime=%d, mw_msgs=%d", req->get_stream_url().c_str(),
|
|
::getpid(), source->source_id(), session_->encrypt, realtime, mw_msgs);
|
|
|
|
SrsPithyPrint* pprint = SrsPithyPrint::create_rtc_play();
|
|
SrsAutoFree(SrsPithyPrint, pprint);
|
|
|
|
srs_trace("rtc session=%s, start play", session_->id().c_str());
|
|
bool stat_enabled = _srs_config->get_rtc_server_perf_stat();
|
|
SrsStatistic* stat = SrsStatistic::instance();
|
|
|
|
// TODO: FIXME: Use cache for performance?
|
|
vector<SrsRtpPacket2*> pkts;
|
|
SrsRtcOutgoingInfo info;
|
|
|
|
if (_srs_rtc_hijacker) {
|
|
if ((err = _srs_rtc_hijacker->on_start_consume(session_, this, session_->req, consumer)) != srs_success) {
|
|
return srs_error_wrap(err, "on start consuming");
|
|
}
|
|
}
|
|
|
|
while (true) {
|
|
if ((err = trd->pull()) != srs_success) {
|
|
return srs_error_wrap(err, "rtc sender thread");
|
|
}
|
|
|
|
// Wait for amount of packets.
|
|
consumer->wait(mw_msgs);
|
|
|
|
// TODO: FIXME: Handle error.
|
|
consumer->dump_packets(pkts);
|
|
|
|
int msg_count = (int)pkts.size();
|
|
if (!msg_count) {
|
|
continue;
|
|
}
|
|
|
|
// Send-out all RTP packets and do cleanup.
|
|
// TODO: FIXME: Handle error.
|
|
send_packets(source, pkts, info);
|
|
|
|
for (int i = 0; i < msg_count; i++) {
|
|
SrsRtpPacket2* pkt = pkts[i];
|
|
srs_freep(pkt);
|
|
}
|
|
pkts.clear();
|
|
|
|
// Stat for performance analysis.
|
|
if (!stat_enabled) {
|
|
continue;
|
|
}
|
|
|
|
// Stat the original RAW AV frame, maybe h264+aac.
|
|
stat->perf_on_msgs(msg_count);
|
|
// Stat the RTC packets, RAW AV frame, maybe h.264+opus.
|
|
int nn_rtc_packets = srs_max(info.nn_audios, info.nn_extras) + info.nn_videos;
|
|
stat->perf_on_rtc_packets(nn_rtc_packets);
|
|
// Stat the RAW RTP packets, which maybe group by GSO.
|
|
stat->perf_on_rtp_packets(msg_count);
|
|
// Stat the bytes and paddings.
|
|
stat->perf_on_rtc_bytes(info.nn_bytes, info.nn_rtp_bytes, info.nn_padding_bytes);
|
|
|
|
pprint->elapse();
|
|
if (pprint->can_print()) {
|
|
// TODO: FIXME: Print stat like frame/s, packet/s, loss_packets.
|
|
srs_trace("-> RTC PLAY %d msgs, %d/%d packets, %d audios, %d extras, %d videos, %d samples, %d/%d/%d bytes, %d pad, %d/%d cache",
|
|
msg_count, msg_count, info.nn_rtp_pkts, info.nn_audios, info.nn_extras, info.nn_videos, info.nn_samples, info.nn_bytes,
|
|
info.nn_rtp_bytes, info.nn_padding_bytes, info.nn_paddings, msg_count, msg_count);
|
|
}
|
|
}
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::send_packets(SrsRtcSource* source, const vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingInfo& info)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// If DTLS is not OK, drop all messages.
|
|
if (!session_->dtls_) {
|
|
return err;
|
|
}
|
|
|
|
// Covert kernel messages to RTP packets.
|
|
for (int i = 0; i < (int)pkts.size(); i++) {
|
|
SrsRtpPacket2* pkt = pkts[i];
|
|
|
|
// Update stats.
|
|
info.nn_bytes += pkt->nb_bytes();
|
|
|
|
// For audio, we transcoded AAC to opus in extra payloads.
|
|
if (pkt->is_audio()) {
|
|
info.nn_audios++;
|
|
|
|
if (!keep_sequence_) {
|
|
pkt->header.set_sequence(audio_sequence++);
|
|
}
|
|
pkt->header.set_ssrc(audio_ssrc);
|
|
pkt->header.set_payload_type(audio_payload_type);
|
|
|
|
// TODO: FIXME: Padding audio to the max payload in RTP packets.
|
|
|
|
continue;
|
|
}
|
|
|
|
// For video, we should process all NALUs in samples.
|
|
info.nn_videos++;
|
|
|
|
// For video, we should set the RTP packet informations about this consumer.
|
|
if (!keep_sequence_) {
|
|
pkt->header.set_sequence(video_sequence++);
|
|
}
|
|
pkt->header.set_ssrc(video_ssrc);
|
|
pkt->header.set_payload_type(video_payload_type);
|
|
}
|
|
|
|
// By default, we send packets by sendmmsg.
|
|
if ((err = do_send_packets(pkts, info)) != srs_success) {
|
|
return srs_error_wrap(err, "raw send");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::do_send_packets(const std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingInfo& info)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// Cache the encrypt flag and sender.
|
|
bool encrypt = session_->encrypt;
|
|
|
|
for (int i = 0; i < (int)pkts.size(); i++) {
|
|
SrsRtpPacket2* pkt = pkts.at(i);
|
|
|
|
// For this message, select the first iovec.
|
|
iovec* iov = new iovec();
|
|
SrsAutoFree(iovec, iov);
|
|
|
|
char* iov_base = new char[kRtpPacketSize];
|
|
SrsAutoFreeA(char, iov_base);
|
|
|
|
iov->iov_base = iov_base;
|
|
iov->iov_len = kRtpPacketSize;
|
|
|
|
// Marshal packet to bytes in iovec.
|
|
if (true) {
|
|
SrsBuffer stream((char*)iov->iov_base, iov->iov_len);
|
|
if ((err = pkt->encode(&stream)) != srs_success) {
|
|
return srs_error_wrap(err, "encode packet");
|
|
}
|
|
iov->iov_len = stream.pos();
|
|
}
|
|
|
|
// Whether encrypt the RTP bytes.
|
|
if (encrypt) {
|
|
int nn_encrypt = (int)iov->iov_len;
|
|
if ((err = session_->dtls_->protect_rtp2(iov->iov_base, &nn_encrypt)) != srs_success) {
|
|
return srs_error_wrap(err, "srtp protect");
|
|
}
|
|
iov->iov_len = (size_t)nn_encrypt;
|
|
}
|
|
|
|
// Put final RTP packet to NACK/ARQ queue.
|
|
if (nack_enabled_) {
|
|
SrsRtpPacket2* nack = new SrsRtpPacket2();
|
|
nack->header = pkt->header;
|
|
|
|
// TODO: FIXME: Should avoid memory copying.
|
|
SrsRtpRawPayload* payload = new SrsRtpRawPayload();
|
|
nack->payload = payload;
|
|
|
|
payload->nn_payload = (int)iov->iov_len;
|
|
payload->payload = new char[payload->nn_payload];
|
|
memcpy((void*)payload->payload, iov->iov_base, iov->iov_len);
|
|
|
|
nack->shared_msg = new SrsSharedPtrMessage();
|
|
nack->shared_msg->wrap(payload->payload, payload->nn_payload);
|
|
|
|
if (nack->header.get_ssrc() == video_ssrc) {
|
|
video_queue_->set(nack->header.get_sequence(), nack);
|
|
} else {
|
|
audio_queue_->set(nack->header.get_sequence(), nack);
|
|
}
|
|
}
|
|
|
|
info.nn_rtp_bytes += (int)iov->iov_len;
|
|
|
|
// When we send out a packet, increase the stat counter.
|
|
info.nn_rtp_pkts++;
|
|
|
|
// For NACK simulator, drop packet.
|
|
if (nn_simulate_nack_drop) {
|
|
simulate_drop_packet(&pkt->header, (int)iov->iov_len);
|
|
iov->iov_len = 0;
|
|
continue;
|
|
}
|
|
|
|
// TODO: FIXME: Handle error.
