mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 02:34:26 +08:00
整理startSendRtp相关接口，新增支持udp被动模式
startSendPassive接口指定is_udp为1时则为开启udp被动模式, 被动模式下需要对方先发数据给zlmediakit，然后才能回复rtp数据； 同时在被动模式指定src_port时，不再等待对方连接成功后再返回(而是直接返回成功和端口号，失败通知请监听on_send_rtp_stopped hook)。
This commit is contained in:
xia-chu
parent
fea7f3647e
commit
81aef25583
@ -159,9 +159,9 @@ API_EXPORT int API_CALL mk_media_source_seek_to(const mk_media_source ctx,uint32
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*/
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typedef void(API_CALL *on_mk_media_source_send_rtp_result)(void *user_data, uint16_t local_port, int err, const char *msg);
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//MediaSource::startSendRtp,请参考mk_media_start_send_rtp,注意ctx参数类型不一样
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API_EXPORT void API_CALL mk_media_source_start_send_rtp(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_source_send_rtp_result cb, void *user_data);
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API_EXPORT void API_CALL mk_media_source_start_send_rtp2(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_source_send_rtp_result cb, void *user_data, on_user_data_free user_data_free);
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// MediaSource::startSendRtp,请参考mk_media_start_send_rtp,注意ctx参数类型不一样
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API_EXPORT void API_CALL mk_media_source_start_send_rtp(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_source_send_rtp_result cb, void *user_data);
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API_EXPORT void API_CALL mk_media_source_start_send_rtp2(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_source_send_rtp_result cb, void *user_data, on_user_data_free user_data_free);
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//MediaSource::stopSendRtp，请参考mk_media_stop_send_rtp,注意ctx参数类型不一样
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API_EXPORT int API_CALL mk_media_source_stop_send_rtp(const mk_media_source ctx);
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@ -269,12 +269,12 @@ typedef on_mk_media_source_send_rtp_result on_mk_media_send_rtp_result;
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* @param dst_url 目标ip或域名
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* @param dst_port 目标端口
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* @param ssrc rtp的ssrc，10进制的字符串打印
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* @param is_udp 是否为udp
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* @param con_type 0: tcp主动，1：udp主动，2：tcp被动，3：udp被动
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* @param cb 启动成功或失败回调
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* @param user_data 回调用户指针
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*/
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API_EXPORT void API_CALL mk_media_start_send_rtp(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_send_rtp_result cb, void *user_data);
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API_EXPORT void API_CALL mk_media_start_send_rtp2(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_send_rtp_result cb, void *user_data, on_user_data_free user_data_free);
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API_EXPORT void API_CALL mk_media_start_send_rtp(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_send_rtp_result cb, void *user_data);
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API_EXPORT void API_CALL mk_media_start_send_rtp2(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_send_rtp_result cb, void *user_data, on_user_data_free user_data_free);
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/**
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* 停止某路或全部ps-rtp发送，此api线程安全
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@ -295,11 +295,11 @@ API_EXPORT int API_CALL mk_media_source_seek_to(const mk_media_source ctx,uint32
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MediaSource *src = (MediaSource *)ctx;
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return src->seekTo(stamp);
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}
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API_EXPORT void API_CALL mk_media_source_start_send_rtp(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_source_send_rtp_result cb, void *user_data) {
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mk_media_source_start_send_rtp2(ctx, dst_url, dst_port, ssrc, is_udp, cb, user_data, nullptr);
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API_EXPORT void API_CALL mk_media_source_start_send_rtp(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_source_send_rtp_result cb, void *user_data) {
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mk_media_source_start_send_rtp2(ctx, dst_url, dst_port, ssrc, con_type, cb, user_data, nullptr);
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}
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API_EXPORT void API_CALL mk_media_source_start_send_rtp2(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_source_send_rtp_result cb, void *user_data, on_user_data_free user_data_free){
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API_EXPORT void API_CALL mk_media_source_start_send_rtp2(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_source_send_rtp_result cb, void *user_data, on_user_data_free user_data_free){
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assert(ctx && dst_url && ssrc);
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MediaSource *src = (MediaSource *)ctx;
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@ -307,7 +307,7 @@ API_EXPORT void API_CALL mk_media_source_start_send_rtp2(const mk_media_source c
