【流媒体】Media soup源码分析（三）传输通道的建立
前面说到channel与router的建立，实际上channel是node服务器与底层C++之间信令交互的通道。transport才是router给用户推流拉流的通道，而创建完Router过后就开始创建Transport了
文章目录
- 一、创建Transport
-
- 1、用户请求
- 2、Worker——把信令传递给下一层
- 3、Router->HandleRequest——确定Transport的类型
- 4、SetNewTransportIdFromInternal——获得TransportID
- 5、RTC::WebRtcTransport——创建Transport
-
- 5.1、DtlsTransport
- 6、default——处理非CreateTransport类型的信令请求
- 二、建立connect连接
-
- 1、main.cpp——各模块的初始化
-
- 1.1、DepOpenSSL——加密算法随机数的产生
- 1.2、DepLibSRTP——初始化srtp
- 1.3、DepUsrSCTP
- 1.4、Utils::Crypto——产生ice_ufrag、ice_password的算法
- 1.5、RTC::DtlsTransport——证书和私钥
-
- 1.5.1、GenerateCertificateAndPrivateKey——产生证书和私钥
- 1.5.2、ReadCertificateAndPrivateKeyFromFiles——已有私钥和证书
- 1.5.3、CreateSslCtx——创建SSL上下文
- 1.5.5、GenerateFingerprints()——生成指纹
- 1.6、RTC::SrtpSession——定义处理不同事件的方法
-
- 1.6.1、OnSrtpEvent()——打印事件异常的信息
- 1.7、Channel::Notifier
- 1.8、PayloadChannel::Notifier
- 2、WebRtcTransport
-
- 2.1、WebRtcTransport::FillJson——配置ICE、dtls参数信息以json传给应用层
- 2.2、WebRtcTransport::HandleRequest——给客户端证书的相关信息
- 2.3、MayRunDtlsTransport()——dtls角色的选择
- 2.4、this->dtlsTransport->Run——dtls握手、建立connect
- 2.5、返回结果
- 三、创建Producer
-
- 1、SetNewProducerIdFromInternal——拿到producer的ID
- 2、RTC::Producer——Producer的构造函数
- 3、Listener——使创建的Producer加入Router的侦听者列表
- 4、Router::OnTransportNewProducer——把producer加入Transport maps中
- 四、创建Consumer
-
- 1、RTC::SimpleConsumer——对data进行数据解析
- 2、CreateRtpStream()——根据data数据确定consumer内部细节
- 3、consumer小结
一、创建Transport
1、用户请求
用户想要加入房间，这时在应用层便会发送creat transport 的信令，传递给Worker
2、Worker——把信令传递给下一层
当Worker处理不了用户的请求时，它便把消息传递下一层，即Router
request便传递给router->HandleRequest这个函数
default:
{
// This may throw.
RTC::Router* router = GetRouterFromInternal(request->internal);
router->HandleRequest(request);
break;
}
3、Router->HandleRequest——确定Transport的类型
进入到Router这层，系统想要create transport，由于transport是个基类，它实际上是有很多个子类
case Channel::Request::MethodId::ROUTER_CREATE_WEBRTC_TRANSPORT:
{
std::string transportId;
// This may throw.
SetNewTransportIdFromInternal(request->internal, transportId);
// This may throw.
auto* webRtcTransport = new RTC::WebRtcTransport(transportId, this, request->data);
// Insert into the map.
this->mapTransports[transportId] = webRtcTransport;
MS_DEBUG_DEV("WebRtcTransport created [transportId:%s]", transportId.c_str());
json data = json::object();
webRtcTransport->FillJson(data);
request->Accept(data);
break;
}
进入SetNewTransportIdFromInternal这个函数
4、SetNewTransportIdFromInternal——获得TransportID
void Router::SetNewTransportIdFromInternal(json& internal, std::string& transportId) const
{
MS_TRACE();
auto jsonTransportIdIt = internal.find("transportId");
if (jsonTransportIdIt == internal.end() || !jsonTransportIdIt->is_string())
MS_THROW_ERROR("missing internal.transportId");
transportId.assign(jsonTransportIdIt->get<std::string>());
if (this->mapTransports.find(transportId) != this->mapTransports.end())
MS_THROW_ERROR("a Transport with same transportId already exists");
}
这段代码就是从internal中搜索transport的ID，找到ID后返回
回到Router后，执行auto* webRtcTransport = new RTC::WebRtcTransport(transportId, this, request->data); 接着进入RTC::WebRtcTransport
5、RTC::WebRtcTransport——创建Transport
从源码来看，这个构造函数的前半部分是解析json数据，它包括
- enableUdp
- enableTcp
- preferUdp
- preferTcp
- listenIps
- 等等
WebRtcTransport::WebRtcTransp ort(const std::string& id, RTC::Transport::Listener* listener, json& data)
: RTC::Transport::Transport(id, listener, data)
{
MS_TRACE();
bool enableUdp{ true };
auto jsonEnableUdpIt = data.find("enableUdp");
if (jsonEnableUdpIt != data.end())
{
if (!jsonEnableUdpIt->is_boolean())
MS_THROW_TYPE_ERROR("wrong enableUdp (not a boolean)");
enableUdp = jsonEnableUdpIt->get<bool>();
}
bool enableTcp{ false };
auto jsonEnableTcpIt = data.find("enableTcp");
if (jsonEnableTcpIt != data.end())
{
if (!jsonEnableTcpIt->is_boolean())
MS_THROW_TYPE_ERROR("wrong enableTcp (not a boolean)");
enableTcp = jsonEnableTcpIt->get<bool>();
}
bool preferUdp{ false };
auto jsonPreferUdpIt = data.find("preferUdp");
if (jsonPreferUdpIt != data.end())
{
if (!jsonPreferUdpIt->is_boolean())
MS_THROW_TYPE_ERROR("wrong preferUdp (not a boolean)");
preferUdp = jsonPreferUdpIt->get<bool>();
}
bool preferTcp{ false };
auto jsonPreferTcpIt = data.find("preferTcp");
if (jsonPreferTcpIt != data.end())
{
if (!jsonPreferTcpIt->is_boolean())
MS_THROW_TYPE_ERROR("wrong preferTcp (not a boolean)");
preferTcp = jsonPreferTcpIt->get<bool>();
}
auto jsonListenIpsIt = data.find("listenIps");
if (jsonListenIpsIt == data.end())
MS_THROW_TYPE_ERROR("missing listenIps");
else if (!jsonListenIpsIt->is_array())
MS_THROW_TYPE_ERROR("wrong listenIps (not an array)");
else if (jsonListenIpsIt->empty())
MS_THROW_TYPE_ERROR("wrong listenIps (empty array)");
else if (jsonListenIpsIt->size() > 8)
MS_THROW_TYPE_ERROR("wrong listenIps (too many IPs)");
std::vector<ListenIp> listenIps(jsonListenIpsIt->size());
for (size_t i{ 0 }; i < jsonListenIpsIt->size(); ++i)
{
auto& jsonListenIp = (*jsonListenIpsIt)[i];
auto& listenIp = listenIps[i];
if (!jsonListenIp.is_object())
MS_THROW_TYPE_ERROR("wrong listenIp (not an object)");
auto jsonIpIt = jsonListenIp.find("ip");
if (jsonIpIt == jsonListenIp.end())
MS_THROW_TYPE_ERROR("missing listenIp.ip");
else if (!jsonIpIt->is_string())
MS_THROW_TYPE_ERROR("wrong listenIp.ip (not an string");
listenIp.ip.assign(jsonIpIt->get<std::string>());
// This may throw.
