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srs/trunk/src/app/srs_app_rtc_conn.cpp
winlin c1464f5aee Use extra queue when set mw_msgs
2020-04-18 10:35:30 +08:00

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/**
* The MIT License (MIT)
*
* Copyright (c) 2013-2020 John
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
* the Software, and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#include <srs_app_rtc_conn.hpp>
using namespace std;
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <stdlib.h>
#include <fcntl.h>
#include <unistd.h>
#include <netinet/udp.h>
#ifndef UDP_SEGMENT
#define UDP_SEGMENT 103
#endif
#include <sstream>
#include <srs_core_autofree.hpp>
#include <srs_kernel_buffer.hpp>
#include <srs_kernel_rtp.hpp>
#include <srs_kernel_error.hpp>
#include <srs_kernel_log.hpp>
#include <srs_stun_stack.hpp>
#include <srs_rtmp_stack.hpp>
#include <srs_rtmp_msg_array.hpp>
#include <srs_app_dtls.hpp>
#include <srs_app_utility.hpp>
#include <srs_app_config.hpp>
#include <srs_app_rtc.hpp>
#include <srs_app_source.hpp>
#include <srs_app_server.hpp>
#include <srs_service_utility.hpp>
#include <srs_http_stack.hpp>
#include <srs_app_http_api.hpp>
#include <srs_app_statistic.hpp>
#include <srs_app_pithy_print.hpp>
// The RTP payload max size, reserved some paddings for SRTP as such:
// kRtpPacketSize = kRtpMaxPayloadSize + paddings
// For example, if kRtpPacketSize is 1500, recommend to set kRtpMaxPayloadSize to 1400,
// which reserves 100 bytes for SRTP or paddings.
const int kRtpMaxPayloadSize = kRtpPacketSize - 200;
static bool is_stun(const uint8_t* data, const int size)
{
return data != NULL && size > 0 && (data[0] == 0 || data[0] == 1);
}
static bool is_dtls(const uint8_t* data, size_t len)
{
return (len >= 13 && (data[0] > 19 && data[0] < 64));
}
static bool is_rtp_or_rtcp(const uint8_t* data, size_t len)
{
return (len >= 12 && (data[0] & 0xC0) == 0x80);
}
static bool is_rtcp(const uint8_t* data, size_t len)
{
return (len >= 12) && (data[0] & 0x80) && (data[1] >= 200 && data[1] <= 209);
}
static string gen_random_str(int len)
{
static string random_table = "0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ";
string ret;
ret.reserve(len);
for (int i = 0; i < len; ++i) {
ret.append(1, random_table[random() % random_table.size()]);
}
return ret;
}
const int SRTP_MASTER_KEY_KEY_LEN = 16;
const int SRTP_MASTER_KEY_SALT_LEN = 14;
static std::vector<std::string> get_candidate_ips()
{
std::vector<std::string> candidate_ips;
string candidate = _srs_config->get_rtc_server_candidates();
if (candidate == "*" || candidate == "0.0.0.0") {
std::vector<std::string> tmp = srs_get_local_ips();
for (int i = 0; i < (int)tmp.size(); ++i) {
if (tmp[i] != "127.0.0.1") {
candidate_ips.push_back(tmp[i]);
}
}
} else {
candidate_ips.push_back(candidate);
}
return candidate_ips;
}
SrsDtlsSession::SrsDtlsSession(SrsRtcSession* s)
{
rtc_session = s;
dtls = NULL;
bio_in = NULL;
bio_out = NULL;
client_key = "";
server_key = "";
srtp_send = NULL;
srtp_recv = NULL;
handshake_done = false;
}
SrsDtlsSession::~SrsDtlsSession()
{
if (dtls) {
// this function will free bio_in and bio_out
SSL_free(dtls);
dtls = NULL;
}
if (srtp_send) {
srtp_dealloc(srtp_send);
}
if (srtp_recv) {
srtp_dealloc(srtp_recv);
}
}
srs_error_t SrsDtlsSession::initialize(const SrsRequest& req)
{
srs_error_t err = srs_success;
if ((err = SrsDtls::instance()->init(req)) != srs_success) {
return srs_error_wrap(err, "DTLS init");
}
// TODO: FIXME: Support config by vhost to use RSA or ECDSA certificate.
if ((dtls = SSL_new(SrsDtls::instance()->get_dtls_ctx())) == NULL) {
return srs_error_new(ERROR_OpenSslCreateSSL, "SSL_new dtls");
}
// Dtls setup passive, as server role.
SSL_set_accept_state(dtls);
if ((bio_in = BIO_new(BIO_s_mem())) == NULL) {
return srs_error_new(ERROR_OpenSslBIONew, "BIO_new in");
}
if ((bio_out = BIO_new(BIO_s_mem())) == NULL) {
BIO_free(bio_in);
return srs_error_new(ERROR_OpenSslBIONew, "BIO_new out");
}
SSL_set_bio(dtls, bio_in, bio_out);
return err;
}
srs_error_t SrsDtlsSession::handshake(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
int ret = SSL_do_handshake(dtls);
unsigned char *out_bio_data;
int out_bio_len = BIO_get_mem_data(bio_out, &out_bio_data);
int ssl_err = SSL_get_error(dtls, ret);
switch(ssl_err) {
case SSL_ERROR_NONE: {
if ((err = on_dtls_handshake_done(skt)) != srs_success) {
return srs_error_wrap(err, "dtls handshake done handle");
}
break;
}
case SSL_ERROR_WANT_READ: {
break;
}
case SSL_ERROR_WANT_WRITE: {
break;
}
default: {
break;
}
}
if (out_bio_len) {
if ((err = skt->sendto(out_bio_data, out_bio_len, 0)) != srs_success) {
return srs_error_wrap(err, "send dtls packet");
}
}
return err;
}
srs_error_t SrsDtlsSession::on_dtls(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
if (BIO_reset(bio_in) != 1) {
return srs_error_new(ERROR_OpenSslBIOReset, "BIO_reset");
}
if (BIO_reset(bio_out) != 1) {
return srs_error_new(ERROR_OpenSslBIOReset, "BIO_reset");
}
if (BIO_write(bio_in, skt->data(), skt->size()) <= 0) {
// TODO: 0 or -1 maybe block, use BIO_should_retry to check.
return srs_error_new(ERROR_OpenSslBIOWrite, "BIO_write");
}
if (! handshake_done) {
err = handshake(skt);
} else {
while (BIO_ctrl_pending(bio_in) > 0) {
char dtls_read_buf[8092];
int nb = SSL_read(dtls, dtls_read_buf, sizeof(dtls_read_buf));
if (nb > 0) {
if ((err =on_dtls_application_data(dtls_read_buf, nb)) != srs_success) {
return srs_error_wrap(err, "dtls application data process");
}
}
}
}
return err;
}
srs_error_t SrsDtlsSession::on_dtls_handshake_done(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
srs_trace("dtls handshake done");
handshake_done = true;
if ((err = srtp_initialize()) != srs_success) {
return srs_error_wrap(err, "srtp init failed");
}
return rtc_session->on_connection_established(skt);
}
srs_error_t SrsDtlsSession::on_dtls_application_data(const char* buf, const int nb_buf)
{
srs_error_t err = srs_success;
// TODO: process SCTP protocol(WebRTC datachannel support)
return err;
}
srs_error_t SrsDtlsSession::srtp_initialize()
{
srs_error_t err = srs_success;
unsigned char material[SRTP_MASTER_KEY_LEN * 2] = {0}; // client(SRTP_MASTER_KEY_KEY_LEN + SRTP_MASTER_KEY_SALT_LEN) + server
static const string dtls_srtp_lable = "EXTRACTOR-dtls_srtp";
if (! SSL_export_keying_material(dtls, material, sizeof(material), dtls_srtp_lable.c_str(), dtls_srtp_lable.size(), NULL, 0, 0)) {
return srs_error_new(ERROR_RTC_SRTP_INIT, "SSL_export_keying_material failed");
}
size_t offset = 0;
std::string client_master_key(reinterpret_cast<char*>(material), SRTP_MASTER_KEY_KEY_LEN);
offset += SRTP_MASTER_KEY_KEY_LEN;
std::string server_master_key(reinterpret_cast<char*>(material + offset), SRTP_MASTER_KEY_KEY_LEN);
offset += SRTP_MASTER_KEY_KEY_LEN;
std::string client_master_salt(reinterpret_cast<char*>(material + offset), SRTP_MASTER_KEY_SALT_LEN);
offset += SRTP_MASTER_KEY_SALT_LEN;
std::string server_master_salt(reinterpret_cast<char*>(material + offset), SRTP_MASTER_KEY_SALT_LEN);
client_key = client_master_key + client_master_salt;
server_key = server_master_key + server_master_salt;
if ((err = srtp_send_init()) != srs_success) {
return srs_error_wrap(err, "srtp send init failed");
}
if ((err = srtp_recv_init()) != srs_success) {
return srs_error_wrap(err, "srtp recv init failed");
}
return err;
}
srs_error_t SrsDtlsSession::srtp_send_init()
{
srs_error_t err = srs_success;
srtp_policy_t policy;
bzero(&policy, sizeof(policy));
// TODO: Maybe we can use SRTP-GCM in future.