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args.dst_url = dst_url;
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args.dst_port = dst_port;
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args.ssrc = ssrc;
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args.is_udp = is_udp;
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args.con_type = (mediakit::MediaSourceEvent::SendRtpArgs::ConType)con_type;
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std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
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src->startSendRtp(args, [cb, ptr](uint16_t local_port, const SockException &ex){
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@ -281,11 +281,11 @@ API_EXPORT int API_CALL mk_media_input_audio(mk_media ctx, const void *data, int
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return (*obj)->getChannel()->inputAudio((const char*)data, len, dts);
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}
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API_EXPORT void API_CALL mk_media_start_send_rtp(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_send_rtp_result cb, void *user_data) {
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mk_media_start_send_rtp2(ctx, dst_url, dst_port, ssrc, is_udp, cb, user_data, nullptr);
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API_EXPORT void API_CALL mk_media_start_send_rtp(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_send_rtp_result cb, void *user_data) {
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mk_media_start_send_rtp2(ctx, dst_url, dst_port, ssrc, con_type, cb, user_data, nullptr);
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}
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API_EXPORT void API_CALL mk_media_start_send_rtp2(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_send_rtp_result cb, void *user_data,
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API_EXPORT void API_CALL mk_media_start_send_rtp2(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int con_type, on_mk_media_send_rtp_result cb, void *user_data,
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on_user_data_free user_data_free) {
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assert(ctx && dst_url && ssrc);
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MediaHelper::Ptr* obj = (MediaHelper::Ptr*) ctx;
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@ -294,7 +294,7 @@ API_EXPORT void API_CALL mk_media_start_send_rtp2(mk_media ctx, const char *dst_
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args.dst_url = dst_url;
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args.dst_port = dst_port;
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args.ssrc = ssrc;
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args.is_udp = is_udp;
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args.con_type = (mediakit::MediaSourceEvent::SendRtpArgs::ConType)con_type;
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// sender参数无用
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auto ref = *obj;
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@ -1881,7 +1881,7 @@
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"response": []
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},
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{
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"name": "开始发送rtp(startSendRtp)",
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"name": "开始active模式发送rtp(startSendRtp)",
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"request": {
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"method": "GET",
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"header": [],
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@ -1940,7 +1940,7 @@
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{
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"key": "is_udp",
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"value": "0",
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"description": "是否为udp模式,否则为tcp模式"
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"description": "1:udp active模式, 0:tcp active模式"
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},
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{
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"key": "src_port",
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@ -1955,9 +1955,9 @@
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"disabled": true
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},
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{
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"key": "use_ps",
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"key": "type",
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"value": "1",
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"description": "rtp打包采用ps还是es模式，默认采用ps模式，该参数非必选参数",
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"description": "rtp打包模式，0:es, 1: ps, 2: ts",
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"disabled": true
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},
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{
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@ -1990,7 +1990,7 @@
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"response": []
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},
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{
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"name": "开始tcp passive被动发送rtp(startSendRtpPassive)",
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"name": "开始passive模式发送rtp(startSendRtpPassive)",
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"request": {
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"method": "GET",