Utils::IP::NormalizeIp(listenIp.ip);
auto jsonAnnouncedIpIt = jsonListenIp.find("announcedIp");
if (jsonAnnouncedIpIt != jsonListenIp.end())
{
if (!jsonAnnouncedIpIt->is_string())
MS_THROW_TYPE_ERROR("wrong listenIp.announcedIp (not an string)");
listenIp.announcedIp.assign(jsonAnnouncedIpIt->get<std::string>());
}
}
解析完数据后，后半部分通过ip生成服务端的candidate，利用listenIp : listenIps来控制，有candidate就有权限
ICE通过IceServer创建，GetRandomString(16)，就是产生一个16位的随机值，32就是32位的随机值
进入IceServer的构造函数可以发现，当有用户连接这个Transport时，将会生成16位的用户名和32位的密码
try
{
uint16_t iceLocalPreferenceDecrement{ 0 };
if (enableUdp && enableTcp)
this->iceCandidates.reserve(2 * jsonListenIpsIt->size());
else
this->iceCandidates.reserve(jsonListenIpsIt->size());
for (auto& listenIp : listenIps)
{
if (enableUdp)
{
uint16_t iceLocalPreference =
IceCandidateDefaultLocalPriority - iceLocalPreferenceDecrement;
if (preferUdp)
iceLocalPreference += 1000;
uint32_t icePriority = generateIceCandidatePriority(iceLocalPreference);
// This may throw.
auto* udpSocket = new RTC::UdpSocket(this, listenIp.ip);
this->udpSockets[udpSocket] = listenIp.announcedIp;
if (listenIp.announcedIp.empty())
this->iceCandidates.emplace_back(udpSocket, icePriority);
else
this->iceCandidates.emplace_back(udpSocket, icePriority, listenIp.announcedIp);
}
if (enableTcp)
{
uint16_t iceLocalPreference =
IceCandidateDefaultLocalPriority - iceLocalPreferenceDecrement;
if (preferTcp)
iceLocalPreference += 1000;
uint32_t icePriority = generateIceCandidatePriority(iceLocalPreference);
// This may throw.
auto* tcpServer = new RTC::TcpServer(this, this, listenIp.ip);
this->tcpServers[tcpServer] = listenIp.announcedIp;
if (listenIp.announcedIp.empty())
this->iceCandidates.emplace_back(tcpServer, icePriority);
else
this->iceCandidates.emplace_back(tcpServer, icePriority, listenIp.announcedIp);
}
// Decrement initial ICE local preference for next IP.
iceLocalPreferenceDecrement += 100;
}
// Create a ICE server.
this->iceServer = new RTC::IceServer(
this, Utils::Crypto::GetRandomString(16), Utils::Crypto::GetRandomString(32));
// Create a DTLS transport.
this->dtlsTransport = new RTC::DtlsTransport(this);
}
catch (const MediaSoupError& error)
{
// Must delete everything since the destructor won't be called.
delete this->dtlsTransport;
this->dtlsTransport = nullptr;
delete this->iceServer;
this->iceServer = nullptr;
for (auto& kv : this->udpSockets)
{
auto* udpSocket = kv.first;
delete udpSocket;
}
this->udpSockets.clear();
for (auto& kv : this->tcpServers)
{
auto* tcpServer = kv.first;
delete tcpServer;
}
this->tcpServers.clear();
this->iceCandidates.clear();
throw;
}
}
它还创建了一个DTLS transport 的对象
5.1、DtlsTransport
在DtlsTransport中，首先创建了一个SSL对象，通过sslCtx上下文，因为在上下文中是有证书和私钥的
SSL就相当于一个连接，类似socket。SSL本身也有一些数据，创建的时候通过this能够获取到这些参数
然后通过创建BIO（Buffer IO）来形成两个管道，通过管道分别实现接、收数据的功能，再通过SSL_set_bio(this->ssl, this->sslBioFromNetwork, this->sslBioToNetwork);完成SSL对两个管道的绑定
接着还设置timer，用来控制DTLS握手的时间，超时后会做其他处理
DtlsTransport::DtlsTransport(Listener* listener) : listener(listener)
{
MS_TRACE();
/* Set SSL. */
this->ssl = SSL_new(DtlsTransport::sslCtx);
if (!this->ssl)
{
LOG_OPENSSL_ERROR("SSL_new() failed");
goto error;
}
// Set this as custom data.
SSL_set_ex_data(this->ssl, 0, static_cast<void*>(this));
this->sslBioFromNetwork = BIO_new(BIO_s_mem());
if (!this->sslBioFromNetwork)
{
LOG_OPENSSL_ERROR("BIO_new() failed");
SSL_free(this->ssl);
goto error;
}
this->sslBioToNetwork = BIO_new(BIO_s_mem());
if (!this->sslBioToNetwork)
{
LOG_OPENSSL_ERROR("BIO_new() failed");
BIO_free(this->sslBioFromNetwork);
SSL_free(this->ssl);
goto error;
}
SSL_set_bio(this->ssl, this->sslBioFromNetwork, this->sslBioToNetwork);
// Set the MTU so that we don't send packets that are too large with no fragmentation.
SSL_set_mtu(this->ssl, DtlsMtu);
DTLS_set_link_mtu(this->ssl, DtlsMtu);
// Set callback handler for setting DTLS timer interval.
DTLS_set_timer_cb(this->ssl, onSslDtlsTimer);
// Set the DTLS timer.
this->timer = new Timer(this);
return;
error:
// NOTE: At this point SSL_set_bio() was not called so we must free BIOs as
// well.
if (this->sslBioFromNetwork)
BIO_free(this->sslBioFromNetwork);
if (this->sslBioToNetwork)
BIO_free(this->sslBioToNetwork);
if (this->ssl)
SSL_free(this->ssl);
// NOTE: If this is not catched by the caller the program will abort, but
// this should never happen.
MS_THROW_ERROR("DtlsTransport instance creation failed");
}
6、default——处理非CreateTransport类型的信令请求
default:
{
// This may throw.
RTC::Transport* transport = GetTransportFromInternal(request->internal);
transport->HandleRequest(request);
break;
}
接下来进入HandleRequest，本文跳转至 第二章第2节
二、建立connect连接
1、main.cpp——各模块的初始化
首先在main.cpp中各对与connect相关的所有模块进行了初始化
// Initialize static stuff.
DepOpenSSL::ClassInit();
DepLibSRTP::ClassInit();
DepUsrSCTP::ClassInit();
DepLibWebRTC::ClassInit();
Utils::Crypto::ClassInit();
RTC::DtlsTransport::ClassInit();
RTC::SrtpSession::ClassInit();
Channel::Notifier::ClassInit(channel);
PayloadChannel::Notifier::ClassInit(payloadChannel);
1.1、DepOpenSSL——加密算法随机数的产生
void DepOpenSSL::ClassInit()
{
MS_TRACE();
MS_DEBUG_TAG(info, "openssl version: \"%s\"", OpenSSL_version(OPENSSL_VERSION));
// Initialize some crypto stuff.
