WebRTC源码之RTCPReceiver源码分析
WebRTC源码之RTCPReceiver源码分析
- WebRTC源码之RTCPReceiver源码分析
- 前言
- 一、 RTCP接受数据的流程的堆栈信息的
- 1、网络io 线程读取数据
- 2、 线程切换的代码
- 3、 线程切换 gcc
- 二、 RTCPReceiver::IncomingPacket方法读取RTCP数据的格式
- 1、 ParseCompoundPacket解析rtcp的头数据
- 2、 rtcp包有统一头格式读取方法
- 三、 rtcp::SenderReport::kPacketType（200）
- 1、SR数据包格式
- 2、 SR的数据解析
- 3、 RTCP report block 反馈包格式
- 4、RTCP report block 反馈包格式解析
- 5、 SR包信息保持处理
- 四、rtcp::ReceiverReport::kPacketType （201）
- 1、 RTCP receiver report包格式
- 2、读取数据包格式
- 3、 RR数据保持与SR包保持一样然后抛到GCC模块中去了
- 总结
WebRTC专题开嗨鸭 ！！！
一、 WebRTC 线程模型
1、WebRTC中线程模型和常见线程模型介绍
2、WebRTC网络PhysicalSocketServer之WSAEventselect模型使用
二、 WebRTC媒体协商
1、WebRTC媒体协商之SDP中JsepSessionDescription类结构分析
2、WebRTC媒体协商之CreatePeerConnectionFactory、CreatePeerConnection、CreateOffer
3、WebRTC之证书(certificate)生成的时机分析
4、WebRTC源码之RtpTransceiver添加视频轨道的AddTrack函数中桥接模式的流程分析
三、 WebRTC 音频数据采集
1、WebRTC源码之音频设备播放流程源码分析
2、WebRTC源码之音频设备的录制流程源码分析
四、 WebRTC 音频引擎(编解码和3A算法)
五、 WebRTC 视频数据采集
六、 WebRTC 视频引擎( 编解码)
七、 WebRTC 网络传输
1、WebRTC的ICE之STUN协议
2、WebRTC的ICE之Dtls/SSL/TLSv1.x协议详解
八、 WebRTC服务质量(Qos)
1、WebRTC中RTCP协议详解
2、WebRTC中RTP协议详解
3、WebRTC之NACK、RTX 在什么时机判断丢包发送NACK请求和RTX丢包重传
4、WebRTC源码之视频质量统计数据的数据结构分析
5、WebRTC源码之RTCPReceiver源码分析
九、 NetEQ
十、 Simulcast与SVC
前言
WebRTC是音视频行业的标杆， 如果要学习音视频， WebRTC是进入音视频行业最好方法， 里面可以有成熟方案， 例如：音频中3A 算法、网络评估、自适应码流、Simulcast、SVC等等 ， 非常适合刚刚进入音视频行业小伙伴哈_ 我也是哦， 以后再音视频行业长期打算的小伙伴的学习项目。 里面有大量知识点
提示：以下是本篇文章正文内容，下面案例可供参考
一、 RTCP接受数据的流程的堆栈信息的
1、网络io 线程读取数据
AllocationSequence::OnReadPacket(rtc::AsyncPacketSocket* socket, const char* data, size_t size, const rtc::SocketAddress& remote_addr, const int64_t& packet_time_us)
UDPPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket, const char* data, size_t size, const rtc::SocketAddress& remote_addr, int64_t packet_time_us)
UDPPort::OnReadPacket(rtc::AsyncPacketSocket* socket, const char* data, size_t size, const rtc::SocketAddress& remote_addr, const int64_t& packet_time_us)
onnection::OnReadPacket(const char* data, size_t size, int64_t packet_time_us)
P2PTransportChannel::OnReadPacket(Connection* connection, const char* data, size_t len, int64_t packet_time_us)
DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport, const char* data, size_t size, const int64_t& packet_time_us, int flags)
RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport, const char* data, size_t len, const int64_t& packet_time_us, int flags)
SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us)
RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us)