// @see https://bugs.chromium.org/p/chromium/issues/detail?id=713701
// @see https://groups.google.com/forum/#!topic/discuss-webrtc/PvCbWSetVAQ
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
policy.ssrc.type = ssrc_any_outbound;
policy.ssrc.value = 0;
// TODO: adjust window_size
policy.window_size = 8192;
policy.allow_repeat_tx = 1;
policy.next = NULL;
uint8_t *key = new uint8_t[server_key.size()];
memcpy(key, server_key.data(), server_key.size());
policy.key = key;
if (srtp_create(&srtp_send, &policy) != srtp_err_status_ok) {
srs_freepa(key);
return srs_error_new(ERROR_RTC_SRTP_INIT, "srtp_create failed");
}
srs_freepa(key);
return err;
}
srs_error_t SrsDtlsSession::srtp_recv_init()
{
srs_error_t err = srs_success;
srtp_policy_t policy;
bzero(&policy, sizeof(policy));
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
policy.ssrc.type = ssrc_any_inbound;
policy.ssrc.value = 0;
// TODO: adjust window_size
policy.window_size = 8192;
policy.allow_repeat_tx = 1;
policy.next = NULL;
uint8_t *key = new uint8_t[client_key.size()];
memcpy(key, client_key.data(), client_key.size());
policy.key = key;
if (srtp_create(&srtp_recv, &policy) != srtp_err_status_ok) {
srs_freepa(key);
return srs_error_new(ERROR_RTC_SRTP_INIT, "srtp_create failed");
}
srs_freepa(key);
return err;
}
srs_error_t SrsDtlsSession::protect_rtp(char* out_buf, const char* in_buf, int& nb_out_buf)
{
srs_error_t err = srs_success;
if (srtp_send) {
memcpy(out_buf, in_buf, nb_out_buf);
if (srtp_protect(srtp_send, out_buf, &nb_out_buf) != 0) {
return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect failed");
}
return err;
}
return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect failed");
}
srs_error_t SrsDtlsSession::protect_rtp2(void* rtp_hdr, int* len_ptr)
{
srs_error_t err = srs_success;
if (!srtp_send) {
return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect");
}
if (srtp_protect(srtp_send, rtp_hdr, len_ptr) != 0) {
return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtp protect");
}
return err;
}
srs_error_t SrsDtlsSession::unprotect_rtp(char* out_buf, const char* in_buf, int& nb_out_buf)
{
srs_error_t err = srs_success;
if (srtp_recv) {
memcpy(out_buf, in_buf, nb_out_buf);
if (srtp_unprotect(srtp_recv, out_buf, &nb_out_buf) != 0) {
return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtp unprotect failed");
}
return err;
}
return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtp unprotect failed");
}
srs_error_t SrsDtlsSession::protect_rtcp(char* out_buf, const char* in_buf, int& nb_out_buf)
{
srs_error_t err = srs_success;
if (srtp_send) {
memcpy(out_buf, in_buf, nb_out_buf);
if (srtp_protect_rtcp(srtp_send, out_buf, &nb_out_buf) != 0) {
return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtcp protect failed");
}
return err;
}
return srs_error_new(ERROR_RTC_SRTP_PROTECT, "rtcp protect failed");
}
srs_error_t SrsDtlsSession::unprotect_rtcp(char* out_buf, const char* in_buf, int& nb_out_buf)
{
srs_error_t err = srs_success;
if (srtp_recv) {
memcpy(out_buf, in_buf, nb_out_buf);
if (srtp_unprotect_rtcp(srtp_recv, out_buf, &nb_out_buf) != srtp_err_status_ok) {
return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtcp unprotect failed");
}
return err;
}
return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtcp unprotect failed");
}
SrsRtcPackets::SrsRtcPackets()
{
#if defined(SRS_DEBUG)
debug_id = 0;
#endif
use_gso = false;
should_merge_nalus = false;
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;
nn_dropped = 0;
cursor = 0;
}
SrsRtcPackets::~SrsRtcPackets()
{
vector<SrsRtpPacket2*>::iterator it;
for (it = packets.begin(); it != packets.end(); ++it) {
SrsRtpPacket2* p = *it;
srs_freep(p);
}
packets.clear();
}
void SrsRtcPackets::reset(bool gso, bool merge_nalus)
{
for (int i = 0; i < cursor; i++) {
SrsRtpPacket2* packet = packets[i];
packet->reset();
}
#if defined(SRS_DEBUG)
debug_id++;
#endif
use_gso = gso;
should_merge_nalus = merge_nalus;
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;
nn_dropped = 0;
cursor = 0;
}
SrsRtpPacket2* SrsRtcPackets::fetch()
{
if (cursor >= (int)packets.size()) {
packets.push_back(new SrsRtpPacket2());
}
return packets[cursor++];
}
SrsRtpPacket2* SrsRtcPackets::back()
{
srs_assert(cursor > 0);
return packets[cursor - 1];
}
int SrsRtcPackets::size()
{
return cursor;
}
int SrsRtcPackets::capacity()
{
return (int)packets.size();
}
SrsRtpPacket2* SrsRtcPackets::at(int index)
{
srs_assert(index < cursor);
return packets[index];
}
SrsRtcSenderThread::SrsRtcSenderThread(SrsRtcSession* s, SrsUdpMuxSocket* u, int parent_cid)
: sendonly_ukt(NULL)
{
_parent_cid = parent_cid;
trd = new SrsDummyCoroutine();
rtc_session = s;
sendonly_ukt = u->copy_sendonly();
sender = u->sender();
gso = false;
merge_nalus = false;
max_padding = 0;
audio_timestamp = 0;
audio_sequence = 0;
video_sequence = 0;
mw_sleep = 0;
mw_msgs = 0;
realtime = true;
_srs_config->subscribe(this);
}
SrsRtcSenderThread::~SrsRtcSenderThread()
{
_srs_config->unsubscribe(this);
srs_freep(trd);
srs_freep(sendonly_ukt);
}
srs_error_t SrsRtcSenderThread::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;
gso = _srs_config->get_rtc_server_gso();
merge_nalus = _srs_config->get_rtc_server_merge_nalus();
max_padding = _srs_config->get_rtc_server_padding();
srs_trace("RTC sender video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), package(gso=%d, merge_nalus=%d), padding=%d",
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, gso, merge_nalus, max_padding);
return err;
}
srs_error_t SrsRtcSenderThread::on_reload_rtc_server()
{
gso = _srs_config->get_rtc_server_gso();
merge_nalus = _srs_config->get_rtc_server_merge_nalus();
max_padding = _srs_config->get_rtc_server_padding();
srs_trace("Reload rtc_server gso=%d, merge_nalus=%d, max_padding=%d", gso, merge_nalus, max_padding);
return srs_success;
}
srs_error_t SrsRtcSenderThread::on_reload_vhost_play(string vhost)
{
SrsRequest* req = &rtc_session->request;
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);
mw_sleep = _srs_config->get_mw_sleep(req->vhost, true);
srs_trace("Reload play realtime=%d, mw_msgs=%d, mw_sleep=%d", realtime, mw_msgs, mw_sleep);
return srs_success;
}
srs_error_t SrsRtcSenderThread::on_reload_vhost_realtime(string vhost)
{
return on_reload_vhost_play(vhost);
}
int SrsRtcSenderThread::cid()
{
return trd->cid();
}
srs_error_t SrsRtcSenderThread::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 SrsRtcSenderThread::stop()
{
trd->stop();
}
void SrsRtcSenderThread::stop_loop()
{
trd->interrupt();
}
void SrsRtcSenderThread::update_sendonly_socket(SrsUdpMuxSocket* skt)
{
srs_trace("session %s address changed, update %s -> %s",
rtc_session->id().c_str(), sendonly_ukt->get_peer_id().c_str(), skt->get_peer_id().c_str());
srs_freep(sendonly_ukt);
sendonly_ukt = skt->copy_sendonly();
sender = skt->sender();
}
srs_error_t SrsRtcSenderThread::cycle()
{
srs_error_t err = srs_success;
SrsSource* source = NULL;
SrsRequest* req = &rtc_session->request;
// TODO: FIXME: Should refactor it, directly use http server as handler.