RAND_poll();
}
这部分主要是对加密算法的随机数的生成
1.2、DepLibSRTP——初始化srtp
void DepLibSRTP::ClassInit()
{
MS_TRACE();
MS_DEBUG_TAG(info, "libsrtp version: \"%s\"", srtp_get_version_string());
srtp_err_status_t err = srtp_init();
if (DepLibSRTP::IsError(err))
MS_THROW_ERROR("srtp_init() failed: %s", DepLibSRTP::GetErrorString(err));
}
这部分调用了srtp_init 这个API，这个API的返回值对后面Session的初始化有着重要的作用
1.3、DepUsrSCTP
void DepUsrSCTP::ClassInit()
{
MS_TRACE();
MS_DEBUG_TAG(info, "usrsctp");
usrsctp_init_nothreads(0, onSendSctpData, sctpDebug);
// Disable explicit congestion notifications (ecn).
usrsctp_sysctl_set_sctp_ecn_enable(0);
#ifdef SCTP_DEBUG
usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
#endif
DepUsrSCTP::checker = new DepUsrSCTP::Checker();
}
1.4、Utils::Crypto——产生ice_ufrag、ice_password的算法
void Crypto::ClassInit()
{
MS_TRACE();
// Init the vrypto seed with a random number taken from the address
// of the seed variable itself (which is random).
Crypto::seed = static_cast<uint32_t>(reinterpret_cast<uintptr_t>(std::addressof(Crypto::seed)));
// Create an OpenSSL HMAC_CTX context for HMAC SHA1 calculation.
Crypto::hmacSha1Ctx = HMAC_CTX_new();
}
创建了Crypto的种子，使用的算法是HMAC，其中H代表的是hash哈希
1.5、RTC::DtlsTransport——证书和私钥
void DtlsTransport::ClassInit()
{
MS_TRACE();
// Generate a X509 certificate and private key (unless PEM files are provided).
if (
Settings::configuration.dtlsCertificateFile.empty() ||
Settings::configuration.dtlsPrivateKeyFile.empty())
{
GenerateCertificateAndPrivateKey();
}
else
{
ReadCertificateAndPrivateKeyFromFiles();
}
// Create a global SSL_CTX.
CreateSslCtx();
// Generate certificate fingerprints.
GenerateFingerprints();
}
在代码中首先判断有没有证书和私钥，若没有便通过 GenerateCertificateAndPrivateKey 这个函数产生一个证书和私钥
1.5.1、GenerateCertificateAndPrivateKey——产生证书和私钥
void DtlsTransport::GenerateCertificateAndPrivateKey()
{
MS_TRACE();
int ret{ 0 };
EC_KEY* ecKey{ nullptr };
X509_NAME* certName{ nullptr };
std::string subject =
std::string("mediasoup") + std::to_string(Utils::Crypto::GetRandomUInt(100000, 999999));
// Create key with curve.
ecKey = EC_KEY_new_by_curve_name(NID_X9_62_prime256v1);
if (ecKey == nullptr)
{
LOG_OPENSSL_ERROR("EC_KEY_new_by_curve_name() failed");
goto error;
}
EC_KEY_set_asn1_flag(ecKey, OPENSSL_EC_NAMED_CURVE);
// NOTE: This can take some time.
ret = EC_KEY_generate_key(ecKey);
if (ret == 0)
{
LOG_OPENSSL_ERROR("EC_KEY_generate_key() failed");
goto error;
}
// Create a private key object.
DtlsTransport::privateKey = EVP_PKEY_new();
if (!DtlsTransport::privateKey)
{
LOG_OPENSSL_ERROR("EVP_PKEY_new() failed");
goto error;
}
// NOLINTNEXTLINE(cppcoreguidelines-pro-type-cstyle-cast)
ret = EVP_PKEY_assign_EC_KEY(DtlsTransport::privateKey, ecKey);
if (ret == 0)
{
LOG_OPENSSL_ERROR("EVP_PKEY_assign_EC_KEY() failed");
goto error;
}
// The EC key now belongs to the private key, so don't clean it up separately.
ecKey = nullptr;
// Create the X509 certificate.
DtlsTransport::certificate = X509_new();
if (!DtlsTransport::certificate)
{
LOG_OPENSSL_ERROR("X509_new() failed");
goto error;
}
// Set version 3 (note that 0 means version 1).
X509_set_version(DtlsTransport::certificate, 2);
// Set serial number (avoid default 0).
ASN1_INTEGER_set(
X509_get_serialNumber(DtlsTransport::certificate),
static_cast<uint64_t>(Utils::Crypto::GetRandomUInt(1000000, 9999999)));
// Set valid period.
X509_gmtime_adj(X509_get_notBefore(DtlsTransport::certificate), -315360000); // -10 years.
X509_gmtime_adj(X509_get_notAfter(DtlsTransport::certificate), 315360000); // 10 years.
// Set the public key for the certificate using the key.
ret = X509_set_pubkey(DtlsTransport::certificate, DtlsTransport::privateKey);
if (ret == 0)
{
LOG_OPENSSL_ERROR("X509_set_pubkey() failed");
goto error;
}
// Set certificate fields.
certName = X509_get_subject_name(DtlsTransport::certificate);
if (!certName)
{
LOG_OPENSSL_ERROR("X509_get_subject_name() failed");
goto error;
}
X509_NAME_add_entry_by_txt(
certName, "O", MBSTRING_ASC, reinterpret_cast<const uint8_t*>(subject.c_str()), -1, -1, 0);
X509_NAME_add_entry_by_txt(
certName, "CN", MBSTRING_ASC, reinterpret_cast<const uint8_t*>(subject.c_str()), -1, -1, 0);
// It is self-signed so set the issuer name to be the same as the subject.
ret = X509_set_issuer_name(DtlsTransport::certificate, certName);
if (ret == 0)
{
LOG_OPENSSL_ERROR("X509_set_issuer_name() failed");
goto error;
}
// Sign the certificate with its own private key.
ret = X509_sign(DtlsTransport::certificate, DtlsTransport::privateKey, EVP_sha1());
if (ret == 0)
{
LOG_OPENSSL_ERROR("X509_sign() failed");
goto error;
}
return;
error:
if (ecKey)
EC_KEY_free(ecKey);
if (DtlsTransport::privateKey)
EVP_PKEY_free(DtlsTransport::privateKey); // NOTE: This also frees the EC key.
if (DtlsTransport::certificate)
X509_free(DtlsTransport::certificate);
MS_THROW_ERROR("DTLS certificate and private key generation failed");
}
这部分代码一连串的if语句，调用API创建了证书，生成其版本、序列号等等
// Create the X509 certificate.