|
|
session_->sendonly_skt->sendto(iov->iov_base, iov->iov_len, 0);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtcPlayer::nack_fetch(vector<SrsRtpPacket2*>& pkts, uint32_t ssrc, uint16_t seq)
|
|
{
|
|
SrsRtpPacket2* pkt = NULL;
|
|
|
|
if (ssrc == video_ssrc) {
|
|
pkt = video_queue_->at(seq);
|
|
} else if (ssrc == audio_ssrc) {
|
|
pkt = audio_queue_->at(seq);
|
|
}
|
|
|
|
if (pkt) {
|
|
pkts.push_back(pkt);
|
|
}
|
|
}
|
|
|
|
void SrsRtcPlayer::simulate_nack_drop(int nn)
|
|
{
|
|
nn_simulate_nack_drop = nn;
|
|
}
|
|
|
|
void SrsRtcPlayer::simulate_drop_packet(SrsRtpHeader* h, int nn_bytes)
|
|
{
|
|
srs_warn("RTC NACK simulator #%d drop seq=%u, ssrc=%u/%s, ts=%u, %d bytes", nn_simulate_nack_drop,
|
|
h->get_sequence(), h->get_ssrc(), (h->get_ssrc()==video_ssrc? "Video":"Audio"), h->get_timestamp(),
|
|
nn_bytes);
|
|
|
|
nn_simulate_nack_drop--;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_rtcp(char* data, int nb_data)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
char* ph = data;
|
|
int nb_left = nb_data;
|
|
while (nb_left) {
|
|
uint8_t payload_type = ph[1];
|
|
uint16_t length_4bytes = (((uint16_t)ph[2]) << 8) | ph[3];
|
|
|
|
int length = (length_4bytes + 1) * 4;
|
|
|
|
if (length > nb_data) {
|
|
return srs_error_new(ERROR_RTC_RTCP, "invalid rtcp packet, length=%u", length);
|
|
}
|
|
|
|
srs_verbose("on rtcp, payload_type=%u", payload_type);
|
|
|
|
switch (payload_type) {
|
|
case kSR: {
|
|
err = on_rtcp_sr(ph, length);
|
|
break;
|
|
}
|
|
case kRR: {
|
|
err = on_rtcp_rr(ph, length);
|
|
break;
|
|
}
|
|
case kSDES: {
|
|
break;
|
|
}
|
|
case kBye: {
|
|
break;
|
|
}
|
|
case kApp: {
|
|
break;
|
|
}
|
|
case kRtpFb: {
|
|
err = on_rtcp_feedback(ph, length);
|
|
break;
|
|
}
|
|
case kPsFb: {
|
|
err = on_rtcp_ps_feedback(ph, length);
|
|
break;
|
|
}
|
|
case kXR: {
|
|
err = on_rtcp_xr(ph, length);
|
|
break;
|
|
}
|
|
default:{
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "unknown rtcp type=%u", payload_type);
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (err != srs_success) {
|
|
return srs_error_wrap(err, "rtcp");
|
|
}
|
|
|
|
ph += length;
|
|
nb_left -= length;
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_rtcp_sr(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
// TODO: FIXME: Implements it.
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_rtcp_xr(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
// TODO: FIXME: Implements it.
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_rtcp_feedback(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (nb_buf < 12) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtp feedback packet, nb_buf=%d", nb_buf);
|
|
}
|
|
|
|
SrsBuffer* stream = new SrsBuffer(buf, nb_buf);
|
|
SrsAutoFree(SrsBuffer, stream);
|
|
|
|
// @see: https://tools.ietf.org/html/rfc4585#section-6.1
|
|
/*
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
|V=2|P| FMT | PT | length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| SSRC of packet sender |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| SSRC of media source |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
: Feedback Control Information (FCI) :
|
|
: :
|
|
*/
|
|
/*uint8_t first = */stream->read_1bytes();
|
|
//uint8_t version = first & 0xC0;
|
|
//uint8_t padding = first & 0x20;
|
|
//uint8_t fmt = first & 0x1F;
|
|
|
|
/*uint8_t payload_type = */stream->read_1bytes();
|
|
/*uint16_t length = */stream->read_2bytes();
|
|
/*uint32_t ssrc_of_sender = */stream->read_4bytes();
|
|
uint32_t ssrc_of_media_source = stream->read_4bytes();
|
|
|
|
/*
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| PID | BLP |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
*/
|
|
|
|
uint16_t pid = stream->read_2bytes();
|
|
int blp = stream->read_2bytes();
|
|
|
|
// TODO: FIXME: Support ARQ.
|
|
vector<SrsRtpPacket2*> resend_pkts;
|
|
nack_fetch(resend_pkts, ssrc_of_media_source, pid);
|
|
|
|
// If NACK disabled, print a log.
|
|
if (!nack_enabled_) {
|
|
srs_trace("RTC NACK seq=%u, ignored", pid);
|
|
return err;
|
|
}
|
|
|
|
uint16_t mask = 0x01;
|
|
for (int i = 1; i < 16 && blp; ++i, mask <<= 1) {
|
|
if (!(blp & mask)) {
|
|
continue;
|
|
}
|
|
|
|
uint32_t loss_seq = pid + i;
|
|
nack_fetch(resend_pkts, ssrc_of_media_source, loss_seq);
|
|
}
|
|
|
|
for (int i = 0; i < (int)resend_pkts.size(); ++i) {
|
|
SrsRtpPacket2* pkt = resend_pkts[i];
|
|
|
|
char* data = new char[pkt->nb_bytes()];
|
|
SrsAutoFreeA(char, data);
|
|
|
|
SrsBuffer buf(data, pkt->nb_bytes());
|
|
|
|
// TODO: FIXME: Check error.
|
|
pkt->encode(&buf);
|
|
session_->sendonly_skt->sendto(data, pkt->nb_bytes(), 0);
|
|
|
|
SrsRtpHeader* h = &pkt->header;
|
|
srs_trace("RTC NACK ARQ seq=%u, ssrc=%u, ts=%u, %d bytes", h->get_sequence(),
|
|
h->get_ssrc(), h->get_timestamp(), pkt->nb_bytes());
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_rtcp_ps_feedback(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (nb_buf < 12) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtp feedback packet, nb_buf=%d", nb_buf);
|
|
}
|
|
|
|
SrsBuffer* stream = new SrsBuffer(buf, nb_buf);
|
|
SrsAutoFree(SrsBuffer, stream);
|
|
|
|
uint8_t first = stream->read_1bytes();
|
|
//uint8_t version = first & 0xC0;
|
|
//uint8_t padding = first & 0x20;
|
|
uint8_t fmt = first & 0x1F;
|
|
|
|
/*uint8_t payload_type = */stream->read_1bytes();
|
|
/*uint16_t length = */stream->read_2bytes();
|
|
/*uint32_t ssrc_of_sender = */stream->read_4bytes();
|
|
/*uint32_t ssrc_of_media_source = */stream->read_4bytes();
|
|
|
|
switch (fmt) {
|
|
case kPLI: {
|
|
ISrsRtcPublisher* publisher = session_->source_->rtc_publisher();
|
|
if (publisher) {
|
|
publisher->request_keyframe();
|
|
srs_trace("RTC request PLI");
|
|
}
|
|
break;
|
|
}
|
|
case kSLI: {
|
|
srs_verbose("sli");
|
|
break;
|
|
}
|
|
case kRPSI: {
|
|
srs_verbose("rpsi");
|
|
break;
|
|
}
|
|
case kAFB: {
|
|
srs_verbose("afb");
|
|
break;
|
|
}
|
|
default: {
|
|
return srs_error_new(ERROR_RTC_RTCP, "unknown payload specific feedback=%u", fmt);
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPlayer::on_rtcp_rr(char* data, int nb_data)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
// TODO: FIXME: Implements it.