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"header": [],
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@ -2030,6 +2030,12 @@
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"value": "1",
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"description": "rtp推流的ssrc，ssrc不同时，可以推流到多个上级服务器"
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},
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{
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"key": "is_udp",
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"value": "0",
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"disabled": true,
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"description": "1:udp passive模式, 0:tcp passive模式"
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},
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{
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"key": "src_port",
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"value": "0",
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@ -2043,9 +2049,9 @@
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"disabled": true
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},
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{
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"key": "use_ps",
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"key": "type",
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"value": "1",
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"description": "rtp打包采用ps还是es模式，默认采用ps模式，该参数非必选参数",
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"description": "rtp打包模式，0:es, 1: ps, 2: ts",
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"disabled": true
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},
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{
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@ -1377,10 +1377,7 @@ void installWebApi() {
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}
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});
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api_regist("/index/api/startSendRtp",[](API_ARGS_MAP_ASYNC){
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CHECK_SECRET();
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CHECK_ARGS("vhost", "app", "stream", "ssrc", "dst_url", "dst_port", "is_udp");
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static auto start_send_rtp = [] (bool passive, API_ARGS_MAP_ASYNC) {
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auto src = MediaSource::find(allArgs["vhost"], allArgs["app"], allArgs["stream"], allArgs["from_mp4"].as<int>());
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if (!src) {
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throw ApiRetException("can not find the source stream", API::NotFound);
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@ -1391,30 +1388,50 @@ void installWebApi() {
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type = allArgs["use_ps"].as<int>();
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}
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MediaSourceEvent::SendRtpArgs args;
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args.passive = false;
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if (passive) {
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args.con_type = allArgs["is_udp"].as<bool>() ? mediakit::MediaSourceEvent::SendRtpArgs::kUdpPassive : mediakit::MediaSourceEvent::SendRtpArgs::kTcpPassive;
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} else {
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args.con_type = allArgs["is_udp"].as<bool>() ? mediakit::MediaSourceEvent::SendRtpArgs::kUdpActive : mediakit::MediaSourceEvent::SendRtpArgs::kTcpActive;
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}
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args.dst_url = allArgs["dst_url"];
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args.dst_port = allArgs["dst_port"];
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args.ssrc_multi_send = allArgs["ssrc_multi_send"].empty() ? false : allArgs["ssrc_multi_send"].as<bool>();
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args.ssrc = allArgs["ssrc"];
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args.is_udp = allArgs["is_udp"];
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args.src_port = allArgs["src_port"];
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args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
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args.type = (MediaSourceEvent::SendRtpArgs::Type)type;
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args.data_type = (MediaSourceEvent::SendRtpArgs::DataType)type;
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args.only_audio = allArgs["only_audio"].as<bool>();
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args.udp_rtcp_timeout = allArgs["udp_rtcp_timeout"];
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args.recv_stream_id = allArgs["recv_stream_id"];
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TraceL << "startSendRtp, pt " << int(args.pt) << " rtp type " << type << " audio " << args.only_audio;
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args.close_delay_ms = allArgs["close_delay_ms"];
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src->getOwnerPoller()->async([=]() mutable {
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src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
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if (ex) {
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val["code"] = API::OtherFailed;
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val["msg"] = ex.what();
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}
