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fix rtmp send timeout, first key frame wait

Browse Source
pull/1748/head
kyxlx550 5 years ago
parent f74a398c1b
commit 28bde1d448
2 changed files with 73 additions and 36 deletions
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95
trunk/src/app/srs_app_gb28181.cpp
Unescape Escape View File

@ -268,7 +268,7 @@ srs_error_t SrsGb28181PsRtpProcessor::on_udp_packet(const sockaddr* from, const
}
if (pprint->can_print()) {
srs_trace("<- " SRS_CONSTS_LOG_STREAM_CASTER " gb28181: client_id %s, peer(%s, %d) ps rtp packet %dB, age=%d, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB",
srs_trace("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, peer(%s, %d) ps rtp packet %dB, age=%d, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB",
channel_id.c_str(), address_string, peer_port, nb_buf, pprint->age(), pkt.version,
pkt.payload_type, pkt.sequence_number, pkt.timestamp, pkt.ssrc,
pkt.payload->length()
@ -357,6 +357,7 @@ SrsPsStreamDemixer::SrsPsStreamDemixer(ISrsPsStreamHander *h, std::string id, bo
audio_enable = a;
wait_first_keyframe = k;
channel_id = id;
first_keyframe_flag = false;
}
SrsPsStreamDemixer::~SrsPsStreamDemixer()
@ -404,7 +405,6 @@ int64_t SrsPsStreamDemixer::parse_ps_timestamp(const uint8_t* p)
srs_error_t SrsPsStreamDemixer::on_ps_stream(char* ps_data, int ps_size, uint32_t timestamp, uint32_t ssrc)
{
srs_error_t err = srs_success;
int complete_len = 0;
int incomplete_len = ps_size;
char *next_ps_pack = ps_data;
@ -589,8 +589,10 @@ srs_error_t SrsPsStreamDemixer::on_ps_stream(char* ps_data, int ps_size, uint32_
//ts=1000 seq=4 mark=true payload= audio
incomplete_len = ps_size - complete_len;
complete_len = complete_len + incomplete_len;
}
first_keyframe_flag = false;
srs_trace("gb28181: client_id %s, unkonw ps data (%#x/%u) %02x %02x %02x %02x\n",
channel_id.c_str(), ssrc, timestamp,
next_ps_pack[0], next_ps_pack[1], next_ps_pack[2], next_ps_pack[3]);
@ -599,7 +601,7 @@ srs_error_t SrsPsStreamDemixer::on_ps_stream(char* ps_data, int ps_size, uint32_
}
if (complete_len != ps_size){
srs_trace("gb28181: client_id %s decode ps packet error (%#x/%u)! ps_size=%d complete=%d \n",
srs_trace("gb28181: client_id %s decode ps packet error (%#x/%u)! ps_size=%d complete=%d \n",
channel_id.c_str(), ssrc, timestamp, ps_size, complete_len);
}else if (hander && video_stream.length() && can_send_ps_av_packet()) {
if ((err = hander->on_rtp_video(&video_stream, video_pts)) != srs_success) {
@ -646,6 +648,7 @@ SrsGb28181Config::SrsGb28181Config(SrsConfDirective* c)
sip_ack_timeout = _srs_config->get_stream_caster_gb28181_ack_timeout(c);
sip_keepalive_timeout = _srs_config->get_stream_caster_gb28181_keepalive_timeout(c);
sip_invite_port_fixed = _srs_config->get_stream_caster_gb28181_sip_invite_port_fixed(c);
sip_query_catalog_interval = _srs_config->get_stream_caster_gb28181_sip_query_catalog_interval(c);
}
SrsGb28181Config::~SrsGb28181Config()
@ -653,7 +656,6 @@ SrsGb28181Config::~SrsGb28181Config()
}
//SrsGb28181RtmpMuxer gb28181 rtmp muxer, process ps stream to rtmp
SrsGb28181RtmpMuxer::SrsGb28181RtmpMuxer(SrsGb28181Manger* c, std::string id, bool a, bool k)
{
@ -675,7 +677,8 @@ SrsGb28181RtmpMuxer::SrsGb28181RtmpMuxer(SrsGb28181Manger* c, std::string id, bo
wait_ps_queue = srs_cond_new();
stream_idle_timeout = -1;
recv_stream_time = 0;
recv_rtp_stream_time = 0;
send_rtmp_stream_time = 0;
_rtmp_url = "";
@ -768,7 +771,7 @@ std::string SrsGb28181RtmpMuxer::rtmp_url()
srs_utime_t SrsGb28181RtmpMuxer::get_recv_stream_time()
{
return recv_stream_time;
return recv_rtp_stream_time;
}
@ -785,7 +788,8 @@ void SrsGb28181RtmpMuxer::destroy()
srs_error_t SrsGb28181RtmpMuxer::do_cycle()
{
srs_error_t err = srs_success;
recv_stream_time = srs_get_system_time();
recv_rtp_stream_time = srs_get_system_time();
send_rtmp_stream_time = srs_get_system_time();
//consume ps stream, and check status
while (true) {
@ -819,7 +823,7 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
}
srs_utime_t now = srs_get_system_time();
srs_utime_t duration = now - recv_stream_time;
srs_utime_t duration = now - recv_rtp_stream_time;
//if no RTP data is received within 2 seconds,
//the peer-port and peer-ip will be cleared and
@ -831,13 +835,28 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
channel->set_rtp_peer_port(0);
channel->set_rtp_peer_ip("");
}
SrsGb28181Config config = gb28181_manger->get_gb28181_config();
if (duration > config.rtp_idle_timeout){
srs_trace("gb28181: client id=%s, stream idle timeout, stop!!!", channel_id.c_str());
break;
}
//RTMP connection is about to timeout without receiving any data.,
//waiting for the next time there is data automatically connected
//it is related to the following two parameter settings of the rtmp server
//the publish 1st packet timeout in srs_utime_t
//publish_1stpkt_timeout default 20000ms
//the publish normal packet timeout in srs_utime_t
//publish_normal_timeout default 5000ms
duration = now - send_rtmp_stream_time;
bool will_timeout = duration > (5 * SRS_UTIME_SECONDS);
if (will_timeout && sdk){
srs_warn("gb28181: client id=%s RTMP connection is about to time out without receiving any data",
