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ZLMediaKit
Commit 307be433 by ziyue
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Merge branch 'dev' of https://github.com/xia-chu/ZLMediaKit into dev_test

parents 8717cd2f 0eecb3be
正在显示 包含 2962 行增加 和 127 行删除
.gitignore
......@@ -46,3 +46,4 @@
/3rdpart/media-server/.idea/
/ios/
/cmake-build-*
/3rdpart/ZLToolKit/cmake-build-mq/
3rdpart/assert.h
......@@ -12,19 +12,22 @@
#define ZLMEDIAKIT_ASSERT_H
#include <stdio.h>
#ifdef __cplusplus
extern "C" {
#endif
extern void Assert_Throw(int failed, const char *exp, const char *func, const char *file, int line);
#ifdef __cplusplus
}
#endif
#define CHECK(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__)
#ifndef NDEBUG
#ifdef assert
#undef assert
#endif//assert
#ifdef __cplusplus
extern "C" {
#endif
extern void Assert_Throw(int failed, const char *exp, const char *func, const char *file, int line);
#ifdef __cplusplus
}
#endif
#define assert(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__);
#else
#define assert(e) ((void)0)
......
CMakeLists.txt
......@@ -55,6 +55,7 @@ option(ENABLE_TESTS "Enable Tests" true)
option(ENABLE_SERVER "Enable Server" true)
option(ENABLE_MEM_DEBUG "Enable Memory Debug" false)
option(ENABLE_ASAN "Enable Address Sanitize" false)
option(ENABLE_WEBRTC "Enable WebRTC" true)
if (ENABLE_MEM_DEBUG)
set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -Wl,-wrap,free -Wl,-wrap,malloc -Wl,-wrap,realloc -Wl,-wrap,calloc")
......@@ -86,6 +87,10 @@ if (OPENSSL_FOUND AND ENABLE_OPENSSL)
include_directories(${OPENSSL_INCLUDE_DIR})
add_definitions(-DENABLE_OPENSSL)
list(APPEND LINK_LIB_LIST ${OPENSSL_LIBRARIES})
if (CMAKE_SYSTEM_NAME MATCHES "Linux" AND OPENSSL_USE_STATIC_LIBS)
list(APPEND LINK_LIB_LIST dl)
endif()
else()
message(WARNING "openssl未找到，rtmp将不支持flash播放器，https/wss/rtsps/rtmps也将失效")
endif ()
......@@ -224,6 +229,24 @@ if(ENABLE_API)
add_subdirectory(api)
endif()
if (ENABLE_WEBRTC)
#查找srtp是否安装
find_package(SRTP QUIET)
if (SRTP_FOUND)
message(STATUS "found library:${SRTP_LIBRARIES}")
include_directories(${SRTP_INCLUDE_DIRS})
list(APPEND LINK_LIB_LIST ${SRTP_LIBRARIES})
add_definitions(-DENABLE_WEBRTC)
include_directories(./webrtc)
file(GLOB SRC_WEBRTC_LIST ./webrtc/*.cpp ./webrtc/*.h ./webrtc/*.hpp)
add_library(webrtc ${SRC_WEBRTC_LIST})
list(APPEND LINK_LIB_LIST webrtc)
else ()
message(WARNING "srtp未找到, webrtc相关功能打开失败")
endif ()
endif ()
if (NOT IOS)
#测试程序
if(ENABLE_TESTS)
......
cmake/FindSRTP.cmake 0 → 100644
############################################################################
# FindSRTP.txt
# Copyright (C) 2014 Belledonne Communications, Grenoble France
#
############################################################################
#
# This program is free software; you can redistribute it and/or
# modify it under the terms of the GNU General Public License
# as published by the Free Software Foundation; either version 2
# of the License, or (at your option) any later version.
#
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU General Public License for more details.
#
# You should have received a copy of the GNU General Public License
# along with this program; if not, write to the Free Software
# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#
############################################################################
#
# - Find the SRTP include file and library
#
# SRTP_FOUND - system has SRTP
# SRTP_INCLUDE_DIRS - the SRTP include directory
# SRTP_LIBRARIES - The libraries needed to use SRTP
set(_SRTP_ROOT_PATHS
${CMAKE_INSTALL_PREFIX}
)
find_path(SRTP_INCLUDE_DIRS
NAMES srtp2/srtp.h
HINTS _SRTP_ROOT_PATHS ${SRTP_PREFIX}
PATH_SUFFIXES include
)
if(SRTP_INCLUDE_DIRS)
set(HAVE_SRTP_SRTP_H 1)
endif()
find_library(SRTP_LIBRARIES
NAMES srtp2
HINTS ${_SRTP_ROOT_PATHS} ${SRTP_PREFIX}
PATH_SUFFIXES bin lib
)
include(FindPackageHandleStandardArgs)
find_package_handle_standard_args(SRTP
DEFAULT_MSG
SRTP_INCLUDE_DIRS SRTP_LIBRARIES HAVE_SRTP_SRTP_H
)
mark_as_advanced(SRTP_INCLUDE_DIRS SRTP_LIBRARIES HAVE_SRTP_SRTP_H)
\ No newline at end of file
conf/config.ini
......@@ -140,7 +140,7 @@ charSet=utf-8
#http链接超时时间
keepAliveSecond=30
#http请求体最大字节数，如果post的body太大，则不适合缓存body在内存
maxReqSize=4096
maxReqSize=40960
#404网页内容，用户可以自定义404网页
notFound=<html><head><title>404 Not Found</title></head><body bgcolor="white"><center><h1>您访问的资源不存在！</h1></center><hr><center>ZLMediaKit-4.0</center></body></html>
#http服务器监听端口
......@@ -210,13 +210,23 @@ port=10000
#rtp超时时间，单位秒
timeoutSec=15
[rtc]
#rtc播放推流、播放超时时间
timeoutSec=15
#本机对rtc客户端的可见ip，作为服务器时一般为公网ip，置空时，会自动获取网卡ip
externIP=
#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
rembBitRate=1000000
[rtsp]
#rtsp专有鉴权方式是采用base64还是md5方式
authBasic=0
#rtsp拉流代理是否是直接代理模式
#rtsp拉流、推流代理是否是直接代理模式
#直接代理后支持任意编码格式，但是会导致GOP缓存无法定位到I帧，可能会导致开播花屏
#并且如果是tcp方式拉流，如果rtp大于mtu会导致无法使用udp方式代理
#假定您的拉流源地址不是264或265或AAC，那么你可以使用直接代理的方式来支持rtsp代理
#如果你是rtsp推拉流，但是rtc播放，也建议关闭直接代理模式，
#因为直接代理时，rtp中可能没有sps pps,会导致rtc无法播放
#默认开启rtsp直接代理，rtmp由于没有这些问题，是强制开启直接代理的
directProxy=1
#rtsp必须在此时间内完成握手，否则服务器会断开链接，单位秒
......
package/rpm/ZLMediaKit.spec
......@@ -2,10 +2,12 @@
%global use_devtoolset 0
%bcond_without faac
%bcond_without x264
%bcond_without webrtc
%else
%global use_devtoolset 1
%bcond_with faac
%bcond_with x264
%bcond_with webrtc
%endif
%bcond_without openssl
......@@ -22,8 +24,12 @@ URL: https://github.com/xia-chu/ZLMediaKit
Source0: %{name}-%{version}.tar.xz
%if %{with openssl}
%if 0%{?rhel} <= 7 && %{with webrtc}
BuildRequires: openssl11-devel
%else
BuildRequires: openssl-devel
%endif
%endif
%if %{with mysql}
BuildRequires: mysql-devel
......@@ -37,6 +43,10 @@ BuildRequires: faac-devel
BuildRequires: x264-devel
%endif
%if %{with webrtc}
BuildRequires: libsrtp-devel >= 2.0
%endif
%if 0%{?use_devtoolset}
BuildRequires: devtoolset-8-gcc-c++
%endif
......@@ -88,6 +98,10 @@ pushd %{_target_platform}
-DENABLE_MYSQL:BOOL=%{with mysql} \
-DENABLE_FAAC:BOOL=%{with faac} \
-DENABLE_X264:BOOL=%{with x264} \
-DENABLE_WEBRTC:BOOL=%{with webrtc} \
%if %{with webrtc} && 0%{?rhel} <= 7
-DOPENSSL_ROOT_DIR:STRING="/usr/lib64/openssl11;/usr/include/openssl11" \
%endif
-DENABLE_MP4:BOOL=ON \
-DENABLE_RTPPROXY:BOOL=ON \
-DENABLE_API:BOOL=ON \
......
server/CMakeLists.txt
......@@ -48,4 +48,8 @@ else()
install(TARGETS MediaServer DESTINATION ${INSTALL_PATH_EXECUTABLE})
endif()
target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST})
if (CMAKE_SYSTEM_NAME MATCHES "Linux")
target_link_libraries(MediaServer -Wl,--start-group jsoncpp ${LINK_LIB_LIST} -Wl,--end-group)
else ()
target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST})
endif ()
server/WebApi.cpp
......@@ -37,6 +37,9 @@
#if defined(ENABLE_RTPPROXY)
#include "Rtp/RtpServer.h"
#endif
#ifdef ENABLE_WEBRTC
#include "../webrtc/WebRtcTransport.h"
#endif
using namespace toolkit;
using namespace mediakit;
......@@ -126,6 +129,26 @@ static HttpApi toApi(const function<void(API_ARGS_JSON)> &cb) {
});
}
static HttpApi toApi(const function<void(API_ARGS_STRING_ASYNC)> &cb) {
return [cb](const Parser &parser, const HttpSession::HttpResponseInvoker &invoker, SockInfo &sender) {
GET_CONFIG(string, charSet, Http::kCharSet);
HttpSession::KeyValue headerOut;
headerOut["Content-Type"] = string("application/json; charset=") + charSet;
Json::Value val;
val["code"] = API::Success;
cb(sender, parser.getHeader(), headerOut, parser, val, invoker);
};
}
static HttpApi toApi(const function<void(API_ARGS_STRING)> &cb) {
return toApi([cb](API_ARGS_STRING_ASYNC) {
cb(API_ARGS_VALUE);
invoker(200, headerOut, val.toStyledString());
});
}
void api_regist(const string &api_path, const function<void(API_ARGS_MAP)> &func) {
s_map_api.emplace(api_path, toApi(func));
}
......@@ -142,6 +165,14 @@ void api_regist(const string &api_path, const function<void(API_ARGS_JSON_ASYNC)
s_map_api.emplace(api_path, toApi(func));
}
void api_regist(const string &api_path, const function<void(API_ARGS_STRING)> &func){
s_map_api.emplace(api_path, toApi(func));
}
void api_regist(const string &api_path, const function<void(API_ARGS_STRING_ASYNC)> &func){
s_map_api.emplace(api_path, toApi(func));
}
//获取HTTP请求中url参数、content参数
static ApiArgsType getAllArgs(const Parser &parser) {
ApiArgsType allArgs;
......@@ -380,8 +411,15 @@ void installWebApi() {
int changed = API::Success;
for (auto &pr : allArgs) {
if (ini.find(pr.first) == ini.end()) {
#if 1
//没有这个key
continue;
#else
// 新增配置选项,为了动态添加多个ffmpeg cmd 模板
ini[pr.first] = pr.second;
// 防止changed变化
continue;
#endif
}
if (ini[pr.first] == pr.second) {
continue;
......@@ -1161,6 +1199,90 @@ void installWebApi() {
#endif
});
#ifdef ENABLE_WEBRTC
api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){
CHECK_ARGS("app", "stream");
auto offer_sdp = allArgs.Content();
auto type = allArgs.getUrlArgs()["type"];
MediaInfo info(StrPrinter << "rtc://" << headerIn["Host"] << "/" << allArgs.getUrlArgs()["app"] << "/" << allArgs.getUrlArgs()["stream"] << "?" << allArgs.Params());
//设置返回类型
headerOut["Content-Type"] = HttpFileManager::getContentType(".json");
//设置跨域
headerOut["Access-Control-Allow-Origin"] = "*";
if (type.empty() || !strcasecmp(type.data(), "play")) {
Broadcast::AuthInvoker authInvoker = [invoker, offer_sdp, val, info, headerOut](const string &err) mutable {
try {
auto src = dynamic_pointer_cast<RtspMediaSource>(MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid));
if (!src) {
throw runtime_error("流不存在");
}
if (!err.empty()) {
throw runtime_error(StrPrinter << "播放鉴权失败:" << err);
}
auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller());
rtc->attach(src, info, true);
val["sdp"] = rtc->getAnswerSdp(offer_sdp);
val["type"] = "answer";
invoker(200, headerOut, val.toStyledString());
} catch (std::exception &ex) {
val["code"] = API::Exception;
val["msg"] = ex.what();
invoker(200, headerOut, val.toStyledString());
}
};
//广播通用播放url鉴权事件
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPlayed, info, authInvoker, sender);
if (!flag) {
//该事件无人监听,默认不鉴权
authInvoker("");
}
return;
}
if (!strcasecmp(type.data(), "push")) {
Broadcast::PublishAuthInvoker authInvoker = [invoker, offer_sdp, val, info, headerOut](const string &err, bool enableHls, bool enableMP4) mutable {
try {
auto src = dynamic_pointer_cast<RtspMediaSource>(MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid));
if (src) {
throw std::runtime_error("已经在推流");
}
if (!err.empty()) {
throw runtime_error(StrPrinter << "推流鉴权失败:" << err);
}
auto push_src = std::make_shared<RtspMediaSourceImp>(info._vhost, info._app, info._streamid);
push_src->setProtocolTranslation(enableHls, enableMP4);
auto rtc = WebRtcTransportImp::create(EventPollerPool::Instance().getPoller());
push_src->setListener(rtc);
rtc->attach(push_src, info, false);
val["sdp"] = rtc->getAnswerSdp(offer_sdp);
val["type"] = "answer";
invoker(200, headerOut, val.toStyledString());
} catch (std::exception &ex) {
val["code"] = API::Exception;
val["msg"] = ex.what();
invoker(200, headerOut, val.toStyledString());
}
};
//rtsp推流需要鉴权
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish, info, authInvoker, sender);
if (!flag) {
//该事件无人监听,默认不鉴权
GET_CONFIG(bool, toHls, General::kPublishToHls);
GET_CONFIG(bool, toMP4, General::kPublishToMP4);
authInvoker("", toHls, toMP4);
}
return;
}
throw ApiRetException("不支持该类型", API::InvalidArgs);
});
#endif
////////////以下是注册的Hook API////////////
api_regist("/index/hook/on_publish",[](API_ARGS_MAP){
//开始推流事件
......