DtlsTransport::certificate = X509_new();
1.5.2、ReadCertificateAndPrivateKeyFromFiles——已有私钥和证书
若有自己的证书和API，则进入这个函数
函数体用的开始是用文件类型打开，接着调用API读取证书和私钥
void DtlsTransport::ReadCertificateAndPrivateKeyFromFiles()
{
MS_TRACE();
FILE* file{ nullptr };
file = fopen(Settings::configuration.dtlsCertificateFile.c_str(), "r");
if (!file)
{
MS_ERROR("error reading DTLS certificate file: %s", std::strerror(errno));
goto error;
}
DtlsTransport::certificate = PEM_read_X509(file, nullptr, nullptr, nullptr);
if (!DtlsTransport::certificate)
{
LOG_OPENSSL_ERROR("PEM_read_X509() failed");
goto error;
}
fclose(file);
file = fopen(Settings::configuration.dtlsPrivateKeyFile.c_str(), "r");
if (!file)
{
MS_ERROR("error reading DTLS private key file: %s", std::strerror(errno));
goto error;
}
DtlsTransport::privateKey = PEM_read_PrivateKey(file, nullptr, nullptr, nullptr);
if (!DtlsTransport::privateKey)
{
LOG_OPENSSL_ERROR("PEM_read_PrivateKey() failed");
goto error;
}
fclose(file);
return;
error:
MS_THROW_ERROR("error reading DTLS certificate and private key PEM files");
}
1.5.3、CreateSslCtx——创建SSL上下文
拿到证书和私钥后，将会创建一个SSL上下文。首先对Dtls的版本做了判断
// Both DTLS 1.0 and 1.2 (requires OpenSSL >= 1.1.0).
DtlsTransport::sslCtx = SSL_CTX_new(DTLS_method());
然后会把证书与上下文绑定
ret = SSL_CTX_use_certificate(DtlsTransport::sslCtx, DtlsTransport::certificate);
key也会与上下文绑定
ret = SSL_CTX_use_PrivateKey(DtlsTransport::sslCtx, DtlsTransport::privateKey);
接着对key做了检查，设置了一些选项
// Set options.
SSL_CTX_set_options(
DtlsTransport::sslCtx,
SSL_OP_CIPHER_SERVER_PREFERENCE | SSL_OP_NO_TICKET | SSL_OP_SINGLE_ECDH_USE |
SSL_OP_NO_QUERY_MTU);
设置ciphers，它是一个密语的套件，具体是哪些加密算法就是通过它来设置的，用一个字符串来描述
// Set ciphers.
ret = SSL_CTX_set_cipher_list(
DtlsTransport::sslCtx, "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK");
除此之外，Dtls使用srtp来加密解密。这个for循环通过对srtpCryptoSuites的遍历，累加在dtlsSrtpCryptoSuites后面，然后利用SSL_CTX_set_tlsext_use_srtp这个API设置进去
// Set the "use_srtp" DTLS extension.
for (auto it = DtlsTransport::srtpCryptoSuites.begin();
it != DtlsTransport::srtpCryptoSuites.end();
++it)
{
if (it != DtlsTransport::srtpCryptoSuites.begin())
dtlsSrtpCryptoSuites += ":";
SrtpCryptoSuiteMapEntry* cryptoSuiteEntry = std::addressof(*it);
dtlsSrtpCryptoSuites += cryptoSuiteEntry->name;
}
1.5.5、GenerateFingerprints()——生成指纹
void DtlsTransport::GenerateFingerprints()
{
MS_TRACE();
for (auto& kv : DtlsTransport::string2FingerprintAlgorithm)
{
const std::string& algorithmString = kv.first;
FingerprintAlgorithm algorithm = kv.second;
uint8_t binaryFingerprint[EVP_MAX_MD_SIZE];
unsigned int size{ 0 };
char hexFingerprint[(EVP_MAX_MD_SIZE * 3) + 1];
const EVP_MD* hashFunction;
int ret;
switch (algorithm)
{
case FingerprintAlgorithm::SHA1:
hashFunction = EVP_sha1();
break;
case FingerprintAlgorithm::SHA224:
hashFunction = EVP_sha224();
break;
case FingerprintAlgorithm::SHA256:
hashFunction = EVP_sha256();
break;
case FingerprintAlgorithm::SHA384:
hashFunction = EVP_sha384();
break;
case FingerprintAlgorithm::SHA512:
hashFunction = EVP_sha512();
break;
default:
MS_THROW_ERROR("unknown algorithm");
}
ret = X509_digest(DtlsTransport::certificate, hashFunction, binaryFingerprint, &size);
if (ret == 0)
{
MS_ERROR("X509_digest() failed");
MS_THROW_ERROR("Fingerprints generation failed");
}
// Convert to hexadecimal format in uppercase with colons.
for (unsigned int i{ 0 }; i < size; ++i)
{
std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]);
}
hexFingerprint[(size * 3) - 1] = '\0';
MS_DEBUG_TAG(dtls, "%-7s fingerprint: %s", algorithmString.c_str(), hexFingerprint);
// Store it in the vector.
DtlsTransport::Fingerprint fingerprint;
fingerprint.algorithm = DtlsTransport::GetFingerprintAlgorithm(algorithmString);
fingerprint.value = hexFingerprint;
DtlsTransport::localFingerprints.push_back(fingerprint);
}
}
指纹是通过证书来产生的，这个函数体内定义了多种产生指纹的算法，如
- SHA1
- SHA224
- SHA256
- SHA384
- SHA512
对于每种不同的算法它都有对应的函数，然后通过ret = X509_digest(DtlsTransport::certificate, hashFunction, binaryFingerprint, &size);这个API计算证书的指纹，结果存放在binaryFingerprint里
接着把binaryFingerprint转化为十六进制的格式std::sprintf(hexFingerprint + (i * 3), "%.2X:", binaryFingerprint[i]);
完成后把结果的值存放进fingerprint.value，使用的算法是fingerprint.algorithm
最后再把fingerprint这个对象存放进localFingerprints
这样指纹便创建好了
1.6、RTC::SrtpSession——定义处理不同事件的方法
void SrtpSession::ClassInit()
{
// Set libsrtp event handler.
srtp_err_status_t err =
srtp_install_event_handler(static_cast<srtp_event_handler_func_t*>(OnSrtpEvent));
if (DepLibSRTP::IsError(err))
{
MS_THROW_ERROR("srtp_install_event_handler() failed: %s", DepLibSRTP::GetErrorString(err));
}
}
Session这块代码主要是在OnSrtpEvent()这个函数中定义了对不同事件的处理
1.6.1、OnSrtpEvent()——打印事件异常的信息
void SrtpSession::OnSrtpEvent(srtp_event_data_t* data)
{
MS_TRACE();
switch (data->event)
{
case event_ssrc_collision:
MS_WARN_TAG(srtp, "SSRC collision occurred");
break;
case event_key_soft_limit:
MS_WARN_TAG(srtp, "stream reached the soft key usage limit and will expire soon");
break;
case event_key_hard_limit:
MS_WARN_TAG(srtp, "stream reached the hard key usage limit and has expired");
break;
case event_packet_index_limit:
MS_WARN_TAG(srtp, "stream reached the hard packet limit (2^48 packets)");
break;
}
}
如ssrc冲突、软件硬件的限制，主要是打印错误信息
1.7、Channel::Notifier
void Notifier::ClassInit(Channel::UnixStreamSocket* channel)
{
MS_TRACE();
Notifier::channel = channel;
}
1.8、PayloadChannel::Notifier
void Notifier::ClassInit(PayloadChannel::UnixStreamSocket* payloadChannel)
{
MS_TRACE();
Notifier::payloadChannel = payloadChannel;
}
2、WebRtcTransport
当以上准备工作做好后了，WebRtcTransport要给客户端返回一个数据data，进入WebRtcTransport::FillJson
2.1、WebRtcTransport::FillJson——配置ICE、dtls参数信息以json传给应用层
// Add iceRole (we are always "controlled").
jsonObject["iceRole"] = "controlled";
// Add iceParameters.