channel_id.c_str());
rtmp_close();
}
if (ps_queue.empty()){
srs_cond_timedwait(wait_ps_queue, 200 * SRS_UTIME_MILLISECONDS);
}else {
@ -862,7 +881,7 @@ void SrsGb28181RtmpMuxer::ps_packet_enqueue(SrsPsRtpPacket *pkt)
{
srs_assert(pkt);
recv_stream_time = srs_get_system_time();
recv_rtp_stream_time = srs_get_system_time();
//prevent consumers from being unable to process data
//and accumulating in the queue
@ -923,7 +942,7 @@ srs_error_t SrsGb28181RtmpMuxer::on_rtp_video(SrsSimpleStream *stream, int64_t f
uint32_t pts = (uint32_t)(fpts / 90);
srs_info("gb28181rtmpmuxer: on_rtp_video dts=%u", dts);
recv_stream_time = srs_get_system_time();
SrsBuffer *avs = new SrsBuffer(stream->bytes(), stream->length());
SrsAutoFree(SrsBuffer, avs);
@ -1010,8 +1029,6 @@ srs_error_t SrsGb28181RtmpMuxer::on_rtp_audio(SrsSimpleStream* stream, int64_t f
return srs_error_wrap(err, "jitter");
}
recv_stream_time = srs_get_system_time();
uint32_t dts = (uint32_t)(fdts / 90);
// send each frame.
@ -1167,6 +1184,8 @@ srs_error_t SrsGb28181RtmpMuxer::rtmp_write_packet(char type, uint32_t timestamp
SrsSharedPtrMessage* msg = NULL;
send_rtmp_stream_time = srs_get_system_time();
if ((err = srs_rtmp_create_msg(type, timestamp, data, size, sdk->sid(), &msg)) != srs_success) {
return srs_error_wrap(err, "create message");
}
@ -1197,8 +1216,9 @@ srs_error_t SrsGb28181RtmpMuxer::connect()
srs_utime_t cto = SRS_CONSTS_RTMP_TIMEOUT;
srs_utime_t sto = SRS_CONSTS_RTMP_PULSE;
sdk = new SrsSimpleRtmpClient(url, cto, sto);
srs_trace("gb28181: rtmp connect url=%s", url.c_str());
if ((err = sdk->connect()) != srs_success) {
close();
return srs_error_wrap(err, "connect %s failed, cto=%dms, sto=%dms.", url.c_str(), srsu2msi(cto), srsu2msi(sto));
@ -1368,8 +1388,8 @@ uint32_t SrsGb28181Manger::generate_ssrc(std::string id)
{
srand(uint(time(0)));
// TODO: SSRC rules can be customized,
//gb28281 live ssrc max value 0999999999(3B9AC9FF)
//gb28281 vod ssrc max value 1999999999(773593FF)
//gb28181 live ssrc max value 0999999999(3B9AC9FF)
//gb28181 vod ssrc max value 1999999999(773593FF)
uint8_t index = uint8_t(rand() % (0x0F - 0x01 + 1) + 0x01);
uint32_t ssrc = 0x2FFFF00 & (hash_code(id) << 8) | index;
//uint32_t ssrc = 0x00FFFFFF & (hash_code(id));
@ -1660,7 +1680,7 @@ uint32_t SrsGb28181Manger::delete_stream_channel(std::string id)
{
//notify the device to stop streaming
//if an internal sip service controlled channel
notify_sip_bye(id);
notify_sip_bye(id, id);
SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(id);
if (muxer){
@ -1673,7 +1693,7 @@ uint32_t SrsGb28181Manger::delete_stream_channel(std::string id)
}
uint32_t SrsGb28181Manger::queue_stream_channel(std::string id, SrsJsonArray* arr)
uint32_t SrsGb28181Manger::query_stream_channel(std::string id, SrsJsonArray* arr)
{
if (!id.empty()){
SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(id);
@ -1696,21 +1716,22 @@ uint32_t SrsGb28181Manger::queue_stream_channel(std::string id, SrsJsonArray* ar
return ERROR_SUCCESS;
}
uint32_t SrsGb28181Manger::notify_sip_invite(std::string id, std::string ip, int port, uint32_t ssrc)
uint32_t SrsGb28181Manger::notify_sip_invite(std::string id, std::string ip, int port, uint32_t ssrc, std::string chid)
{
if (!sip_service){
return ERROR_GB28181_SIP_NOT_RUN;
}
//if RTMP Muxer does not exist, you need to create
SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(id);
std::string key = id+"@"+chid;
SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(key);
if (!muxer){
//if there is an invalid parameter, the channel will be created automatically
if (ip.empty() || port == 0 || ssrc == 0){
//channel not exist
SrsGb28181StreamChannel channel;
channel.set_channel_id(id);
channel.set_channel_id(key);
int code = create_stream_channel(&channel);
if (code != ERROR_SUCCESS){
return code;
@ -1730,24 +1751,18 @@ uint32_t SrsGb28181Manger::notify_sip_invite(std::string id, std::string ip, int
SrsSipRequest req;
req.sip_auth_id = id;
return sip_service->send_invite(&req, ip, port, ssrc);
return sip_service->send_invite(&req, ip, port, ssrc, chid);
}
uint32_t SrsGb28181Manger::notify_sip_bye(std::string id)
uint32_t SrsGb28181Manger::notify_sip_bye(std::string id, std::string chid)
{
if (!sip_service){
return ERROR_GB28181_SIP_NOT_RUN;
}
SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(id);
if (muxer){
muxer->rtmp_close();
}
SrsSipRequest req;
req.sip_auth_id = id;
return sip_service->send_bye(&req);
return sip_service->send_bye(&req, chid);
}
uint32_t SrsGb28181Manger::notify_sip_raw_data(std::string id, std::string data)
@ -1772,3 +1787,23 @@ uint32_t SrsGb28181Manger::notify_sip_unregister(std::string id)
sip_service->remove_session(id);
return ERROR_SUCCESS;
}
uint32_t SrsGb28181Manger::notify_sip_query_catalog(std::string id)
{
if (!sip_service){
return ERROR_GB28181_SIP_NOT_RUN;
}
SrsSipRequest req;
req.sip_auth_id = id;
return sip_service->send_query_catalog(&req);
}
uint32_t SrsGb28181Manger::query_sip_session(std::string id, SrsJsonArray* arr)
{
if (!sip_service){
return ERROR_GB28181_SIP_NOT_RUN;
}
return sip_service->query_sip_session(id, arr);
}