server/WebApi.h
......@@ -85,6 +85,8 @@ using ApiArgsType = map<string, variant, StrCaseCompare>;
#define API_ARGS_MAP_ASYNC API_ARGS_MAP, const HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_JSON SockInfo &sender, HttpSession::KeyValue &headerIn, HttpSession::KeyValue &headerOut, Json::Value &allArgs, Json::Value &val
#define API_ARGS_JSON_ASYNC API_ARGS_JSON, const HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_STRING SockInfo &sender, HttpSession::KeyValue &headerIn, HttpSession::KeyValue &headerOut, const Parser &allArgs, Json::Value &val
#define API_ARGS_STRING_ASYNC API_ARGS_STRING, const HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_VALUE sender, headerIn, headerOut, allArgs, val
//注册http请求参数是map<string, variant, StrCaseCompare>类型的http api
......@@ -97,6 +99,11 @@ void api_regist(const string &api_path, const function<void(API_ARGS_JSON)> &fun
//注册http请求参数是Json::Value类型，但是可以异步回复的的http api
void api_regist(const string &api_path, const function<void(API_ARGS_JSON_ASYNC)> &func);
//注册http请求参数是http原始请求信息的http api
void api_regist(const string &api_path, const function<void(API_ARGS_STRING)> &func);
//注册http请求参数是http原始请求信息的异步回复的http api
void api_regist(const string &api_path, const function<void(API_ARGS_STRING_ASYNC)> &func);
template<typename Args, typename First>
bool checkArgs(Args &&args, First &&first) {
return !args[first].empty();
......@@ -107,6 +114,16 @@ bool checkArgs(Args &&args, First &&first, KeyTypes &&...keys) {
return !args[first].empty() && checkArgs(std::forward<Args>(args), std::forward<KeyTypes>(keys)...);
}
template<typename First>
bool checkArgs(const Parser &args, First &&first) {
return !args.getUrlArgs()[first].empty();
}
template<typename First, typename ...KeyTypes>
bool checkArgs(const Parser &args, First &&first, KeyTypes &&...keys) {
return !args.getUrlArgs()[first].empty() && checkArgs(args, std::forward<KeyTypes>(keys)...);
}
//检查http参数是否为空的宏
#define CHECK_ARGS(...) \
if(!checkArgs(allArgs,##__VA_ARGS__)){ \
......
server/main.cpp
......@@ -25,6 +25,7 @@
#include "Rtp/RtpServer.h"
#include "WebApi.h"
#include "WebHook.h"
#include "../webrtc/Sdp.h"
#if defined(ENABLE_VERSION)
#include "Version.h"
......
src/Common/MediaSource.cpp
......@@ -35,6 +35,7 @@ string getOriginTypeString(MediaOriginType type){
SWITCH_CASE(ffmpeg_pull);
SWITCH_CASE(mp4_vod);
SWITCH_CASE(device_chn);
SWITCH_CASE(rtc_push);
default : return "unknown";
}
}
......
src/Common/MediaSource.h
......@@ -45,7 +45,8 @@ enum class MediaOriginType : uint8_t {
pull,
ffmpeg_pull,
mp4_vod,
device_chn
device_chn,
rtc_push,
};
string getOriginTypeString(MediaOriginType type);
......
src/Extension/AAC.cpp
......@@ -205,7 +205,7 @@ public:
if (bitrate) {
_printer << "b=AS:" << bitrate << "\r\n";
}
_printer << "a=rtpmap:" << payload_type << " mpeg4-generic/" << sample_rate << "/" << channels << "\r\n";
_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n";
string configStr;
char buf[4] = {0};
......
src/Extension/CommonRtp.cpp
......@@ -69,7 +69,7 @@ CommonRtpEncoder::CommonRtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_si
}
void CommonRtpEncoder::inputFrame(const Frame::Ptr &frame){
auto stamp = frame->dts();
auto stamp = frame->pts();
auto ptr = frame->data() + frame->prefixSize();
auto len = frame->size() - frame->prefixSize();
auto remain_size = len;
......
src/Extension/Frame.cpp
......@@ -11,7 +11,7 @@
#include "Frame.h"
#include "H264.h"
#include "H265.h"
#include "Common/Parser.h"
using namespace std;
using namespace toolkit;
......@@ -106,30 +106,50 @@ Frame::Ptr Frame::getCacheAbleFrame(const Frame::Ptr &frame){
return std::make_shared<FrameCacheAble>(frame);
}
#define SWITCH_CASE(codec_id) case codec_id : return #codec_id
const char *getCodecName(CodecId codecId) {
TrackType getTrackType(CodecId codecId) {
switch (codecId) {
SWITCH_CASE(CodecH264);
SWITCH_CASE(CodecH265);
SWITCH_CASE(CodecAAC);
SWITCH_CASE(CodecG711A);
SWITCH_CASE(CodecG711U);
SWITCH_CASE(CodecOpus);
SWITCH_CASE(CodecL16);
default : return "unknown codec";
#define XX(name, type, value, str) case name : return type;
CODEC_MAP(XX)
#undef XX
default : return TrackInvalid;
}
}
TrackType getTrackType(CodecId codecId){
switch (codecId){
case CodecH264:
case CodecH265: return TrackVideo;
case CodecAAC:
case CodecG711A:
case CodecG711U:
case CodecOpus:
case CodecL16: return TrackAudio;
default: return TrackInvalid;
const char *getCodecName(CodecId codec) {
switch (codec) {
#define XX(name, type, value, str) case name : return str;
CODEC_MAP(XX)
#undef XX
default : return "invalid";
}
}
#define XX(name, type, value, str) {str, name},
static map<string, CodecId, StrCaseCompare> codec_map = {CODEC_MAP(XX)};
#undef XX
CodecId getCodecId(const string &str){
auto it = codec_map.find(str);
return it == codec_map.end() ? CodecInvalid : it->second;
}
static map<string, TrackType, StrCaseCompare> track_str_map = {
{"video", TrackVideo},
{"audio", TrackAudio},
{"application", TrackApplication}
};
TrackType getTrackType(const string &str) {
auto it = track_str_map.find(str);
return it == track_str_map.end() ? TrackInvalid : it->second;
}
const char* getTrackString(TrackType type){
switch (type) {
case TrackVideo : return "video";
case TrackAudio : return "audio";
case TrackApplication : return "application";
default: return "invalid";
}
}
......@@ -171,16 +191,23 @@ bool FrameMerger::willFlush(const Frame::Ptr &frame) const{
}
switch (frame->getCodecId()) {
case CodecH264 : {
if (H264_TYPE(frame->data()[frame->prefixSize()]) == H264Frame::NAL_B_P) {
//如果是264的b/p帧，那么也刷新输出
auto type = H264_TYPE(frame->data()[frame->prefixSize()]);
if ((frame->data()[frame->prefixSize()+1]&0x80) !=0 && type >=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR ) {// sei aud pps sps 不判断
//264 新一帧的开始，刷新输出
return true;
}else{
// 不刷新输出
return false;
}
break;
}
case CodecH265 : {
if (H265_TYPE(frame->data()[frame->prefixSize()]) == H265Frame::NAL_TRAIL_R) {
//如果是265的TRAIL_R帧，那么也刷新输出
auto type = H265_TYPE(frame->data()[frame->prefixSize()]);
if ((type>=H265Frame::NAL_TRAIL_R &&type<= H265Frame::NAL_RSV_IRAP_VCL23) && ( (frame->data()[frame->prefixSize()+2]>>7 & 0x01) != 0)) {
//first_slice_segment_in_pic_flag is frame start
return true;
}else{
return false;
}
break;
}
......@@ -217,8 +244,33 @@ void FrameMerger::doMerge(BufferLikeString &merged, const Frame::Ptr &frame) con
default: /*不可达*/ assert(0); break;
}
}
bool FrameMerger::shouldDrop(const Frame::Ptr &frame) const{
switch (frame->getCodecId()) {
case CodecH264:{
auto type = H264_TYPE(frame->data()[frame->prefixSize()]);
if(type == H264Frame::NAL_SEI || type == H264Frame::NAL_AUD){
// 防止吧AUD或者SEI当成一帧
return true;
}
break;
}
case CodecH265: {
//如果是新的一帧，前面的缓存需要输出
auto type = H265_TYPE(frame->data()[frame->prefixSize()]);
if(type == H265Frame::NAL_AUD || type == H265Frame::NAL_SEI_PREFIX || type == H265Frame::NAL_SEI_SUFFIX){
// 防止吧AUD或者SEI当成一帧
return true;
}
break;
}
default: break;
}
return false;
}
void FrameMerger::inputFrame(const Frame::Ptr &frame, const onOutput &cb) {
if(shouldDrop(frame)){
return;
}
if (willFlush(frame)) {
Frame::Ptr back = _frameCached.back();
Buffer::Ptr merged_frame = back;
......
src/Extension/Frame.h
......@@ -22,25 +22,51 @@ using namespace toolkit;
namespace mediakit{
typedef enum {
CodecInvalid = -1,
CodecH264 = 0,
CodecH265,
CodecAAC,
CodecG711A,
CodecG711U,
CodecOpus,
CodecL16,
CodecMax = 0x7FFF
} CodecId;
typedef enum {
TrackInvalid = -1,
TrackVideo = 0,
TrackAudio,
TrackTitle,
TrackMax = 3
TrackApplication,
TrackMax
} TrackType;
#define CODEC_MAP(XX) \
XX(CodecH264, TrackVideo, 0, "H264") \
XX(CodecH265, TrackVideo, 1, "H265") \
XX(CodecAAC, TrackAudio, 2, "mpeg4-generic") \
XX(CodecG711A, TrackAudio, 3, "PCMA") \
XX(CodecG711U, TrackAudio, 4, "PCMU") \
XX(CodecOpus, TrackAudio, 5, "opus") \
XX(CodecL16, TrackAudio, 6, "L16") \
XX(CodecVP8, TrackVideo, 7, "VP8") \
XX(CodecVP9, TrackVideo, 8, "VP9") \
XX(CodecAV1, TrackVideo, 9, "AV1X")
typedef enum {
CodecInvalid = -1,
#define XX(name, type, value, str) name = value,
CODEC_MAP(XX)
#undef XX
CodecMax
} CodecId;
/**
* 字符串转媒体类型转
*/
TrackType getTrackType(const string &str);
/**
* 媒体类型转字符串
*/
const char* getTrackString(TrackType type);
/**
* 根据SDP中描述获取codec_id
* @param str
* @return
*/
CodecId getCodecId(const string &str);
/**
* 获取编码器名称
*/
......@@ -449,6 +475,7 @@ public:
private:
bool willFlush(const Frame::Ptr &frame) const;
void doMerge(BufferLikeString &buffer, const Frame::Ptr &frame) const;
bool shouldDrop(const Frame::Ptr &frame) const;
private:
int _type;
......
src/Extension/G711.cpp
......@@ -33,7 +33,7 @@ public:
if (bitrate) {
_printer << "b=AS:" << bitrate << "\r\n";
}
_printer << "a=rtpmap:" << payload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate << "/" << channels << "\r\n";
_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n";
_printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
}
......
src/Extension/H264.cpp
......@@ -181,17 +181,15 @@ void H264Track::inputFrame_l(const Frame::Ptr &frame){
_pps = string(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
break;
}
case H264Frame::NAL_IDR: {
insertConfigFrame(frame);
VideoTrack::inputFrame(frame);
break;
}
case H264Frame::NAL_AUD: {
//忽略AUD帧;
break;
}
default:
if (frame->keyFrame()) {
insertConfigFrame(frame);
}
VideoTrack::inputFrame(frame);
break;
}
......@@ -235,7 +233,7 @@ public:
if (bitrate) {
_printer << "b=AS:" << bitrate << "\r\n";
}
_printer << "a=rtpmap:" << payload_type << " H264/" << 90000 << "\r\n";
_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << 90000 << "\r\n";
_printer << "a=fmtp:" << payload_type << " packetization-mode=1; profile-level-id=";
char strTemp[1024];
......@@ -281,7 +279,8 @@ Sdp::Ptr H264Track::getSdp() {
////////////////////////////////////////////////////////////////////////////////////////////////////
bool H264Frame::keyFrame() const {
return H264_TYPE(_buffer[_prefix_size]) == H264Frame::NAL_IDR;
//多slice 一帧的情况下检查 first_mb_in_slice 是否为0 表示其为一帧的开始
return H264_TYPE(_buffer[_prefix_size]) == H264Frame::NAL_IDR && (_buffer[_prefix_size + 1] & 0x80);
}
bool H264Frame::configFrame() const {
......