|
|
return err;
|
|
}
|
|
|
|
SrsRtcPublisher::SrsRtcPublisher(SrsRtcSession* session)
|
|
{
|
|
report_timer = new SrsHourGlass(this, 200 * SRS_UTIME_MILLISECONDS);
|
|
|
|
session_ = session;
|
|
request_keyframe_ = false;
|
|
video_queue_ = new SrsRtpRingBuffer(1000);
|
|
video_nack_ = new SrsRtpNackForReceiver(video_queue_, 1000 * 2 / 3);
|
|
audio_queue_ = new SrsRtpRingBuffer(100);
|
|
audio_nack_ = new SrsRtpNackForReceiver(audio_queue_, 100 * 2 / 3);
|
|
|
|
source = NULL;
|
|
nn_simulate_nack_drop = 0;
|
|
nack_enabled_ = false;
|
|
|
|
nn_audio_frames = 0;
|
|
twcc_ext_id_ = 0;
|
|
last_twcc_feedback_time_ = 0;
|
|
twcc_fb_count_ = 0;
|
|
}
|
|
|
|
SrsRtcPublisher::~SrsRtcPublisher()
|
|
{
|
|
// TODO: FIXME: Do unpublish when session timeout.
|
|
if (source) {
|
|
source->set_rtc_publisher(NULL);
|
|
source->on_unpublish();
|
|
}
|
|
|
|
srs_freep(report_timer);
|
|
srs_freep(video_nack_);
|
|
srs_freep(video_queue_);
|
|
srs_freep(audio_nack_);
|
|
srs_freep(audio_queue_);
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::initialize(uint32_t vssrc, uint32_t assrc, uint8_t twcc_ext_id, SrsRequest* r)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
video_ssrc = vssrc;
|
|
audio_ssrc = assrc;
|
|
twcc_ext_id_ = twcc_ext_id;
|
|
rtcp_twcc_.set_media_ssrc(video_ssrc);
|
|
req = r;
|
|
|
|
if (twcc_ext_id_ != 0) {
|
|
extension_map_.register_by_uri(twcc_ext_id_, kTWCCExt);
|
|
}
|
|
// TODO: FIXME: Support reload.
|
|
nack_enabled_ = _srs_config->get_rtc_nack_enabled(session_->req->vhost);
|
|
|
|
srs_trace("RTC publisher video(ssrc=%u), audio(ssrc=%u), nack=%d",
|
|
video_ssrc, audio_ssrc, nack_enabled_);
|
|
|
|
if ((err = report_timer->tick(0 * SRS_UTIME_MILLISECONDS)) != srs_success) {
|
|
return srs_error_wrap(err, "hourglass tick");
|
|
}
|
|
|
|
if ((err = report_timer->start()) != srs_success) {
|
|
return srs_error_wrap(err, "start report_timer");
|
|
}
|
|
|
|
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
|
|
return srs_error_wrap(err, "create source");
|
|
}
|
|
|
|
if ((err = source->on_publish()) != srs_success) {
|
|
return srs_error_wrap(err, "on publish");
|
|
}
|
|
|
|
source->set_rtc_publisher(this);
|
|
|
|
if (_srs_rtc_hijacker) {
|
|
if ((err = _srs_rtc_hijacker->on_start_publish(session_, this, req)) != srs_success) {
|
|
return srs_error_wrap(err, "on start publish");
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtcPublisher::check_send_nacks(SrsRtpNackForReceiver* nack, uint32_t ssrc)
|
|
{
|
|
// If DTLS is not OK, drop all messages.
|
|
if (!session_->dtls_) {
|
|
return;
|
|
}
|
|
|
|
// @see: https://tools.ietf.org/html/rfc4585#section-6.1
|
|
vector<uint16_t> nack_seqs;
|
|
nack->get_nack_seqs(nack_seqs);
|
|
|
|
vector<uint16_t>::iterator iter = nack_seqs.begin();
|
|
while (iter != nack_seqs.end()) {
|
|
char buf[kRtpPacketSize];
|
|
SrsBuffer stream(buf, sizeof(buf));
|
|
// FIXME: Replace magic number.
|
|
stream.write_1bytes(0x81);
|
|
stream.write_1bytes(kRtpFb);
|
|
stream.write_2bytes(3);
|
|
stream.write_4bytes(ssrc); // TODO: FIXME: Should be 1?
|
|
stream.write_4bytes(ssrc); // TODO: FIXME: Should be 0?
|
|
uint16_t pid = *iter;
|
|
uint16_t blp = 0;
|
|
while (iter + 1 != nack_seqs.end() && (*(iter + 1) - pid <= 15)) {
|
|
blp |= (1 << (*(iter + 1) - pid - 1));
|
|
++iter;
|
|
}
|
|
|
|
stream.write_2bytes(pid);
|
|
stream.write_2bytes(blp);
|
|
|
|
if (session_->blackhole && session_->blackhole_addr && session_->blackhole_stfd) {
|
|
// Ignore any error for black-hole.
|
|
void* p = stream.data(); int len = stream.pos(); SrsRtcSession* s = session_;
|
|
srs_sendto(s->blackhole_stfd, p, len, (sockaddr*)s->blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
|
|
}
|
|
|
|
char protected_buf[kRtpPacketSize];
|
|
int nb_protected_buf = stream.pos();
|
|
|
|
// FIXME: Merge nack rtcp into one packets.
|
|
if (session_->dtls_->protect_rtcp(protected_buf, stream.data(), nb_protected_buf) == srs_success) {
|
|
// TODO: FIXME: Check error.
|
|
session_->sendonly_skt->sendto(protected_buf, nb_protected_buf, 0);
|
|
}
|
|
|
|
++iter;
|
|
}
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::send_rtcp_rr(uint32_t ssrc, SrsRtpRingBuffer* rtp_queue)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// If DTLS is not OK, drop all messages.
|
|
if (!session_->dtls_) {
|
|
return err;
|
|
}
|
|
|
|
// @see https://tools.ietf.org/html/rfc3550#section-6.4.2
|
|
char buf[kRtpPacketSize];
|
|
SrsBuffer stream(buf, sizeof(buf));
|
|
stream.write_1bytes(0x81);
|
|
stream.write_1bytes(kRR);
|
|
stream.write_2bytes(7);
|
|
stream.write_4bytes(ssrc); // TODO: FIXME: Should be 1?
|
|
|
|
// TODO: FIXME: Implements it.
|
|
// TODO: FIXME: See https://github.com/ossrs/srs/blob/f81d35d20f04ebec01915cb78a882e45b7ee8800/trunk/src/app/srs_app_rtc_queue.cpp
|
|
uint8_t fraction_lost = 0;
|
|
uint32_t cumulative_number_of_packets_lost = 0 & 0x7FFFFF;
|
|
uint32_t extended_highest_sequence = rtp_queue->get_extended_highest_sequence();
|
|
uint32_t interarrival_jitter = 0;
|
|
|
|
uint32_t rr_lsr = 0;
|
|
uint32_t rr_dlsr = 0;
|
|
|
|
const uint64_t& lsr_systime = last_sender_report_sys_time[ssrc];
|
|
const SrsNtp& lsr_ntp = last_sender_report_ntp[ssrc];
|
|
|
|
if (lsr_systime > 0) {
|
|
rr_lsr = (lsr_ntp.ntp_second_ << 16) | (lsr_ntp.ntp_fractions_ >> 16);
|
|
uint32_t dlsr = (srs_update_system_time() - lsr_systime) / 1000;
|
|
rr_dlsr = ((dlsr / 1000) << 16) | ((dlsr % 1000) * 65536 / 1000);
|
|
}
|
|
|
|
stream.write_4bytes(ssrc);
|
|
stream.write_1bytes(fraction_lost);
|
|
stream.write_3bytes(cumulative_number_of_packets_lost);
|
|
stream.write_4bytes(extended_highest_sequence);
|
|
stream.write_4bytes(interarrival_jitter);
|
|
stream.write_4bytes(rr_lsr);
|
|
stream.write_4bytes(rr_dlsr);
|
|
|
|
srs_verbose("RR ssrc=%u, fraction_lost=%u, cumulative_number_of_packets_lost=%u, extended_highest_sequence=%u, interarrival_jitter=%u",
|
|
ssrc, fraction_lost, cumulative_number_of_packets_lost, extended_highest_sequence, interarrival_jitter);
|
|
|
|
char protected_buf[kRtpPacketSize];
|
|
int nb_protected_buf = stream.pos();
|
|
if ((err = session_->dtls_->protect_rtcp(protected_buf, stream.data(), nb_protected_buf)) != srs_success) {
|
|
return srs_error_wrap(err, "protect rtcp rr");
|
|
}
|
|
|
|
// TDOO: FIXME: Check error.