ISrsSourceHandler* handler = _srs_hybrid->srs()->instance();
if ((err = _srs_sources->fetch_or_create(req, handler, &source)) != srs_success) {
return srs_error_wrap(err, "rtc fetch source failed");
}
SrsConsumer* consumer = NULL;
SrsAutoFree(SrsConsumer, consumer);
if ((err = source->create_consumer(NULL, consumer)) != srs_success) {
return srs_error_wrap(err, "rtc create consumer, source url=%s", req->get_stream_url().c_str());
}
realtime = _srs_config->get_realtime_enabled(req->vhost, true);
mw_sleep = _srs_config->get_mw_sleep(req->vhost, true);
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime, true);
// We merged write more messages, so we need larger queue.
if (mw_msgs > 2) {
sender->set_extra_ratio(150);
} else if (mw_msgs > 0) {
sender->set_extra_ratio(80);
}
srs_trace("RTC source url=%s, source_id=[%d][%d], encrypt=%d, realtime=%d, mw_sleep=%dms, mw_msgs=%d", req->get_stream_url().c_str(),
::getpid(), source->source_id(), rtc_session->encrypt, realtime, srsu2msi(mw_sleep), mw_msgs);
SrsRtcPackets pkts;
SrsMessageArray msgs(SRS_PERF_MW_MSGS);
SrsPithyPrint* pprint = SrsPithyPrint::create_rtc_play();
SrsAutoFree(SrsPithyPrint, pprint);
bool stat_enabled = _srs_config->get_rtc_server_perf_stat();
SrsStatistic* stat = SrsStatistic::instance();
while (true) {
if ((err = trd->pull()) != srs_success) {
return srs_error_wrap(err, "rtc sender thread");
}
#ifdef SRS_PERF_QUEUE_COND_WAIT
// Wait for amount of messages or a duration.
consumer->wait(mw_msgs, mw_sleep);
#endif
// Try to read some messages.
int msg_count = 0;
if ((err = consumer->dump_packets(&msgs, msg_count)) != srs_success) {
continue;
}
if (msg_count <= 0) {
#ifndef SRS_PERF_QUEUE_COND_WAIT
srs_usleep(mw_sleep);
#endif
continue;
}
// Transmux and send out messages.
pkts.reset(gso, merge_nalus);
if ((err = send_messages(source, msgs.msgs, msg_count, pkts)) != srs_success) {
srs_warn("send err %s", srs_error_summary(err).c_str()); srs_error_reset(err);
}
// Do cleanup messages.
for (int i = 0; i < msg_count; i++) {
SrsSharedPtrMessage* msg = msgs.msgs[i];
srs_freep(msg);
}
// Stat for performance analysis.
if (stat_enabled) {
// 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(pkts.nn_audios, pkts.nn_extras) + pkts.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(pkts.size());
// Stat the RTP packets going into kernel.
stat->perf_on_gso_packets(pkts.nn_rtp_pkts);
// Stat the bytes and paddings.
stat->perf_on_rtc_bytes(pkts.nn_bytes, pkts.nn_rtp_bytes, pkts.nn_padding_bytes);
// Stat the messages and dropped count.
stat->perf_on_dropped(msg_count, nn_rtc_packets, pkts.nn_dropped);
#if defined(SRS_DEBUG)
srs_trace("RTC PLAY perf, msgs %d/%d, rtp %d, gso %d, %d audios, %d extras, %d videos, %d samples, %d/%d/%d bytes",
msg_count, nn_rtc_packets, pkts.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos,
pkts.nn_samples, pkts.nn_bytes, pkts.nn_rtp_bytes, pkts.nn_padding_bytes);
#endif
}
pprint->elapse();
if (pprint->can_print()) {
// TODO: FIXME: Print stat like frame/s, packet/s, loss_packets.
srs_trace("-> RTC PLAY %d/%d msgs, %d/%d packets, %d audios, %d extras, %d videos, %d samples, %d/%d/%d bytes, %d pad, %d/%d cache",
msg_count, pkts.nn_dropped, pkts.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos, pkts.nn_samples, pkts.nn_bytes,
pkts.nn_rtp_bytes, pkts.nn_padding_bytes, pkts.nn_paddings, pkts.size(), pkts.capacity());
}
}
}
srs_error_t SrsRtcSenderThread::send_messages(
SrsSource* source, SrsSharedPtrMessage** msgs, int nb_msgs, SrsRtcPackets& packets
) {
srs_error_t err = srs_success;
// If DTLS is not OK, drop all messages.
if (!rtc_session->dtls_session) {
return err;
}
// Covert kernel messages to RTP packets.
if ((err = messages_to_packets(source, msgs, nb_msgs, packets)) != srs_success) {
return srs_error_wrap(err, "messages to packets");
}
#ifndef SRS_AUTO_OSX
// If enabled GSO, send out some packets in a msghdr.
if (packets.use_gso) {
if ((err = send_packets_gso(packets)) != srs_success) {
return srs_error_wrap(err, "gso send");
}
return err;
}
#endif
// By default, we send packets by sendmmsg.
if ((err = send_packets(packets)) != srs_success) {
return srs_error_wrap(err, "raw send");
}
return err;
}
srs_error_t SrsRtcSenderThread::messages_to_packets(
SrsSource* source, SrsSharedPtrMessage** msgs, int nb_msgs, SrsRtcPackets& packets
) {
srs_error_t err = srs_success;
for (int i = 0; i < nb_msgs; i++) {
SrsSharedPtrMessage* msg = msgs[i];
// If overflow, drop all messages.
if (sender->overflow()) {
packets.nn_dropped += nb_msgs - i;
return err;
}
// Update stats.
packets.nn_bytes += msg->size;
int nn_extra_payloads = msg->nn_extra_payloads();
packets.nn_extras += nn_extra_payloads;
int nn_samples = msg->nn_samples();
packets.nn_samples += nn_samples;
// For audio, we transcoded AAC to opus in extra payloads.
if (msg->is_audio()) {
packets.nn_audios++;
for (int i = 0; i < nn_extra_payloads; i++) {
SrsSample* sample = msg->extra_payloads() + i;
if ((err = packet_opus(sample, packets, msg->nn_max_extra_payloads())) != srs_success) {
return srs_error_wrap(err, "opus package");
}
}
continue;
}
// For video, we should process all NALUs in samples.
packets.nn_videos++;
// Well, for each IDR, we append a SPS/PPS before it, which is packaged in STAP-A.
if (msg->has_idr()) {
if ((err = packet_stap_a(source, msg, packets)) != srs_success) {
return srs_error_wrap(err, "packet stap-a");
}
}
// If merge Nalus, we pcakges all NALUs(samples) as one NALU, in a RTP or FUA packet.
if (packets.should_merge_nalus && nn_samples > 1) {
if ((err = packet_nalus(msg, packets)) != srs_success) {
return srs_error_wrap(err, "packet stap-a");
}
continue;
}
// By default, we package each NALU(sample) to a RTP or FUA packet.
for (int i = 0; i < nn_samples; i++) {
SrsSample* sample = msg->samples() + i;
// We always ignore bframe here, if config to discard bframe,
// the bframe flag will not be set.
if (sample->bframe) {
continue;
}
if (sample->size <= kRtpMaxPayloadSize) {
if ((err = packet_single_nalu(msg, sample, packets)) != srs_success) {
return srs_error_wrap(err, "packet single nalu");
}
} else {
if ((err = packet_fu_a(msg, sample, kRtpMaxPayloadSize, packets)) != srs_success) {
return srs_error_wrap(err, "packet fu-a");
}
}
if (i == nn_samples - 1) {
packets.back()->rtp_header.set_marker(true);
}
}
}
return err;
}
srs_error_t SrsRtcSenderThread::send_packets(SrsRtcPackets& packets)
{
srs_error_t err = srs_success;
// Cache the encrypt flag.
bool encrypt = rtc_session->encrypt;
int nn_packets = packets.size();
for (int i = 0; i < nn_packets; i++) {
SrsRtpPacket2* packet = packets.at(i);
// Fetch a cached message from queue.