src/Extension/H264Rtmp.cpp
......@@ -183,7 +183,7 @@ void H264RtmpEncoder::inputFrame(const Frame::Ptr &frame) {
}
}
if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || type == H264Frame::NAL_B_P)) {
if(_lastPacket && (_lastPacket->time_stamp != frame->dts() || ((pcData[1]&0x80) != 0 && type>=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR))) {
RtmpCodec::inputRtmp(_lastPacket);
_lastPacket = nullptr;
}
......@@ -214,10 +214,6 @@ void H264RtmpEncoder::inputFrame(const Frame::Ptr &frame) {
_lastPacket->buffer.append((char *) &size, 4);
_lastPacket->buffer.append(pcData, iLen);
_lastPacket->body_size = _lastPacket->buffer.size();
if (type == H264Frame::NAL_B_P) {
RtmpCodec::inputRtmp(_lastPacket);
_lastPacket = nullptr;
}
}
void H264RtmpEncoder::makeVideoConfigPkt() {
......
src/Extension/H264Rtp.cpp
......@@ -184,53 +184,109 @@ H264RtpEncoder::H264RtpEncoder(uint32_t ssrc, uint32_t mtu, uint32_t sample_rate
: RtpInfo(ssrc, mtu, sample_rate, pt, interleaved) {
}
void H264RtpEncoder::inputFrame(const Frame::Ptr &frame) {
auto ptr = frame->data() + frame->prefixSize();
auto len = frame->size() - frame->prefixSize();
auto pts = frame->pts();
auto nal_type = H264_TYPE(ptr[0]);
void H264RtpEncoder::insertConfigFrame(uint32_t pts){
if (!_sps || !_pps) {
return;
}
//gop缓存从sps开始，sps、pps后面还有时间戳相同的关键帧，所以mark bit为false
packRtp(_sps->data() + _sps->prefixSize(), _sps->size() - _sps->prefixSize(), pts, false, true);
packRtp(_pps->data() + _pps->prefixSize(), _pps->size() - _pps->prefixSize(), pts, false, false);
}
void H264RtpEncoder::packRtp(const char *ptr, size_t len, uint32_t pts, bool is_mark, bool gop_pos){
if (len + 3 <= getMaxSize()) {
//STAP-A模式打包小于MTU
packRtpStapA(ptr, len, pts, is_mark, gop_pos);
} else {
//STAP-A模式打包会大于MTU,所以采用FU-A模式
packRtpFu(ptr, len, pts, is_mark, gop_pos);
}
}
void H264RtpEncoder::packRtpFu(const char *ptr, size_t len, uint32_t pts, bool is_mark, bool gop_pos){
auto packet_size = getMaxSize() - 2;
if (len <= packet_size + 1) {
//小于FU-A打包最小字节长度要求，采用STAP-A模式
packRtpStapA(ptr, len, pts, is_mark, gop_pos);
return;
}
//末尾5bit为nalu type，固定为28(FU-A)
auto fu_char_0 = (ptr[0] & (~0x1F)) | 28;
auto fu_char_1 = nal_type;
auto fu_char_1 = H264_TYPE(ptr[0]);
FuFlags *fu_flags = (FuFlags *) (&fu_char_1);
fu_flags->start_bit = 1;
//超过MTU则按照FU-A模式打包
if (len > packet_size + 1) {
size_t offset = 1;
while (!fu_flags->end_bit) {
if (!fu_flags->start_bit && len <= offset + packet_size) {
//FU-A end
packet_size = len - offset;
fu_flags->end_bit = 1;
}
size_t offset = 1;
while (!fu_flags->end_bit) {
if (!fu_flags->start_bit && len <= offset + packet_size) {
//FU-A end
packet_size = len - offset;
fu_flags->end_bit = 1;
}
//传入nullptr先不做payload的内存拷贝
auto rtp = makeRtp(getTrackType(), nullptr, packet_size + 2, fu_flags->end_bit && is_mark, pts);
//rtp payload 负载部分
uint8_t *payload = rtp->getPayload();
//FU-A 第1个字节
payload[0] = fu_char_0;
//FU-A 第2个字节
payload[1] = fu_char_1;
//H264 数据
memcpy(payload + 2, (uint8_t *) ptr + offset, packet_size);
//输入到rtp环形缓存
RtpCodec::inputRtp(rtp, gop_pos);
//传入nullptr先不做payload的内存拷贝
auto rtp = makeRtp(getTrackType(), nullptr, packet_size + 2, fu_flags->end_bit, pts);
//rtp payload 负载部分
uint8_t *payload = rtp->getPayload();
//FU-A 第1个字节
payload[0] = fu_char_0;
//FU-A 第2个字节
payload[1] = fu_char_1;
//H264 数据
memcpy(payload + 2, (uint8_t *) ptr + offset, packet_size);
//输入到rtp环形缓存
RtpCodec::inputRtp(rtp, fu_flags->start_bit && nal_type == H264Frame::NAL_IDR);
offset += packet_size;
fu_flags->start_bit = 0;
offset += packet_size;
fu_flags->start_bit = 0;
}
}
void H264RtpEncoder::packRtpStapA(const char *ptr, size_t len, uint32_t pts, bool is_mark, bool gop_pos){
//如果帧长度不超过mtu,为了兼容性 webrtc，采用STAP-A模式打包
auto rtp = makeRtp(getTrackType(), nullptr, len + 3, is_mark, pts);
uint8_t *payload = rtp->getPayload();
//STAP-A
payload[0] = (ptr[0] & (~0x1F)) | 24;
payload[1] = (len >> 8) & 0xFF;
payload[2] = len & 0xff;
memcpy(payload + 3, (uint8_t *) ptr, len);
RtpCodec::inputRtp(rtp, gop_pos);
}
void H264RtpEncoder::inputFrame(const Frame::Ptr &frame) {
auto ptr = frame->data() + frame->prefixSize();
switch (H264_TYPE(ptr[0])) {
case H264Frame::NAL_AUD:
case H264Frame::NAL_SEI : {
return;
}
} else {
//如果帧长度不超过mtu, 则按照Single NAL unit packet per H.264 方式打包
makeH264Rtp(ptr, len, false, false, pts);
case H264Frame::NAL_SPS: {
_sps = Frame::getCacheAbleFrame(frame);
return;
}
case H264Frame::NAL_PPS: {
_pps = Frame::getCacheAbleFrame(frame);
return;
}
default: break;
}
if (_last_frame) {
//如果时间戳发生了变化，那么markbit才置true
inputFrame_l(_last_frame, _last_frame->pts() != frame->pts());
}
_last_frame = Frame::getCacheAbleFrame(frame);
}
void H264RtpEncoder::makeH264Rtp(const void* data, size_t len, bool mark, bool gop_pos, uint32_t uiStamp) {
RtpCodec::inputRtp(makeRtp(getTrackType(), data, len, mark, uiStamp), gop_pos);
void H264RtpEncoder::inputFrame_l(const Frame::Ptr &frame, bool is_mark){
if (frame->keyFrame()) {
//保证每一个关键帧前都有SPS与PPS
insertConfigFrame(frame->pts());
}
packRtp(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize(), frame->pts(), is_mark, false);
}
}//namespace mediakit
src/Extension/H264Rtp.h
......@@ -82,7 +82,16 @@ public:
void inputFrame(const Frame::Ptr &frame) override;
private:
void makeH264Rtp(const void *pData, size_t uiLen, bool bMark, bool gop_pos, uint32_t uiStamp);
void insertConfigFrame(uint32_t pts);
void inputFrame_l(const Frame::Ptr &frame, bool is_mark);
void packRtp(const char *data, size_t len, uint32_t pts, bool is_mark, bool gop_pos);
void packRtpFu(const char *data, size_t len, uint32_t pts, bool is_mark, bool gop_pos);
void packRtpStapA(const char *data, size_t len, uint32_t pts, bool is_mark, bool gop_pos);
private:
Frame::Ptr _sps;
Frame::Ptr _pps;
Frame::Ptr _last_frame;
};
}//namespace mediakit{
......
src/Extension/H265.cpp
......@@ -64,10 +64,11 @@ bool H265Frame::configFrame() const {
}
bool H265Frame::isKeyFrame(int type, const char *ptr) {
if (!ptr || type != NAL_IDR_W_RADL) {
return type >= NAL_BLA_W_LP && type <= NAL_RSV_IRAP_VCL23;
if(ptr){
return (((*((uint8_t *) ptr + 2)) >> 7) & 0x01) == 1 && (type == NAL_IDR_N_LP || type == NAL_IDR_W_RADL);
}
return (((*((uint8_t *) ptr + 2)) >> 7) & 0x01) == 1;
return false;
}
H265Frame::H265Frame(){
......@@ -254,7 +255,7 @@ public:
if (bitrate) {
_printer << "b=AS:" << bitrate << "\r\n";
}
_printer << "a=rtpmap:" << payload_type << " H265/" << 90000 << "\r\n";
_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << 90000 << "\r\n";
_printer << "a=fmtp:" << payload_type << " ";
_printer << "sprop-vps=";
_printer << encodeBase64(strVPS) << "; ";
......
src/Extension/H265Rtmp.cpp
......@@ -165,11 +165,11 @@ void H265RtmpEncoder::inputFrame(const Frame::Ptr &frame) {
}
}
if(type == H265Frame::NAL_SEI_PREFIX || type == H265Frame::NAL_SEI_SUFFIX){
return;
if(type == H265Frame::NAL_SEI_PREFIX || type == H265Frame::NAL_SEI_SUFFIX || type == H265Frame::NAL_AUD){
return;// 防止sei aud 作为一帧
}
if (_lastPacket && (_lastPacket->time_stamp != frame->dts() || type == H265Frame::NAL_TRAIL_R)) {
if (_lastPacket && (_lastPacket->time_stamp != frame->dts() || (type >=H264Frame::NAL_B_P && type<=H264Frame::NAL_IDR && (pcData[2]>>7 &0x01) !=0))) {
RtmpCodec::inputRtmp(_lastPacket);
_lastPacket = nullptr;
}
......
src/Extension/L16.cpp
......@@ -33,7 +33,7 @@ public:
if (bitrate) {
_printer << "b=AS:" << bitrate << "\r\n";
}
_printer << "a=rtpmap:" << payload_type << " L16/" << sample_rate << "/" << channels << "\r\n";
_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n";
_printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
}
......
src/Extension/Opus.cpp
......@@ -31,7 +31,7 @@ public:
if (bitrate) {
_printer << "b=AS:" << bitrate << "\r\n";
}
_printer << "a=rtpmap:" << payload_type << " opus/" << sample_rate << "/" << channels << "\r\n";
_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n";
_printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
}
......
src/Rtcp/Rtcp.h
......@@ -25,8 +25,7 @@ namespace mediakit {
#pragma pack(push, 1)
#endif // defined(_WIN32)
//https://datatracker.ietf.org/doc/rfc3550
//http://www.networksorcery.com/enp/protocol/rtcp.htm
#define RTCP_PT_MAP(XX) \
XX(RTCP_FIR, 192) \
XX(RTCP_NACK, 193) \
......@@ -43,7 +42,8 @@ namespace mediakit {
XX(RTCP_AVB, 208) \
XX(RTCP_RSI, 209) \
XX(RTCP_TOKEN, 210)
//https://tools.ietf.org/html/rfc3550#section-6.5
#define SDES_TYPE_MAP(XX) \
XX(RTCP_SDES_END, 0) \
XX(RTCP_SDES_CNAME, 1) \
......@@ -55,6 +55,63 @@ namespace mediakit {
XX(RTCP_SDES_NOTE, 7) \
XX(RTCP_SDES_PRIVATE, 8)
//https://datatracker.ietf.org/doc/rfc4585/?include_text=1
//6.3. Payload-Specific Feedback Messages
//
// Payload-Specific FB messages are identified by the value PT=PSFB as
// RTCP message type.