RtpDemuxer::OnRtpPacket(const RtpPacketReceived& packet)
BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet)
BaseChannel::OnPacketReceived(bool rtcp, const rtc::CopyOnWriteBuffer& packet, int64_t packet_time_us)
2、 线程切换的代码
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time_us));
]
// received
BaseChannel::ProcessPacket(bool rtcp, const rtc::CopyOnWriteBuffer& packet, int64_t packet_time_us)
WebRtcVideoChannel::OnRtcpReceived
Call::DeliverPacket
Call::DeliverRtcp
VideoSendStream::DeliverRtcp
VideoSendStreamImpl::DeliverRtcp
RtpVideoSender::DeliverRtcp
ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet, const size_t length)
RTCPReceiver::IncomingPacket(const uint8_t* packet, size_t packet_size) --> RTCPReceiver模块接受数据并读取数据格式
RTCPReceiver::TriggerCallbacksFromRtcpPacket(const PacketInformation& packet_information)
RtpTransportControllerSend::OnTransportFeedback(const rtcp::TransportFeedback& feedback)
3、 线程切换 gcc
GoogCcNetworkController::OnTransportPacketsFeedback(TransportPacketsFeedback report)
二、 RTCPReceiver::IncomingPacket方法读取RTCP数据的格式
1、 ParseCompoundPacket解析rtcp的头数据
void RTCPReceiver::IncomingPacket(const uint8_t* packet, size_t packet_size) {
if (packet_size == 0) {
RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
return;
}
// TODO@chensong 20220909 根据对端反馈信息处理
// TODO@chensong 2022-10-19 解析RTCP 数据包的格式
PacketInformation packet_information;
if (!ParseCompoundPacket(packet, packet + packet_size, &packet_information)) {
return;
}
TriggerCallbacksFromRtcpPacket(packet_information);
}
CommonHeader rtcp_block;
for (const uint8_t* next_block = packet_begin; next_block != packet_end; next_block = rtcp_block.NextPacket())
{
ptrdiff_t remaining_blocks_size = packet_end - next_block;
RTC_DCHECK_GT(remaining_blocks_size, 0);
//rtcp包有统一头格式读取方法
if (!rtcp_block.Parse(next_block, remaining_blocks_size))
{
if (next_block == packet_begin)
{
// Failed to parse 1st header, nothing was extracted from this packet.
RTC_LOG(LS_WARNING) << "Incoming invalid RTCP packet";
return false;
}
++num_skipped_packets_;
break;
}
***
}
2、 rtcp包有统一头格式读取方法
// webrtc\src\modules\rtp_rtcp\source\rtcp_packet\common_header.cc
// 0 1 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 0 |V=2|P| C/F |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 1 | Packet Type |
// ----------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 2 | length |
// --------------------------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// TODO@chensong 2022-12-25
// Common header for all RTCP packets, 4 octets.