|
|
session_->sendonly_skt->sendto(protected_buf, nb_protected_buf, 0);
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::send_rtcp_xr_rrtr(uint32_t ssrc)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// If DTLS is not OK, drop all messages.
|
|
if (!session_->dtls_) {
|
|
return err;
|
|
}
|
|
|
|
/*
|
|
@see: http://www.rfc-editor.org/rfc/rfc3611.html#section-2
|
|
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
|V=2|P|reserved | PT=XR=207 | length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| SSRC |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
: report blocks :
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
|
|
@see: http://www.rfc-editor.org/rfc/rfc3611.html#section-4.4
|
|
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| BT=4 | reserved | block length = 2 |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| NTP timestamp, most significant word |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| NTP timestamp, least significant word |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
*/
|
|
srs_utime_t now = srs_update_system_time();
|
|
SrsNtp cur_ntp = SrsNtp::from_time_ms(now / 1000);
|
|
|
|
char buf[kRtpPacketSize];
|
|
SrsBuffer stream(buf, sizeof(buf));
|
|
stream.write_1bytes(0x80);
|
|
stream.write_1bytes(kXR);
|
|
stream.write_2bytes(4);
|
|
stream.write_4bytes(ssrc);
|
|
stream.write_1bytes(4);
|
|
stream.write_1bytes(0);
|
|
stream.write_2bytes(2);
|
|
stream.write_4bytes(cur_ntp.ntp_second_);
|
|
stream.write_4bytes(cur_ntp.ntp_fractions_);
|
|
|
|
char protected_buf[kRtpPacketSize];
|
|
int nb_protected_buf = stream.pos();
|
|
if ((err = session_->dtls_->protect_rtcp(protected_buf, stream.data(), nb_protected_buf)) != srs_success) {
|
|
return srs_error_wrap(err, "protect rtcp xr");
|
|
}
|
|
|
|
// TDOO: FIXME: Check error.
|
|
session_->sendonly_skt->sendto(protected_buf, nb_protected_buf, 0);
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::send_rtcp_fb_pli(uint32_t ssrc)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// If DTLS is not OK, drop all messages.
|
|
if (!session_->dtls_) {
|
|
return err;
|
|
}
|
|
|
|
char buf[kRtpPacketSize];
|
|
SrsBuffer stream(buf, sizeof(buf));
|
|
stream.write_1bytes(0x81);
|
|
stream.write_1bytes(kPsFb);
|
|
stream.write_2bytes(2);
|
|
stream.write_4bytes(ssrc);
|
|
stream.write_4bytes(ssrc);
|
|
|
|
srs_trace("RTC PLI ssrc=%u", ssrc);
|
|
|
|
if (session_->blackhole && session_->blackhole_addr && session_->blackhole_stfd) {
|
|
// Ignore any error for black-hole.
|
|
void* p = stream.data(); int len = stream.pos(); SrsRtcSession* s = session_;
|
|
srs_sendto(s->blackhole_stfd, p, len, (sockaddr*)s->blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
|
|
}
|
|
|
|
char protected_buf[kRtpPacketSize];
|
|
int nb_protected_buf = stream.pos();
|
|
if ((err = session_->dtls_->protect_rtcp(protected_buf, stream.data(), nb_protected_buf)) != srs_success) {
|
|
return srs_error_wrap(err, "protect rtcp psfb pli");
|
|
}
|
|
|
|
// TDOO: FIXME: Check error.
|
|
session_->sendonly_skt->sendto(protected_buf, nb_protected_buf, 0);
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_twcc(uint16_t sn) {
|
|
srs_error_t err = srs_success;
|
|
srs_utime_t now = srs_get_system_time();
|
|
rtcp_twcc_.recv_packet(sn, now);
|
|
if(0 == last_twcc_feedback_time_) {
|
|
last_twcc_feedback_time_ = now;
|
|
return err;
|
|
}
|
|
srs_utime_t diff = now - last_twcc_feedback_time_;
|
|
if( diff >= 50 * SRS_UTIME_MILLISECONDS) {
|
|
last_twcc_feedback_time_ = now;
|
|
char pkt[kRtcpPacketSize];
|
|
SrsBuffer *buffer = new SrsBuffer(pkt, sizeof(pkt));
|
|
SrsAutoFree(SrsBuffer, buffer);
|
|
rtcp_twcc_.set_feedback_count(twcc_fb_count_);
|
|
twcc_fb_count_++;
|
|
if((err = rtcp_twcc_.encode(buffer)) != srs_success) {
|
|
return srs_error_wrap(err, "fail to generate twcc feedback packet");
|
|
}
|
|
int nb_protected_buf = buffer->pos();
|
|
char protected_buf[kRtpPacketSize];
|
|
if (session_->dtls_->protect_rtcp(protected_buf, pkt, nb_protected_buf) == srs_success) {
|
|
session_->sendonly_skt->sendto(protected_buf, nb_protected_buf, 0);
|
|
}
|
|
}
|
|
return err;
|
|
}
|
|
srs_error_t SrsRtcPublisher::on_rtp(char* data, int nb_data)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// For NACK simulator, drop packet.
|
|
if (nn_simulate_nack_drop) {
|
|
SrsBuffer b0(data, nb_data); SrsRtpHeader h0; h0.decode(&b0);
|
|
simulate_drop_packet(&h0, nb_data);
|
|
return err;
|
|
}
|
|
|
|
// Decrypt the cipher to plaintext RTP data.
|
|
int nb_unprotected_buf = nb_data;
|
|
char* unprotected_buf = new char[kRtpPacketSize];
|
|
if ((err = session_->dtls_->unprotect_rtp(unprotected_buf, data, nb_unprotected_buf)) != srs_success) {
|
|
// We try to decode the RTP header for more detail error informations.
|
|
SrsBuffer b0(data, nb_data); SrsRtpHeader h0; h0.decode(&b0);
|
|
err = srs_error_wrap(err, "marker=%u, pt=%u, seq=%u, ts=%u, ssrc=%u, pad=%u, payload=%uB", h0.get_marker(), h0.get_payload_type(),
|
|
h0.get_sequence(), h0.get_timestamp(), h0.get_ssrc(), h0.get_padding(), nb_data - b0.pos());
|
|
|
|
srs_freepa(unprotected_buf);
|
|
return err;
|
|
}
|
|
|
|
if (session_->blackhole && session_->blackhole_addr && session_->blackhole_stfd) {
|
|
// Ignore any error for black-hole.
|
|
void* p = unprotected_buf; int len = nb_unprotected_buf; SrsRtcSession* s = session_;
|
|
srs_sendto(s->blackhole_stfd, p, len, (sockaddr*)s->blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
|
|
}
|
|
|
|
char* buf = unprotected_buf;
|
|
int nb_buf = nb_unprotected_buf;
|
|
|
|
// Decode the RTP packet from buffer.
|
|
SrsRtpPacket2* pkt = new SrsRtpPacket2();
|
|
SrsAutoFree(SrsRtpPacket2, pkt);
|
|
|
|
if (true) {
|
|
pkt->set_decode_handler(this);
|
|
pkt->set_rtp_header_extensions(&extension_map_);
|
|
pkt->shared_msg = new SrsSharedPtrMessage();
|
|
pkt->shared_msg->wrap(buf, nb_buf);
|
|
|
|
SrsBuffer b(buf, nb_buf);
|
|
if ((err = pkt->decode(&b)) != srs_success) {
|
|
return srs_error_wrap(err, "decode rtp packet");
|
|
}
|
|
|
|
if (0 != twcc_ext_id_) {
|
|
uint16_t twcc_sn = 0;
|
|
if ((err = pkt->header.get_twcc_sequence_number(twcc_sn)) == srs_success) {
|
|
if((err = on_twcc(twcc_sn))) {
|
|
return srs_error_wrap(err, "fail to process twcc packet");
|
|
}
|
|
} else {
|
|
// TODO: FIXME: process no twcc seq number for audio ssrc
|
|
srs_error_reset(err);
|
|
}
|
|
}
|
|
}
|
|
|
|
// For source to consume packet.