//
// Three payload-specific FB messages are defined so far plus an
// application layer FB message. They are identified by means of the
// FMT parameter as follows:
//
// 0: unassigned
// 1: Picture Loss Indication (PLI)
// 2: Slice Loss Indication (SLI)
// 3: Reference Picture Selection Indication (RPSI)
// 4: FIR https://tools.ietf.org/html/rfc5104#section-4.3.1.1
// 5: TSTR https://tools.ietf.org/html/rfc5104#section-4.3.2.1
// 6: TSTN https://tools.ietf.org/html/rfc5104#section-4.3.2.1
// 7: VBCM https://tools.ietf.org/html/rfc5104#section-4.3.4.1
// 8-14: unassigned
// 15: REMB / Application layer FB (AFB) message, https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
// 16-30: unassigned
// 31: reserved for future expansion of the sequence number space
#define PSFB_TYPE_MAP(XX) \
XX(RTCP_PSFB_PLI, 1) \
XX(RTCP_PSFB_SLI, 2) \
XX(RTCP_PSFB_RPSI, 3) \
XX(RTCP_PSFB_FIR, 4) \
XX(RTCP_PSFB_TSTR, 5)\
XX(RTCP_PSFB_TSTN, 6)\
XX(RTCP_PSFB_VBCM, 7) \
XX(RTCP_PSFB_REMB, 15)
//https://tools.ietf.org/html/rfc4585#section-6.2
//6.2. Transport Layer Feedback Messages
//
// Transport layer FB messages are identified by the value RTPFB as RTCP
// message type.
//
// A single general purpose transport layer FB message is defined in
// this document: Generic NACK. It is identified by means of the FMT
// parameter as follows:
//
// 0: unassigned
// 1: Generic NACK
// 2: reserved https://tools.ietf.org/html/rfc5104#section-4.2
// 3: TMMBR https://tools.ietf.org/html/rfc5104#section-4.2.1.1
// 4: TMMBN https://tools.ietf.org/html/rfc5104#section-4.2.2.1
// 5-14: unassigned
// 15 transport-cc https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
// 16-30: unassigned
// 31: reserved for future expansion of the identifier number space
#define RTPFB_TYPE_MAP(XX) \
XX(RTCP_RTPFB_NACK, 1) \
XX(RTCP_RTPFB_TMMBR, 3) \
XX(RTCP_RTPFB_TMMBN, 4) \
XX(RTCP_RTPFB_TWCC, 15)
//rtcp类型枚举
enum class RtcpType : uint8_t {
#define XX(key, value) key = value,
......@@ -69,6 +126,20 @@ enum class SdesType : uint8_t {
#undef XX
};
//psfb类型枚举
enum class PSFBType : uint8_t {
#define XX(key, value) key = value,
PSFB_TYPE_MAP(XX)
#undef XX
};
//rtpfb类型枚举
enum class RTPFBType : uint8_t {
#define XX(key, value) key = value,
RTPFB_TYPE_MAP(XX)
#undef XX
};
/**
* RtcpType转描述字符串
*/
......@@ -79,6 +150,16 @@ const char *rtcpTypeToStr(RtcpType type);
*/
const char *sdesTypeToStr(SdesType type);
/**
* psfb枚举转描述字符串
*/
const char *psfbTypeToStr(PSFBType type);
/**
* rtpfb枚举转描述字符串
*/
const char *rtpfbTypeToStr(RTPFBType type);
class RtcpHeader {
public:
#if __BYTE_ORDER == __BIG_ENDIAN
......@@ -91,13 +172,15 @@ public:
#else
//reception report count
uint32_t report_count: 5;
//padding，固定为0
//padding，末尾是否有追加填充
uint32_t padding: 1;
//版本号，固定为2
uint32_t version: 2;
#endif
//rtcp类型,RtcpType
uint32_t pt: 8;
private:
//长度
uint32_t length: 16;
......@@ -124,11 +207,23 @@ public:
*/
string dumpString() const;
protected:
/**
* 网络字节序转换为主机字节序
* 根据length字段获取rtcp总长度
*/
void net2Host();
size_t getSize() const;
/**
* 后面追加padding数据长度
*/
size_t getPaddingSize() const;
/**
* 设置rtcp length字段
* @param size rtcp总长度，单位字节
*/
void setSize(size_t size);
protected:
/**
* 打印字段详情
......@@ -389,9 +484,9 @@ public:
//SdesType
uint8_t type;
//text长度股，可以为0
uint8_t length;
uint8_t txt_len;
//不定长
char text;
char text[1];
//最后以RTCP_SDES_END结尾
//只字段为占位字段，不代表真实位置
uint8_t end;
......@@ -434,7 +529,7 @@ public:
* @param item_text SdesItem列表，只赋值length和text部分
* @return SDES包
*/
static std::shared_ptr<RtcpSdes> create(const std::initializer_list<string> &item_text);
static std::shared_ptr<RtcpSdes> create(const std::vector<string> &item_text);
/**
* 获取SdesItem对象指针列表
......@@ -456,6 +551,142 @@ private:
void net2Host(size_t size);
} PACKED;
// https://tools.ietf.org/html/rfc4585#section-6.1
// 6.1. Common Packet Format for Feedback Messages
//
// All FB messages MUST use a common packet format that is depicted in
// Figure 3:
//
// 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| FMT | PT | length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | SSRC of packet sender |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | SSRC of media source |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// : Feedback Control Information (FCI) :
// : :
// rtcpfb和psfb的数据结构一致
class RtcpFB : public RtcpHeader {
public:
friend class RtcpHeader;
uint32_t ssrc;
uint32_t ssrc_media;
public:
/**
* 创建psfb类型的反馈包
*/
static std::shared_ptr<RtcpFB> create(PSFBType fmt, const void *fci = nullptr, size_t fci_len = 0);
/**
* 创建rtpfb类型的反馈包
*/
static std::shared_ptr<RtcpFB> create(RTPFBType fmt, const void *fci = nullptr, size_t fci_len = 0);
/**
* fci转换成某对象指针
* @tparam Type 对象类型
* @return 对象指针
*/
template<typename Type>
const Type& getFci() const{
auto fci_data = getFciPtr();
auto fci_len = getFciSize();
Type *fci = (Type *) fci_data;
fci->check(fci_len);
return *fci;
}
/**
* 获取fci指针
*/
const void *getFciPtr() const;
/**
* 获取fci数据长度
*/
size_t getFciSize() const;
private:
/**
* 打印字段详情
* 使用net2Host转换成主机字节序后才可使用此函数
*/
string dumpString() const;
/**
* 网络字节序转换为主机字节序
* @param size 字节长度，防止内存越界
*/
void net2Host(size_t size);
private:
static std::shared_ptr<RtcpFB> create_l(RtcpType type, int fmt, const void *fci, size_t fci_len);
} PACKED;
//BYE
/*
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| SC | PT=BYE=203 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC/CSRC |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
(opt) | length | reason for leaving ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
class RtcpBye : public RtcpHeader {
public:
friend class RtcpHeader;
/* 变长，根据count决定有多少个ssrc */
uint32_t ssrc[1];
/** 中间可能有若干个 ssrc **/
/* 可选 */
uint8_t reason_len;
char reason[1];
public:
/**
* 创建bye包
* @param ssrc ssrc列表
* @param reason 原因
* @return rtcp bye包
*/
static std::shared_ptr<RtcpBye> create(const std::vector<uint32_t> &ssrc, const string &reason);
/**
* 获取ssrc列表
*/
vector<uint32_t *> getSSRC();
/**
* 获取原因
*/
string getReason() const;
private:
/**
* 打印字段详情
* 使用net2Host转换成主机字节序后才可使用此函数
*/
string dumpString() const;
/**
* 网络字节序转换为主机字节序
* @param size 字节长度，防止内存越界
*/
void net2Host(size_t size);
} PACKED;
#if defined(_WIN32)
#pragma pack(pop)
#endif // defined(_WIN32)
......
src/Rtmp/RtmpSession.cpp
......@@ -463,6 +463,9 @@ void RtmpSession::onRtmpChunk(RtmpPacket::Ptr packet) {
std::string type = dec.load<std::string>();
if (type == "@setDataFrame") {
setMetaData(dec);
} else if (type == "onMetaData") {
//兼容某些不规范的推流器
_publisher_metadata = dec.load<AMFValue>();
} else {
TraceP(this) << "unknown notify:" << type;
}
......
src/Rtsp/RtpReceiver.h
......@@ -21,7 +21,7 @@ using namespace toolkit;
namespace mediakit {
template<typename T, typename SEQ = uint16_t, size_t kMax = 256, size_t kMin = 10>
template<typename T, typename SEQ = uint16_t, size_t kMax = 1024, size_t kMin = 32>
class PacketSortor {
public:
PacketSortor() = default;
......
src/Rtsp/Rtsp.cpp
......@@ -458,18 +458,27 @@ size_t RtpHeader::getExtSize() const {
return 0;
}
auto ext_ptr = &payload + getCsrcSize();
uint16_t reserved = AV_RB16(ext_ptr);
//uint16_t reserved = AV_RB16(ext_ptr);
//每个ext占用4字节
return AV_RB16(ext_ptr + 2) << 2;
}
uint16_t RtpHeader::getExtReserved() const{
//rtp有ext
if (!ext) {
return 0;
}
auto ext_ptr = &payload + getCsrcSize();
return AV_RB16(ext_ptr);
}
uint8_t *RtpHeader::getExtData() {
if (!ext) {
return nullptr;
}
auto ext_ptr = &payload + getCsrcSize();
//多出的4个字节分别为reserved、ext_len
return ext_ptr + 4 + getExtSize();
return ext_ptr + 4;
}
size_t RtpHeader::getPayloadOffset() const {
......@@ -521,6 +530,10 @@ RtpHeader* RtpPacket::getHeader(){
return (RtpHeader*)(data() + RtpPacket::kRtpTcpHeaderSize);
}
string RtpPacket::dumpString() const{
return ((RtpPacket *) this)->getHeader()->dumpString(size() - RtpPacket::kRtpTcpHeaderSize);
}
uint16_t RtpPacket::getSeq(){
return ntohs(getHeader()->seq);
}
......
src/Rtsp/Rtsp.h
......@@ -119,6 +119,8 @@ public:
//返回ext字段字节长度
size_t getExtSize() const;
//返回ext reserved值
uint16_t getExtReserved() const;
//返回ext段首地址，不存在时返回nullptr
uint8_t *getExtData();
......@@ -150,7 +152,11 @@ public:
kRtpTcpHeaderSize = 4
};
//获取rtp头
RtpHeader* getHeader();
//打印调试信息
string dumpString() const;
//主机字节序的seq
uint16_t getSeq();
//主机字节序的时间戳，已经转换为毫秒
......
src/Rtsp/RtspMediaSourceImp.h
......@@ -56,7 +56,11 @@ public:
//需要解复用rtp
key_pos = _demuxer->inputRtp(rtp);
}
RtspMediaSource::onWrite(std::move(rtp), key_pos);
GET_CONFIG(bool, directProxy, Rtsp::kDirectProxy);
if (directProxy) {
//直接代理模式才直接使用原始rtp
RtspMediaSource::onWrite(std::move(rtp), key_pos);
}
}
/**
......@@ -72,8 +76,10 @@ public:
* @param enableMP4 是否mp4录制
*/
void setProtocolTranslation(bool enableHls,bool enableMP4){
//不重复生成rtsp
_muxer = std::make_shared<MultiMediaSourceMuxer>(getVhost(), getApp(), getId(), _demuxer->getDuration(), false, true, enableHls, enableMP4);
GET_CONFIG(bool, directProxy, Rtsp::kDirectProxy);
//开启直接代理模式时，rtsp直接代理，不重复产生；但是有些rtsp推流端，由于sdp中已有sps pps，rtp中就不再包括sps pps,
//导致rtc无法播放，所以在rtsp推流rtc播放时，建议关闭直接代理模式
_muxer = std::make_shared<MultiMediaSourceMuxer>(getVhost(), getApp(), getId(), _demuxer->getDuration(), !directProxy, true, enableHls, enableMP4);
_muxer->setMediaListener(getListener());
_muxer->setTrackListener(static_pointer_cast<RtspMediaSourceImp>(shared_from_this()));
//让_muxer对象拦截一部分事件(比如说录像相关事件)
......
tests/test_rtcp_fci.cpp 0 → 100644
/*
* Copyright (c) 2016 The ZLToolKit project authors. All Rights Reserved.
*
* This file is part of ZLToolKit(https://github.com/xia-chu/ZLToolKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#include <iostream>
#include "Util/logger.h"
#include "Rtcp/RtcpFCI.h"
using namespace std;
using namespace toolkit;
using namespace mediakit;
extern void testFCI();
int main() {
Logger::Instance().add(std::make_shared<ConsoleChannel>());
testFCI();
return 0;
}
tests/test_rtcp_nack.cpp 0 → 100644
/*
* Copyright (c) 2016 The ZLToolKit project authors. All Rights Reserved.