// TODO: FIXME: Maybe encrypt in async, so the state of mhdr maybe not ready.
mmsghdr* mhdr = NULL;
if ((err = sender->fetch(&mhdr)) != srs_success) {
return srs_error_wrap(err, "fetch msghdr");
}
// For this message, select the first iovec.
iovec* iov = mhdr->msg_hdr.msg_iov;
mhdr->msg_hdr.msg_iovlen = 1;
if (!iov->iov_base) {
iov->iov_base = new char[kRtpPacketSize];
}
iov->iov_len = kRtpPacketSize;
// Marshal packet to bytes in iovec.
if (true) {
SrsBuffer stream((char*)iov->iov_base, iov->iov_len);
if ((err = packet->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 = rtc_session->dtls_session->protect_rtp2(iov->iov_base, &nn_encrypt)) != srs_success) {
return srs_error_wrap(err, "srtp protect");
}
iov->iov_len = (size_t)nn_encrypt;
}
packets.nn_rtp_bytes += (int)iov->iov_len;
// Set the address and control information.
sockaddr_in* addr = (sockaddr_in*)sendonly_ukt->peer_addr();
socklen_t addrlen = (socklen_t)sendonly_ukt->peer_addrlen();
mhdr->msg_hdr.msg_name = (sockaddr_in*)addr;
mhdr->msg_hdr.msg_namelen = (socklen_t)addrlen;
mhdr->msg_hdr.msg_controllen = 0;
// When we send out a packet, we commit a RTP packet.
packets.nn_rtp_pkts++;
if ((err = sender->sendmmsg(mhdr)) != srs_success) {
return srs_error_wrap(err, "send msghdr");
}
}
return err;
}
// TODO: FIXME: We can gather and pad audios, because they have similar size.
srs_error_t SrsRtcSenderThread::send_packets_gso(SrsRtcPackets& packets)
{
srs_error_t err = srs_success;
// Cache the encrypt flag.
bool encrypt = rtc_session->encrypt;
// Previous handler, if has the same size, we can use GSO.
mmsghdr* gso_mhdr = NULL; int gso_size = 0; int gso_encrypt = 0; int gso_cursor = 0;
// GSO, N packets has same length, the final one may not.
bool using_gso = false; bool gso_final = false;
// The message will marshal in iovec.
iovec* iov = NULL;
int nn_packets = packets.size();
for (int i = 0; i < nn_packets; i++) {
SrsRtpPacket2* packet = packets.at(i);
int nn_packet = packet->nb_bytes();
int padding = 0;
SrsRtpPacket2* next_packet = NULL;
int nn_next_packet = 0;
if (max_padding > 0) {
if (i < nn_packets - 1) {
next_packet = (i < nn_packets - 1)? packets.at(i + 1):NULL;
nn_next_packet = next_packet? next_packet->nb_bytes() : 0;
}
// Padding the packet to next or GSO size.
if (next_packet) {
if (!using_gso) {
// Padding to the next packet to merge with it.
if (nn_next_packet > nn_packet) {
padding = nn_next_packet - nn_packet;
}
} else {
// Padding to GSO size for next one to merge with us.
if (nn_next_packet < gso_size) {
padding = gso_size - nn_packet;
}
}
// Reset padding if exceed max.
if (padding > max_padding) {
padding = 0;
}
if (padding > 0) {
#if defined(SRS_DEBUG)
srs_trace("#%d, Padding %d bytes %d=>%d, packets %d, max_padding %d", packets.debug_id,
padding, nn_packet, nn_packet + padding, nn_packets, max_padding);
#endif
packet->add_padding(padding);
nn_packet += padding;
packets.nn_paddings++;
packets.nn_padding_bytes += padding;
}
}
}
// Check whether we can use GSO to send it.
if (using_gso && !gso_final) {
gso_final = (gso_size != nn_packet);
}
if (next_packet) {
// If not GSO, maybe the first fresh packet, we should see whether the next packet is smaller than this one,
// if smaller, we can still enter GSO.
if (!using_gso) {
using_gso = (nn_packet >= nn_next_packet);
}
// If GSO, but next is bigger than this one, we must enter the final state.
if (using_gso && !gso_final) {
gso_final = (nn_packet < nn_next_packet);
}
}
// For GSO, reuse mhdr if possible.
mmsghdr* mhdr = gso_mhdr;
if (!mhdr) {
// Fetch a cached message from queue.
// TODO: FIXME: Maybe encrypt in async, so the state of mhdr maybe not ready.
if ((err = sender->fetch(&mhdr)) != srs_success) {
return srs_error_wrap(err, "fetch msghdr");
}
// Now, GSO will use this message and size.
gso_mhdr = mhdr;
gso_size = nn_packet;
}
// For this message, select a new iovec.
if (!iov) {
iov = mhdr->msg_hdr.msg_iov;
} else {
iov++;
}
gso_cursor++;
mhdr->msg_hdr.msg_iovlen = gso_cursor;
if (gso_cursor > SRS_PERF_RTC_GSO_IOVS && !iov->iov_base) {
iov->iov_base = new char[kRtpPacketSize];
}
iov->iov_len = kRtpPacketSize;
// Marshal packet to bytes in iovec.
if (true) {
SrsBuffer stream((char*)iov->iov_base, iov->iov_len);
if ((err = packet->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 = rtc_session->dtls_session->protect_rtp2(iov->iov_base, &nn_encrypt)) != srs_success) {
return srs_error_wrap(err, "srtp protect");
}
iov->iov_len = (size_t)nn_encrypt;
}
packets.nn_rtp_bytes += (int)iov->iov_len;
// If GSO, they must has same size, except the final one.
if (using_gso && !gso_final && gso_encrypt && gso_encrypt != (int)iov->iov_len) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "GSO size=%d/%d, encrypt=%d/%d", gso_size, nn_packet, gso_encrypt, iov->iov_len);
}
if (using_gso && !gso_final) {
gso_encrypt = iov->iov_len;
}
// If exceed the max GSO size, set to final.
if (using_gso && gso_cursor + 1 >= SRS_PERF_RTC_GSO_MAX) {
gso_final = true;
}
// For last message, or final gso, or determined not using GSO, send it now.
bool do_send = (i == nn_packets - 1 || gso_final || !using_gso);
#if defined(SRS_DEBUG)
bool is_video = packet->rtp_header.get_payload_type() == video_payload_type;
srs_trace("#%d, Packet %s SSRC=%d, SN=%d, %d/%d bytes", packets.debug_id, is_video? "Video":"Audio",
packet->rtp_header.get_ssrc(), packet->rtp_header.get_sequence(), nn_packet - padding, padding);
if (do_send) {
for (int j = 0; j < (int)mhdr->msg_hdr.msg_iovlen; j++) {
iovec* iov = mhdr->msg_hdr.msg_iov + j;
srs_trace("#%d, %s #%d/%d/%d, %d/%d bytes, size %d/%d", packets.debug_id, (using_gso? "GSO":"RAW"), j,
gso_cursor + 1, mhdr->msg_hdr.msg_iovlen, iov->iov_len, padding, gso_size, gso_encrypt);
}
}
#endif
if (do_send) {
// Set the address and control information.
sockaddr_in* addr = (sockaddr_in*)sendonly_ukt->peer_addr();
socklen_t addrlen = (socklen_t)sendonly_ukt->peer_addrlen();
mhdr->msg_hdr.msg_name = (sockaddr_in*)addr;
mhdr->msg_hdr.msg_namelen = (socklen_t)addrlen;
mhdr->msg_hdr.msg_controllen = 0;
#ifndef SRS_AUTO_OSX
if (using_gso) {
mhdr->msg_hdr.msg_controllen = CMSG_SPACE(sizeof(uint16_t));
if (!mhdr->msg_hdr.msg_control) {
mhdr->msg_hdr.msg_control = new char[mhdr->msg_hdr.msg_controllen];
}
cmsghdr* cm = CMSG_FIRSTHDR(&mhdr->msg_hdr);
cm->cmsg_level = SOL_UDP;
cm->cmsg_type = UDP_SEGMENT;
cm->cmsg_len = CMSG_LEN(sizeof(uint16_t));
*((uint16_t*)CMSG_DATA(cm)) = gso_encrypt;
}
#endif
// When we send out a packet, we commit a RTP packet.
packets.nn_rtp_pkts++;
if ((err = sender->sendmmsg(mhdr)) != srs_success) {
return srs_error_wrap(err, "send msghdr");
}
// Reset the GSO flag.
gso_mhdr = NULL; gso_size = 0; gso_encrypt = 0; gso_cursor = 0;
using_gso = gso_final = false; iov = NULL;
}
}
#if defined(SRS_DEBUG)
srs_trace("#%d, RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d, pad %d/%d/%d", packets.debug_id, packets.size(),
packets.nn_rtp_pkts, packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras, packets.nn_paddings,
packets.nn_padding_bytes, packets.nn_rtp_bytes);
#endif
return err;
}
srs_error_t SrsRtcSenderThread::packet_nalus(SrsSharedPtrMessage* msg, SrsRtcPackets& packets)
{
srs_error_t err = srs_success;
SrsRtpRawNALUs* raw = new SrsRtpRawNALUs();
for (int i = 0; i < msg->nn_samples(); i++) {
SrsSample* sample = msg->samples() + i;
// We always ignore bframe here, if config to discard bframe,
// the bframe flag will not be set.
if (sample->bframe) {
continue;
}
raw->push_back(sample->copy());
}
// Ignore empty.
int nn_bytes = raw->nb_bytes();
if (nn_bytes <= 0) {
srs_freep(raw);
return err;
}
if (nn_bytes < kRtpMaxPayloadSize) {
// Package NALUs in a single RTP packet.