|
|
uint32_t ssrc = pkt->header.get_ssrc();
|
|
if (ssrc == audio_ssrc) {
|
|
pkt->frame_type = SrsFrameTypeAudio;
|
|
if ((err = on_audio(pkt)) != srs_success) {
|
|
return srs_error_wrap(err, "on audio");
|
|
}
|
|
} else if (ssrc == video_ssrc) {
|
|
pkt->frame_type = SrsFrameTypeVideo;
|
|
if ((err = on_video(pkt)) != srs_success) {
|
|
return srs_error_wrap(err, "on video");
|
|
}
|
|
} else {
|
|
return srs_error_new(ERROR_RTC_RTP, "unknown ssrc=%u", ssrc);
|
|
}
|
|
|
|
// For NACK to handle packet.
|
|
if (nack_enabled_ && (err = on_nack(pkt)) != srs_success) {
|
|
return srs_error_wrap(err, "on nack");
|
|
}
|
|
|
|
if (_srs_rtc_hijacker) {
|
|
if ((err = _srs_rtc_hijacker->on_rtp_packet(session_, this, req, pkt->copy())) != srs_success) {
|
|
return srs_error_wrap(err, "on rtp packet");
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtcPublisher::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload)
|
|
{
|
|
// No payload, ignore.
|
|
if (buf->empty()) {
|
|
return;
|
|
}
|
|
|
|
uint32_t ssrc = pkt->header.get_ssrc();
|
|
if (ssrc == audio_ssrc) {
|
|
*ppayload = new SrsRtpRawPayload();
|
|
} else if (ssrc == video_ssrc) {
|
|
uint8_t v = (uint8_t)pkt->nalu_type;
|
|
if (v == kStapA) {
|
|
*ppayload = new SrsRtpSTAPPayload();
|
|
} else if (v == kFuA) {
|
|
*ppayload = new SrsRtpFUAPayload2();
|
|
} else {
|
|
*ppayload = new SrsRtpRawPayload();
|
|
}
|
|
}
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_audio(SrsRtpPacket2* pkt)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
pkt->frame_type = SrsFrameTypeAudio;
|
|
|
|
// TODO: FIXME: Error check.
|
|
source->on_rtp(pkt);
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_video(SrsRtpPacket2* pkt)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
pkt->frame_type = SrsFrameTypeVideo;
|
|
|
|
// TODO: FIXME: Error check.
|
|
source->on_rtp(pkt);
|
|
|
|
if (request_keyframe_) {
|
|
request_keyframe_ = false;
|
|
|
|
// TODO: FIXME: Check error.
|
|
send_rtcp_fb_pli(video_ssrc);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_nack(SrsRtpPacket2* pkt)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRtpNackForReceiver* nack_receiver = audio_nack_;
|
|
SrsRtpRingBuffer* ring_queue = audio_queue_;
|
|
|
|
// TODO: FIXME: use is_audio() to jugdement
|
|
uint32_t ssrc = pkt->header.get_ssrc();
|
|
uint16_t seq = pkt->header.get_sequence();
|
|
bool video = (ssrc == video_ssrc) ? true : false;
|
|
if (video) {
|
|
nack_receiver = video_nack_;
|
|
ring_queue = video_queue_;
|
|
}
|
|
|
|
// TODO: check whether is necessary?
|
|
nack_receiver->remove_timeout_packets();
|
|
|
|
SrsRtpNackInfo* nack_info = nack_receiver->find(seq);
|
|
if (nack_info) {
|
|
// seq had been received.
|
|
nack_receiver->remove(seq);
|
|
return err;
|
|
}
|
|
|
|
// insert check nack list
|
|
uint16_t nack_first = 0, nack_last = 0;
|
|
if (!ring_queue->update(seq, nack_first, nack_last)) {
|
|
srs_warn("too old seq %u, range [%u, %u]", seq, ring_queue->begin, ring_queue->end);
|
|
}
|
|
if (srs_rtp_seq_distance(nack_first, nack_last) > 0) {
|
|
srs_trace("update seq=%u, nack range [%u, %u]", seq, nack_first, nack_last);
|
|
nack_receiver->insert(nack_first, nack_last);
|
|
nack_receiver->check_queue_size();
|
|
}
|
|
|
|
// insert into video_queue and audio_queue
|
|
ring_queue->set(seq, pkt->copy());
|
|
// send_nack
|
|
check_send_nacks(nack_receiver, ssrc);
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_rtcp(char* data, int nb_data)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
char* ph = data;
|
|
int nb_left = nb_data;
|
|
while (nb_left) {
|
|
uint8_t payload_type = ph[1];
|
|
uint16_t length_4bytes = (((uint16_t)ph[2]) << 8) | ph[3];
|
|
|
|
int length = (length_4bytes + 1) * 4;
|
|
|
|
if (length > nb_data) {
|
|
return srs_error_new(ERROR_RTC_RTCP, "invalid rtcp packet, length=%u", length);
|
|
}
|
|
|
|
srs_verbose("on rtcp, payload_type=%u", payload_type);
|
|
|
|
switch (payload_type) {
|
|
case kSR: {
|
|
err = on_rtcp_sr(ph, length);
|
|
break;
|
|
}
|
|
case kRR: {
|
|
err = on_rtcp_rr(ph, length);
|
|
break;
|
|
}
|
|
case kSDES: {
|
|
break;
|
|
}
|
|
case kBye: {
|
|
break;
|
|
}
|
|
case kApp: {
|
|
break;
|
|
}
|
|
case kRtpFb: {
|
|
err = on_rtcp_feedback(ph, length);
|
|
break;
|
|
}
|
|
case kPsFb: {
|
|
err = on_rtcp_ps_feedback(ph, length);
|
|
break;
|
|
}
|
|
case kXR: {
|
|
err = on_rtcp_xr(ph, length);
|
|
break;
|
|
}
|
|
default:{
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "unknown rtcp type=%u", payload_type);
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (err != srs_success) {
|
|
return srs_error_wrap(err, "rtcp");
|
|
}
|
|
|
|
ph += length;
|
|
nb_left -= length;
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_rtcp_sr(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (nb_buf < 28) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtp sender report packet, nb_buf=%d", nb_buf);
|
|
}
|
|
|
|
SrsBuffer* stream = new SrsBuffer(buf, nb_buf);
|
|
SrsAutoFree(SrsBuffer, stream);
|
|
|
|
// @see: https://tools.ietf.org/html/rfc3550#section-6.4.1
|
|
/*
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
header |V=2|P| RC | PT=SR=200 | length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| SSRC of sender |
|
|
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|
|
sender | NTP timestamp, most significant word |
|
|
info +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| NTP timestamp, least significant word |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| RTP timestamp |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| sender's packet count |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| sender's octet count |
|
|
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|
|
report | SSRC_1 (SSRC of first source) |
|
|
block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
1 | fraction lost | cumulative number of packets lost |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| extended highest sequence number received |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| interarrival jitter |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| last SR (LSR) |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| delay since last SR (DLSR) |
|
|
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|
|
report | SSRC_2 (SSRC of second source) |
|
|
block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
2 : ... :
|
|
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|
|
| profile-specific extensions |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
*/
|
|
uint8_t first = stream->read_1bytes();
|
|
uint8_t rc = first & 0x1F;
|
|
|
|
uint8_t payload_type = stream->read_1bytes();
|
|
srs_assert(payload_type == kSR);
|
|
uint16_t length = stream->read_2bytes();
|
|
|
|
if (((length + 1) * 4) != (rc * 24 + 28)) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtcp sender report packet, length=%u, rc=%u", length, rc);
|
|
}
|
|
|
|
uint32_t ssrc_of_sender = stream->read_4bytes();
|
|
uint64_t ntp = stream->read_8bytes();
|
|
SrsNtp srs_ntp = SrsNtp::to_time_ms(ntp);
|
|
uint32_t rtp_time = stream->read_4bytes();
|
|
uint32_t sender_packet_count = stream->read_4bytes();
|
|
uint32_t sender_octec_count = stream->read_4bytes();
|
|
|
|
(void)sender_packet_count; (void)sender_octec_count; (void)rtp_time;
|
|
