external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-03-09 15:49:59 +00:00
Code Issues Releases Wiki Activity

RTC: Always keep and use original sequence.

Browse source
This commit is contained in:
winlin 2020-06-28 11:40:49 +08:00
parent 6662568c11
commit 75fbcba71d
6 changed files with 7 additions and 76 deletions
Download patch file Download diff file Expand all files Collapse all files

4
trunk/conf/full.conf
View file

@ -491,10 +491,6 @@ vhost rtc.vhost.srs.com {
# The strick check when process stun.
# default: off
stun_strict_check on;
# Whether keep original sequence number.
# If off, we will regenerate the sequence number for RTP packet.
# default: off
keep_sequence off;
# The role of dtls when peer is actpass: passive or active
# default: passive
dtls_role passive;

20
trunk/src/app/srs_app_config.cpp
View file

@ -3935,7 +3935,7 @@ srs_error_t SrsConfig::check_normal_config()
for (int j = 0; j < (int)conf->directives.size(); j++) {
string m = conf->at(j)->name;
if (m != "enabled" && m != "bframe" && m != "aac" && m != "stun_timeout" && m != "stun_strict_check"
&& m != "keep_sequence" && m != "dtls_role" && m != "dtls_version") {
&& m != "dtls_role" && m != "dtls_version") {
return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
}
}
@ -5020,24 +5020,6 @@ bool SrsConfig::get_rtc_stun_strict_check(string vhost)
return SRS_CONF_PERFER_FALSE(conf->arg0());
}
bool SrsConfig::get_rtc_keep_sequence(string vhost)
{
static bool DEFAULT = false;
SrsConfDirective* conf = get_rtc(vhost);
if (!conf) {
return DEFAULT;
}
conf = conf->get("keep_sequence");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}
return SRS_CONF_PERFER_FALSE(conf->arg0());
}
std::string SrsConfig::get_rtc_dtls_role(string vhost)
{
static std::string DEFAULT = "passive";

1
trunk/src/app/srs_app_config.hpp
View file

@ -544,7 +544,6 @@ public:
bool get_rtc_aac_discard(std::string vhost);
srs_utime_t get_rtc_stun_timeout(std::string vhost);
bool get_rtc_stun_strict_check(std::string vhost);
bool get_rtc_keep_sequence(std::string vhost);
std::string get_rtc_dtls_role(std::string vhost);
std::string get_rtc_dtls_version(std::string vhost);
bool get_rtc_nack_enabled(std::string vhost);

16
trunk/src/app/srs_app_rtc_api.cpp
View file

@ -136,16 +136,9 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
string eip = r->query_get("eip");
// For client to specifies whether encrypt by SRTP.
string encrypt = r->query_get("encrypt");
// If keep_sequence is off, for client to specifies the startup sequence.
string sequence_startup = r->query_get("sequence_startup");
// If keep_sequence is on, for client to specifies the delta value for sequence.
string sequence_delta = r->query_get("sequence_delta");
// Whether keep sequence, overwrite the config for debugging each session.
string sequence_keep = r->query_get("sequence_keep");
srs_trace("RTC play %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, encrypt=%s, sequence(startup=%s,delta=%s,keep=%s)",
streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(), encrypt.c_str(),
sequence_startup.c_str(), sequence_delta.c_str(), sequence_keep.c_str());
srs_trace("RTC play %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, encrypt=%s",
streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(), encrypt.c_str());
// TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
SrsSdp remote_sdp;
@ -200,11 +193,6 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
session->set_encrypt(encrypt != "false");
}
// Set the optional parameters from client.
session->sequence_startup = sequence_startup;
session->sequence_delta = sequence_delta;
session->sequence_keep = sequence_keep;
ostringstream os;
if ((err = local_sdp.encode(os)) != srs_success) {
return srs_error_wrap(err, "encode sdp");