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val["local_port"] = local_port;
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try {
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src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
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if (ex) {
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val["code"] = API::OtherFailed;
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val["msg"] = ex.what();
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}
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val["local_port"] = local_port;
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invoker(200, headerOut, val.toStyledString());
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});
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} catch (std::exception &ex) {
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val["code"] = API::Exception;
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val["msg"] = ex.what();
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invoker(200, headerOut, val.toStyledString());
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});
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}
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});
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};
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api_regist("/index/api/startSendRtp",[](API_ARGS_MAP_ASYNC){
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CHECK_SECRET();
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CHECK_ARGS("vhost", "app", "stream", "ssrc", "dst_url", "dst_port", "is_udp");
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start_send_rtp(false, API_ARGS_VALUE, invoker);
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});
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api_regist("/index/api/startSendRtpPassive",[](API_ARGS_MAP_ASYNC){
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CHECK_SECRET();
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CHECK_ARGS("vhost", "app", "stream", "ssrc");
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start_send_rtp(true, API_ARGS_VALUE, invoker);
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});
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api_regist("/index/api/listRtpSender",[](API_ARGS_MAP_ASYNC){
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@ -1437,45 +1454,6 @@ void installWebApi() {
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});
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});
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api_regist("/index/api/startSendRtpPassive",[](API_ARGS_MAP_ASYNC){
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CHECK_SECRET();
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CHECK_ARGS("vhost", "app", "stream", "ssrc");
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auto src = MediaSource::find(allArgs["vhost"], allArgs["app"], allArgs["stream"], allArgs["from_mp4"].as<int>());
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if (!src) {
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throw ApiRetException("can not find the source stream", API::NotFound);
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}
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auto type = allArgs["type"].empty() ? (int)MediaSourceEvent::SendRtpArgs::kRtpPS : allArgs["type"].as<int>();
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if (!allArgs["use_ps"].empty()) {
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// 兼容之前的use_ps参数
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type = allArgs["use_ps"].as<int>();
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}
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MediaSourceEvent::SendRtpArgs args;
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args.passive = true;
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args.ssrc = allArgs["ssrc"];
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args.is_udp = false;
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args.src_port = allArgs["src_port"];
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args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
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args.type = (MediaSourceEvent::SendRtpArgs::Type)type;
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args.only_audio = allArgs["only_audio"].as<bool>();
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args.recv_stream_id = allArgs["recv_stream_id"];
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//tcp被动服务器等待链接超时时间
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args.tcp_passive_close_delay_ms = allArgs["close_delay_ms"];
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TraceL << "startSendRtpPassive, pt " << int(args.pt) << " rtp type " << type << " audio " << args.only_audio;
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src->getOwnerPoller()->async([=]() mutable {
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src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
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if (ex) {
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val["code"] = API::OtherFailed;
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val["msg"] = ex.what();
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}
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val["local_port"] = local_port;
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invoker(200, headerOut, val.toStyledString());
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});
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});
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});