srs_verbose("sender report, ssrc_of_sender=%u, rtp_time=%u, sender_packet_count=%u, sender_octec_count=%u",
|
|
ssrc_of_sender, rtp_time, sender_packet_count, sender_octec_count);
|
|
|
|
for (int i = 0; i < rc; ++i) {
|
|
uint32_t ssrc = stream->read_4bytes();
|
|
uint8_t fraction_lost = stream->read_1bytes();
|
|
uint32_t cumulative_number_of_packets_lost = stream->read_3bytes();
|
|
uint32_t highest_seq = stream->read_4bytes();
|
|
uint32_t jitter = stream->read_4bytes();
|
|
uint32_t lst = stream->read_4bytes();
|
|
uint32_t dlsr = stream->read_4bytes();
|
|
|
|
(void)ssrc; (void)fraction_lost; (void)cumulative_number_of_packets_lost; (void)highest_seq; (void)jitter; (void)lst; (void)dlsr;
|
|
srs_verbose("sender report, ssrc=%u, fraction_lost=%u, cumulative_number_of_packets_lost=%u, highest_seq=%u, jitter=%u, lst=%u, dlst=%u",
|
|
ssrc, fraction_lost, cumulative_number_of_packets_lost, highest_seq, jitter, lst, dlsr);
|
|
}
|
|
|
|
last_sender_report_ntp[ssrc_of_sender] = srs_ntp;
|
|
last_sender_report_sys_time[ssrc_of_sender] = srs_update_system_time();
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_rtcp_xr(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
/*
|
|
@see: http://www.rfc-editor.org/rfc/rfc3611.html#section-2
|
|
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
|V=2|P|reserved | PT=XR=207 | length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| SSRC |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
: report blocks :
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
*/
|
|
|
|
SrsBuffer stream(buf, nb_buf);
|
|
/*uint8_t first = */stream.read_1bytes();
|
|
uint8_t pt = stream.read_1bytes();
|
|
srs_assert(pt == kXR);
|
|
uint16_t length = (stream.read_2bytes() + 1) * 4;
|
|
/*uint32_t ssrc = */stream.read_4bytes();
|
|
|
|
if (length != nb_buf) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid XR packet, length=%u, nb_buf=%d", length, nb_buf);
|
|
}
|
|
|
|
while (stream.pos() + 4 < length) {
|
|
uint8_t bt = stream.read_1bytes();
|
|
stream.skip(1);
|
|
uint16_t block_length = (stream.read_2bytes() + 1) * 4;
|
|
|
|
if (stream.pos() + block_length - 4 > nb_buf) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid XR packet block, block_length=%u, nb_buf=%d", block_length, nb_buf);
|
|
}
|
|
|
|
if (bt == 5) {
|
|
for (int i = 4; i < block_length; i += 12) {
|
|
uint32_t ssrc = stream.read_4bytes();
|
|
uint32_t lrr = stream.read_4bytes();
|
|
uint32_t dlrr = stream.read_4bytes();
|
|
|
|
SrsNtp cur_ntp = SrsNtp::from_time_ms(srs_update_system_time() / 1000);
|
|
uint32_t compact_ntp = (cur_ntp.ntp_second_ << 16) | (cur_ntp.ntp_fractions_ >> 16);
|
|
|
|
int rtt_ntp = compact_ntp - lrr - dlrr;
|
|
int rtt = ((rtt_ntp * 1000) >> 16) + ((rtt_ntp >> 16) * 1000);
|
|
srs_verbose("ssrc=%u, compact_ntp=%u, lrr=%u, dlrr=%u, rtt=%d",
|
|
ssrc, compact_ntp, lrr, dlrr, rtt);
|
|
|
|
if (ssrc == video_ssrc) {
|
|
video_nack_->update_rtt(rtt);
|
|
} else if (ssrc == audio_ssrc) {
|
|
audio_nack_->update_rtt(rtt);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_rtcp_feedback(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
// TODO: FIXME: Implements it.
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_rtcp_ps_feedback(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (nb_buf < 12) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtp feedback packet, nb_buf=%d", nb_buf);
|
|
}
|
|
|
|
SrsBuffer* stream = new SrsBuffer(buf, nb_buf);
|
|
SrsAutoFree(SrsBuffer, stream);
|
|
|
|
uint8_t first = stream->read_1bytes();
|
|
//uint8_t version = first & 0xC0;
|
|
//uint8_t padding = first & 0x20;
|
|
uint8_t fmt = first & 0x1F;
|
|
|
|
/*uint8_t payload_type = */stream->read_1bytes();
|
|
/*uint16_t length = */stream->read_2bytes();
|
|
/*uint32_t ssrc_of_sender = */stream->read_4bytes();
|
|
/*uint32_t ssrc_of_media_source = */stream->read_4bytes();
|
|
|
|
switch (fmt) {
|
|
case kPLI: {
|
|
srs_verbose("pli");
|
|
break;
|
|
}
|
|
case kSLI: {
|
|
srs_verbose("sli");
|
|
break;
|
|
}
|
|
case kRPSI: {
|
|
srs_verbose("rpsi");
|
|
break;
|
|
}
|
|
case kAFB: {
|
|
srs_verbose("afb");
|
|
break;
|
|
}
|
|
default: {
|
|
return srs_error_new(ERROR_RTC_RTCP, "unknown payload specific feedback=%u", fmt);
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::on_rtcp_rr(char* buf, int nb_buf)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (nb_buf < 8) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtp receiver report packet, nb_buf=%d", nb_buf);
|
|
}
|
|
|
|
SrsBuffer* stream = new SrsBuffer(buf, nb_buf);
|
|
SrsAutoFree(SrsBuffer, stream);
|
|
|
|
// @see: https://tools.ietf.org/html/rfc3550#section-6.4.2
|
|
/*
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
header |V=2|P| RC | PT=RR=201 | length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| SSRC of packet sender |
|
|
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|
|
report | SSRC_1 (SSRC of first source) |
|
|
block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
1 | fraction lost | cumulative number of packets lost |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| extended highest sequence number received |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| interarrival jitter |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| last SR (LSR) |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| delay since last SR (DLSR) |
|
|
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|
|
report | SSRC_2 (SSRC of second source) |
|
|
block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
2 : ... :
|
|
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|
|
| profile-specific extensions |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
*/
|
|
uint8_t first = stream->read_1bytes();
|
|
//uint8_t version = first & 0xC0;
|
|
//uint8_t padding = first & 0x20;
|
|
uint8_t rc = first & 0x1F;
|
|
|
|
/*uint8_t payload_type = */stream->read_1bytes();
|
|
uint16_t length = stream->read_2bytes();
|
|
/*uint32_t ssrc_of_sender = */stream->read_4bytes();
|
|
|
|
if (((length + 1) * 4) != (rc * 24 + 8)) {
|
|
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtcp receiver packet, length=%u, rc=%u", length, rc);
|
|
}
|
|
|
|
for (int i = 0; i < rc; ++i) {
|
|
uint32_t ssrc = stream->read_4bytes();
|
|
uint8_t fraction_lost = stream->read_1bytes();
|
|
uint32_t cumulative_number_of_packets_lost = stream->read_3bytes();
|
|
uint32_t highest_seq = stream->read_4bytes();
|
|
uint32_t jitter = stream->read_4bytes();
|
|
uint32_t lst = stream->read_4bytes();
|
|
uint32_t dlsr = stream->read_4bytes();
|
|
|
|
(void)ssrc; (void)fraction_lost; (void)cumulative_number_of_packets_lost; (void)highest_seq; (void)jitter; (void)lst; (void)dlsr;
|
|
srs_verbose("ssrc=%u, fraction_lost=%u, cumulative_number_of_packets_lost=%u, highest_seq=%u, jitter=%u, lst=%u, dlst=%u",
|
|
ssrc, fraction_lost, cumulative_number_of_packets_lost, highest_seq, jitter, lst, dlsr);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtcPublisher::request_keyframe()
|
|
{
|
|
int scid = _srs_context->get_id();
|
|
int pcid = session_->context_id();
|
|
srs_trace("RTC play=[%d][%d] request keyframe from publish=[%d][%d]", ::getpid(), scid, ::getpid(), pcid);
|
|
|
|
request_keyframe_ = true;
|
|
}
|
|
|
|
srs_error_t SrsRtcPublisher::notify(int type, srs_utime_t interval, srs_utime_t tick)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
// TODO: FIXME: Check error.