*
* This file is part of ZLToolKit(https://github.com/xia-chu/ZLToolKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#include <iostream>
#include "Util/logger.h"
#include "Rtcp/RtcpFCI.h"
#include "../webrtc/WebRtcTransport.h"
using namespace std;
using namespace toolkit;
using namespace mediakit;
extern void testFCI();
int main() {
Logger::Instance().add(std::make_shared<ConsoleChannel>());
srand((unsigned) time(NULL));
NackContext ctx;
for (int i = 1; i < 1000; ++i) {
if (i % (1 + (rand() % 30)) == 0) {
DebugL << "drop:" << i;
} else {
ctx.received(i);
}
}
sleep(1);
return 0;
}
webrtc/DtlsTransport.hpp 0 → 100644
/**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_DTLS_TRANSPORT_HPP
#define MS_RTC_DTLS_TRANSPORT_HPP
#include "SrtpSession.hpp"
#include <openssl/bio.h>
#include <openssl/ssl.h>
#include <openssl/x509.h>
#include <map>
#include <string>
#include <vector>
#include "Poller/Timer.h"
#include "Poller/EventPoller.h"
using namespace toolkit;
namespace RTC
{
class DtlsTransport : public std::enable_shared_from_this<DtlsTransport>
{
public:
enum class DtlsState
{
NEW = 1,
CONNECTING,
CONNECTED,
FAILED,
CLOSED
};
public:
enum class Role
{
NONE = 0,
AUTO = 1,
CLIENT,
SERVER
};
public:
enum class FingerprintAlgorithm
{
NONE = 0,
SHA1 = 1,
SHA224,
SHA256,
SHA384,
SHA512
};
public:
struct Fingerprint
{
FingerprintAlgorithm algorithm{ FingerprintAlgorithm::NONE };
std::string value;
};
private:
struct SrtpCryptoSuiteMapEntry
{
RTC::SrtpSession::CryptoSuite cryptoSuite;
const char* name;
};
class DtlsEnvironment : public std::enable_shared_from_this<DtlsEnvironment>
{
public:
using Ptr = std::shared_ptr<DtlsEnvironment>;
~DtlsEnvironment();
static DtlsEnvironment& Instance();
private:
DtlsEnvironment();
void GenerateCertificateAndPrivateKey();
void ReadCertificateAndPrivateKeyFromFiles();
void CreateSslCtx();
void GenerateFingerprints();
public:
X509* certificate{ nullptr };
EVP_PKEY* privateKey{ nullptr };
SSL_CTX* sslCtx{ nullptr };
std::vector<Fingerprint> localFingerprints;
};
public:
class Listener
{
public:
// DTLS is in the process of negotiating a secure connection. Incoming
// media can flow through.
// NOTE: The caller MUST NOT call any method during this callback.
virtual void OnDtlsTransportConnecting(const RTC::DtlsTransport* dtlsTransport) = 0;
// DTLS has completed negotiation of a secure connection (including DTLS-SRTP
// and remote fingerprint verification). Outgoing media can now flow through.
// NOTE: The caller MUST NOT call any method during this callback.
virtual void OnDtlsTransportConnected(
const RTC::DtlsTransport* dtlsTransport,
RTC::SrtpSession::CryptoSuite srtpCryptoSuite,
uint8_t* srtpLocalKey,
size_t srtpLocalKeyLen,
uint8_t* srtpRemoteKey,
size_t srtpRemoteKeyLen,
std::string& remoteCert) = 0;
// The DTLS connection has been closed as the result of an error (such as a
// DTLS alert or a failure to validate the remote fingerprint).
virtual void OnDtlsTransportFailed(const RTC::DtlsTransport* dtlsTransport) = 0;
// The DTLS connection has been closed due to receipt of a close_notify alert.
virtual void OnDtlsTransportClosed(const RTC::DtlsTransport* dtlsTransport) = 0;
// Need to send DTLS data to the peer.
virtual void OnDtlsTransportSendData(
const RTC::DtlsTransport* dtlsTransport, const uint8_t* data, size_t len) = 0;
// DTLS application data received.
virtual void OnDtlsTransportApplicationDataReceived(
const RTC::DtlsTransport* dtlsTransport, const uint8_t* data, size_t len) = 0;
};
public:
static Role StringToRole(const std::string& role)
{
auto it = DtlsTransport::string2Role.find(role);
if (it != DtlsTransport::string2Role.end())
return it->second;
else
return DtlsTransport::Role::NONE;
}
static FingerprintAlgorithm GetFingerprintAlgorithm(const std::string& fingerprint)
{
auto it = DtlsTransport::string2FingerprintAlgorithm.find(fingerprint);
if (it != DtlsTransport::string2FingerprintAlgorithm.end())
return it->second;
else
return DtlsTransport::FingerprintAlgorithm::NONE;
}
static std::string& GetFingerprintAlgorithmString(FingerprintAlgorithm fingerprint)
{
auto it = DtlsTransport::fingerprintAlgorithm2String.find(fingerprint);
return it->second;
}
static bool IsDtls(const uint8_t* data, size_t len)
{
// clang-format off
return (
// Minimum DTLS record length is 13 bytes.
(len >= 13) &&
// DOC: https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes
(data[0] > 19 && data[0] < 64)
);
// clang-format on
}
private:
static std::map<std::string, Role> string2Role;
static std::map<std::string, FingerprintAlgorithm> string2FingerprintAlgorithm;
static std::map<FingerprintAlgorithm, std::string> fingerprintAlgorithm2String;
static std::vector<SrtpCryptoSuiteMapEntry> srtpCryptoSuites;
public:
DtlsTransport(EventPoller::Ptr poller, Listener* listener);
~DtlsTransport();
public:
void Dump() const;
void Run(Role localRole);
std::vector<Fingerprint>& GetLocalFingerprints() const
{
return env->localFingerprints;
}
bool SetRemoteFingerprint(Fingerprint fingerprint);
void ProcessDtlsData(const uint8_t* data, size_t len);
DtlsState GetState() const
{
return this->state;
}
Role GetLocalRole() const
{
return this->localRole;
}
void SendApplicationData(const uint8_t* data, size_t len);
private:
bool IsRunning() const
{
switch (this->state)
{
case DtlsState::NEW:
return false;
case DtlsState::CONNECTING:
case DtlsState::CONNECTED:
return true;
case DtlsState::FAILED:
case DtlsState::CLOSED:
return false;
}
// Make GCC 4.9 happy.
return false;
}
void Reset();
bool CheckStatus(int returnCode);
void SendPendingOutgoingDtlsData();
bool SetTimeout();
bool ProcessHandshake();
bool CheckRemoteFingerprint();
void ExtractSrtpKeys(RTC::SrtpSession::CryptoSuite srtpCryptoSuite);
RTC::SrtpSession::CryptoSuite GetNegotiatedSrtpCryptoSuite();
private:
void OnSslInfo(int where, int ret);
void OnTimer();
private:
DtlsEnvironment::Ptr env;
EventPoller::Ptr poller;
// Passed by argument.
Listener* listener{ nullptr };
// Allocated by this.
SSL* ssl{ nullptr };
BIO* sslBioFromNetwork{ nullptr }; // The BIO from which ssl reads.
BIO* sslBioToNetwork{ nullptr }; // The BIO in which ssl writes.
Timer::Ptr timer;
// Others.
DtlsState state{ DtlsState::NEW };
Role localRole{ Role::NONE };
Fingerprint remoteFingerprint;
bool handshakeDone{ false };
bool handshakeDoneNow{ false };
std::string remoteCert;
//最大不超过mtu
static constexpr int SslReadBufferSize{ 2000 };
uint8_t sslReadBuffer[SslReadBufferSize];
};
} // namespace RTC
#endif
webrtc/IceServer.hpp 0 → 100644
/**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_ICE_SERVER_HPP
#define MS_RTC_ICE_SERVER_HPP
#include "StunPacket.hpp"
#include "logger.h"
#include "Utils.hpp"
#include <list>
#include <string>
#include <functional>
#include <memory>
using _TransportTuple = struct sockaddr;
namespace RTC
{
using TransportTuple = _TransportTuple;
class IceServer
{
public:
enum class IceState
{
NEW = 1,
CONNECTED,
COMPLETED,
DISCONNECTED
};
public:
class Listener
{
public:
virtual ~Listener() = default;
public:
/**
* These callbacks are guaranteed to be called before ProcessStunPacket()
* returns, so the given pointers are still usable.
*/
virtual void OnIceServerSendStunPacket(
const RTC::IceServer* iceServer, const RTC::StunPacket* packet, RTC::TransportTuple* tuple) = 0;
virtual void OnIceServerSelectedTuple(
const RTC::IceServer* iceServer, RTC::TransportTuple* tuple) = 0;
virtual void OnIceServerConnected(const RTC::IceServer* iceServer) = 0;
virtual void OnIceServerCompleted(const RTC::IceServer* iceServer) = 0;
virtual void OnIceServerDisconnected(const RTC::IceServer* iceServer) = 0;
};
public:
IceServer(Listener* listener, const std::string& usernameFragment, const std::string& password);
public:
void ProcessStunPacket(RTC::StunPacket* packet, RTC::TransportTuple* tuple);
const std::string& GetUsernameFragment() const
{
return this->usernameFragment;
}
const std::string& GetPassword() const
{
return this->password;
}
IceState GetState() const
{
return this->state;
}
RTC::TransportTuple* GetSelectedTuple() const
{
return this->selectedTuple;
}
void SetUsernameFragment(const std::string& usernameFragment)
{
this->oldUsernameFragment = this->usernameFragment;
this->usernameFragment = usernameFragment;
}
void SetPassword(const std::string& password)
{
this->oldPassword = this->password;
this->password = password;
}
bool IsValidTuple(const RTC::TransportTuple* tuple) const;
void RemoveTuple(RTC::TransportTuple* tuple);
// This should be just called in 'connected' or completed' state
// and the given tuple must be an already valid tuple.
void ForceSelectedTuple(const RTC::TransportTuple* tuple);
private:
void HandleTuple(RTC::TransportTuple* tuple, bool hasUseCandidate);
/**
* Store the given tuple and return its stored address.
*/
RTC::TransportTuple* AddTuple(RTC::TransportTuple* tuple);
/**
* If the given tuple exists return its stored address, nullptr otherwise.
*/
RTC::TransportTuple* HasTuple(const RTC::TransportTuple* tuple) const;
/**
* Set the given tuple as the selected tuple.
* NOTE: The given tuple MUST be already stored within the list.
*/
void SetSelectedTuple(RTC::TransportTuple* storedTuple);
private:
// Passed by argument.
Listener* listener{ nullptr };
// Others.
std::string usernameFragment;
std::string password;
std::string oldUsernameFragment;
std::string oldPassword;
IceState state{ IceState::NEW };
std::list<RTC::TransportTuple> tuples;
RTC::TransportTuple* selectedTuple{ nullptr };
//最大不超过mtu
static constexpr size_t StunSerializeBufferSize{ 1600 };
uint8_t StunSerializeBuffer[StunSerializeBufferSize];
};
} // namespace RTC
#endif
webrtc/RtpExt.h 0 → 100644
/*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#ifndef ZLMEDIAKIT_RTPEXT_H
#define ZLMEDIAKIT_RTPEXT_H
#include <stdint.h>
#include <map>
#include <string>
#include "Common/macros.h"
#include "Rtsp/Rtsp.h"
using namespace std;
using namespace mediakit;
enum class RtpExtType : uint8_t {
padding = 0,
ssrc_audio_level = 1,
abs_send_time = 2,
transport_cc = 3,
sdes_mid = 4,
sdes_rtp_stream_id = 5,
sdes_repaired_rtp_stream_id = 6,
video_timing = 7,
color_space = 8,
//for firefox
csrc_audio_level = 9,
//svc ?
framemarking = 10,
video_content_type = 11,
playout_delay = 12,
video_orientation = 13,
toffset = 14,
reserved = 15,
// e2e ?
encrypt = reserved
};
class RtcMedia;
//使用次对象的方法前需保证RtpHeader内存未释放
class RtpExt {
public:
template<typename Type>
friend void appendExt(map<uint8_t, RtpExt> &ret, uint8_t *ptr, const uint8_t *end);
~RtpExt() = default;
static map<uint8_t/*id*/, RtpExt/*data*/> getExtValue(const RtpHeader *header);
static RtpExtType getExtType(const string &url);
static const string& getExtUrl(RtpExtType type);
static const char *getExtName(RtpExtType type);
void setType(RtpExtType type);
RtpExtType getType() const;
string dumpString() const;
uint8_t getAudioLevel(bool *vad) const;
uint32_t getAbsSendTime() const;
uint16_t getTransportCCSeq() const;
string getSdesMid() const;
string getRtpStreamId() const;
string getRepairedRtpStreamId() const;
void getVideoTiming(uint8_t &flags,
uint16_t &encode_start,
uint16_t &encode_finish,
uint16_t &packetization_complete,
uint16_t &last_pkt_left_pacer,
uint16_t &reserved_net0,
uint16_t &reserved_net1) const;
uint8_t getVideoContentType() const;
void getVideoOrientation(bool &camera_bit,
bool &flip_bit,
bool &first_rotation,
bool &second_rotation) const;
void getPlayoutDelay(uint16_t &min_delay, uint16_t &max_delay) const;
uint32_t getTransmissionOffset() const;
uint8_t getFramemarkingTID() const;
void setExtId(uint8_t ext_id);
void clearExt();
private:
RtpExt(void *ptr, bool one_byte_ext, const char *str, size_t size);
const char *data() const;
size_t size() const;
const char& operator[](size_t pos) const;
operator std::string() const;
private:
void *_ext = nullptr;
const char *_data;
size_t _size;
bool _one_byte_ext = true;
RtpExtType _type = RtpExtType::padding;
};
#endif //ZLMEDIAKIT_RTPEXT_H
webrtc/SrtpSession.cpp 0 → 100644
/**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#define MS_CLASS "RTC::SrtpSession"
// #define MS_LOG_DEV_LEVEL 3
#include "SrtpSession.hpp"
#include <cstring> // std::memset(), std::memcpy()
#include "logger.h"
#include "Util/util.h"
#include "Util/logger.h"
using namespace toolkit;
namespace RTC
{
/* Static. */
static std::vector<const char *> errors = {
// From 0 (srtp_err_status_ok) to 24 (srtp_err_status_pfkey_err).