jsonObject["iceParameters"] = json::object();
auto jsonIceParametersIt = jsonObject.find("iceParameters");
(*jsonIceParametersIt)["usernameFragment"] = this->iceServer->GetUsernameFragment();
(*jsonIceParametersIt)["password"] = this->iceServer->GetPassword();
(*jsonIceParametersIt)["iceLite"] = true;
在FillJson里，iceRole总是设为被控制端
当创建一个transport时，会收到从服务端返回的ice_ufrag和ice_password
iceLite是指使用的ICE不是完整的ICE，而是其中部分ICE协议
// Add iceCandidates.
jsonObject["iceCandidates"] = json::array();
auto jsonIceCandidatesIt = jsonObject.find("iceCandidates");
for (size_t i{ 0 }; i < this->iceCandidates.size(); ++i)
{
jsonIceCandidatesIt->emplace_back(json::value_t::object);
auto& jsonEntry = (*jsonIceCandidatesIt)[i];
auto& iceCandidate = this->iceCandidates[i];
iceCandidate.FillJson(jsonEntry);
}
我们还从服务端收到了许多IP，它们形成了candidate，然后把这些candidate组成数组放入json中
// Add iceState.
switch (this->iceServer->GetState())
{
case RTC::IceServer::IceState::NEW:
jsonObject["iceState"] = "new";
break;
case RTC::IceServer::IceState::CONNECTED:
jsonObject["iceState"] = "connected";
break;
case RTC::IceServer::IceState::COMPLETED:
jsonObject["iceState"] = "completed";
break;
case RTC::IceServer::IceState::DISCONNECTED:
jsonObject["iceState"] = "disconnected";
break;
}
ICE的状态，一般为new
// Add dtlsParameters.
jsonObject["dtlsParameters"] = json::object();
auto jsonDtlsParametersIt = jsonObject.find("dtlsParameters");
// Add dtlsParameters.fingerprints.
(*jsonDtlsParametersIt)["fingerprints"] = json::array();
auto jsonDtlsParametersFingerprintsIt = jsonDtlsParametersIt->find("fingerprints");
auto& fingerprints = this->dtlsTransport->GetLocalFingerprints();
for (size_t i{ 0 }; i < fingerprints.size(); ++i)
{
jsonDtlsParametersFingerprintsIt->emplace_back(json::value_t::object);
auto& jsonEntry = (*jsonDtlsParametersFingerprintsIt)[i];
auto& fingerprint = fingerprints[i];
jsonEntry["algorithm"] =
RTC::DtlsTransport::GetFingerprintAlgorithmString(fingerprint.algorithm);
jsonEntry["value"] = fingerprint.value;
}
dtls的参数，其中最重要的是fingerprint，它是一组数据（因为用了不同的加密算法），然后形成一个数组把它塞入fingerprints，同时传给客户端，客户端就拿到了各种指纹和证书了
// Add dtlsParameters.role.
switch (this->dtlsRole)
{
case RTC::DtlsTransport::Role::NONE:
(*jsonDtlsParametersIt)["role"] = "none";
break;
case RTC::DtlsTransport::Role::AUTO:
(*jsonDtlsParametersIt)["role"] = "auto";
break;
case RTC::DtlsTransport::Role::CLIENT:
(*jsonDtlsParametersIt)["role"] = "client";
break;
case RTC::DtlsTransport::Role::SERVER:
(*jsonDtlsParametersIt)["role"] = "server";
break;
}
// Add dtlsState.
switch (this->dtlsTransport->GetState())
{
case RTC::DtlsTransport::DtlsState::NEW:
jsonObject["dtlsState"] = "new";
break;
case RTC::DtlsTransport::DtlsState::CONNECTING:
jsonObject["dtlsState"] = "connecting";
break;
case RTC::DtlsTransport::DtlsState::CONNECTED:
jsonObject["dtlsState"] = "connected";
break;
case RTC::DtlsTransport::DtlsState::FAILED:
jsonObject["dtlsState"] = "failed";
break;
case RTC::DtlsTransport::DtlsState::CLOSED:
jsonObject["dtlsState"] = "closed";
break;
}
}
除此之外还有dtls的角色和传输状态
以上所有信息都会一起打包传给应用层，应用层就拿到了服务端的信息，然后就可以进行验证
2.2、WebRtcTransport::HandleRequest——给客户端证书的相关信息
case Channel::Request::MethodId::TRANSPORT_CONNECT:
{
// Ensure this method is not called twice.
if (this->connectCalled)
MS_THROW_ERROR("connect() already called");
RTC::DtlsTransport::Fingerprint dtlsRemoteFingerprint;
RTC::DtlsTransport::Role dtlsRemoteRole;
auto jsonDtlsParametersIt = request->data.find("dtlsParameters");
if (jsonDtlsParametersIt == request->data.end() || !jsonDtlsParametersIt->is_object())
MS_THROW_TYPE_ERROR("missing dtlsParameters");
auto jsonFingerprintsIt = jsonDtlsParametersIt->find("fingerprints");
if (jsonFingerprintsIt == jsonDtlsParametersIt->end() || !jsonFingerprintsIt->is_array())
MS_THROW_TYPE_ERROR("missing dtlsParameters.fingerprints");
// NOTE: Just take the first fingerprint.
for (auto& jsonFingerprint : *jsonFingerprintsIt)
{
if (!jsonFingerprint.is_object())
MS_THROW_TYPE_ERROR("wrong entry in dtlsParameters.fingerprints (not an object)");
auto jsonAlgorithmIt = jsonFingerprint.find("algorithm");
if (jsonAlgorithmIt == jsonFingerprint.end())
MS_THROW_TYPE_ERROR("missing fingerprint.algorithm");
else if (!jsonAlgorithmIt->is_string())
MS_THROW_TYPE_ERROR("wrong fingerprint.algorithm (not a string)");
dtlsRemoteFingerprint.algorithm =
RTC::DtlsTransport::GetFingerprintAlgorithm(jsonAlgorithmIt->get<std::string>());
if (dtlsRemoteFingerprint.algorithm == RTC::DtlsTransport::FingerprintAlgorithm::NONE)
MS_THROW_TYPE_ERROR("invalid fingerprint.algorithm value");
auto jsonValueIt = jsonFingerprint.find("value");
if (jsonValueIt == jsonFingerprint.end())
MS_THROW_TYPE_ERROR("missing fingerprint.value");
else if (!jsonValueIt->is_string())
MS_THROW_TYPE_ERROR("wrong fingerprint.value (not a string)");
dtlsRemoteFingerprint.value = jsonValueIt->get<std::string>();
// Just use the first fingerprint.
break;
}
auto jsonRoleIt = jsonDtlsParametersIt->find("role");
if (jsonRoleIt != jsonDtlsParametersIt->end())
{
if (!jsonRoleIt->is_string())
MS_THROW_TYPE_ERROR("wrong dtlsParameters.role (not a string)");
dtlsRemoteRole = RTC::DtlsTransport::StringToRole(jsonRoleIt->get<std::string>());
if (dtlsRemoteRole == RTC::DtlsTransport::Role::NONE)
MS_THROW_TYPE_ERROR("invalid dtlsParameters.role value");
}
else
{
dtlsRemoteRole = RTC::DtlsTransport::Role::AUTO;
}
// Set local DTLS role.
switch (dtlsRemoteRole)
{
case RTC::DtlsTransport::Role::CLIENT:
{
this->dtlsRole = RTC::DtlsTransport::Role::SERVER;
break;
}
// If the peer has role "auto" we become "client" since we are ICE controlled.
case RTC::DtlsTransport::Role::SERVER:
case RTC::DtlsTransport::Role::AUTO:
{
this->dtlsRole = RTC::DtlsTransport::Role::CLIENT;
break;
}
case RTC::DtlsTransport::Role::NONE:
{
MS_THROW_TYPE_ERROR("invalid remote DTLS role");
}
}
this->connectCalled = true;
// Pass the remote fingerprint to the DTLS transport.
if (this->dtlsTransport->SetRemoteFingerprint(dtlsRemoteFingerprint))
{
// If everything is fine, we may run the DTLS transport if ready.