|
|
send_rtcp_rr(video_ssrc, video_queue_);
|
|
send_rtcp_rr(audio_ssrc, audio_queue_);
|
|
send_rtcp_xr_rrtr(video_ssrc);
|
|
send_rtcp_xr_rrtr(audio_ssrc);
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtcPublisher::simulate_nack_drop(int nn)
|
|
{
|
|
nn_simulate_nack_drop = nn;
|
|
}
|
|
|
|
void SrsRtcPublisher::simulate_drop_packet(SrsRtpHeader* h, int nn_bytes)
|
|
{
|
|
srs_warn("RTC NACK simulator #%d drop seq=%u, ssrc=%u/%s, ts=%u, %d bytes", nn_simulate_nack_drop,
|
|
h->get_sequence(), h->get_ssrc(), (h->get_ssrc()==video_ssrc? "Video":"Audio"), h->get_timestamp(),
|
|
nn_bytes);
|
|
|
|
nn_simulate_nack_drop--;
|
|
}
|
|
|
|
SrsRtcSession::SrsRtcSession(SrsRtcServer* s)
|
|
{
|
|
req = NULL;
|
|
cid = 0;
|
|
is_publisher_ = false;
|
|
encrypt = true;
|
|
|
|
source_ = NULL;
|
|
publisher_ = NULL;
|
|
player_ = NULL;
|
|
sendonly_skt = NULL;
|
|
server_ = s;
|
|
dtls_ = new SrsRtcDtls(this);
|
|
|
|
state_ = INIT;
|
|
last_stun_time = 0;
|
|
sessionStunTimeout = 0;
|
|
disposing_ = false;
|
|
|
|
blackhole = false;
|
|
blackhole_addr = NULL;
|
|
blackhole_stfd = NULL;
|
|
}
|
|
|
|
SrsRtcSession::~SrsRtcSession()
|
|
{
|
|
srs_freep(player_);
|
|
srs_freep(publisher_);
|
|
srs_freep(dtls_);
|
|
srs_freep(req);
|
|
srs_close_stfd(blackhole_stfd);
|
|
srs_freep(blackhole_addr);
|
|
srs_freep(sendonly_skt);
|
|
}
|
|
|
|
SrsSdp* SrsRtcSession::get_local_sdp()
|
|
{
|
|
return &local_sdp;
|
|
}
|
|
|
|
void SrsRtcSession::set_local_sdp(const SrsSdp& sdp)
|
|
{
|
|
local_sdp = sdp;
|
|
}
|
|
|
|
SrsSdp* SrsRtcSession::get_remote_sdp()
|
|
{
|
|
return &remote_sdp;
|
|
}
|
|
|
|
void SrsRtcSession::set_remote_sdp(const SrsSdp& sdp)
|
|
{
|
|
remote_sdp = sdp;
|
|
}
|
|
|
|
SrsRtcSessionStateType SrsRtcSession::state()
|
|
{
|
|
return state_;
|
|
}
|
|
|
|
void SrsRtcSession::set_state(SrsRtcSessionStateType state)
|
|
{
|
|
state_ = state;
|
|
}
|
|
|
|
string SrsRtcSession::id()
|
|
{
|
|
return peer_id_ + "/" + username_;
|
|
}
|
|
|
|
|
|
string SrsRtcSession::peer_id()
|
|
{
|
|
return peer_id_;
|
|
}
|
|
|
|
void SrsRtcSession::set_peer_id(string v)
|
|
{
|
|
peer_id_ = v;
|
|
}
|
|
|
|
string SrsRtcSession::username()
|
|
{
|
|
return username_;
|
|
}
|
|
|
|
void SrsRtcSession::set_encrypt(bool v)
|
|
{
|
|
encrypt = v;
|
|
}
|
|
|
|
void SrsRtcSession::switch_to_context()
|
|
{
|
|
_srs_context->set_id(cid);
|
|
}
|
|
|
|
int SrsRtcSession::context_id()
|
|
{
|
|
return cid;
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::initialize(SrsRtcSource* source, SrsRequest* r, bool is_publisher, string username, int context_id)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
username_ = username;
|
|
req = r->copy();
|
|
cid = context_id;
|
|
is_publisher_ = is_publisher;
|
|
source_ = source;
|
|
|
|
if ((err = dtls_->initialize(req)) != srs_success) {
|
|
return srs_error_wrap(err, "init");
|
|
}
|
|
|
|
// TODO: FIXME: Support reload.
|
|
sessionStunTimeout = _srs_config->get_rtc_stun_timeout(req->vhost);
|
|
last_stun_time = srs_get_system_time();
|
|
|
|
blackhole = _srs_config->get_rtc_server_black_hole();
|
|
|
|
srs_trace("RTC init session, timeout=%dms, blackhole=%d", srsu2msi(sessionStunTimeout), blackhole);
|
|
|
|
if (blackhole) {
|
|
string blackhole_ep = _srs_config->get_rtc_server_black_hole_addr();
|
|
if (!blackhole_ep.empty()) {
|
|
string host; int port;
|
|
srs_parse_hostport(blackhole_ep, host, port);
|
|
|
|
srs_freep(blackhole_addr);
|
|
blackhole_addr = new sockaddr_in();
|
|
blackhole_addr->sin_family = AF_INET;
|
|
blackhole_addr->sin_addr.s_addr = inet_addr(host.c_str());
|
|
blackhole_addr->sin_port = htons(port);
|
|
|
|
int fd = socket(AF_INET, SOCK_DGRAM, 0);
|
|
blackhole_stfd = srs_netfd_open_socket(fd);
|
|
srs_assert(blackhole_stfd);
|
|
|
|
srs_trace("RTC blackhole %s:%d, fd=%d", host.c_str(), port, fd);
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::on_stun(SrsUdpMuxSocket* skt, SrsStunPacket* r)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (!r->is_binding_request()) {
|
|
return err;
|
|
}
|
|
|
|
last_stun_time = srs_get_system_time();
|
|
|
|
// We are running in the ice-lite(server) mode. If client have multi network interface,
|
|
// we only choose one candidate pair which is determined by client.
|
|
if (!sendonly_skt || sendonly_skt->peer_id() != skt->peer_id()) {
|
|
update_sendonly_socket(skt);
|
|
}
|
|
|
|
// Write STUN messages to blackhole.
|
|
if (blackhole && blackhole_addr && blackhole_stfd) {
|
|
// Ignore any error for black-hole.
|
|
void* p = skt->data(); int len = skt->size();
|
|
srs_sendto(blackhole_stfd, p, len, (sockaddr*)blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
|
|
}
|
|
|
|
if ((err = on_binding_request(r)) != srs_success) {
|
|
return srs_error_wrap(err, "stun binding request failed");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::on_dtls(char* data, int nb_data)
|
|
{
|
|
return dtls_->on_dtls(data, nb_data);
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::on_rtcp(char* data, int nb_data)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (dtls_ == NULL) {
|
|
return srs_error_new(ERROR_RTC_RTCP, "recv unexpect rtp packet before dtls done");
|
|
}
|
|
|
|
char unprotected_buf[kRtpPacketSize];
|
|
int nb_unprotected_buf = nb_data;
|
|
if ((err = dtls_->unprotect_rtcp(unprotected_buf, data, nb_unprotected_buf)) != srs_success) {
|
|
return srs_error_wrap(err, "rtcp unprotect failed");
|
|
}
|
|
|
|
if (blackhole && blackhole_addr && blackhole_stfd) {
|
|
// Ignore any error for black-hole.
