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webrtc支持通过http参数指定是否优先tcp模式 (#2105)

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* webrtc push/play支持通过http参数指定tcp

* force_tcp改成perferred_tcp

Co-authored-by: xiongziliang <771730766@qq.com>
This commit is contained in:
Dw9 2022-11-27 12:43:16 +08:00 committed by GitHub
parent 50fa671564
commit fef9d31631
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GPG Key ID: 4AEE18F83AFDEB23
7 changed files with 32 additions and 20 deletions
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8
webrtc/WebRtcPlayer.cpp
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@ -16,8 +16,9 @@ namespace mediakit {
WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const MediaInfo &info) {
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
const MediaInfo &info,
bool perferred_tcp) {
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, perferred_tcp), [](WebRtcPlayer *ptr) {
ptr->onDestory();
delete ptr;
});
@ -27,7 +28,8 @@ WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const MediaInfo &info) : WebRtcTransportImp(poller) {
const MediaInfo &info,
bool perferred_tcp) : WebRtcTransportImp(poller,perferred_tcp) {
_media_info = info;
_play_src = src;
CHECK(_play_src);

4
webrtc/WebRtcPlayer.h
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@ -19,7 +19,7 @@ class WebRtcPlayer : public WebRtcTransportImp {
public:
using Ptr = std::shared_ptr<WebRtcPlayer>;
~WebRtcPlayer() override = default;
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool perferred_tcp = false);
protected:
///////WebRtcTransportImp override///////
@ -29,7 +29,7 @@ protected:
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
private:
WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool perferred_tcp);
private:
//媒体相关元数据

8
webrtc/WebRtcPusher.cpp
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@ -18,8 +18,9 @@ WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const ProtocolOption &option) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
const ProtocolOption &option,
bool perferred_tcp) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,perferred_tcp), [](WebRtcPusher *ptr) {
ptr->onDestory();
delete ptr;
});
@ -31,7 +32,8 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const ProtocolOption &option) : WebRtcTransportImp(poller) {
const ProtocolOption &option,
bool perferred_tcp) : WebRtcTransportImp(poller,perferred_tcp) {
_media_info = info;
_push_src = src;
_push_src_ownership = ownership;

4
webrtc/WebRtcPusher.h
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@ -20,7 +20,7 @@ public:
using Ptr = std::shared_ptr<WebRtcPusher>;
~WebRtcPusher() override = default;
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool perferred_tcp = false);
protected:
///////WebRtcTransportImp override///////
@ -51,7 +51,7 @@ protected:
private:
WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool perferred_tcp);
private:
bool _simulcast = false;

3
webrtc/WebRtcSession.cpp
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@ -69,6 +69,7 @@ void WebRtcSession::onRecv_l(const char *data, size_t len) {
//WebRtcTransport在其他poller线程上，需要切换poller线程并重新创建WebRtcSession对象
if (!transport->getPoller()->isCurrentThread()) {
auto sock = Socket::createSocket(transport->getPoller());
//1、克隆socket(fd不变)，切换poller线程到WebRtcTransport所在线程
sock->cloneFromPeerSocket(*(getSock()));
auto server = _server;
std::string str(data, len);
@ -76,9 +77,11 @@ void WebRtcSession::onRecv_l(const char *data, size_t len) {
auto strong_server = server.lock();
if (strong_server) {
auto session = static_pointer_cast<WebRtcSession>(strong_server->createSession(sock));
//2、创建新的WebRtcSession对象(绑定到WebRtcTransport所在线程)，重新处理一遍ice binding request命令
session->onRecv_l(str.data(), str.size());
}
});
//3、销毁原先的socket和WebRtcSession(原先的对象跟WebRtcTransport不在同一条线程)
throw std::runtime_error("webrtc over tcp change poller: " + getPoller()->getThreadName() + " -> " + sock->getPoller()->getThreadName());
}

22
webrtc/WebRtcTransport.cpp
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@ -402,8 +402,8 @@ void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsT
#endif
}
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller)
: WebRtcTransport(poller) {
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool perferred_tcp)
: WebRtcTransport(poller), _perferred_tcp(perferred_tcp) {
InfoL << getIdentifier();
}
@ -629,13 +629,13 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
if (extern_ips.empty()) {
std::string local_ip = SockUtil::get_local_ip();
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, 110, "tcp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _perferred_tcp ? 125 : 115, "tcp")); }
} else {
const uint32_t delta = 10;
uint32_t priority = 100 + delta * extern_ips.size();
for (auto ip : extern_ips) {
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(ip, local_udp_port, priority + 5, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(ip, local_tcp_port, priority, "tcp")); }
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(ip, local_udp_port, priority, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(ip, local_tcp_port, priority - (_perferred_tcp ? -5 : 5), "tcp")); }
priority -= delta;
}
}
@ -1153,7 +1153,9 @@ void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
MediaInfo info(args["url"]);
Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
bool perferred_tcp = args["perferred_tcp"];
Broadcast::PublishAuthInvoker invoker = [cb, info, perferred_tcp](const string &err, const ProtocolOption &option) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@ -1192,7 +1194,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
push_src_ownership = push_src->getOwnership();
push_src->setProtocolOption(option);
}
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, perferred_tcp);
push_src->setListener(rtc);
cb(*rtc);
};
@ -1207,8 +1209,10 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
MediaInfo info(args["url"]);
bool perferred_tcp = args["perferred_tcp"];
auto session_ptr = sender.shared_from_this();
Broadcast::AuthInvoker invoker = [cb, info, session_ptr](const string &err) mutable {
Broadcast::AuthInvoker invoker = [cb, info, session_ptr, perferred_tcp](const string &err) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@ -1224,7 +1228,7 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
// 还原成rtc，目的是为了hook时识别哪种播放协议
info._schema = RTC_SCHEMA;
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, perferred_tcp);
cb(*rtc);
});
};

3
webrtc/WebRtcTransport.h
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@ -251,7 +251,7 @@ public:
void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
protected:
WebRtcTransportImp(const EventPoller::Ptr &poller);
WebRtcTransportImp(const EventPoller::Ptr &poller,bool perferred_tcp = false);
void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
void onStartWebRTC() override;
void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
@ -281,6 +281,7 @@ private:
void onCheckAnswer(RtcSession &sdp);
private:
bool _perferred_tcp;
uint16_t _rtx_seq[2] = {0, 0};
//用掉的总流量
uint64_t _bytes_usage = 0;