SrsRtpPacket2* packet = packets.fetch();
packet->rtp_header.set_timestamp(msg->timestamp * 90);
packet->rtp_header.set_sequence(video_sequence++);
packet->rtp_header.set_ssrc(video_ssrc);
packet->rtp_header.set_payload_type(video_payload_type);
packet->payload = raw;
} else {
SrsAutoFree(SrsRtpRawNALUs, raw);
// Package NALUs in FU-A RTP packets.
int fu_payload_size = kRtpMaxPayloadSize;
// The first byte is store in FU-A header.
uint8_t header = raw->skip_first_byte();
uint8_t nal_type = header & kNalTypeMask;
int nb_left = nn_bytes - 1;
int num_of_packet = 1 + (nn_bytes - 1) / fu_payload_size;
for (int i = 0; i < num_of_packet; ++i) {
int packet_size = srs_min(nb_left, fu_payload_size);
SrsRtpPacket2* packet = packets.fetch();
packet->rtp_header.set_timestamp(msg->timestamp * 90);
packet->rtp_header.set_sequence(video_sequence++);
packet->rtp_header.set_ssrc(video_ssrc);
packet->rtp_header.set_payload_type(video_payload_type);
SrsRtpFUAPayload* fua = packet->reuse_fua();
fua->nri = (SrsAvcNaluType)header;
fua->nalu_type = (SrsAvcNaluType)nal_type;
fua->start = bool(i == 0);
fua->end = bool(i == num_of_packet - 1);
if ((err = raw->read_samples(fua->nalus, packet_size)) != srs_success) {
return srs_error_wrap(err, "read samples %d bytes, left %d, total %d", packet_size, nb_left, nn_bytes);
}
nb_left -= packet_size;
}
}
if (packets.size() > 0) {
packets.back()->rtp_header.set_marker(true);
}
return err;
}
srs_error_t SrsRtcSenderThread::packet_opus(SrsSample* sample, SrsRtcPackets& packets, int nn_max_payload)
{
srs_error_t err = srs_success;
SrsRtpPacket2* packet = packets.fetch();
packet->rtp_header.set_marker(true);
packet->rtp_header.set_timestamp(audio_timestamp);
packet->rtp_header.set_sequence(audio_sequence++);
packet->rtp_header.set_ssrc(audio_ssrc);
packet->rtp_header.set_payload_type(audio_payload_type);
SrsRtpRawPayload* raw = packet->reuse_raw();
raw->payload = sample->bytes;
raw->nn_payload = sample->size;
if (max_padding > 0) {
if (sample->size < nn_max_payload && nn_max_payload - sample->size < max_padding) {
int padding = nn_max_payload - sample->size;
packet->set_padding(padding);
#if defined(SRS_DEBUG)
srs_trace("#%d, Fast Padding %d bytes %d=>%d, SN=%d, max_payload %d, max_padding %d", packets.debug_id,
padding, sample->size, sample->size + padding, packet->rtp_header.get_sequence(), nn_max_payload, max_padding);
#endif
}
}
// TODO: FIXME: Why 960? Need Refactoring?
audio_timestamp += 960;
return err;
}
srs_error_t SrsRtcSenderThread::packet_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, SrsRtcPackets& packets)
{
srs_error_t err = srs_success;
char* p = sample->bytes + 1;
int nb_left = sample->size - 1;
uint8_t header = sample->bytes[0];
uint8_t nal_type = header & kNalTypeMask;
int num_of_packet = 1 + (sample->size - 1) / fu_payload_size;
for (int i = 0; i < num_of_packet; ++i) {
int packet_size = srs_min(nb_left, fu_payload_size);
SrsRtpPacket2* packet = packets.fetch();
packet->rtp_header.set_timestamp(msg->timestamp * 90);
packet->rtp_header.set_sequence(video_sequence++);
packet->rtp_header.set_ssrc(video_ssrc);
packet->rtp_header.set_payload_type(video_payload_type);
SrsRtpFUAPayload* fua = packet->reuse_fua();
fua->nri = (SrsAvcNaluType)header;
fua->nalu_type = (SrsAvcNaluType)nal_type;
fua->start = bool(i == 0);
fua->end = bool(i == num_of_packet - 1);
SrsSample* fragment_sample = new SrsSample();
fragment_sample->bytes = p;
fragment_sample->size = packet_size;
fua->nalus.push_back(fragment_sample);
p += packet_size;
nb_left -= packet_size;
}
return err;
}
// Single NAL Unit Packet @see https://tools.ietf.org/html/rfc6184#section-5.6
srs_error_t SrsRtcSenderThread::packet_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, SrsRtcPackets& packets)
{
srs_error_t err = srs_success;
SrsRtpPacket2* packet = packets.fetch();
packet->rtp_header.set_timestamp(msg->timestamp * 90);
packet->rtp_header.set_sequence(video_sequence++);
packet->rtp_header.set_ssrc(video_ssrc);
packet->rtp_header.set_payload_type(video_payload_type);
SrsRtpRawPayload* raw = packet->reuse_raw();
raw->payload = sample->bytes;
raw->nn_payload = sample->size;
return err;
}
srs_error_t SrsRtcSenderThread::packet_stap_a(SrsSource* source, SrsSharedPtrMessage* msg, SrsRtcPackets& packets)
{
srs_error_t err = srs_success;
SrsMetaCache* meta = source->cached_meta();
if (!meta) {
return err;
}
SrsFormat* format = meta->vsh_format();
if (!format || !format->vcodec) {
return err;
}
const vector<char>& sps = format->vcodec->sequenceParameterSetNALUnit;
const vector<char>& pps = format->vcodec->pictureParameterSetNALUnit;
if (sps.empty() || pps.empty()) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "sps/pps empty");
}
SrsRtpPacket2* packet = packets.fetch();
packet->rtp_header.set_marker(false);
packet->rtp_header.set_timestamp(msg->timestamp * 90);
packet->rtp_header.set_sequence(video_sequence++);
packet->rtp_header.set_ssrc(video_ssrc);
packet->rtp_header.set_payload_type(video_payload_type);
SrsRtpSTAPPayload* stap = new SrsRtpSTAPPayload();
packet->payload = stap;
uint8_t header = sps[0];
stap->nri = (SrsAvcNaluType)header;
if (true) {
SrsSample* sample = new SrsSample();
sample->bytes = (char*)&sps[0];
sample->size = (int)sps.size();
stap->nalus.push_back(sample);
}
if (true) {
SrsSample* sample = new SrsSample();
sample->bytes = (char*)&pps[0];
sample->size = (int)pps.size();
stap->nalus.push_back(sample);
}
return err;
}
SrsRtcSession::SrsRtcSession(SrsRtcServer* rtc_svr, const SrsRequest& req, const std::string& un, int context_id)
{
rtc_server = rtc_svr;
session_state = INIT;
dtls_session = new SrsDtlsSession(this);
dtls_session->initialize(req);
strd = NULL;
username = un;
last_stun_time = srs_get_system_time();
request = req;
source = NULL;
cid = context_id;
encrypt = true;
// TODO: FIXME: Support reload.
sessionStunTimeout = _srs_config->get_rtc_stun_timeout(req.vhost);
}
SrsRtcSession::~SrsRtcSession()
{
srs_freep(dtls_session);
if (strd) {
strd->stop();
}
srs_freep(strd);
}
void SrsRtcSession::set_local_sdp(const SrsSdp& sdp)
{
local_sdp = sdp;
}
void SrsRtcSession::switch_to_context()
{
_srs_context->set_id(cid);
}
srs_error_t SrsRtcSession::on_stun(SrsUdpMuxSocket* skt, SrsStunPacket* stun_req)
{
srs_error_t err = srs_success;
if (stun_req->is_binding_request()) {
if ((err = on_binding_request(skt, stun_req)) != srs_success) {
return srs_error_wrap(err, "stun binding request failed");
}
last_stun_time = srs_get_system_time();
if (strd && strd->sendonly_ukt) {
// 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 (stun_req->get_use_candidate() && strd->sendonly_ukt->get_peer_id() != skt->get_peer_id()) {
strd->update_sendonly_socket(skt);
}
}
}
return err;
}
srs_error_t SrsRtcSession::check_source()
{
srs_error_t err = srs_success;
if (source == NULL) {
// TODO: FIXME: Should refactor it, directly use http server as handler.