MayRunDtlsTransport();
}
// Tell the caller about the selected local DTLS role.
json data = json::object();
switch (this->dtlsRole)
{
case RTC::DtlsTransport::Role::CLIENT:
data["dtlsLocalRole"] = "client";
break;
case RTC::DtlsTransport::Role::SERVER:
data["dtlsLocalRole"] = "server";
break;
default:
MS_ABORT("invalid local DTLS role");
}
request->Accept(data);
break;
}
当收到客户端发送过来的TRANSPORT_CONNECT请求时，会创建远端的dtlsRemoteFingerprint和远端的dtlsRemoteRole，然后在for循环中找寻dtlsRemoteFingerprint.algorithm所使用的算法，以及对应的值jsonValueIt
这样客户端就拿到了证书的指纹、证书的值以及使用的算法
这些都配置好后便进入MayRunDtlsTransport()
2.3、MayRunDtlsTransport()——dtls角色的选择
这个函数会根据Dtls的角色来进行不同的操作
void WebRtcTransport::MayRunDtlsTransport()
{
MS_TRACE();
// Do nothing if we have the same local DTLS role as the DTLS transport.
// NOTE: local role in DTLS transport can be NONE, but not ours.
if (this->dtlsTransport->GetLocalRole() == this->dtlsRole)
return;
// Check our local DTLS role.
switch (this->dtlsRole)
{
// If still 'auto' then transition to 'server' if ICE is 'connected' or
// 'completed'.
case RTC::DtlsTransport::Role::AUTO:
{
// clang-format off
if (
this->iceServer->GetState() == RTC::IceServer::IceState::CONNECTED ||
this->iceServer->GetState() == RTC::IceServer::IceState::COMPLETED
)
// clang-format on
{
MS_DEBUG_TAG(
dtls, "transition from DTLS local role 'auto' to 'server' and running DTLS transport");
this->dtlsRole = RTC::DtlsTransport::Role::SERVER;
this->dtlsTransport->Run(RTC::DtlsTransport::Role::SERVER);
}
break;
}
// 'client' is only set if a 'connect' request was previously called with
// remote DTLS role 'server'.
//
// If 'client' then wait for ICE to be 'completed' (got USE-CANDIDATE).
//
// NOTE: This is the theory, however let's be more flexible as told here:
// https://bugs.chromium.org/p/webrtc/issues/detail?id=3661
case RTC::DtlsTransport::Role::CLIENT:
{
// clang-format off
if (
this->iceServer->GetState() == RTC::IceServer::IceState::CONNECTED ||
this->iceServer->GetState() == RTC::IceServer::IceState::COMPLETED
)
// clang-format on
{
MS_DEBUG_TAG(dtls, "running DTLS transport in local role 'client'");
this->dtlsTransport->Run(RTC::DtlsTransport::Role::CLIENT);
}
break;
}
// If 'server' then run the DTLS transport if ICE is 'connected' (not yet
// USE-CANDIDATE) or 'completed'.
case RTC::DtlsTransport::Role::SERVER:
{
// clang-format off
if (
this->iceServer->GetState() == RTC::IceServer::IceState::CONNECTED ||
this->iceServer->GetState() == RTC::IceServer::IceState::COMPLETED
)
// clang-format on
{
MS_DEBUG_TAG(dtls, "running DTLS transport in local role 'server'");
this->dtlsTransport->Run(RTC::DtlsTransport::Role::SERVER);
}
break;
}
case RTC::DtlsTransport::Role::NONE:
{
MS_ABORT("local DTLS role not set");
}
}
}
2.4、this->dtlsTransport->Run——dtls握手、建立connect
以客户端为例，进入this->dtlsTransport->Run()
// Set state and notify the listener.
this->state = DtlsState::CONNECTING;
this->listener->OnDtlsTransportConnecting(this);
先将Dtls状态设置为CONNECTING，然后把dtlsTransport对象设置给侦听者（WebRtcTransport）
case Role::CLIENT:
{
MS_DEBUG_TAG(dtls, "running [role:client]");
SSL_set_connect_state(this->ssl);
SSL_do_handshake(this->ssl);
SendPendingOutgoingDtlsData();
SetTimeout();
break;
}
当用户发送连接请求，dtlsTransport的角色为客户端时，dtls就会主动发起连接请求调用SSL_set_connect_state这个API，将客户端当作dtls的服务端，对方收到这个连接请求时，就开始握手，调用SSL_do_handshake这个API主动握手，接着调用SendPendingOutgoingDtlsData()发送dtls的一些数据
case Role::SERVER:
{
MS_DEBUG_TAG(dtls, "running [role:server]");
SSL_set_accept_state(this->ssl);
SSL_do_handshake(this->ssl);
break;
}
若dtlsTransport的角色为服务端时，就会把SSL设置为阻塞模式，然后等待握手
通过以上便建立起了连接
2.5、返回结果
建立连接的结果会赋给json data = json::object();
然后通过request->Accept(data);反馈给应用层
三、创建Producer
SetNewProducerIdFromInternal(request->internal, producerId);
进入SetNewProducerIdFromInternal这个函数
1、SetNewProducerIdFromInternal——拿到producer的ID
void Transport::SetNewProducerIdFromInternal(json& internal, std::string& producerId) const
{
MS_TRACE();
auto jsonProducerIdIt = internal.find("producerId");
if (jsonProducerIdIt == internal.end() || !jsonProducerIdIt->is_string())
MS_THROW_ERROR("missing internal.producerId");
producerId.assign(jsonProducerIdIt->get<std::string>());
if (this->mapProducers.find(producerId) != this->mapProducers.end())
MS_THROW_ERROR("a Producer with same producerId already exists");
}
搜索producerId，拿到ID后返回，进入构造函数RTC::Producer
2、RTC::Producer——Producer的构造函数
在这个构造函数中，主要也是对json数据data进行解析，它包括
- kind：audio和video
- rtpparameters：mid、codecs、headerExtensions、encodings、rtcp
- 等等
Producer::Producer(const std::string& id, RTC::Producer::Listener* listener, json& data)
: id(id), listener(listener)
{
MS_TRACE();
auto jsonKindIt = data.find("kind");
if (jsonKindIt == data.end() || !jsonKindIt->is_string())
MS_THROW_TYPE_ERROR("missing kind");
// This may throw.
this->kind = RTC::Media::GetKind(jsonKindIt->get<std::string>());
if (this->kind == RTC::Media::Kind::ALL)
MS_THROW_TYPE_ERROR("invalid empty kind");
auto jsonRtpParametersIt = data.find("rtpParameters");
if (jsonRtpParametersIt == data.end() || !jsonRtpParametersIt->is_object())
MS_THROW_TYPE_ERROR("missing rtpParameters");
// This may throw.