14
trunk/src/app/srs_app_gb28181.hpp
Unescape Escape View File

@ -218,7 +218,8 @@ private:
SrsPithyPrint* pprint;
SrsGb28181StreamChannel *channel;
int stream_idle_timeout;
srs_utime_t recv_stream_time;
srs_utime_t recv_rtp_stream_time;
srs_utime_t send_rtmp_stream_time;
private:
std::string channel_id;
std::string _rtmp_url;
@ -313,6 +314,7 @@ public:
srs_utime_t sip_keepalive_timeout;
bool sip_auto_play;
bool sip_invite_port_fixed;
srs_utime_t sip_query_catalog_interval;
public:
SrsGb28181Config(SrsConfDirective* c);
@ -392,9 +394,7 @@ private:
std::map<uint32_t, SrsGb28181RtmpMuxer*> rtmpmuxers_ssrc;
std::map<std::string, SrsGb28181RtmpMuxer*> rtmpmuxers;
SrsCoroutineManager* manager;
SrsGb28181SipService* sip_service;
public:
SrsGb28181Manger(SrsConfDirective* c);
virtual ~SrsGb28181Manger();
@ -415,12 +415,14 @@ public:
//stream channel api
uint32_t create_stream_channel(SrsGb28181StreamChannel *channel);
uint32_t delete_stream_channel(std::string id);
uint32_t queue_stream_channel(std::string id, SrsJsonArray* arr);
uint32_t query_stream_channel(std::string id, SrsJsonArray* arr);
//sip api
uint32_t notify_sip_invite(std::string id, std::string ip, int port, uint32_t ssrc);
uint32_t notify_sip_bye(std::string id);
uint32_t notify_sip_invite(std::string id, std::string ip, int port, uint32_t ssrc, std::string chid);
uint32_t notify_sip_bye(std::string id, std::string chid);
uint32_t notify_sip_raw_data(std::string id, std::string data);
uint32_t notify_sip_unregister(std::string id);
uint32_t notify_sip_query_catalog(std::string id);
uint32_t query_sip_session(std::string id, SrsJsonArray* arr);
private:
void destroy();

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