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api_regist("/index/api/stopSendRtp",[](API_ARGS_MAP_ASYNC){
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CHECK_SECRET();
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CHECK_ARGS("vhost", "app", "stream");
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@ -96,18 +96,29 @@ public:
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class SendRtpArgs {
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public:
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enum Type { kRtpRAW = 0, kRtpPS = 1, kRtpTS = 2 };
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// 是否采用udp方式发送rtp
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bool is_udp = true;
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enum DataType {
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kRtpES = 0, // 发送ES流
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kRtpPS = 1, // 发送PS流
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kRtpTS = 2 // 发送TS流
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};
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enum ConType {
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kTcpActive = 0, // tcp主动模式，tcp客户端主动连接对方并发送rtp
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kUdpActive = 1, // udp主动模式，主动发送数据给对方
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kTcpPassive = 2, // tcp被动模式，tcp服务器，等待对方连接并回复rtp
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kUdpPassive = 3 // udp被动方式，等待对方发送nat打洞包，然后回复rtp至打洞包源地址
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};
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// rtp类型
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Type type = kRtpPS;
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//发送es流时指定是否只发送纯音频流
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DataType data_type = kRtpPS;
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// 连接类型
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ConType con_type = kUdpActive;
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// 发送es流时指定是否只发送纯音频流
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bool only_audio = false;
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//tcp被动方式
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bool passive = false;
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// rtp payload type
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uint8_t pt = 96;
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//是否支持同ssrc多服务器发送
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// 是否支持同ssrc多服务器发送
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bool ssrc_multi_send = false;
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// 指定rtp ssrc
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std::string ssrc;
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@ -118,16 +129,16 @@ public:
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// 发送目标主机地址，可以是ip或域名
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std::string dst_url;
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//udp发送时，是否开启rr rtcp接收超时判断
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// udp发送时，是否开启rr rtcp接收超时判断
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bool udp_rtcp_timeout = false;
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//tcp被动发送服务器延时关闭事件，单位毫秒；设置为0时，则使用默认值5000ms
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uint32_t tcp_passive_close_delay_ms = 0;
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//udp 发送时，rr rtcp包接收超时时间，单位毫秒
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// passive被动、tcp主动模式超时时间
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uint32_t close_delay_ms = 0;
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// udp 发送时，rr rtcp包接收超时时间，单位毫秒
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uint32_t rtcp_timeout_ms = 30 * 1000;
|
||||
//udp 发送时，发送sr rtcp包间隔，单位毫秒
|
||||
// udp 发送时，发送sr rtcp包间隔，单位毫秒
|
||||
uint32_t rtcp_send_interval_ms = 5 * 1000;
|
||||
|
||||
//发送rtp同时接收，一般用于双向语言对讲, 如果不为空，说明开启接收
|
||||
// 发送rtp同时接收，一般用于双向语言对讲, 如果不为空，说明开启接收
|
||||
std::string recv_stream_id;
|
||||
};
|
||||
|
||||
|
||||
@ -386,6 +386,17 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &sender, const MediaSourceE
|
||||
|
||||
weak_ptr<MultiMediaSourceMuxer> weak_self = shared_from_this();
|
||||
|
||||
rtp_sender->setOnClose([weak_self, ssrc](const toolkit::SockException &ex) {
|
||||
if (auto strong_self = weak_self.lock()) {
|
||||
// 可能归属线程发生变更
|
||||
strong_self->getOwnerPoller(MediaSource::NullMediaSource())->async([=]() {
|
||||
WarnL << "stream:" << strong_self->shortUrl() << " stop send rtp:" << ssrc << ", reason:" << ex;
|
||||
strong_self->_rtp_sender.erase(ssrc);
|
||||
NOTICE_EMIT(BroadcastSendRtpStoppedArgs, Broadcast::kBroadcastSendRtpStopped, *strong_self, ssrc, ex);
|
||||
});
|
||||
}
|
||||
});
|
||||
|
||||
rtp_sender->startSend(args, [ssrc,ssrc_multi_send, weak_self, rtp_sender, cb, tracks, ring, poller](uint16_t local_port, const SockException &ex) mutable {
|
||||
cb(local_port, ex);
|
||||
auto strong_self = weak_self.lock();
|
||||
@ -397,16 +408,6 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &sender, const MediaSourceE
|
||||
rtp_sender->addTrack(track);
|
||||
}
|
||||
rtp_sender->addTrackCompleted();
|
||||
rtp_sender->setOnClose([weak_self, ssrc](const toolkit::SockException &ex) {
|
||||
if (auto strong_self = weak_self.lock()) {
|
||||
// 可能归属线程发生变更
|
||||
strong_self->getOwnerPoller(MediaSource::NullMediaSource())->async([=]() {
|
||||
WarnL << "stream:" << strong_self->shortUrl() << " stop send rtp:" << ssrc << ", reason:" << ex;
|
||||
strong_self->_rtp_sender.erase(ssrc);
|
||||
NOTICE_EMIT(BroadcastSendRtpStoppedArgs, Broadcast::kBroadcastSendRtpStopped, *strong_self, ssrc, ex);
|
||||
});
|
||||
}
|
||||
});
|
||||