bool CommonHeader::Parse(const uint8_t* buffer, size_t size_bytes)
{
const uint8_t kVersion = 2;
if (size_bytes < kHeaderSizeBytes)
{
RTC_LOG(LS_WARNING)
<< "Too little data (" << size_bytes << " byte"
<< (size_bytes != 1 ? "s" : "")
<< ") remaining in buffer to parse RTCP header (4 bytes).";
return false;
}
// rtcp 版本
uint8_t version = buffer[0] >> 6;
if (version != kVersion)
{
RTC_LOG(LS_WARNING) << "Invalid RTCP header: Version must be "
<< static_cast<int>(kVersion) << " but was "
<< static_cast<int>(version);
return false;
}
// 是否有扩展的数据包
bool has_padding = (buffer[0] & 0x20) != 0;
count_or_format_ = buffer[0] & 0x1F;
// rtcp 包类型
packet_type_ = buffer[1];
// rtcp 包中数据大小 读取4个字节 就是32bit
payload_size_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) * 4;
// rtcp 包中实际数据开始地址的位置
payload_ = buffer + kHeaderSizeBytes /*default kHeaderSizeBytes = 4*/;
padding_size_ = 0;
if (size_bytes < kHeaderSizeBytes + payload_size_)
{
RTC_LOG(LS_WARNING) << "Buffer too small (" << size_bytes
<< " bytes) to fit an RtcpPacket with a header and "
<< payload_size_ << " bytes.";
return false;
}
if (has_padding)
{
if (payload_size_ == 0)
{
RTC_LOG(LS_WARNING)
<< "Invalid RTCP header: Padding bit set but 0 payload "
"size specified.";
return false;
}
padding_size_ = payload_[payload_size_ - 1];
if (padding_size_ == 0)
{
RTC_LOG(LS_WARNING)
<< "Invalid RTCP header: Padding bit set but 0 padding "
"size specified.";
return false;
}
if (padding_size_ > payload_size_)
{
RTC_LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes ("
<< padding_size_ << ") for a packet payload size of "
<< payload_size_ << " bytes.";
return false;
}
payload_size_ -= padding_size_;
}
return true;
}
然后根据packet_type的不同类型进行处理
三、 rtcp::SenderReport::kPacketType（200）
1、SR数据包格式
Sender report (SR) (RFC 3550).
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P| RC | PT=SR=200 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
0 | SSRC of sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
4 | NTP timestamp, most significant word |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
8 | NTP timestamp, least significant word |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
12 | RTP timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
16 | sender's packet count |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
20 | sender's octet count |
24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
2、 SR的数据解析
bool SenderReport::Parse(const CommonHeader& packet)
{
RTC_DCHECK_EQ(packet.type(), kPacketType);
const uint8_t report_block_count = packet.count();
if (packet.payload_size_bytes() < kSenderBaseLength/*24*/ + report_block_count * ReportBlock::kLength /*24*/)
{
RTC_LOG(LS_WARNING) << "Packet is too small to contain all the data.";
return false;
}
// Read SenderReport header.
const uint8_t* const payload = packet.payload();
// 发送端ssrc
sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
uint32_t secs = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
uint32_t frac = ByteReader<uint32_t>::ReadBigEndian(&payload[8]);
ntp_.Set(secs, frac);
// rtp 网络时间戳
rtp_timestamp_ = ByteReader<uint32_t>::ReadBigEndian(&payload[12]);
// 发送的总包数
sender_packet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[16]);
// 总共发送数据包量
sender_octet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[20]);
report_blocks_.resize(report_block_count);
const uint8_t* next_block = payload + kSenderBaseLength;
for (ReportBlock& block : report_blocks_)
{
bool block_parsed = block.Parse(next_block, ReportBlock::kLength);
RTC_DCHECK(block_parsed);
next_block += ReportBlock::kLength;
}
// Double check we didn't read beyond provided buffer.
RTC_DCHECK_LE(next_block - payload, static_cast<ptrdiff_t>(packet.payload_size_bytes()));
return true;
}
3、 RTCP report block 反馈包格式
From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
RTCP report block (RFC 3550).