this->rtpParameters = RTC::RtpParameters(*jsonRtpParametersIt);
// Evaluate type.
this->type = RTC::RtpParameters::GetType(this->rtpParameters);
// Reserve a slot in rtpStreamByEncodingIdx and rtpStreamsScores vectors
// for each RTP stream.
this->rtpStreamByEncodingIdx.resize(this->rtpParameters.encodings.size(), nullptr);
this->rtpStreamScores.resize(this->rtpParameters.encodings.size(), 0u);
auto& encoding = this->rtpParameters.encodings[0];
auto* mediaCodec = this->rtpParameters.GetCodecForEncoding(encoding);
if (!RTC::Codecs::Tools::IsValidTypeForCodec(this->type, mediaCodec->mimeType))
{
MS_THROW_TYPE_ERROR(
"%s codec not supported for %s",
mediaCodec->mimeType.ToString().c_str(),
RTC::RtpParameters::GetTypeString(this->type).c_str());
}
auto jsonRtpMappingIt = data.find("rtpMapping");
if (jsonRtpMappingIt == data.end() || !jsonRtpMappingIt->is_object())
MS_THROW_TYPE_ERROR("missing rtpMapping");
auto jsonCodecsIt = jsonRtpMappingIt->find("codecs");
if (jsonCodecsIt == jsonRtpMappingIt->end() || !jsonCodecsIt->is_array())
MS_THROW_TYPE_ERROR("missing rtpMapping.codecs");
for (auto& codec : *jsonCodecsIt)
{
if (!codec.is_object())
MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.codecs (not an object)");
auto jsonPayloadTypeIt = codec.find("payloadType");
// clang-format off
if (
jsonPayloadTypeIt == codec.end() ||
!Utils::Json::IsPositiveInteger(*jsonPayloadTypeIt)
)
// clang-format on
{
MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.codecs (missing payloadType)");
}
auto jsonMappedPayloadTypeIt = codec.find("mappedPayloadType");
// clang-format off
if (
jsonMappedPayloadTypeIt == codec.end() ||
!Utils::Json::IsPositiveInteger(*jsonMappedPayloadTypeIt)
)
// clang-format on
{
MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.codecs (missing mappedPayloadType)");
}
this->rtpMapping.codecs[jsonPayloadTypeIt->get<uint8_t>()] =
jsonMappedPayloadTypeIt->get<uint8_t>();
}
auto jsonEncodingsIt = jsonRtpMappingIt->find("encodings");
if (jsonEncodingsIt == jsonRtpMappingIt->end() || !jsonEncodingsIt->is_array())
{
MS_THROW_TYPE_ERROR("missing rtpMapping.encodings");
}
this->rtpMapping.encodings.reserve(jsonEncodingsIt->size());
for (auto& encoding : *jsonEncodingsIt)
{
if (!encoding.is_object())
MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.encodings");
this->rtpMapping.encodings.emplace_back();
auto& encodingMapping = this->rtpMapping.encodings.back();
// ssrc is optional.
auto jsonSsrcIt = encoding.find("ssrc");
// clang-format off
if (
jsonSsrcIt != encoding.end() &&
Utils::Json::IsPositiveInteger(*jsonSsrcIt)
)
// clang-format on
{
encodingMapping.ssrc = jsonSsrcIt->get<uint32_t>();
}
// rid is optional.
auto jsonRidIt = encoding.find("rid");
if (jsonRidIt != encoding.end() && jsonRidIt->is_string())
encodingMapping.rid = jsonRidIt->get<std::string>();
// However ssrc or rid must be present (if more than 1 encoding).
// clang-format off
if (
jsonEncodingsIt->size() > 1 &&
jsonSsrcIt == encoding.end() &&
jsonRidIt == encoding.end()
)
// clang-format on
{
MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.encodings (missing ssrc or rid)");
}
// If there is no mid and a single encoding, ssrc or rid must be present.
// clang-format off
if (
this->rtpParameters.mid.empty() &&
jsonEncodingsIt->size() == 1 &&
jsonSsrcIt == encoding.end() &&
jsonRidIt == encoding.end()
)
// clang-format on
{
MS_THROW_TYPE_ERROR(
"wrong entry in rtpMapping.encodings (missing ssrc or rid, or rtpParameters.mid)");
}
// mappedSsrc is mandatory.
auto jsonMappedSsrcIt = encoding.find("mappedSsrc");
// clang-format off
if (
jsonMappedSsrcIt == encoding.end() ||
!Utils::Json::IsPositiveInteger(*jsonMappedSsrcIt)
)
// clang-format on
{
MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.encodings (missing mappedSsrc)");
}
encodingMapping.mappedSsrc = jsonMappedSsrcIt->get<uint32_t>();
}
auto jsonPausedIt = data.find("paused");
if (jsonPausedIt != data.end() && jsonPausedIt->is_boolean())
this->paused = jsonPausedIt->get<bool>();
// The number of encodings in rtpParameters must match the number of encodings
// in rtpMapping.
if (this->rtpParameters.encodings.size() != this->rtpMapping.encodings.size())
{
MS_THROW_TYPE_ERROR("rtpParameters.encodings size does not match rtpMapping.encodings size");
}
// Fill RTP header extension ids.
// This may throw.
for (auto& exten : this->rtpParameters.headerExtensions)
{
if (exten.id == 0u)
MS_THROW_TYPE_ERROR("RTP extension id cannot be 0");
if (this->rtpHeaderExtensionIds.mid == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::MID)
{
this->rtpHeaderExtensionIds.mid = exten.id;
}
if (this->rtpHeaderExtensionIds.rid == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::RTP_STREAM_ID)
{
this->rtpHeaderExtensionIds.rid = exten.id;
}
if (this->rtpHeaderExtensionIds.rrid == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::REPAIRED_RTP_STREAM_ID)
{
this->rtpHeaderExtensionIds.rrid = exten.id;
}
if (this->rtpHeaderExtensionIds.absSendTime == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::ABS_SEND_TIME)
{
this->rtpHeaderExtensionIds.absSendTime = exten.id;
}
if (this->rtpHeaderExtensionIds.transportWideCc01 == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::TRANSPORT_WIDE_CC_01)
{
this->rtpHeaderExtensionIds.transportWideCc01 = exten.id;
}
// NOTE: Remove this once framemarking draft becomes RFC.