|
|
void* p = unprotected_buf; int len = nb_unprotected_buf;
|
|
srs_sendto(blackhole_stfd, p, len, (sockaddr*)blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
|
|
}
|
|
|
|
if (player_) {
|
|
return player_->on_rtcp(unprotected_buf, nb_unprotected_buf);
|
|
}
|
|
|
|
if (publisher_) {
|
|
return publisher_->on_rtcp(unprotected_buf, nb_unprotected_buf);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::on_rtp(char* data, int nb_data)
|
|
{
|
|
if (publisher_ == NULL) {
|
|
return srs_error_new(ERROR_RTC_RTCP, "rtc publisher null");
|
|
}
|
|
|
|
if (dtls_ == NULL) {
|
|
return srs_error_new(ERROR_RTC_RTCP, "recv unexpect rtp packet before dtls done");
|
|
}
|
|
|
|
return publisher_->on_rtp(data, nb_data);
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::on_connection_established()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
srs_trace("RTC %s session=%s, to=%dms connection established", (is_publisher_? "Publisher":"Subscriber"),
|
|
id().c_str(), srsu2msi(sessionStunTimeout));
|
|
|
|
if (is_publisher_) {
|
|
if ((err = start_publish()) != srs_success) {
|
|
return srs_error_wrap(err, "start publish");
|
|
}
|
|
} else {
|
|
if ((err = start_play()) != srs_success) {
|
|
return srs_error_wrap(err, "start play");
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::start_play()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
srs_freep(player_);
|
|
player_ = new SrsRtcPlayer(this, _srs_context->get_id());
|
|
|
|
uint32_t video_ssrc = 0;
|
|
uint32_t audio_ssrc = 0;
|
|
uint16_t video_payload_type = 0;
|
|
uint16_t audio_payload_type = 0;
|
|
for (size_t i = 0; i < local_sdp.media_descs_.size(); ++i) {
|
|
const SrsMediaDesc& media_desc = local_sdp.media_descs_[i];
|
|
if (media_desc.is_audio()) {
|
|
audio_ssrc = media_desc.ssrc_infos_[0].ssrc_;
|
|
audio_payload_type = media_desc.payload_types_[0].payload_type_;
|
|
} else if (media_desc.is_video()) {
|
|
video_ssrc = media_desc.ssrc_infos_[0].ssrc_;
|
|
video_payload_type = media_desc.payload_types_[0].payload_type_;
|
|
}
|
|
}
|
|
|
|
if ((err = player_->initialize(video_ssrc, audio_ssrc, video_payload_type, audio_payload_type)) != srs_success) {
|
|
return srs_error_wrap(err, "SrsRtcPlayer init");
|
|
}
|
|
|
|
if ((err = player_->start()) != srs_success) {
|
|
return srs_error_wrap(err, "start SrsRtcPlayer");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtcSession::start_publish()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
srs_freep(publisher_);
|
|
publisher_ = new SrsRtcPublisher(this);
|
|
// Request PLI for exists players?
|
|
//publisher_->request_keyframe();
|
|
|
|
uint32_t video_ssrc = 0;
|
|
uint32_t audio_ssrc = 0;
|
|
for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
|
|
const SrsMediaDesc& media_desc = remote_sdp.media_descs_[i];
|
|
if (media_desc.is_audio()) {
|
|
if (!media_desc.ssrc_infos_.empty()) {
|
|
audio_ssrc = media_desc.ssrc_infos_[0].ssrc_;
|
|
}
|
|
} else if (media_desc.is_video()) {
|
|
if (!media_desc.ssrc_infos_.empty()) {
|
|
video_ssrc = media_desc.ssrc_infos_[0].ssrc_;
|
|
}
|
|
}
|
|
}
|
|
|
|
uint32_t twcc_ext_id = 0;
|
|
for (size_t i = 0; i < local_sdp.media_descs_.size(); ++i) {
|
|
const SrsMediaDesc& media_desc = remote_sdp.media_descs_[i];
|
|
map<int, string> extmaps = media_desc.get_extmaps();
|
|
for(map<int, string>::iterator it_ext = extmaps.begin(); it_ext != extmaps.end(); ++it_ext) {
|
|
if(kTWCCExt == it_ext->second) {
|
|
twcc_ext_id = it_ext->first;
|
|
break;
|
|
}
|
|
}
|
|
if (twcc_ext_id != 0){
|
|
break;
|
|
}
|
|
}
|
|
|
|
// FIXME: err process.
|
|
if ((err = publisher_->initialize(video_ssrc, audio_ssrc, twcc_ext_id, req)) != srs_success) {
|
|
return srs_error_wrap(err, "rtc publisher init");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
bool SrsRtcSession::is_stun_timeout()
|
|
{
|
|
return last_stun_time + sessionStunTimeout < srs_get_system_time();
|
|
}
|
|
|
|
void SrsRtcSession::update_sendonly_socket(SrsUdpMuxSocket* skt)
|
|
{
|
|
if (sendonly_skt) {
|
|
srs_trace("session %s address changed, update %s -> %s",
|
|
id().c_str(), sendonly_skt->peer_id().c_str(), skt->peer_id().c_str());
|
|
}
|
|
|
|
srs_freep(sendonly_skt);
|
|
sendonly_skt = skt->copy_sendonly();
|
|
}
|
|
|
|
void SrsRtcSession::simulate_nack_drop(int nn)
|
|
{
|
|
if (player_) {
|
|
player_->simulate_nack_drop(nn);
|
|
}
|
|
|
|
if (publisher_) {
|
|
publisher_->simulate_nack_drop(nn);
|
|
}
|
|
}
|
|
|
|
#ifdef SRS_OSX
|
|
// These functions are similar to the older byteorder(3) family of functions.
|
|
// For example, be32toh() is identical to ntohl().
|
|
// @see https://linux.die.net/man/3/be32toh
|
|
#define be32toh ntohl
|
|
#endif
|
|
|
|
srs_error_t SrsRtcSession::on_binding_request(SrsStunPacket* r)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
bool strict_check = _srs_config->get_rtc_stun_strict_check(req->vhost);
|
|
if (strict_check && r->get_ice_controlled()) {
|
|
// @see: https://tools.ietf.org/html/draft-ietf-ice-rfc5245bis-00#section-6.1.3.1
|
|
// TODO: Send 487 (Role Conflict) error response.
|
|
return srs_error_new(ERROR_RTC_STUN, "Peer must not in ice-controlled role in ice-lite mode.");
|
|
}
|
|
|
|
SrsStunPacket stun_binding_response;
|
|
char buf[kRtpPacketSize];
|
|
SrsBuffer* stream = new SrsBuffer(buf, sizeof(buf));
|
|
SrsAutoFree(SrsBuffer, stream);
|
|
|
|
stun_binding_response.set_message_type(BindingResponse);
|
|
stun_binding_response.set_local_ufrag(r->get_remote_ufrag());
|
|
stun_binding_response.set_remote_ufrag(r->get_local_ufrag());
|
|
stun_binding_response.set_transcation_id(r->get_transcation_id());
|
|
// FIXME: inet_addr is deprecated, IPV6 support
|
|
stun_binding_response.set_mapped_address(be32toh(inet_addr(sendonly_skt->get_peer_ip().c_str())));
|
|
stun_binding_response.set_mapped_port(sendonly_skt->get_peer_port());
|
|
|
|
if ((err = stun_binding_response.encode(get_local_sdp()->get_ice_pwd(), stream)) != srs_success) {
|
|
return srs_error_wrap(err, "stun binding response encode failed");
|
|
}
|
|
|
|
if ((err = sendonly_skt->sendto(stream->data(), stream->pos(), 0)) != srs_success) {
|
|
return srs_error_wrap(err, "stun binding response send failed");
|
|
}
|
|
|
|
if (state_ == WAITING_STUN) {
|
|
state_ = DOING_DTLS_HANDSHAKE;
|
|
|
|
peer_id_ = sendonly_skt->peer_id();
|
|
server_->insert_into_id_sessions(peer_id_, this);
|
|
|
|
state_ = DOING_DTLS_HANDSHAKE;
|
|
srs_trace("rtc session=%s, STUN done, waitting DTLS handshake.", id().c_str());
|
|
}
|
|
|
|
if (blackhole && blackhole_addr && blackhole_stfd) {
|
|
// Ignore any error for black-hole.
|
|
void* p = stream->data(); int len = stream->pos();
|
|
srs_sendto(blackhole_stfd, p, len, (sockaddr*)blackhole_addr, sizeof(sockaddr_in), SRS_UTIME_NO_TIMEOUT);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
ISrsRtcHijacker::ISrsRtcHijacker()
|
|
{
|
|
}
|
|
|
|
ISrsRtcHijacker::~ISrsRtcHijacker()
|
|
{
|
|
}
|
|
|
|
ISrsRtcHijacker* _srs_rtc_hijacker = NULL;
|
|
|