"success (srtp_err_status_ok)",
"unspecified failure (srtp_err_status_fail)",
"unsupported parameter (srtp_err_status_bad_param)",
"couldn't allocate memory (srtp_err_status_alloc_fail)",
"couldn't deallocate memory (srtp_err_status_dealloc_fail)",
"couldn't initialize (srtp_err_status_init_fail)",
"can’t process as much data as requested (srtp_err_status_terminus)",
"authentication failure (srtp_err_status_auth_fail)",
"cipher failure (srtp_err_status_cipher_fail)",
"replay check failed (bad index) (srtp_err_status_replay_fail)",
"replay check failed (index too old) (srtp_err_status_replay_old)",
"algorithm failed test routine (srtp_err_status_algo_fail)",
"unsupported operation (srtp_err_status_no_such_op)",
"no appropriate context found (srtp_err_status_no_ctx)",
"unable to perform desired validation (srtp_err_status_cant_check)",
"can’t use key any more (srtp_err_status_key_expired)",
"error in use of socket (srtp_err_status_socket_err)",
"error in use POSIX signals (srtp_err_status_signal_err)",
"nonce check failed (srtp_err_status_nonce_bad)",
"couldn’t read data (srtp_err_status_read_fail)",
"couldn’t write data (srtp_err_status_write_fail)",
"error parsing data (srtp_err_status_parse_err)",
"error encoding data (srtp_err_status_encode_err)",
"error while using semaphores (srtp_err_status_semaphore_err)",
"error while using pfkey (srtp_err_status_pfkey_err)"};
// clang-format on
/* Static methods. */
const char *DepLibSRTP::GetErrorString(srtp_err_status_t code) {
// This throws out_of_range if the given index is not in the vector.
return errors.at(code);
}
bool DepLibSRTP::IsError(srtp_err_status_t code) {
return (code != srtp_err_status_ok);
}
INSTANCE_IMP(DepLibSRTP);
DepLibSRTP::DepLibSRTP(){
MS_TRACE();
MS_DEBUG_TAG(info, "libsrtp version: \"%s\"", srtp_get_version_string());
srtp_err_status_t err = srtp_init();
#if 0
srtp_install_log_handler([](srtp_log_level_t level,
const char *msg,
void *data) {
printf("%s\n", msg);
}, nullptr);
srtp_set_debug_module("srtp", 1);
srtp_set_debug_module("hmac sha-1", 1);
srtp_set_debug_module("aes icm", 1);
srtp_set_debug_module("alloc", 1);
srtp_set_debug_module("stat test", 1);
srtp_set_debug_module("cipher", 1);
srtp_set_debug_module("auth func", 1);
srtp_set_debug_module("crypto kernel", 1);
srtp_list_debug_modules();
#endif
if (DepLibSRTP::IsError(err)) {
MS_THROW_ERROR("srtp_init() failed: %s", DepLibSRTP::GetErrorString(err));
}
// Set libsrtp event handler.
err = srtp_install_event_handler([](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;
}
});
if (DepLibSRTP::IsError(err))
{
MS_THROW_ERROR("srtp_install_event_handler() failed: %s", DepLibSRTP::GetErrorString(err));
}
}
DepLibSRTP::~DepLibSRTP(){
MS_TRACE();
srtp_shutdown();
}
/////////////////////////////////////////////////////////////////////////////////////
/* Instance methods. */
SrtpSession::SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t* key, size_t keyLen)
{
_env = DepLibSRTP::Instance().shared_from_this();
MS_TRACE();
srtp_policy_t policy; // NOLINT(cppcoreguidelines-pro-type-member-init)
// Set all policy fields to 0.
std::memset(&policy, 0, sizeof(srtp_policy_t));
switch (cryptoSuite)
{
case CryptoSuite::AES_CM_128_HMAC_SHA1_80:
{
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
break;
}
case CryptoSuite::AES_CM_128_HMAC_SHA1_32:
{
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
// NOTE: Must be 80 for RTCP.
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
break;
}
case CryptoSuite::AEAD_AES_256_GCM:
{
srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp);
srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp);
break;
}
case CryptoSuite::AEAD_AES_128_GCM:
{
srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp);
srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp);
break;
}
default:
{
MS_ABORT("unknown SRTP crypto suite");
}
}
MS_ASSERT(
(int)keyLen == policy.rtp.cipher_key_len,
"given keyLen does not match policy.rtp.cipher_keyLen");
switch (type)
{
case Type::INBOUND:
policy.ssrc.type = ssrc_any_inbound;
break;
case Type::OUTBOUND:
policy.ssrc.type = ssrc_any_outbound;
break;
}
policy.ssrc.value = 0;
policy.key = key;
// Required for sending RTP retransmission without RTX.
policy.allow_repeat_tx = 1;
policy.window_size = 1024;
policy.next = nullptr;
// Set the SRTP session.
srtp_err_status_t err = srtp_create(&this->session, &policy);
if (DepLibSRTP::IsError(err))
MS_THROW_ERROR("srtp_create() failed: %s", DepLibSRTP::GetErrorString(err));
}
SrtpSession::~SrtpSession()
{
MS_TRACE();
if (this->session != nullptr)
{
srtp_err_status_t err = srtp_dealloc(this->session);
if (DepLibSRTP::IsError(err))
MS_ABORT("srtp_dealloc() failed: %s", DepLibSRTP::GetErrorString(err));
}
}
bool SrtpSession::EncryptRtp(uint8_t* data, size_t* len)
{
MS_TRACE();
srtp_err_status_t err =
srtp_protect(this->session, static_cast<void*>(data), reinterpret_cast<int*>(len));
if (DepLibSRTP::IsError(err))
{
MS_WARN_TAG(srtp, "srtp_protect() failed: %s", DepLibSRTP::GetErrorString(err));
return false;
}
return true;
}
bool SrtpSession::DecryptSrtp(uint8_t* data, size_t* len)
{
MS_TRACE();
srtp_err_status_t err =
srtp_unprotect(this->session, static_cast<void*>(data), reinterpret_cast<int*>(len));
if (DepLibSRTP::IsError(err))
{
MS_DEBUG_TAG(srtp, "srtp_unprotect() failed: %s", DepLibSRTP::GetErrorString(err));
return false;
}
return true;
}
bool SrtpSession::EncryptRtcp(uint8_t* data, size_t* len)
{
MS_TRACE();
srtp_err_status_t err = srtp_protect_rtcp(
this->session, static_cast<void*>(data), reinterpret_cast<int*>(len));
if (DepLibSRTP::IsError(err))
{
MS_WARN_TAG(srtp, "srtp_protect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err));
return false;
}
return true;
}
bool SrtpSession::DecryptSrtcp(uint8_t* data, size_t* len)
{
MS_TRACE();
srtp_err_status_t err =
srtp_unprotect_rtcp(this->session, static_cast<void*>(data), reinterpret_cast<int*>(len));
if (DepLibSRTP::IsError(err))
{
MS_DEBUG_TAG(srtp, "srtp_unprotect_rtcp() failed: %s", DepLibSRTP::GetErrorString(err));
return false;
}
return true;
}
} // namespace RTC
webrtc/SrtpSession.hpp 0 → 100644
/**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_SRTP_SESSION_HPP
#define MS_RTC_SRTP_SESSION_HPP
#include "Utils.hpp"
#include <srtp2/srtp.h>
#include <vector>
#include <memory>
namespace RTC
{
class DepLibSRTP : public std::enable_shared_from_this<DepLibSRTP>
{
public:
using Ptr = std::shared_ptr<DepLibSRTP>;
~DepLibSRTP();
static bool IsError(srtp_err_status_t code);
static const char *GetErrorString(srtp_err_status_t code);
static DepLibSRTP &Instance();
private:
DepLibSRTP();
};
class SrtpSession
{
public:
enum class CryptoSuite
{
NONE = 0,
AES_CM_128_HMAC_SHA1_80 = 1,
AES_CM_128_HMAC_SHA1_32,
AEAD_AES_256_GCM,
AEAD_AES_128_GCM
};
public:
enum class Type
{
INBOUND = 1,
OUTBOUND
};
public:
SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t* key, size_t keyLen);
~SrtpSession();
public:
bool EncryptRtp(uint8_t* data, size_t* len);
bool DecryptSrtp(uint8_t* data, size_t* len);
bool EncryptRtcp(uint8_t* data, size_t* len);
bool DecryptSrtcp(uint8_t* data, size_t* len);
void RemoveStream(uint32_t ssrc)
{
srtp_remove_stream(this->session, uint32_t{ htonl(ssrc) });
}
private:
// Allocated by this.
srtp_t session{ nullptr };
DepLibSRTP::Ptr _env;
};
} // namespace RTC
#endif
webrtc/StunPacket.hpp 0 → 100644
/**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_STUN_PACKET_HPP
#define MS_RTC_STUN_PACKET_HPP
#include "logger.h"
#include "Utils.hpp"
#include <string>
namespace RTC
{
class StunPacket
{
public:
// STUN message class.
enum class Class : uint16_t
{
REQUEST = 0,
INDICATION = 1,
SUCCESS_RESPONSE = 2,
ERROR_RESPONSE = 3
};
// STUN message method.
enum class Method : uint16_t
{
BINDING = 1
};
// Attribute type.
enum class Attribute : uint16_t
{
MAPPED_ADDRESS = 0x0001,
USERNAME = 0x0006,
MESSAGE_INTEGRITY = 0x0008,
ERROR_CODE = 0x0009,
UNKNOWN_ATTRIBUTES = 0x000A,
REALM = 0x0014,
NONCE = 0x0015,
XOR_MAPPED_ADDRESS = 0x0020,
PRIORITY = 0x0024,
USE_CANDIDATE = 0x0025,
SOFTWARE = 0x8022,
ALTERNATE_SERVER = 0x8023,
FINGERPRINT = 0x8028,
ICE_CONTROLLED = 0x8029,
ICE_CONTROLLING = 0x802A
};
// Authentication result.
enum class Authentication
{
OK = 0,
UNAUTHORIZED = 1,
BAD_REQUEST = 2
};
public:
static bool IsStun(const uint8_t* data, size_t len)
{
// clang-format off
return (
// STUN headers are 20 bytes.
(len >= 20) &&
// DOC: https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes
(data[0] < 3) &&
// Magic cookie must match.
(data[4] == StunPacket::magicCookie[0]) && (data[5] == StunPacket::magicCookie[1]) &&
(data[6] == StunPacket::magicCookie[2]) && (data[7] == StunPacket::magicCookie[3])
);
// clang-format on
}
static StunPacket* Parse(const uint8_t* data, size_t len);
private:
static const uint8_t magicCookie[];
public:
StunPacket(
Class klass, Method method, const uint8_t* transactionId, const uint8_t* data, size_t size);
~StunPacket();
void Dump() const;
Class GetClass() const
{
return this->klass;
}
Method GetMethod() const
{
return this->method;
}
const uint8_t* GetData() const
{
return this->data;
}
size_t GetSize() const
{
return this->size;
}
void SetUsername(const char* username, size_t len)
{
this->username.assign(username, len);
}
void SetPriority(uint32_t priority)
{
this->priority = priority;
}
void SetIceControlling(uint64_t iceControlling)
{
this->iceControlling = iceControlling;
}
void SetIceControlled(uint64_t iceControlled)
{
this->iceControlled = iceControlled;
}
void SetUseCandidate()
{
this->hasUseCandidate = true;
}
void SetXorMappedAddress(const struct sockaddr* xorMappedAddress)
{
this->xorMappedAddress = xorMappedAddress;
}
void SetErrorCode(uint16_t errorCode)
{
this->errorCode = errorCode;
}
void SetMessageIntegrity(const uint8_t* messageIntegrity)
{
this->messageIntegrity = messageIntegrity;
}
void SetFingerprint()
{
this->hasFingerprint = true;
}
const std::string& GetUsername() const
{
return this->username;
}
uint32_t GetPriority() const
{
return this->priority;
}
uint64_t GetIceControlling() const
{
return this->iceControlling;
}
uint64_t GetIceControlled() const
{
return this->iceControlled;
}
bool HasUseCandidate() const
{
return this->hasUseCandidate;
}
uint16_t GetErrorCode() const
{
return this->errorCode;
}
bool HasMessageIntegrity() const
{
return (this->messageIntegrity ? true : false);
}
bool HasFingerprint() const
{
return this->hasFingerprint;
}
Authentication CheckAuthentication(
const std::string& localUsername, const std::string& localPassword);
StunPacket* CreateSuccessResponse();
StunPacket* CreateErrorResponse(uint16_t errorCode);
void Authenticate(const std::string& password);
void Serialize(uint8_t* buffer);
private:
// Passed by argument.
Class klass; // 2 bytes.
Method method; // 2 bytes.
const uint8_t* transactionId{ nullptr }; // 12 bytes.
uint8_t* data{ nullptr }; // Pointer to binary data.
size_t size{ 0u }; // The full message size (including header).