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
0 | SSRC_1 (SSRC of first source) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4 | fraction lost | cumulative number of packets lost |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
8 | extended highest sequence number received |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
12 | interarrival jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
16 | last SR (LSR) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
20 | delay since last SR (DLSR) |
24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
4、RTCP report block 反馈包格式解析
bool ReportBlock::Parse(const uint8_t* buffer, size_t length)
{
RTC_DCHECK(buffer != nullptr);
if (length < ReportBlock::kLength)
{
RTC_LOG(LS_ERROR) << "Report Block should be 24 bytes long";
return false;
}
// 接收到的媒体源ssrc
source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
// TODO@chensong 2022-10-19 丢包率 fraction_lost
fraction_lost_ = buffer[4];
// 接收开始丢包总数， 迟到包不算丢包，重传有可以导致负数
cumulative_lost_ = ByteReader<int32_t, 3>::ReadBigEndian(&buffer[5]);
// 低16位表示收到的最大seq，高16位表示seq循环次数
extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
// rtp包到达时间间隔的统计方差
jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
// ntp时间戳的中间32位
last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
// 记录上一个接收SR的时间与上一个发送SR的时间差
delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]);
return true;
}
5、 SR包信息保持处理
大致是包RTCP report block包数据进行再加工 重要处理rtt, 然后把 RTCP report blocb包全部数据包抛到gcc模块中去
void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block, PacketInformation* packet_information)
{
rtcp::SenderReport sender_report;
if (!sender_report.Parse(rtcp_block))
{
++num_skipped_packets_;
return;
}
const uint32_t remote_ssrc = sender_report.sender_ssrc();
packet_information->remote_ssrc = remote_ssrc;
UpdateTmmbrRemoteIsAlive(remote_ssrc);
// Have I received RTP packets from this party?
if (remote_ssrc_ == remote_ssrc)
{
// Only signal that we have received a SR when we accept one.
packet_information->packet_type_flags |= kRtcpSr;
remote_sender_ntp_time_ = sender_report.ntp();
remote_sender_rtp_time_ = sender_report.rtp_timestamp();
last_received_sr_ntp_ = TimeMicrosToNtp(clock_->TimeInMicroseconds());
}
else
{
// We will only store the send report from one source, but
// we will store all the receive blocks.
packet_information->packet_type_flags |= kRtcpRr;
}
for (const rtcp::ReportBlock& report_block : sender_report.report_blocks())
{
HandleReportBlock(report_block, packet_information, remote_ssrc);
}
}
void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block, PacketInformation* packet_information, uint32_t remote_ssrc)
{
// This will be called once per report block in the RTCP packet.
// We filter out all report blocks that are not for us.
// Each packet has max 31 RR blocks.
//
// We can calc RTT if we send a send report and get a report block back.
// |report_block.source_ssrc()| is the SSRC identifier of the source to
// which the information in this reception report block pertains.
// Filter out all report blocks that are not for us.
// TODO@chensong 2022-12-26
//这将在RTCP数据包中的每个报告块调用一次。
//我们过滤掉所有不适合我们的报告块。
//每个数据包最多有31个RR块。
//如果我们发送发送报告并得到报告块，我们可以计算RTT。
//|report_block.source_ssrc（）|是源的ssrc标识符
//该接收报告块中的信息与之相关。
//过滤掉所有不适合我们的报告块。
if (registered_ssrcs_.count(report_block.source_ssrc()) == 0)
{
return;
}
last_received_rb_ms_ = clock_->TimeInMilliseconds();
// TODO@chensong 2022-12-26 没有该ssrc在received_report_blocks_中map中正好插入
ReportBlockWithRtt* report_block_info = &received_report_blocks_[report_block.source_ssrc()][remote_ssrc];
report_block_info->report_block.sender_ssrc = remote_ssrc;
report_block_info->report_block.source_ssrc = report_block.source_ssrc();
report_block_info->report_block.fraction_lost = report_block.fraction_lost();
report_block_info->report_block.packets_lost = report_block.cumulative_lost_signed();
if (report_block.extended_high_seq_num() > report_block_info->report_block.extended_highest_sequence_number)
{
// We have successfully delivered new RTP packets to the remote side after
// the last RR was sent from the remote side.
last_increased_sequence_number_ms_ = clock_->TimeInMilliseconds();
}
report_block_info->report_block.extended_highest_sequence_number = report_block.extended_high_seq_num();
report_block_info->report_block.jitter = report_block.jitter();
report_block_info->report_block.delay_since_last_sender_report = report_block.delay_since_last_sr();
report_block_info->report_block.last_sender_report_timestamp = report_block.last_sr();
int64_t rtt_ms = 0;
uint32_t send_time_ntp = report_block.last_sr();
// RFC3550, section 6.4.1, LSR field discription states:
// If no SR has been received yet, the field is set to zero.