ISrsSourceHandler* handler = _srs_hybrid->srs()->instance();
if ((err = _srs_sources->fetch_or_create(&request, handler, &source)) != srs_success) {
return srs_error_wrap(err, "create source");
}
}
return err;
}
#ifdef SRS_AUTO_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(SrsUdpMuxSocket* skt, SrsStunPacket* stun_req)
{
srs_error_t err = srs_success;
bool strict_check = _srs_config->get_rtc_stun_strict_check(request.vhost);
if (strict_check && stun_req->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(stun_req->get_remote_ufrag());
stun_binding_response.set_remote_ufrag(stun_req->get_local_ufrag());
stun_binding_response.set_transcation_id(stun_req->get_transcation_id());
// FIXME: inet_addr is deprecated, IPV6 support
stun_binding_response.set_mapped_address(be32toh(inet_addr(skt->get_peer_ip().c_str())));
stun_binding_response.set_mapped_port(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 = skt->sendto(stream->data(), stream->pos(), 0)) != srs_success) {
return srs_error_wrap(err, "stun binding response send failed");
}
if (get_session_state() == WAITING_STUN) {
set_session_state(DOING_DTLS_HANDSHAKE);
peer_id = skt->get_peer_id();
rtc_server->insert_into_id_sessions(peer_id, this);
}
return err;
}
srs_error_t SrsRtcSession::on_rtcp_feedback(char* buf, int nb_buf, SrsUdpMuxSocket* skt)
{
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();
srs_verbose("pid=%u, blp=%d", pid, blp);
if ((err = check_source()) != srs_success) {
return srs_error_wrap(err, "check");
}
if (! source) {
return srs_error_new(ERROR_RTC_SOURCE_CHECK, "can not found source");
}
vector<SrsRtpSharedPacket*> resend_pkts;
SrsRtpSharedPacket* pkt = source->find_rtp_packet(pid);
if (pkt) {
resend_pkts.push_back(pkt);
}
uint16_t mask = 0x01;
for (int i = 1; i < 16 && blp; ++i, mask <<= 1) {
if (! (blp & mask)) {
continue;
}
uint32_t loss_seq = pid + i;
SrsRtpSharedPacket* pkt = source->find_rtp_packet(loss_seq);
if (! pkt) {
continue;
}
resend_pkts.push_back(pkt);
}
for (int i = 0; i < (int)resend_pkts.size(); ++i) {
if (dtls_session) {
char protected_buf[kRtpPacketSize];
int nb_protected_buf = resend_pkts[i]->size;
srs_verbose("resend pkt sequence=%u", resend_pkts[i]->rtp_header.get_sequence());
dtls_session->protect_rtp(protected_buf, resend_pkts[i]->payload, nb_protected_buf);
skt->sendto(protected_buf, nb_protected_buf, 0);
}
}
return err;
}
srs_error_t SrsRtcSession::on_rtcp_ps_feedback(char* buf, int nb_buf, SrsUdpMuxSocket* skt)
{
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;
// TODO: FIXME: Dead code?
/*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 SrsRtcSession::on_rtcp_receiver_report(char* buf, int nb_buf, SrsUdpMuxSocket* skt)
{
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;
}
srs_error_t SrsRtcSession::on_connection_established(SrsUdpMuxSocket* skt)
{
srs_trace("rtc session=%s, to=%dms connection established", id().c_str(), srsu2msi(sessionStunTimeout));
return start_play(skt);
}
srs_error_t SrsRtcSession::start_play(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
srs_freep(strd);
strd = new SrsRtcSenderThread(this, skt, _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 =strd->initialize(video_ssrc, audio_ssrc, video_payload_type, audio_payload_type)) != srs_success) {
return srs_error_wrap(err, "SrsRtcSenderThread init");
}
if ((err = strd->start()) != srs_success) {
return srs_error_wrap(err, "start SrsRtcSenderThread");
}
return err;
}
bool SrsRtcSession::is_stun_timeout()
{
return last_stun_time + sessionStunTimeout < srs_get_system_time();
}
srs_error_t SrsRtcSession::on_dtls(SrsUdpMuxSocket* skt)
{
return dtls_session->on_dtls(skt);
}
srs_error_t SrsRtcSession::on_rtcp(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
if (dtls_session == NULL) {
return srs_error_new(ERROR_RTC_RTCP, "recv unexpect rtp packet before dtls done");
}
char unprotected_buf[kRtpPacketSize];
int nb_unprotected_buf = skt->size();
if ((err = dtls_session->unprotect_rtcp(unprotected_buf, skt->data(), nb_unprotected_buf)) != srs_success) {
return srs_error_wrap(err, "rtcp unprotect failed");
}
char* ph = unprotected_buf;
int nb_left = nb_unprotected_buf;
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_unprotected_buf) {
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: {
break;
}
case kRR: {
err = on_rtcp_receiver_report(ph, length, skt);
break;
}
case kSDES: {
break;
}
case kBye: {
break;
}
case kApp: {
break;
}
case kRtpFb: {
err = on_rtcp_feedback(ph, length, skt);
break;
}
case kPsFb: {
err = on_rtcp_ps_feedback(ph, length, skt);
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;
}
SrsUdpMuxSender::SrsUdpMuxSender(SrsRtcServer* s)
{
lfd = NULL;
server = s;
waiting_msgs = false;
cond = srs_cond_new();
trd = new SrsDummyCoroutine();
cache_pos = 0;
max_sendmmsg = 0;
queue_length = 0;
extra_ratio = 0;
extra_queue = 0;
gso = false;
nn_senders = 0;
_srs_config->subscribe(this);
}
SrsUdpMuxSender::~SrsUdpMuxSender()
{
_srs_config->unsubscribe(this);
srs_freep(trd);
srs_cond_destroy(cond);
free_mhdrs(hotspot);
hotspot.clear();
free_mhdrs(cache);
cache.clear();
}
srs_error_t SrsUdpMuxSender::initialize(srs_netfd_t fd, int senders)
{
srs_error_t err = srs_success;
lfd = fd;
srs_freep(trd);
trd = new SrsSTCoroutine("udp", this);
if ((err = trd->start()) != srs_success) {
return srs_error_wrap(err, "start coroutine");
}
max_sendmmsg = _srs_config->get_rtc_server_sendmmsg();
gso = _srs_config->get_rtc_server_gso();
queue_length = srs_max(128, _srs_config->get_rtc_server_queue_length());
nn_senders = senders;
// For no GSO, we need larger queue.
if (!gso) {
queue_length *= 2;
}
srs_trace("RTC sender #%d init ok, max_sendmmsg=%d, gso=%d, queue_max=%dx%d, extra_ratio=%d/%d", srs_netfd_fileno(fd),
max_sendmmsg, gso, queue_length, nn_senders, extra_ratio, extra_queue);
return err;
}
void SrsUdpMuxSender::free_mhdrs(std::vector<mmsghdr>& mhdrs)
{
int nn_mhdrs = (int)mhdrs.size();
for (int i = 0; i < nn_mhdrs; i++) {
// @see https://linux.die.net/man/2/sendmmsg
// @see https://linux.die.net/man/2/sendmsg
mmsghdr* hdr = &mhdrs[i];
// Free control for GSO.
char* msg_control = (char*)hdr->msg_hdr.msg_control;
srs_freepa(msg_control);
// Free iovec.