|
||||
auto reader = ring->attach(poller);
|
||||
reader->setReadCB([rtp_sender](const Frame::Ptr &frame) {
|
||||
|
||||
@ -40,86 +40,99 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
|
||||
if (!_interface) {
|
||||
//重连时不重新创建对象
|
||||
auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
|
||||
switch (args.type) {
|
||||
switch (args.data_type) {
|
||||
case MediaSourceEvent::SendRtpArgs::kRtpPS: _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt, true); break;
|
||||
case MediaSourceEvent::SendRtpArgs::kRtpTS: _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt, false); break;
|
||||
case MediaSourceEvent::SendRtpArgs::kRtpRAW: _interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio); break;
|
||||
default: CHECK(0, "invalid rtp type:" + to_string(args.type)); break;
|
||||
case MediaSourceEvent::SendRtpArgs::kRtpES: _interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio); break;
|
||||
default: CHECK(0, "invalid rtp type: " + to_string(args.data_type)); break;
|
||||
}
|
||||
}
|
||||
|
||||
auto delay_ms = _args.close_delay_ms ? _args.close_delay_ms : 5000;
|
||||
weak_ptr<RtpSender> weak_self = shared_from_this();
|
||||
if (args.passive) {
|
||||
// tcp被动发流模式
|
||||
_args.is_udp = false;
|
||||
// 默认等待链接
|
||||
bool is_wait = true;
|
||||
try {
|
||||
auto tcp_listener = Socket::createSocket(_poller, false);
|
||||
if (args.src_port) {
|
||||
//指定端口
|
||||
if (!tcp_listener->listen(args.src_port)) {
|
||||
throw std::invalid_argument(StrPrinter << "open tcp passive server failed on port:" << args.src_port
|
||||
<< ", err:" << get_uv_errmsg(true));
|
||||
}
|
||||
is_wait = true;
|
||||
} else {
|
||||
auto pr = std::make_pair(tcp_listener, Socket::createSocket(_poller, false));
|
||||
//从端口池获取随机端口
|
||||
makeSockPair(pr, "::", false, false);
|
||||
// 随机端口不等待，保证调用者可以知道端口
|
||||
is_wait = false;
|
||||
if (args.con_type == MediaSourceEvent::SendRtpArgs::kTcpPassive) {
|
||||
auto tcp_listener = Socket::createSocket(_poller, false);
|
||||
if (args.src_port) {
|
||||
// 指定端口
|
||||
if (!tcp_listener->listen(args.src_port)) {
|
||||
throw std::invalid_argument(StrPrinter << "open tcp passive server failed on port: " << args.src_port << ", err: " << get_uv_errmsg(true));
|
||||
}
|
||||
// tcp服务器默认开启5秒
|
||||
auto delay = _args.tcp_passive_close_delay_ms ? _args.tcp_passive_close_delay_ms : 5000;
|
||||
auto delay_task = _poller->doDelayTask(delay, [tcp_listener, cb, is_wait]() mutable {
|
||||
if (is_wait) {
|
||||
cb(0, SockException(Err_timeout, "wait tcp connection timeout"));
|
||||
}
|
||||
tcp_listener = nullptr;
|
||||
return 0;
|
||||
});
|
||||
tcp_listener->setOnAccept([weak_self, cb, delay_task,is_wait](Socket::Ptr &sock, std::shared_ptr<void> &complete) {
|
||||
auto strong_self = weak_self.lock();
|
||||
if (!strong_self) {
|
||||
return;
|
||||
}
|
||||
//立即关闭tcp服务器
|
||||
delay_task->cancel();
|
||||
strong_self->_socket_rtp = sock;
|
||||
strong_self->onConnect();
|
||||
if (is_wait) {
|
||||
cb(sock->get_local_port(), SockException());
|
||||
}
|
||||
InfoL << "accept connection from:" << sock->get_peer_ip() << ":" << sock->get_peer_port();
|
||||
});
|
||||
InfoL << "start tcp passive server on:" << tcp_listener->get_local_port();
|
||||
if (!is_wait) {
|
||||
// 随机端口马上返回端口，保证调用者知道端口
|
||||
cb(tcp_listener->get_local_port(), SockException());
|
||||
}
|
||||
} catch (std::exception &ex) {
|
||||
cb(0, SockException(Err_other, ex.what()));
|
||||
return;
|
||||
} else {
|
||||
auto pr = std::make_pair(tcp_listener, Socket::createSocket(_poller, false));
|
||||
// 从端口池获取随机端口
|
||||
makeSockPair(pr, "::", true, false);
|
||||
}
|
||||
return;
|
||||
}
|
||||
if (args.is_udp) {
|
||||
// 定时器持有tcp_listener，保证超时时间内保持监听
|
||||
auto delay_task = _poller->doDelayTask(delay_ms, [weak_self, tcp_listener]() mutable {
|
||||
// 防止循环引用
|
||||
tcp_listener = nullptr;
|
||||
if (auto strong_self = weak_self.lock()) {
|
||||
strong_self->onClose(SockException(Err_timeout, "wait tcp connection timeout"));
|
||||
}
|
||||
return 0;
|
||||
});
|
||||
tcp_listener->setOnAccept([weak_self, delay_task](Socket::Ptr &sock, std::shared_ptr<void> &complete) {
|
||||
auto strong_self = weak_self.lock();
|
||||
if (!strong_self) {
|
||||
return;
|
||||
}
|
||||
delay_task->cancel();
|
||||
strong_self->_socket_rtp = sock;
|
||||
strong_self->onConnect();
|
||||
InfoL << "accept tcp connection from: " << sock->get_peer_ip() << ":" << sock->get_peer_port();
|
||||
});
|
||||
InfoL << "start tcp passive server on: " << tcp_listener->get_local_port();
|
||||
cb(tcp_listener->get_local_port(), SockException());
|
||||
|
||||
} else if (args.con_type == MediaSourceEvent::SendRtpArgs::kUdpPassive) {
|
||||
if (args.src_port) {
|
||||
// 指定端口
|
||||
if (!_socket_rtp->bindUdpSock(args.src_port, "::", true)) {
|
||||
throw std::invalid_argument(StrPrinter << "open udp passive server failed on port: " << args.src_port << ", err: " << get_uv_errmsg(true));
|
||||
}
|
||||
} else {
|
||||
auto pr = std::make_pair(_socket_rtp, Socket::createSocket(_poller, false));
|
||||
// 从端口池获取随机端口
|
||||
makeSockPair(pr, "::", true, true);
|
||||
}
|
||||
auto delay_task = _poller->doDelayTask(delay_ms, [weak_self]() mutable {
|
||||
if (auto strong_self = weak_self.lock()) {
|
||||
// 关闭端口
|
||||
strong_self->_socket_rtp->closeSock();
|
||||
strong_self->onClose(SockException(Err_timeout, "wait udp connection timeout"));
|
||||
}
|
||||
return 0;
|
||||
});
|
||||
_socket_rtp->setOnRead([weak_self, delay_task](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) {
|
||||
auto strong_self = weak_self.lock();
|
||||
if (!strong_self) {
|
||||
return;
|
||||
}
|
||||
delay_task->cancel();
|
||||
strong_self->_socket_rtp->bindPeerAddr(addr, addr_len, true);
|
||||
// 异步执行onConnect，防止在OnRead回调中调用setOnRead
|
||||