// STUN attributes.
std::string username; // Less than 513 bytes.
uint32_t priority{ 0u }; // 4 bytes unsigned integer.
uint64_t iceControlling{ 0u }; // 8 bytes unsigned integer.
uint64_t iceControlled{ 0u }; // 8 bytes unsigned integer.
bool hasUseCandidate{ false }; // 0 bytes.
const uint8_t* messageIntegrity{ nullptr }; // 20 bytes.
bool hasFingerprint{ false }; // 4 bytes.
const struct sockaddr* xorMappedAddress{ nullptr }; // 8 or 20 bytes.
uint16_t errorCode{ 0u }; // 4 bytes (no reason phrase).
std::string password;
};
} // namespace RTC
#endif
webrtc/Utils.hpp 0 → 100644
/**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_UTILS_HPP
#define MS_UTILS_HPP
#if defined(_WIN32)
#include <winsock2.h>
#include <ws2tcpip.h>
#include <Iphlpapi.h>
#pragma comment (lib, "Ws2_32.lib")
#pragma comment(lib,"Iphlpapi.lib")
#else
#include <netdb.h>
#include <arpa/inet.h>
#include <sys/ioctl.h>
#include <sys/socket.h>
#include <net/if.h>
#include <netinet/in.h>
#include <netinet/tcp.h>
#endif // defined(_WIN32)
#include <algorithm>// std::transform(), std::find(), std::min(), std::max()
#include <cinttypes>// PRIu64, etc
#include <cmath>
#include <cstddef>// size_t
#include <cstdint>// uint8_t, etc
#include <cstring>// std::memcmp(), std::memcpy()
#include <memory>
#include <openssl/bio.h>
#include <openssl/ssl.h>
#include <openssl/x509.h>
#include <string>
namespace Utils {
class Byte {
public:
/**
* Getters below get value in Host Byte Order.
* Setters below set value in Network Byte Order.
*/
static uint8_t Get1Byte(const uint8_t *data, size_t i);
static uint16_t Get2Bytes(const uint8_t *data, size_t i);
static uint32_t Get3Bytes(const uint8_t *data, size_t i);
static uint32_t Get4Bytes(const uint8_t *data, size_t i);
static uint64_t Get8Bytes(const uint8_t *data, size_t i);
static void Set1Byte(uint8_t *data, size_t i, uint8_t value);
static void Set2Bytes(uint8_t *data, size_t i, uint16_t value);
static void Set3Bytes(uint8_t *data, size_t i, uint32_t value);
static void Set4Bytes(uint8_t *data, size_t i, uint32_t value);
static void Set8Bytes(uint8_t *data, size_t i, uint64_t value);
static uint16_t PadTo4Bytes(uint16_t size);
static uint32_t PadTo4Bytes(uint32_t size);
};
/* Inline static methods. */
inline uint8_t Byte::Get1Byte(const uint8_t *data, size_t i) { return data[i]; }
inline uint16_t Byte::Get2Bytes(const uint8_t *data, size_t i) {
return uint16_t{data[i + 1]} | uint16_t{data[i]} << 8;
}
inline uint32_t Byte::Get3Bytes(const uint8_t *data, size_t i) {
return uint32_t{data[i + 2]} | uint32_t{data[i + 1]} << 8 | uint32_t{data[i]} << 16;
}
inline uint32_t Byte::Get4Bytes(const uint8_t *data, size_t i) {
return uint32_t{data[i + 3]} | uint32_t{data[i + 2]} << 8 | uint32_t{data[i + 1]} << 16 |
uint32_t{data[i]} << 24;
}
inline uint64_t Byte::Get8Bytes(const uint8_t *data, size_t i) {
return uint64_t{Byte::Get4Bytes(data, i)} << 32 | Byte::Get4Bytes(data, i + 4);
}
inline void Byte::Set1Byte(uint8_t *data, size_t i, uint8_t value) { data[i] = value; }
inline void Byte::Set2Bytes(uint8_t *data, size_t i, uint16_t value) {
data[i + 1] = static_cast<uint8_t>(value);
data[i] = static_cast<uint8_t>(value >> 8);
}
inline void Byte::Set3Bytes(uint8_t *data, size_t i, uint32_t value) {
data[i + 2] = static_cast<uint8_t>(value);
data[i + 1] = static_cast<uint8_t>(value >> 8);
data[i] = static_cast<uint8_t>(value >> 16);
}
inline void Byte::Set4Bytes(uint8_t *data, size_t i, uint32_t value) {
data[i + 3] = static_cast<uint8_t>(value);
data[i + 2] = static_cast<uint8_t>(value >> 8);
data[i + 1] = static_cast<uint8_t>(value >> 16);
data[i] = static_cast<uint8_t>(value >> 24);
}
inline void Byte::Set8Bytes(uint8_t *data, size_t i, uint64_t value) {
data[i + 7] = static_cast<uint8_t>(value);
data[i + 6] = static_cast<uint8_t>(value >> 8);
data[i + 5] = static_cast<uint8_t>(value >> 16);
data[i + 4] = static_cast<uint8_t>(value >> 24);
data[i + 3] = static_cast<uint8_t>(value >> 32);
data[i + 2] = static_cast<uint8_t>(value >> 40);
data[i + 1] = static_cast<uint8_t>(value >> 48);
data[i] = static_cast<uint8_t>(value >> 56);
}
inline uint16_t Byte::PadTo4Bytes(uint16_t size) {
// If size is not multiple of 32 bits then pad it.
if (size & 0x03)
return (size & 0xFFFC) + 4;
else
return size;
}
}// namespace Utils
#endif
webrtc/answer.sdp 0 → 100644
v=0
o=- 2712902958023202213 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:68ws
a=ice-pwd:68wscAOadU4rdMwZ4Ow14M5u
a=ice-options:trickle
a=fingerprint:sha-256 BF:F8:C2:35:19:2E:C3:17:44:5A:BA:DF:08:C4:65:87:AB:0C:85:B9:91:69:80:8E:55:D9:2F:EF:5C:D6:F3:A2
a=setup:active
a=mid:0
a=ice-lite
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
m=video 9 UDP/TLS/RTP/SAVPF 102 121
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:68ws
a=ice-pwd:68wscAOadU4rdMwZ4Ow14M5u
a=ice-options:trickle
a=fingerprint:sha-256 BF:F8:C2:35:19:2E:C3:17:44:5A:BA:DF:08:C4:65:87:AB:0C:85:B9:91:69:80:8E:55:D9:2F:EF:5C:D6:F3:A2
a=setup:active
a=mid:1
a=ice-lite
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=recvonly
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
webrtc/janus_answer.sdp 0 → 100644
v=0
o=mozilla...THIS_IS_SDPARTA-87.0 5715694060540081921 0 IN IP4 0.0.0.0
s=-
t=0 0
a=fingerprint:sha-256 9C:9C:D9:4B:2B:82:84:95:C2:23:BC:3A:0B:BE:3B:8B:D5:6A:DA:A6:B2:F4:43:5E:C1:07:48:F8:AF:CE:FC:3D
a=group:BUNDLE video data
a=ice-options:trickle
a=msid-semantic:WMS *
m=video 9 UDP/TLS/RTP/SAVPF 120 122 121 123 126 127 97 98
c=IN IP4 0.0.0.0
a=sendrecv
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:5 urn:ietf:params:rtp-hdrext:toffset
a=extmap:7 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
a=fmtp:120 max-fs=12288;max-fr=60
a=fmtp:122 apt=120
a=fmtp:121 max-fs=12288;max-fr=60
a=fmtp:123 apt=121
a=fmtp:127 apt=126
a=fmtp:98 apt=97
a=ice-pwd:518cb4fc626f82bef2ada4e9221dfb50
a=ice-ufrag:d5484fed
a=mid:video
a=msid:{9cd17d8b-f96d-4663-866b-72ddd3b8b54b} {3dcd399d-4933-41d5-89fd-87070cfc3cf6}
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:120 goog-remb
a=rtcp-fb:120 transport-cc
a=rtcp-fb:121 nack
a=rtcp-fb:121 nack pli
a=rtcp-fb:121 ccm fir
a=rtcp-fb:121 goog-remb
a=rtcp-fb:121 transport-cc
a=rtcp-fb:126 nack
a=rtcp-fb:126 nack pli
a=rtcp-fb:126 ccm fir
a=rtcp-fb:126 goog-remb
a=rtcp-fb:126 transport-cc
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=rtcp-fb:97 goog-remb
a=rtcp-fb:97 transport-cc
a=rtcp-mux
a=rtpmap:120 VP8/90000
a=rtpmap:122 rtx/90000
a=rtpmap:121 VP9/90000
a=rtpmap:123 rtx/90000
a=rtpmap:126 H264/90000
a=rtpmap:127 rtx/90000
a=rtpmap:97 H264/90000
a=rtpmap:98 rtx/90000
a=setup:active
a=ssrc:697402393 cname:{f1ba68ef-ad59-4fff-a1f3-dbbb0cda3118}
a=ssrc:258662392 cname:{f1ba68ef-ad59-4fff-a1f3-dbbb0cda3118}
a=ssrc-group:FID 697402393 258662392
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=sendrecv
a=ice-pwd:518cb4fc626f82bef2ada4e9221dfb50
a=ice-ufrag:d5484fed
a=mid:data
a=setup:active
a=sctp-port:5000
a=max-message-size:1073741823
webrtc/janus_offer.sdp 0 → 100644
v=0
o=mozilla...THIS_IS_SDPARTA-87.0 1617247876363377 1 IN IP4 23.101.8.213
s=-
t=0 0
a=group:BUNDLE video data
a=msid-semantic: WMS janus
a=ice-lite
a=sendrecv
m=video 9 UDP/TLS/RTP/SAVPF 120 121 126 97 122 123 127 98
c=IN IP4 23.101.8.213
a=sendrecv
a=mid:video
a=rtcp-mux
a=ice-ufrag:ePgh
a=ice-pwd:ZURiB67/xs69E76aOa7JDw
a=ice-options:trickle
a=fingerprint:sha-256 EF:7A:50:9C:05:8C:EF:84:4D:72:B2:74:30:BA:FD:82:76:D1:C3:FE:0C:A0:10:43:B8:6C:B2:ED:B3:F7:77:8B
a=setup:actpass
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:5 urn:ietf:params:rtp-hdrext:toffset
a=extmap:6/recvonly http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
a=fmtp:120 max-fs=12288;max-fr=60
a=fmtp:121 max-fs=12288;max-fr=60
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:120 goog-remb
a=rtcp-fb:120 transport-cc
a=rtcp-fb:121 nack
a=rtcp-fb:121 nack pli
a=rtcp-fb:121 ccm fir
a=rtcp-fb:121 goog-remb
a=rtcp-fb:121 transport-cc
a=rtcp-fb:126 nack
a=rtcp-fb:126 nack pli
a=rtcp-fb:126 ccm fir
a=rtcp-fb:126 goog-remb
a=rtcp-fb:126 transport-cc
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=rtcp-fb:97 goog-remb
a=rtcp-fb:97 transport-cc
a=rtpmap:120 VP8/90000
a=rtpmap:121 VP9/90000
a=rtpmap:126 H264/90000
a=rtpmap:97 H264/90000
a=rtpmap:122 rtx/90000
a=fmtp:122 apt=120
a=rtpmap:123 rtx/90000
a=fmtp:123 apt=121
a=rtpmap:127 rtx/90000
a=fmtp:127 apt=126
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=97
a=ssrc-group:FID 3581519470 2258376012
a=msid:janus janusv0
a=ssrc:3581519470 cname:janus
a=ssrc:3581519470 msid:janus janusv0
a=ssrc:3581519470 mslabel:janus
a=ssrc:3581519470 label:janusv0
a=ssrc:2258376012 cname:janus
a=ssrc:2258376012 msid:janus janusv0
a=ssrc:2258376012 mslabel:janus
a=ssrc:2258376012 label:janusv0
a=candidate:1 1 udp 2013266431 23.101.8.213 41901 typ host
a=end-of-candidates
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 23.101.8.213
a=sendrecv
a=sctp-port:5000
a=mid:data
a=ice-ufrag:ePgh
a=ice-pwd:ZURiB67/xs69E76aOa7JDw
a=ice-options:trickle
a=fingerprint:sha-256 EF:7A:50:9C:05:8C:EF:84:4D:72:B2:74:30:BA:FD:82:76:D1:C3:FE:0C:A0:10:43:B8:6C:B2:ED:B3:F7:77:8B
a=setup:actpass
a=candidate:1 1 udp 2013266431 23.101.8.213 41901 typ host
a=end-of-candidates
webrtc/logger.h 0 → 100644
#pragma once
#include <stdio.h>
#include <assert.h>
#if 0
#define MS_TRACE()
#define MS_ERROR(fmt, ...) printf("error:" fmt "\n", ##__VA_ARGS__)
#define MS_THROW_ERROR(fmt, ...) do{ printf("throw:" fmt "\n", ##__VA_ARGS__); throw std::runtime_error("error"); } while(false);
#define MS_DUMP(fmt, ...) printf("dump:" fmt "\n", ##__VA_ARGS__)
#define MS_DEBUG_2TAGS(tag1, tag2,fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__)
#define MS_WARN_2TAGS(tag1, tag2,fmt, ...) printf("warn:" fmt "\n", ##__VA_ARGS__)
#define MS_DEBUG_TAG(tag,fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__)
#define MS_ASSERT(con, fmt, ...) do{if(!(con)) { printf("assert failed:%s" fmt "\n", #con, ##__VA_ARGS__);} assert(con); } while(false);
#define MS_ABORT(fmt, ...) do{ printf("abort:" fmt "\n", ##__VA_ARGS__); abort(); } while(false);
#define MS_WARN_TAG(tag,fmt, ...) printf("warn:" fmt "\n", ##__VA_ARGS__)
#define MS_DEBUG_DEV(fmt, ...) printf("debug:" fmt "\n", ##__VA_ARGS__)
#else
#define MS_TRACE()
#define MS_ERROR(fmt, ...)