for (int j = SRS_PERF_RTC_GSO_MAX - 1; j >= 0 ; j--) {
iovec* iov = hdr->msg_hdr.msg_iov + j;
char* data = (char*)iov->iov_base;
srs_freepa(data);
srs_freepa(iov);
}
}
mhdrs.clear();
}
srs_error_t SrsUdpMuxSender::fetch(mmsghdr** pphdr)
{
// TODO: FIXME: Maybe need to shrink?
if (cache_pos >= (int)cache.size()) {
// @see https://linux.die.net/man/2/sendmmsg
// @see https://linux.die.net/man/2/sendmsg
mmsghdr mhdr;
mhdr.msg_len = 0;
mhdr.msg_hdr.msg_flags = 0;
mhdr.msg_hdr.msg_control = NULL;
mhdr.msg_hdr.msg_iovlen = SRS_PERF_RTC_GSO_MAX;
mhdr.msg_hdr.msg_iov = new iovec[mhdr.msg_hdr.msg_iovlen];
memset((void*)mhdr.msg_hdr.msg_iov, 0, sizeof(iovec) * mhdr.msg_hdr.msg_iovlen);
for (int i = 0; i < SRS_PERF_RTC_GSO_IOVS; i++) {
iovec* p = mhdr.msg_hdr.msg_iov + i;
p->iov_base = new char[kRtpPacketSize];
}
cache.push_back(mhdr);
}
*pphdr = &cache[cache_pos++];
return srs_success;
}
bool SrsUdpMuxSender::overflow()
{
return cache_pos > queue_length + extra_queue;
}
void SrsUdpMuxSender::set_extra_ratio(int r)
{
// We use the larger extra ratio, because all vhosts shares the senders.
if (extra_ratio > r) {
return;
}
extra_ratio = r;
extra_queue = queue_length * r / 100;
srs_trace("RTC sender #%d extra queue, max_sendmmsg=%d, gso=%d, queue_max=%dx%d, extra_ratio=%d/%d, cache=%d/%d/%d", srs_netfd_fileno(lfd),
max_sendmmsg, gso, queue_length, nn_senders, extra_ratio, extra_queue, cache_pos, (int)cache.size(), (int)hotspot.size());
}
srs_error_t SrsUdpMuxSender::sendmmsg(mmsghdr* hdr)
{
if (waiting_msgs) {
waiting_msgs = false;
srs_cond_signal(cond);
}
return srs_success;
}
srs_error_t SrsUdpMuxSender::cycle()
{
srs_error_t err = srs_success;
uint64_t nn_msgs = 0; uint64_t nn_msgs_last = 0; int nn_msgs_max = 0;
uint64_t nn_gso_msgs = 0; uint64_t nn_gso_iovs = 0; int nn_gso_msgs_max = 0; int nn_gso_iovs_max = 0;
int nn_loop = 0; int nn_wait = 0;
srs_utime_t time_last = srs_get_system_time();
bool stat_enabled = _srs_config->get_rtc_server_perf_stat();
SrsStatistic* stat = SrsStatistic::instance();
SrsPithyPrint* pprint = SrsPithyPrint::create_rtc_send(srs_netfd_fileno(lfd));
SrsAutoFree(SrsPithyPrint, pprint);
while (true) {
if ((err = trd->pull()) != srs_success) {
return err;
}
nn_loop++;
int pos = cache_pos;
int gso_iovs = 0;
if (pos <= 0) {
waiting_msgs = true;
nn_wait++;
srs_cond_wait(cond);
continue;
}
// We are working on hotspot now.
cache.swap(hotspot);
cache_pos = 0;
// Collect informations for GSO.
int gso_pos = 0;
if (pos > 0 && stat_enabled) {
// For shared GSO cache, stat the messages.
// @see https://linux.die.net/man/2/sendmmsg
// @see https://linux.die.net/man/2/sendmsg
for (int i = 0; i < pos; i++) {
mmsghdr* mhdr = &hotspot[i];
int real_iovs = mhdr->msg_hdr.msg_iovlen;
gso_pos++; nn_gso_msgs++; nn_gso_iovs += real_iovs;
gso_iovs += real_iovs;
}
}
// Send out all messages.
if (pos > 0) {
// Send out all messages.
// @see https://linux.die.net/man/2/sendmmsg
// @see https://linux.die.net/man/2/sendmsg
mmsghdr* p = &hotspot[0]; mmsghdr* end = p + pos;
for (p = &hotspot[0]; p < end; p += max_sendmmsg) {
int vlen = (int)(end - p);
vlen = srs_min(max_sendmmsg, vlen);
int r0 = srs_sendmmsg(lfd, p, (unsigned int)vlen, 0, SRS_UTIME_NO_TIMEOUT);
if (r0 != vlen) {
srs_warn("sendmmsg %d msgs, %d done", vlen, r0);
}
if (stat_enabled) {
stat->perf_on_sendmmsg_packets(vlen);
}
}
}
// Increase total messages.
nn_msgs += pos + gso_iovs;
nn_msgs_max = srs_max(pos, nn_msgs_max);
nn_gso_msgs_max = srs_max(gso_pos, nn_gso_msgs_max);
nn_gso_iovs_max = srs_max(gso_iovs, nn_gso_iovs_max);
pprint->elapse();
if (pprint->can_print()) {
// TODO: FIXME: Extract a PPS calculator.
int pps_average = 0; int pps_last = 0;
if (true) {
if (srs_get_system_time() > srs_get_system_startup_time()) {
pps_average = (int)(nn_msgs * SRS_UTIME_SECONDS / (srs_get_system_time() - srs_get_system_startup_time()));
}
if (srs_get_system_time() > time_last) {
pps_last = (int)((nn_msgs - nn_msgs_last) * SRS_UTIME_SECONDS / (srs_get_system_time() - time_last));
}
}
string pps_unit = "";
if (pps_last > 10000 || pps_average > 10000) {
pps_unit = "(w)"; pps_last /= 10000; pps_average /= 10000;
} else if (pps_last > 1000 || pps_average > 1000) {
pps_unit = "(k)"; pps_last /= 1000; pps_average /= 1000;
}
int nn_cache = 0;
int nn_hotspot_size = (int)hotspot.size();
for (int i = 0; i < nn_hotspot_size; i++) {
mmsghdr* hdr = &hotspot[i];
nn_cache += hdr->msg_hdr.msg_iovlen;
}
srs_trace("-> RTC SEND #%d, sessions %d, udp %d/%d/%" PRId64 ", gso %d/%d/%" PRId64 ", iovs %d/%d/%" PRId64 ", pps %d/%d%s, cache %d/%d",
srs_netfd_fileno(lfd), (int)server->nn_sessions(), pos, nn_msgs_max, nn_msgs, gso_pos, nn_gso_msgs_max, nn_gso_msgs, gso_iovs,
nn_gso_iovs_max, nn_gso_iovs, pps_average, pps_last, pps_unit.c_str(), (int)hotspot.size(), nn_cache);
nn_msgs_last = nn_msgs; time_last = srs_get_system_time();
nn_loop = nn_wait = nn_msgs_max = 0;
nn_gso_msgs_max = 0; nn_gso_iovs_max = 0;
}
}
return err;
}
srs_error_t SrsUdpMuxSender::on_reload_rtc_server()
{
if (true) {
int v = _srs_config->get_rtc_server_sendmmsg();
if (max_sendmmsg != v) {
srs_trace("Reload max_sendmmsg %d=>%d", max_sendmmsg, v);
max_sendmmsg = v;
}
}
return srs_success;
}
SrsRtcServer::SrsRtcServer()
{
timer = new SrsHourGlass(this, 1 * SRS_UTIME_SECONDS);
}
SrsRtcServer::~SrsRtcServer()
{
srs_freep(timer);
if (true) {
vector<SrsUdpMuxListener*>::iterator it;
for (it = listeners.begin(); it != listeners.end(); ++it) {
SrsUdpMuxListener* listener = *it;
srs_freep(listener);
}
}
if (true) {
vector<SrsUdpMuxSender*>::iterator it;
for (it = senders.begin(); it != senders.end(); ++it) {
SrsUdpMuxSender* sender = *it;
srs_freep(sender);
}
}
}
srs_error_t SrsRtcServer::initialize()
{
srs_error_t err = srs_success;
if ((err = timer->tick(1 * SRS_UTIME_SECONDS)) != srs_success) {
return srs_error_wrap(err, "hourglass tick");
}
if ((err = timer->start()) != srs_success) {
return srs_error_wrap(err, "start timer");
}
srs_trace("RTC server init ok");
return err;
}
srs_error_t SrsRtcServer::listen_udp()
{
srs_error_t err = srs_success;
if (!_srs_config->get_rtc_server_enabled()) {
return err;
}
int port = _srs_config->get_rtc_server_listen();
if (port <= 0) {
return srs_error_new(ERROR_RTC_PORT, "invalid port=%d", port);
}
string ip = srs_any_address_for_listener();
srs_assert(listeners.empty());
int nn_listeners = _srs_config->get_rtc_server_reuseport();
for (int i = 0; i < nn_listeners; i++) {
SrsUdpMuxSender* sender = new SrsUdpMuxSender(this);
SrsUdpMuxListener* listener = new SrsUdpMuxListener(this, sender, ip, port);
if ((err = listener->listen()) != srs_success) {
srs_freep(listener);
return srs_error_wrap(err, "listen %s:%d", ip.c_str(), port);
}
if ((err = sender->initialize(listener->stfd(), nn_listeners)) != srs_success) {
return srs_error_wrap(err, "init sender");
}
srs_trace("rtc listen at udp://%s:%d, fd=%d", ip.c_str(), port, listener->fd());
listeners.push_back(listener);
senders.push_back(sender);
}
return err;
}
srs_error_t SrsRtcServer::on_udp_packet(SrsUdpMuxSocket* skt)
{
if (is_stun(reinterpret_cast<const uint8_t*>(skt->data()), skt->size())) {
return on_stun(skt);
} else if (is_dtls(reinterpret_cast<const uint8_t*>(skt->data()), skt->size())) {
return on_dtls(skt);
} else if (is_rtp_or_rtcp(reinterpret_cast<const uint8_t*>(skt->data()), skt->size())) {
return on_rtp_or_rtcp(skt);
}
return srs_error_new(ERROR_RTC_UDP, "unknown udp packet type");
}
srs_error_t SrsRtcServer::listen_api()
{
srs_error_t err = srs_success;
// TODO: FIXME: Fetch api from hybrid manager.