37
trunk/src/app/srs_app_rtc_conn.cpp
View file

@ -273,9 +273,6 @@ SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, string parent_cid)
session_ = s;
audio_sequence = 0;
video_sequence = 0;
sequence_delta = 0;
mw_msgs = 0;
realtime = true;
@ -285,7 +282,6 @@ SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, string parent_cid)
nn_simulate_nack_drop = 0;
nack_enabled_ = false;
keep_sequence_ = false;
_srs_config->subscribe(this);
}
@ -311,18 +307,8 @@ srs_error_t SrsRtcPlayer::initialize(const uint32_t& vssrc, const uint32_t& assr
// 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);
if (!session_->sequence_startup.empty()) {
audio_sequence = video_sequence = uint16_t(::atoi(session_->sequence_startup.c_str()));
}
if (!session_->sequence_delta.empty()) {
sequence_delta = uint16_t(::atoi(session_->sequence_delta.c_str()));
}
if (!session_->sequence_keep.empty()) {
keep_sequence_ = (session_->sequence_keep == "true");
}
srs_trace("RTC player video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), nack=%d, keep-seq=%d, sequence(audio=%u,video=%u,delta=%u)",
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, nack_enabled_, keep_sequence_, audio_sequence, video_sequence, sequence_delta);
srs_trace("RTC player video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), nack=%d",
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, nack_enabled_);
if (_srs_rtc_hijacker) {
if ((err = _srs_rtc_hijacker->on_start_play(session_, this, session_->req)) != srs_success) {
@ -495,36 +481,21 @@ srs_error_t SrsRtcPlayer::send_packets(SrsRtcSource* source, const vector<SrsRtp
// Update stats.
info.nn_bytes += pkt->nb_bytes();
uint16_t oseq = pkt->header.get_sequence();
if (pkt->is_audio()) {
info.nn_audios++;
if (!keep_sequence_) {
// TODO: FIXME: Should keep the order by original sequence.
pkt->header.set_sequence(sequence_delta + audio_sequence++);
} else {
pkt->header.set_sequence(sequence_delta + oseq);
}
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.
} else {
info.nn_videos++;
if (!keep_sequence_) {
// TODO: FIXME: Should keep the order by original sequence.
pkt->header.set_sequence(sequence_delta + video_sequence++);
} else {
pkt->header.set_sequence(sequence_delta + oseq);
}
pkt->header.set_ssrc(video_ssrc);
pkt->header.set_payload_type(video_payload_type);
}
// Detail log, should disable it in release version.
srs_info("RTC: Update PT=%u, SSRC=%#x, OSEQ=%u, SEQ=%u, Time=%u, %u bytes", pkt->header.get_payload_type(), pkt->header.get_ssrc(),
oseq, pkt->header.get_sequence(), pkt->header.get_timestamp(), pkt->nb_bytes());
srs_info("RTC: Update PT=%u, SSRC=%#x, Time=%u, %u bytes", pkt->header.get_payload_type(), pkt->header.get_ssrc(),
pkt->header.get_timestamp(), pkt->nb_bytes());
}
// By default, we send packets by sendmmsg.

5
trunk/src/app/srs_app_rtc_conn.hpp
View file

@ -191,18 +191,15 @@ protected:
private:
// TODO: FIXME: How to handle timestamp overflow?
// Information for audio.
uint16_t audio_sequence;
uint32_t audio_ssrc;
uint16_t audio_payload_type;
// Information for video.
uint16_t video_sequence;
uint16_t video_payload_type;
uint32_t video_ssrc;
// NACK ARQ ring buffer.
SrsRtpRingBuffer* audio_queue_;
SrsRtpRingBuffer* video_queue_;
// Simulators.
uint16_t sequence_delta;
int nn_simulate_nack_drop;
private:
// For merged-write messages.
@ -210,8 +207,6 @@ private:
bool realtime;
// Whether enabled nack.
bool nack_enabled_;
// Whether keep original sequence number.
bool keep_sequence_;
public:
SrsRtcPlayer(SrsRtcSession* s, std::string parent_cid);
virtual ~SrsRtcPlayer();