strong_self->_poller->async([strong_self]() { strong_self->onConnect(); }, false);
|
||||
InfoL << "accept udp connection from: " << strong_self->_socket_rtp->get_peer_ip() << ":" << strong_self->_socket_rtp->get_peer_port();
|
||||
});
|
||||
InfoL << "start udp passive server on: " << _socket_rtp->get_local_port();
|
||||
cb(_socket_rtp->get_local_port(), SockException());
|
||||
|
||||
} else if (args.con_type == MediaSourceEvent::SendRtpArgs::kUdpActive) {
|
||||
auto poller = _poller;
|
||||
WorkThreadPool::Instance().getPoller()->async([cb, args, weak_self, poller]() {
|
||||
struct sockaddr_storage addr;
|
||||
//切换线程目的是为了dns解析放在后台线程执行
|
||||
// 切换线程目的是为了dns解析放在后台线程执行
|
||||
if (!SockUtil::getDomainIP(args.dst_url.data(), args.dst_port, addr, AF_INET, SOCK_DGRAM, IPPROTO_UDP)) {
|
||||
poller->async([args, cb]() {
|
||||
//切回自己的线程
|
||||
cb(0, SockException(Err_dns, StrPrinter << "dns解析域名失败:" << args.dst_url));
|
||||
// 切回自己的线程
|
||||
cb(0, SockException(Err_dns, StrPrinter << "dns resolution failed: " << args.dst_url));
|
||||
});
|
||||
return;
|
||||
}
|
||||
|
||||
//dns解析成功
|
||||
// dns解析成功
|
||||
poller->async([args, addr, weak_self, cb]() {
|
||||
//切回自己的线程
|
||||
// 切回自己的线程
|
||||
auto strong_self = weak_self.lock();
|
||||
if (!strong_self) {
|
||||
return;
|
||||
@ -127,15 +140,14 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
|
||||
string ifr_ip = addr.ss_family == AF_INET ? "0.0.0.0" : "::";
|
||||
try {
|
||||
if (args.src_port) {
|
||||
//指定端口
|
||||
if (!strong_self->_socket_rtp->bindUdpSock(args.src_port, ifr_ip)) {
|
||||
throw std::invalid_argument(StrPrinter << "bindUdpSock failed on port:" << args.src_port
|
||||
<< ", err:" << get_uv_errmsg(true));
|
||||
// 指定端口
|
||||
if (!strong_self->_socket_rtp->bindUdpSock(args.src_port, ifr_ip, true)) {
|
||||
throw std::invalid_argument(StrPrinter << "open udp active client failed on port: " << args.src_port << ", err: " << get_uv_errmsg(true));
|
||||
}
|
||||
} else {
|
||||
auto pr = std::make_pair(strong_self->_socket_rtp, Socket::createSocket(strong_self->_poller, false));
|
||||
//从端口池获取随机端口
|
||||
makeSockPair(pr, ifr_ip, true);
|
||||
// 从端口池获取随机端口
|
||||
makeSockPair(pr, ifr_ip, true, true);
|
||||
}
|
||||
} catch (std::exception &ex) {
|
||||
cb(0, SockException(Err_other, ex.what()));
|
||||
@ -146,19 +158,24 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
|
||||
cb(strong_self->_socket_rtp->get_local_port(), SockException());
|
||||
});
|
||||
});
|
||||
} else {
|
||||
_socket_rtp->connect(args.dst_url, args.dst_port, [cb, weak_self](const SockException &err) {
|
||||
InfoL << "start udp active send rtp to: " << args.dst_url << ":" << args.dst_port;
|
||||
|
||||
} else if (args.con_type == MediaSourceEvent::SendRtpArgs::kTcpActive) {
|
||||
_socket_rtp->connect(args.dst_url, args.dst_port,[cb, weak_self](const SockException &err) {
|
||||
auto strong_self = weak_self.lock();
|
||||
if (strong_self) {
|
||||
if (!err) {
|
||||
//tcp连接成功
|
||||
// tcp连接成功
|
||||
strong_self->onConnect();
|
||||
}
|
||||
cb(strong_self->_socket_rtp->get_local_port(), err);
|
||||
} else {
|
||||
cb(0, err);
|
||||
}
|
||||
}, 5.0F, "::", args.src_port);
|
||||
}, delay_ms / 1000.0, "::", args.src_port);
|
||||
InfoL << "start tcp active send rtp to: " << args.dst_url << ":" << args.dst_port;
|
||||
} else {
|
||||
CHECK(0, "invalid con type");
|
||||
}
|
||||
}
|
||||
|
||||
@ -168,8 +185,8 @@ void RtpSender::createRtcpSocket() {
|
||||
}
|
||||
_socket_rtcp = Socket::createSocket(_socket_rtp->getPoller(), false);
|
||||
//rtcp端口使用户rtp端口+1
|
||||
if(!_socket_rtcp->bindUdpSock(_socket_rtp->get_local_port() + 1, _socket_rtp->get_local_ip(), false)){
|
||||
WarnL << "bind rtcp udp socket failed:" << get_uv_errmsg(true);
|
||||
if(!_socket_rtcp->bindUdpSock(_socket_rtp->get_local_port() + 1, _socket_rtp->get_local_ip(), true)){
|
||||
WarnL << "bind rtcp udp socket failed: " << get_uv_errmsg(true);
|
||||
_socket_rtcp = nullptr;
|
||||
return;
|
||||
}
|
||||
@ -180,12 +197,18 @@ void RtpSender::createRtcpSocket() {
|
||||
|
||||
_rtcp_context = std::make_shared<RtcpContextForSend>();
|
||||
weak_ptr<RtpSender> weak_self = shared_from_this();
|
||||
_socket_rtcp->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *, int) {
|
||||
bool bind_addr = false;
|
||||
_socket_rtcp->setOnRead([weak_self, bind_addr](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
|
||||
//接收receive report rtcp
|
||||
auto strong_self = weak_self.lock();
|
||||
if (!strong_self) {
|
||||
return;
|
||||
}
|
||||
if (!bind_addr) {
|
||||
// 收到对方rtcp打洞包后，再回复rtcp
|
||||
bind_addr = true;
|
||||
strong_self->_socket_rtcp->bindPeerAddr(addr, addr_len, true);
|
||||
}
|
||||
auto rtcp_arr = RtcpHeader::loadFromBytes(buf->data(), buf->size());
|
||||
for (auto &rtcp : rtcp_arr) {
|
||||
strong_self->onRecvRtcp(rtcp);
|
||||
@ -199,19 +222,19 @@ void RtpSender::onRecvRtcp(RtcpHeader *rtcp) {
|
||||
_rtcp_recv_ticker.resetTime();
|
||||
}
|
||||
|
||||
//连接建立成功事件
|
||||
void RtpSender::onConnect(){
|
||||
// 连接建立成功事件
|
||||
void RtpSender::onConnect() {
|
||||
_is_connect = true;
|
||||
//加大发送缓存,防止udp丢包之类的问题
|
||||
// 加大发送缓存,防止udp丢包之类的问题
|
||||
SockUtil::setSendBuf(_socket_rtp->rawFD(), 4 * 1024 * 1024);
|
||||
if (!_args.is_udp) {
|
||||
//关闭tcp no_delay并开启MSG_MORE, 提高发送性能
|
||||
if (_args.con_type == MediaSourceEvent::SendRtpArgs::kTcpActive || _args.con_type == MediaSourceEvent::SendRtpArgs::kTcpPassive) {
|
||||
// 关闭tcp no_delay并开启MSG_MORE, 提高发送性能
|
||||
SockUtil::setNoDelay(_socket_rtp->rawFD(), false);
|
||||
_socket_rtp->setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
|
||||
} else if (_args.udp_rtcp_timeout) {
|
||||
createRtcpSocket();
|
||||
}
|
||||
//连接建立成功事件
|
||||
// 连接建立成功事件
|
||||
weak_ptr<RtpSender> weak_self = shared_from_this();
|
||||