if (this->rtpHeaderExtensionIds.frameMarking07 == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::FRAME_MARKING_07)
{
this->rtpHeaderExtensionIds.frameMarking07 = exten.id;
}
if (this->rtpHeaderExtensionIds.frameMarking == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::FRAME_MARKING)
{
this->rtpHeaderExtensionIds.frameMarking = exten.id;
}
if (this->rtpHeaderExtensionIds.ssrcAudioLevel == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::SSRC_AUDIO_LEVEL)
{
this->rtpHeaderExtensionIds.ssrcAudioLevel = exten.id;
}
if (this->rtpHeaderExtensionIds.videoOrientation == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::VIDEO_ORIENTATION)
{
this->rtpHeaderExtensionIds.videoOrientation = exten.id;
}
if (this->rtpHeaderExtensionIds.toffset == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::TOFFSET)
{
this->rtpHeaderExtensionIds.toffset = exten.id;
}
}
3、Listener——使创建的Producer加入Router的侦听者列表
当RTC::Producer构造完成，返回Transport.cpp执行cpp this->rtpListener.AddProducer(producer);
这会把producer加入rtpListener，接着继续执行cpp this->listener->OnTransportNewProducer(this, producer);
这个listener是Router的侦听者，返回Router::OnTransportNewProducer
4、Router::OnTransportNewProducer——把producer加入Transport maps中
inline void Router::OnTransportNewProducer(RTC::Transport* /*transport*/, RTC::Producer* producer)
{
MS_TRACE();
MS_ASSERT(
this->mapProducerConsumers.find(producer) == this->mapProducerConsumers.end(),
"Producer already present in mapProducerConsumers");
if (this->mapProducers.find(producer->id) != this->mapProducers.end())
{
MS_THROW_ERROR("Producer already present in mapProducers [producerId:%s]", producer->id.c_str());
}
// Insert the Producer in the maps.
this->mapProducers[producer->id] = producer;
this->mapProducerConsumers[producer];
this->mapProducerRtpObservers[producer];
}
创建好producer之后，当有数据传来时，便可以根据ssrc找到它对应的producer，然后由producer查看哪些consumer订阅了它，接着就转发数据流了
四、创建Consumer
在Transport.cpp里进入下面的case
case Channel::Request::MethodId::TRANSPORT_CONSUME
在这个case里，会获取到producerId auto jsonProducerIdIt = request->internal.find("producerId");
然后还会获取到consumerId，不过它是调用的一个函数 SetNewConsumerIdFromInternal(request->internal, consumerId);
接着就开始创建consumer，由于它是多态的，所以会进入一个switch case，它有4种类型：
- SIMPLE：简单的consumer
- SIMULCAST：
- SVC
- PIPE
switch (type)
{
case RTC::RtpParameters::Type::NONE:
{
MS_THROW_TYPE_ERROR("invalid type 'none'");
break;
}
case RTC::RtpParameters::Type::SIMPLE:
{
// This may throw.
consumer = new RTC::SimpleConsumer(consumerId, producerId, this, request->data);
break;
}
case RTC::RtpParameters::Type::SIMULCAST:
{
// This may throw.
consumer = new RTC::SimulcastConsumer(consumerId, producerId, this, request->data);
break;
}
case RTC::RtpParameters::Type::SVC:
{
// This may throw.
consumer = new RTC::SvcConsumer(consumerId, producerId, this, request->data);
break;
}
case RTC::RtpParameters::Type::PIPE:
{
// This may throw.
consumer = new RTC::PipeConsumer(consumerId, producerId, this, request->data);
break;
}
}
1、RTC::SimpleConsumer——对data进行数据解析
以SIMPLE为例，进入RTC::SimpleConsumer
SimpleConsumer::SimpleConsumer(
const std::string& id, const std::string& producerId, RTC::Consumer::Listener* listener, json& data)
: RTC::Consumer::Consumer(id, producerId, listener, data, RTC::RtpParameters::Type::SIMPLE)
{
MS_TRACE();
// Ensure there is a single encoding.
if (this->consumableRtpEncodings.size() != 1u)
MS_THROW_TYPE_ERROR("invalid consumableRtpEncodings with size != 1");
auto& encoding = this->rtpParameters.encodings[0];
const auto* mediaCodec = this->rtpParameters.GetCodecForEncoding(encoding);
this->keyFrameSupported = RTC::Codecs::Tools::CanBeKeyFrame(mediaCodec->mimeType);
// Create RtpStreamSend instance for sending a single stream to the remote.
CreateRtpStream();
}
这段代码主要也是对data数据的解析，然后在代码的末尾段构造了一个Rtp数据流，进入CreateRtpStream()
2、CreateRtpStream()——根据data数据确定consumer内部细节
Consumer的本质就是一个Rtp数据流
void SimpleConsumer::CreateRtpStream()
{
MS_TRACE();
auto& encoding = this->rtpParameters.encodings[0];
const auto* mediaCodec = this->rtpParameters.GetCodecForEncoding(encoding);
MS_DEBUG_TAG(
rtp, "[ssrc:%" PRIu32 ", payloadType:%" PRIu8 "]", encoding.ssrc, mediaCodec->payloadType);
// Set stream params.
RTC::RtpStream::Params params;
params.ssrc = encoding.ssrc;
params.payloadType = mediaCodec->payloadType;
params.mimeType = mediaCodec->mimeType;
params.clockRate = mediaCodec->clockRate;
params.cname = this->rtpParameters.rtcp.cname;
// Check in band FEC in codec parameters.
if (mediaCodec->parameters.HasInteger("useinbandfec") && mediaCodec->parameters.GetInteger("useinbandfec") == 1)
{
MS_DEBUG_TAG(rtp, "in band FEC enabled");
params.useInBandFec = true;
}
// Check DTX in codec parameters.
if (mediaCodec->parameters.HasInteger("usedtx") && mediaCodec->parameters.GetInteger("usedtx") == 1)
{
MS_DEBUG_TAG(rtp, "DTX enabled");
params.useDtx = true;
}
// Check DTX in the encoding.
if (encoding.dtx)
{
MS_DEBUG_TAG(rtp, "DTX enabled");
params.useDtx = true;
}
for (const auto& fb : mediaCodec->rtcpFeedback)
{
if (!params.useNack && fb.type == "nack" && fb.parameter.empty())
{
MS_DEBUG_2TAGS(rtp, rtcp, "NACK supported");
params.useNack = true;
}
else if (!params.usePli && fb.type == "nack" && fb.parameter == "pli")
{
MS_DEBUG_2TAGS(rtp, rtcp, "PLI supported");
params.usePli = true;
}
else if (!params.useFir && fb.type == "ccm" && fb.parameter == "fir")
{
MS_DEBUG_2TAGS(rtp, rtcp, "FIR supported");
params.useFir = true;
}
}
// Create a RtpStreamSend for sending a single media stream.
size_t bufferSize = params.useNack ? 600u : 0u;
this->rtpStream = new RTC::RtpStreamSend(this, params, bufferSize);
this->rtpStreams.push_back(this->rtpStream);
// If the Consumer is paused, tell the RtpStreamSend.
if (IsPaused() || IsProducerPaused())
this->rtpStream->Pause();
const auto* rtxCodec = this->rtpParameters.GetRtxCodecForEncoding(encoding);
if (rtxCodec && encoding.hasRtx)
this->rtpStream->SetRtx(rtxCodec->payloadType, encoding.rtx.ssrc);
}
通过创建params获得了5个参数，分别是
- ssrc
- payloadType
- mimeType
- clockRate
- cname
以及后面通过if判断来确定是否使用FEC、DTX，以及Feedback丢包重传机制是使用NACK、PLI、FIR的哪一种
接着创建了一个RTC::RtpStreamSend对象，传入的三个参数分别是
- this——这个RtpStreamSend对象
- params——之前获得了5个参数
- bufferSize——缓存数组
创建好了这个对象后，就把它压入rtpStreams中保存起来
3、consumer小结
当共享者的数据源源不断地传过来，mediasoup就会根据它的ssrc找到对应的生产者，通过生产者再找到对应的消费者，接着再通过RtpStreamSend发送给用户