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add rtc tcp port config and ignore candidate when port is 0

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This commit is contained in:
xiongguangjie 2022-11-19 01:51:53 +08:00
parent 89959ef02f
commit 0d6fa1281a
5 changed files with 19 additions and 4 deletions
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4
conf/config.ini
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@ -313,6 +313,10 @@ externIP=
#该端口是多线程的，同时支持客户端网络切换导致的连接迁移
#需要注意的是，如果服务器在nat内，需要做端口映射时，必须确保外网映射端口跟该端口一致
port=8000
#rtc tcp服务器监听端口号，在udp 不通的情况下，会使用tcp传输数据
#该端口是多线程的，同时支持客户端网络切换导致的连接迁移
#需要注意的是，如果服务器在nat内，需要做端口映射时，必须确保外网映射端口跟该端口一致
tcpPort = 8000
#设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质
#目前已经实现twcc自动调整码率，关闭remb根据真实网络状况调整码率
rembBitRate=0

6
server/main.cpp
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@ -292,6 +292,7 @@ int start_main(int argc,char *argv[]) {
return Socket::createSocket(new_poller, false);
});
uint16_t rtcPort = mINI::Instance()[Rtc::kPort];
uint16_t rtcTcpPort = mINI::Instance()[Rtc::kTcpPort];
#endif//defined(ENABLE_WEBRTC)
@ -338,7 +339,10 @@ int start_main(int argc,char *argv[]) {
#if defined(ENABLE_WEBRTC)
//webrtc udp服务器
if (rtcPort) { rtcSrv_udp->start<WebRtcSession>(rtcPort); rtcSrv_tcp->start<WebRtcSession>(rtcPort); }
if (rtcPort) { rtcSrv_udp->start<WebRtcSession>(rtcPort);}
if (rtcTcpPort) { rtcSrv_tcp->start<WebRtcSession>(rtcTcpPort);}
#endif//defined(ENABLE_WEBRTC)
#if defined(ENABLE_SRT)

4
webrtc/Sdp.cpp
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@ -1204,7 +1204,9 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{
}
for (auto &cand : m.candidate) {
sdp_media.addAttr(std::make_shared<SdpAttrCandidate>(cand));
if(cand.port){
sdp_media.addAttr(std::make_shared<SdpAttrCandidate>(cand));
}
}
}
return ret;

8
webrtc/WebRtcTransport.cpp
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@ -44,11 +44,14 @@ const string kRembBitRate = RTC_FIELD "rembBitRate";
// webrtc单端口udp服务器
const string kPort = RTC_FIELD "port";
const string kTcpPort = RTC_FIELD "tcpPort";
static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 15;
mINI::Instance()[kExternIP] = "";
mINI::Instance()[kRembBitRate] = 0;
mINI::Instance()[kPort] = 8000;
mINI::Instance()[kTcpPort] = 8000;
});
} // namespace RTC
@ -612,6 +615,7 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransport::onRtcConfigure(configure);
GET_CONFIG(uint16_t, local_port, Rtc::kPort);
GET_CONFIG(uint16_t, local_tcp_port, Rtc::kTcpPort);
// 添加接收端口candidate信息
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, Rtc::kExternIP, [](string str) {
std::vector<std::string> ret;
@ -624,13 +628,13 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
if (extern_ips.empty()) {
std::string localIp = SockUtil::get_local_ip();
configure.addCandidate(*makeIceCandidate(localIp, local_port, 120, "udp"));
configure.addCandidate(*makeIceCandidate(localIp, local_port, 110, "tcp"));
configure.addCandidate(*makeIceCandidate(localIp, local_tcp_port, 110, "tcp"));
} else {
const uint32_t delta = 10;
uint32_t priority = 100 + delta * extern_ips.size();
for (auto ip : extern_ips) {
configure.addCandidate(*makeIceCandidate(ip, local_port, priority + 5, "udp"));
configure.addCandidate(*makeIceCandidate(ip, local_port, priority, "tcp"));
configure.addCandidate(*makeIceCandidate(ip, local_tcp_port, priority, "tcp"));
priority -= delta;
}
}

1
webrtc/WebRtcTransport.h
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@ -32,6 +32,7 @@ namespace mediakit {
//RTC配置项目
namespace Rtc {
extern const std::string kPort;
extern const std::string kTcpPort;
extern const std::string kTimeOutSec;
}//namespace RTC