// Receiver rtp_rtcp module is not expected to calculate rtt using
// Sender Reports even if it accidentally can.
// TODO(nisse): Use this way to determine the RTT only when |receiver_only_|
// is false. However, that currently breaks the tests of the
// googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
// delete all dependencies on RTT measurements for audio receive streams, or
// ensure that audio receive streams that need RTT and stats that depend on it
// are configured with an associated audio send stream.
if (send_time_ntp != 0)
{
uint32_t delay_ntp = report_block.delay_since_last_sr();
// Local NTP time.
// 微妙
uint32_t receive_time_ntp = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
// RTT in 1/(2^16) seconds.
uint32_t rtt_ntp = receive_time_ntp - delay_ntp /*发送时间与接收到时间差值*/ - send_time_ntp;
// Convert to 1/1000 seconds (milliseconds).
// 微妙转换 毫秒级
rtt_ms = CompactNtpRttToMs(rtt_ntp);
if (rtt_ms > report_block_info->max_rtt_ms)
{
report_block_info->max_rtt_ms = rtt_ms;
}
if (report_block_info->num_rtts == 0 || rtt_ms < report_block_info->min_rtt_ms)
{
report_block_info->min_rtt_ms = rtt_ms;
}
report_block_info->last_rtt_ms = rtt_ms;
report_block_info->sum_rtt_ms += rtt_ms;
++report_block_info->num_rtts;
packet_information->rtt_ms = rtt_ms;
}
packet_information->report_blocks.push_back(report_block_info->report_block);
}
四、rtcp::ReceiverReport::kPacketType （201）
1、 RTCP receiver report包格式
RTCP receiver report (RFC 3550).
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P| RC | PT=RR=201 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of packet sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| report block(s) |
| .... |
2、读取数据包格式
bool ReceiverReport::Parse(const CommonHeader& packet)
{
RTC_DCHECK_EQ(packet.type(), kPacketType);
const uint8_t report_blocks_count = packet.count();
if (packet.payload_size_bytes() < kRrBaseLength + report_blocks_count * ReportBlock::kLength)
{
RTC_LOG(LS_WARNING) << "Packet is too small to contain all the data.";
return false;
}
// 发送者ssrc
sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(packet.payload());
const uint8_t* next_report_block = packet.payload() + kRrBaseLength;
report_blocks_.resize(report_blocks_count);
for (ReportBlock& block : report_blocks_)
{
block.Parse(next_report_block, ReportBlock::kLength);
next_report_block += ReportBlock::kLength;
}
RTC_DCHECK_LE(next_report_block - packet.payload(),
static_cast<ptrdiff_t>(packet.payload_size_bytes()));
return true;
}
3、 RR数据保持与SR包保持一样然后抛到GCC模块中去了
void RTCPReceiver::HandleReceiverReport(const CommonHeader& rtcp_block, PacketInformation* packet_information)
{
rtcp::ReceiverReport receiver_report;
if (!receiver_report.Parse(rtcp_block))
{
++num_skipped_packets_;
return;
}
const uint32_t remote_ssrc = receiver_report.sender_ssrc();
packet_information->remote_ssrc = remote_ssrc;
UpdateTmmbrRemoteIsAlive(remote_ssrc);
packet_information->packet_type_flags |= kRtcpRr;
for (const ReportBlock& report_block : receiver_report.report_blocks())
{
// TODO@chensong 2022-12-26 和SR包反馈包的处理流程一样的然后抛到GCC模块中去了
HandleReportBlock(report_block, packet_information, remote_ssrc);
}
}
总结
WebRTC源码分析地址：https://github.com/chensongpoixs/cwebrtc