SrsHttpServeMux* http_api_mux = _srs_hybrid->srs()->instance()->api_server();
if ((err = http_api_mux->handle("/rtc/v1/play/", new SrsGoApiRtcPlay(this))) != srs_success) {
return srs_error_wrap(err, "handle sdp");
}
return err;
}
SrsRtcSession* SrsRtcServer::create_rtc_session(const SrsRequest& req, const SrsSdp& remote_sdp, SrsSdp& local_sdp, const string& mock_eip)
{
std::string local_pwd = gen_random_str(32);
std::string local_ufrag = "";
std::string username = "";
while (true) {
local_ufrag = gen_random_str(8);
username = local_ufrag + ":" + remote_sdp.get_ice_ufrag();
if (! map_username_session.count(username))
break;
}
int cid = _srs_context->get_id();
SrsRtcSession* session = new SrsRtcSession(this, req, username, cid);
map_username_session.insert(make_pair(username, session));
local_sdp.set_ice_ufrag(local_ufrag);
local_sdp.set_ice_pwd(local_pwd);
local_sdp.set_fingerprint_algo("sha-256");
local_sdp.set_fingerprint(SrsDtls::instance()->get_fingerprint());
// We allows to mock the eip of server.
if (!mock_eip.empty()) {
local_sdp.add_candidate(mock_eip, _srs_config->get_rtc_server_listen(), "host");
} else {
std::vector<string> candidate_ips = get_candidate_ips();
for (int i = 0; i < (int)candidate_ips.size(); ++i) {
local_sdp.add_candidate(candidate_ips[i], _srs_config->get_rtc_server_listen(), "host");
}
}
session->set_remote_sdp(remote_sdp);
session->set_local_sdp(local_sdp);
session->set_session_state(WAITING_STUN);
return session;
}
SrsRtcSession* SrsRtcServer::find_rtc_session_by_peer_id(const string& peer_id)
{
map<string, SrsRtcSession*>::iterator iter = map_id_session.find(peer_id);
if (iter == map_id_session.end()) {
return NULL;
}
return iter->second;
}
srs_error_t SrsRtcServer::on_stun(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
SrsStunPacket stun_req;
if ((err = stun_req.decode(skt->data(), skt->size())) != srs_success) {
return srs_error_wrap(err, "decode stun packet failed");
}
srs_verbose("recv stun packet from %s, use-candidate=%d, ice-controlled=%d, ice-controlling=%d",
skt->get_peer_id().c_str(), stun_req.get_use_candidate(), stun_req.get_ice_controlled(), stun_req.get_ice_controlling());
std::string username = stun_req.get_username();
SrsRtcSession* rtc_session = find_rtc_session_by_username(username);
if (rtc_session == NULL) {
return srs_error_new(ERROR_RTC_STUN, "can not find rtc_session, stun username=%s", username.c_str());
}
// Now, we got the RTC session to handle the packet, switch to its context
// to make all logs write to the "correct" pid+cid.
rtc_session->switch_to_context();
return rtc_session->on_stun(skt, &stun_req);
}
srs_error_t SrsRtcServer::on_dtls(SrsUdpMuxSocket* skt)
{
SrsRtcSession* rtc_session = find_rtc_session_by_peer_id(skt->get_peer_id());
if (rtc_session == NULL) {
return srs_error_new(ERROR_RTC_DTLS, "can not find rtc session by peer_id=%s", skt->get_peer_id().c_str());
}
// Now, we got the RTC session to handle the packet, switch to its context
// to make all logs write to the "correct" pid+cid.
rtc_session->switch_to_context();
return rtc_session->on_dtls(skt);
}
srs_error_t SrsRtcServer::on_rtp_or_rtcp(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
SrsRtcSession* rtc_session = find_rtc_session_by_peer_id(skt->get_peer_id());
if (rtc_session == NULL) {
return srs_error_new(ERROR_RTC_RTP, "can not find rtc session by peer_id=%s", skt->get_peer_id().c_str());
}
// Now, we got the RTC session to handle the packet, switch to its context
// to make all logs write to the "correct" pid+cid.
rtc_session->switch_to_context();
if (is_rtcp(reinterpret_cast<const uint8_t*>(skt->data()), skt->size())) {
err = rtc_session->on_rtcp(skt);
} else {
// We disable it because no RTP for player.
// see https://github.com/ossrs/srs/blob/018577e685a07d9de7a47354e7a9c5f77f5f4202/trunk/src/app/srs_app_rtc_conn.cpp#L1081
// err = rtc_session->on_rtp(skt);
}
return err;
}
SrsRtcSession* SrsRtcServer::find_rtc_session_by_username(const std::string& username)
{
map<string, SrsRtcSession*>::iterator iter = map_username_session.find(username);
if (iter == map_username_session.end()) {
return NULL;
}
return iter->second;
}
bool SrsRtcServer::insert_into_id_sessions(const string& peer_id, SrsRtcSession* rtc_session)
{
return map_id_session.insert(make_pair(peer_id, rtc_session)).second;
}
void SrsRtcServer::check_and_clean_timeout_session()
{
map<string, SrsRtcSession*>::iterator iter = map_username_session.begin();
while (iter != map_username_session.end()) {
SrsRtcSession* session = iter->second;
if (session == NULL) {
map_username_session.erase(iter++);
continue;
}
if (session->is_stun_timeout()) {
// Now, we got the RTC session to cleanup, switch to its context
// to make all logs write to the "correct" pid+cid.
session->switch_to_context();
srs_trace("rtc session=%s, stun timeout", session->id().c_str());
map_username_session.erase(iter++);
map_id_session.erase(session->get_peer_id());
delete session;
continue;
}
++iter;
}
}
srs_error_t SrsRtcServer::notify(int type, srs_utime_t interval, srs_utime_t tick)
{
check_and_clean_timeout_session();
return srs_success;
}
RtcServerAdapter::RtcServerAdapter()
{
rtc = new SrsRtcServer();
}
RtcServerAdapter::~RtcServerAdapter()
{
srs_freep(rtc);
}
srs_error_t RtcServerAdapter::initialize()
{
srs_error_t err = srs_success;
if ((err = rtc->initialize()) != srs_success) {
return srs_error_wrap(err, "rtc server initialize");
}
return err;
}
srs_error_t RtcServerAdapter::run()
{
srs_error_t err = srs_success;
if ((err = rtc->listen_udp()) != srs_success) {
return srs_error_wrap(err, "listen udp");
}
if ((err = rtc->listen_api()) != srs_success) {
return srs_error_wrap(err, "listen api");
}
return err;
}
void RtcServerAdapter::stop()
{
}
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