if (!_args.recv_stream_id.empty()) {
|
||||
mINI ini;
|
||||
@ -226,11 +249,13 @@ void RtpSender::onConnect(){
|
||||
}
|
||||
try {
|
||||
strong_self->_rtp_session->onRecv(buf);
|
||||
} catch (std::exception &ex){
|
||||
} catch (std::exception &ex) {
|
||||
SockException err(toolkit::Err_shutdown, ex.what());
|
||||
strong_self->_rtp_session->shutdown(err);
|
||||
}
|
||||
});
|
||||
} else {
|
||||
_socket_rtp->setOnRead(nullptr);
|
||||
}
|
||||
_socket_rtp->setOnErr([weak_self](const SockException &err) {
|
||||
auto strong_self = weak_self.lock();
|
||||
@ -238,12 +263,10 @@ void RtpSender::onConnect(){
|
||||
strong_self->onErr(err);
|
||||
}
|
||||
});
|
||||
//获取本地端口，断开重连后确保端口不变
|
||||
_args.src_port = _socket_rtp->get_local_port();
|
||||
InfoL << "开始发送 rtp:" << _socket_rtp->get_peer_ip() << ":" << _socket_rtp->get_peer_port() << ", 是否为udp方式:" << _args.is_udp;
|
||||
InfoL << "startSend rtp success: " << _socket_rtp->get_peer_ip() << ":" << _socket_rtp->get_peer_port() << ", data_type: " << _args.data_type << ", con_type: " << _args.con_type;
|
||||
}
|
||||
|
||||
bool RtpSender::addTrack(const Track::Ptr &track){
|
||||
bool RtpSender::addTrack(const Track::Ptr &track) {
|
||||
if (_args.only_audio && track->getTrackType() == TrackVideo) {
|
||||
// 如果只发送音频则忽略视频
|
||||
return false;
|
||||
@ -251,11 +274,11 @@ bool RtpSender::addTrack(const Track::Ptr &track){
|
||||
return _interface->addTrack(track);
|
||||
}
|
||||
|
||||
void RtpSender::addTrackCompleted(){
|
||||
void RtpSender::addTrackCompleted() {
|
||||
_interface->addTrackCompleted();
|
||||
}
|
||||
|
||||
void RtpSender::resetTracks(){
|
||||
void RtpSender::resetTracks() {
|
||||
_interface->resetTracks();
|
||||
}
|
||||
|
||||
@ -265,13 +288,12 @@ void RtpSender::flush() {
|
||||
}
|
||||
}
|
||||
|
||||
//此函数在其他线程执行
|
||||
bool RtpSender::inputFrame(const Frame::Ptr &frame) {
|
||||
if (_args.only_audio && frame->getTrackType() == TrackVideo) {
|
||||
// 如果只发送音频则忽略视频
|
||||
return false;
|
||||
}
|
||||
//连接成功后才做实质操作(节省cpu资源)
|
||||
// 连接成功后才做实质操作(节省cpu资源)
|
||||
return _is_connect ? _interface->inputFrame(frame) : false;
|
||||
}
|
||||
|
||||
@ -283,20 +305,20 @@ void RtpSender::onSendRtpUdp(const toolkit::Buffer::Ptr &buf, bool check) {
|
||||
_rtcp_context->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, 90000 /*not used*/, rtp->size());
|
||||
|
||||
if (!check) {
|
||||
//减少判断次数
|
||||
// 减少判断次数
|
||||
return;
|
||||
}
|
||||
//每5秒发送一次rtcp
|
||||
// 每5秒发送一次rtcp
|
||||
if (_rtcp_send_ticker.elapsedTime() > _args.rtcp_send_interval_ms) {
|
||||
_rtcp_send_ticker.resetTime();
|
||||
//rtcp ssrc为rtp ssrc + 1
|
||||
auto sr = _rtcp_context->createRtcpSR(atoi(_args.ssrc.data()) + 1);
|
||||
//send sender report rtcp
|
||||
_socket_rtcp->send(sr);
|
||||
// rtcp ssrc为rtp ssrc + 1
|
||||
auto sr = _rtcp_context->createRtcpSR(atoi(_args.ssrc.data()) + 1);
|
||||
// send sender report rtcp
|
||||
_socket_rtcp->send(sr);
|
||||
}
|
||||
|
||||
if (_rtcp_recv_ticker.elapsedTime() > _args.rtcp_timeout_ms) {
|
||||
//接收rr rtcp超时
|
||||
// 接收rr rtcp超时
|
||||
WarnL << "recv rr rtcp timeout";
|
||||
_rtcp_recv_ticker.resetTime();
|
||||
onClose(SockException(Err_timeout, "recv rr rtcp timeout"));
|
||||
@ -306,28 +328,36 @@ void RtpSender::onSendRtpUdp(const toolkit::Buffer::Ptr &buf, bool check) {
|
||||
void RtpSender::onClose(const SockException &ex) {
|
||||
auto cb = _on_close;
|
||||
if (cb) {
|
||||
//在下次循环时触发onClose，原因是防止遍历map时删除元素
|
||||
// 在下次循环时触发onClose，原因是防止遍历map时删除元素
|
||||
_poller->async([cb, ex]() { cb(ex); }, false);
|
||||
}
|
||||
}
|
||||
|
||||
//此函数在其他线程执行
|
||||
void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr> > rtp_list) {
|
||||
if(!_is_connect){
|
||||
//连接成功后才能发送数据
|
||||
// 此函数在其他线程执行
|
||||
void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr>> rtp_list) {
|
||||
if (!_is_connect) {
|
||||
// 连接成功后才能发送数据
|
||||
return;
|
||||
}
|
||||
|
||||
size_t i = 0;
|
||||
auto size = rtp_list->size();
|
||||
rtp_list->for_each([&](Buffer::Ptr &packet) {
|
||||
if (_args.is_udp) {
|
||||
onSendRtpUdp(packet, i == 0);
|
||||
// udp模式，rtp over tcp前4个字节可以忽略
|
||||
_socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
|
||||
} else {
|
||||
// tcp模式, rtp over tcp前2个字节可以忽略,只保留后续rtp长度的2个字节
|
||||
_socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), 2), nullptr, 0, ++i == size);
|
||||
switch (_args.con_type) {
|
||||
case MediaSourceEvent::SendRtpArgs::kUdpActive:
|
||||
case MediaSourceEvent::SendRtpArgs::kUdpPassive: {
|
||||
onSendRtpUdp(packet, i == 0);
|
||||
// udp模式，rtp over tcp前4个字节可以忽略
|
||||
_socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
|
||||
break;
|
||||
}
|
||||
case MediaSourceEvent::SendRtpArgs::kTcpActive:
|
||||
case MediaSourceEvent::SendRtpArgs::kTcpPassive: {
|
||||
// tcp模式, rtp over tcp前2个字节可以忽略,只保留后续rtp长度的2个字节
|
||||
_socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), 2), nullptr, 0, ++i == size);
|
||||
break;
|
||||
}
|
||||
default: CHECK(0);
|
||||
}
|
||||
});
|
||||
}
|
||||
@ -338,9 +368,9 @@ void RtpSender::onErr(const SockException &ex) {
|
||||
onClose(ex);
|
||||
}
|
||||
|
||||
void RtpSender::setOnClose(std::function<void(const toolkit::SockException &ex)> on_close){
|
||||
void RtpSender::setOnClose(std::function<void(const toolkit::SockException &ex)> on_close) {
|
||||
_on_close = std::move(on_close);
|
||||
}
|
||||
|
||||
}//namespace mediakit
|
||||
#endif// defined(ENABLE_RTPPROXY)
|
||||
} // namespace mediakit
|
||||
#endif // defined(ENABLE_RTPPROXY)
|
||||
|
||||
@ -63,6 +63,9 @@ public:
|
||||
*/
|
||||
virtual void resetTracks() override;
|
||||
|
||||
/**
|
||||
* 设置发送rtp停止回调
|
||||
*/
|
||||
void setOnClose(std::function<void(const toolkit::SockException &ex)> on_close);
|
||||
|
||||
private:
|
||||
|
||||
Loading…
Reference in New Issue
Block a user