#define MS_THROW_ERROR(fmt, ...)
#define MS_DUMP(fmt, ...)
#define MS_DEBUG_2TAGS(tag1, tag2,fmt, ...)
#define MS_WARN_2TAGS(tag1, tag2,fmt, ...)
#define MS_DEBUG_TAG(tag,fmt, ...)
#define MS_ASSERT(con, fmt, ...)
#define MS_ABORT(fmt, ...)
#define MS_WARN_TAG(tag,fmt, ...)
#define MS_DEBUG_DEV(fmt, ...)
#endif
\ No newline at end of file
webrtc/offer-simulcast.sdp 0 → 100644
# chrome的sdp
v=0
o=- 403371946498103831 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:pW4Z
a=ice-pwd:S38S++HW3eTcPTyytsNI1XVp
a=ice-options:trickle
a=fingerprint:sha-256 04:32:7B:56:7D:F7:D4:EC:65:7C:04:6C:F8:0B:03:F0:35:A9:1A:C3:43:3E:18:95:67:E6:0D:D1:EE:C9:16:8C
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- a3c6a137-1291-45cd-b985-07a9bd365452
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3626257331 cname:JSFJMbaE9Pu5tevN
a=ssrc:3626257331 msid:- a3c6a137-1291-45cd-b985-07a9bd365452
a=ssrc:3626257331 mslabel:-
a=ssrc:3626257331 label:a3c6a137-1291-45cd-b985-07a9bd365452
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:pW4Z
a=ice-pwd:S38S++HW3eTcPTyytsNI1XVp
a=ice-options:trickle
a=fingerprint:sha-256 04:32:7B:56:7D:F7:D4:EC:65:7C:04:6C:F8:0B:03:F0:35:A9:1A:C3:43:3E:18:95:67:E6:0D:D1:EE:C9:16:8C
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- 261e5384-9cf6-479d-9d59-aaf924d1a2ea
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:35 AV1X/90000
a=rtcp-fb:35 goog-remb
a=rtcp-fb:35 transport-cc
a=rtcp-fb:35 ccm fir
a=rtcp-fb:35 nack
a=rtcp-fb:35 nack pli
a=rtpmap:36 rtx/90000
a=fmtp:36 apt=35
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=124
a=rtpmap:123 H264/90000
a=rtcp-fb:123 goog-remb
a=rtcp-fb:123 transport-cc
a=rtcp-fb:123 ccm fir
a=rtcp-fb:123 nack
a=rtcp-fb:123 nack pli
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f
a=rtpmap:118 rtx/90000
a=fmtp:118 apt=123
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=rid:q send
a=rid:h send
a=rid:f send
a=simulcast:send q;h;f
#firefox的sdp
v=0
o=mozilla...THIS_IS_SDPARTA-88.0.1 3954544078885279475 0 IN IP4 0.0.0.0
s=-
t=0 0
a=fingerprint:sha-256 9B:4F:D1:D2:A5:ED:08:BC:E8:D7:DD:D8:59:2C:E6:3D:19:F9:4C:67:9C:D9:9B:7B:C9:47:7A:3A:1F:05:C8:96
a=group:BUNDLE 0 1
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:92a9ced6d734f7ff2a45cde8b29572a9
a=ice-ufrag:b986b945
a=mid:0
a=msid:- {ea61729a-c244-4c79-aeb7-b57765fefa26}
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000/1
a=setup:actpass
a=ssrc:3000327501 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc}
m=video 9 UDP/TLS/RTP/SAVPF 120 124 121 125 126 127 97 98
c=IN IP4 0.0.0.0
a=sendrecv
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:5 urn:ietf:params:rtp-hdrext:toffset
a=extmap:6/recvonly http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:8/sendonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:9/sendonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
a=fmtp:120 max-fs=12288;max-fr=60
a=fmtp:124 apt=120
a=fmtp:121 max-fs=12288;max-fr=60
a=fmtp:125 apt=121
a=fmtp:127 apt=126
a=fmtp:98 apt=97
a=ice-pwd:92a9ced6d734f7ff2a45cde8b29572a9
a=ice-ufrag:b986b945
a=mid:1
a=msid:- {3bfe1b80-20eb-4b42-b8b7-fac45fb281bf}
a=rid:q send
a=rid:h send
a=rid:f send
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:120 goog-remb
a=rtcp-fb:120 transport-cc
a=rtcp-fb:121 nack
a=rtcp-fb:121 nack pli
a=rtcp-fb:121 ccm fir
a=rtcp-fb:121 goog-remb
a=rtcp-fb:121 transport-cc
a=rtcp-fb:126 nack
a=rtcp-fb:126 nack pli
a=rtcp-fb:126 ccm fir
a=rtcp-fb:126 goog-remb
a=rtcp-fb:126 transport-cc
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=rtcp-fb:97 goog-remb
a=rtcp-fb:97 transport-cc
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:120 VP8/90000
a=rtpmap:124 rtx/90000
a=rtpmap:121 VP9/90000
a=rtpmap:125 rtx/90000
a=rtpmap:126 H264/90000
a=rtpmap:127 rtx/90000
a=rtpmap:97 H264/90000
a=rtpmap:98 rtx/90000
a=setup:actpass
a=simulcast:send q;h;f
a=ssrc:2581133096 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc}
a=ssrc:773854125 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc}
a=ssrc:4100728001 cname:{12e7c547-559e-46e2-94db-f1ad474c95dc}
webrtc/offer.sdp 0 → 100644
v=0
o=- 8056465047193717905 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:LtFR
a=ice-pwd:sUVVlvhNoL2g/GL36TyfZGwP
a=ice-options:trickle
a=fingerprint:sha-256 21:21:07:E8:3C:D0:3B:45:87:9A:31:86:DE:4F:C1:BA:E1:0E:96:BA:41:36:6E:3A:3F:C6:C8:92:95:5B:71:5F
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- 2ebeb64c-2eb3-4c4f-b5d5-d578245b969e
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:905965261 cname:7iEkMV0/MMfqSEce
a=ssrc:905965261 msid:- 2ebeb64c-2eb3-4c4f-b5d5-d578245b969e
a=ssrc:905965261 mslabel:-
a=ssrc:905965261 label:2ebeb64c-2eb3-4c4f-b5d5-d578245b969e
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:LtFR
a=ice-pwd:sUVVlvhNoL2g/GL36TyfZGwP
a=ice-options:trickle
a=fingerprint:sha-256 21:21:07:E8:3C:D0:3B:45:87:9A:31:86:DE:4F:C1:BA:E1:0E:96:BA:41:36:6E:3A:3F:C6:C8:92:95:5B:71:5F
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- f36bb41d-d05d-4310-b05b-7913d0029b18
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:35 AV1X/90000
a=rtcp-fb:35 goog-remb
a=rtcp-fb:35 transport-cc
a=rtcp-fb:35 ccm fir
a=rtcp-fb:35 nack
a=rtcp-fb:35 nack pli
a=rtpmap:36 rtx/90000
a=fmtp:36 apt=35
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=124
a=rtpmap:123 H264/90000
a=rtcp-fb:123 goog-remb
a=rtcp-fb:123 transport-cc
a=rtcp-fb:123 ccm fir
a=rtcp-fb:123 nack
a=rtcp-fb:123 nack pli
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032
a=rtpmap:118 rtx/90000
a=fmtp:118 apt=123
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 2678501654 361960375
a=ssrc:2678501654 cname:7iEkMV0/MMfqSEce
a=ssrc:2678501654 msid:- f36bb41d-d05d-4310-b05b-7913d0029b18
a=ssrc:2678501654 mslabel:-
a=ssrc:2678501654 label:f36bb41d-d05d-4310-b05b-7913d0029b18
a=ssrc:361960375 cname:7iEkMV0/MMfqSEce
a=ssrc:361960375 msid:- f36bb41d-d05d-4310-b05b-7913d0029b18
a=ssrc:361960375 mslabel:-
a=ssrc:361960375 label:f36bb41d-d05d-4310-b05b-7913d0029b18
\ No newline at end of file
webrtc/readme.md 0 → 100644
# 致谢与声明
本文件夹下部分文件提取自[MediaSoup](https://github.com/versatica/mediasoup) ，分别为：
- ice相关功能：
- IceServer.cpp
- IceServer.hpp
- StunPacket.cpp
- StunPacket.hpp
- Utils.hpp
- dtls相关功能：
- DtlsTransport.cpp
- DtlsTransport.hpp
- srtp相关功能：
- SrtpSession.cpp
- SrtpSession.hpp
以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者，
用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。
同时,在此也感谢开源项目[easy_webrtc_server](https://github.com/Mihawk086/easy_webrtc_server) 及作者，
在集成MediaSoup相关代码前期，主要参考这个项目。
另外，感谢[big panda](<2381267071@qq.com>) 开发并贡献的webrtc js测试客户端(www/webrtc目录下文件)，
其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js
# 现状与规划
ZLMediaKit的WebRTC相关功能目前仅供测试与开发，现在还不成熟，后续主要工作有：
- 1、完善webrtc rtcp相关功能，包括丢包重传、带宽检测等功能。
- 2、实现rtp重传等相关功能。
- 3、实现simulcast相关功能。
- 4、fec、rtp扩展等其他功能。
- 5、如果精力允许，逐步替换MediaSoup相关代码，改用自有版权代码。
www/webrtc/ZLMRTCClient.js 0 → 100644
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www/webrtc/ZLMRTCClient.js.map 0 → 100644
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www/webrtc/index.html 0 → 100644
<html>
<meta charset="utf-8">
<head>
<title>ZLM RTC demo</title>
<script src="./ZLMRTCClient.js"></script>
</head>
<body>
<div style="text-align: center;">
<div>
<video id='video' controls autoplay style="text-align:left;">
Your browser is too old which doesn't support HTML5 video.
</video>
<video id='selfVideo' controls autoplay style="text-align:right;">
Your browser is too old which doesn't support HTML5 video.
</video>
</div>
<div>
<p>
<label for="streamUrl">url:</label>
<input type="text" style="co" id='streamUrl' value="http://127.0.0.1/index/api/webrtc?app=live&stream=test&type=play">
</p>
<p>
<label for="simulcast">simulcast:</label>
<input type="checkbox" id='simulcast' checked="checked">
</p>
<p>
<label for="useCamera">useCamera:</label>
<input type="checkbox" id='useCamera' checked="checked">
</p>
<p>
<label for="audioEnable">audioEnable:</label>
<input type="checkbox" id='audioEnable' checked="checked">
</p>
<p>
<label for="videoEnable">videoEnable:</label>
<input type="checkbox" id='videoEnable' checked="checked">
</p>
<p>
<label for="method">method(play or push):</label>
<input type="radio" name="method" value="push" >push
<input type="radio" name="method" value="play" checked = true>play
</p>
<p>
<label for="resolution">resolution:</label>
<select id="resolution">
</select>
</p>
<button onclick="start()">开始</button>
<button onclick="stop()">停止</button>
</div>
</div>
<script>
var player = null
var recvOnly = true
var resArr = []
document.getElementsByName("method").forEach((el,idx)=>{
el.onclick=function(e){
if(el.value == "play")
{
let url = document.getElementById('streamUrl').value;
document.getElementById('streamUrl').value = url.replace("&type=push","&type=play");
recvOnly = true;
}
else
{
let url = document.getElementById('streamUrl').value;
document.getElementById('streamUrl').value = url.replace("&type=play","&type=push");
recvOnly = false;
}
};
});
ZLMRTCClient.GetAllScanResolution().forEach((r,i)=>{
opt = document.createElement('option');
opt.text = r.label +"("+r.width+"x"+r.height+")";
opt.value = r;
document.getElementById("resolution").add(opt,null)
//console.log(opt.text.match(/\d+/g))
})
function start_play(){
let elr = document.getElementById("resolution");
let res = elr.options[elr.selectedIndex].text.match(/\d+/g);
let h = parseInt(res.pop());
let w = parseInt(res.pop());
player = new ZLMRTCClient.Endpoint(
{
element: document.getElementById('video'),// video 标签
debug: true,// 是否打印日志
zlmsdpUrl:document.getElementById('streamUrl').value,//流地址
simulcast:false,//document.getElementById('simulcast').checked,
useCamera:document.getElementById('useCamera').checked,
audioEnable:document.getElementById('audioEnable').checked,
videoEnable:document.getElementById('videoEnable').checked,
recvOnly:recvOnly,
resolution:{w:w,h:h}
}
);
player.on(ZLMRTCClient.Events.WEBRTC_ICE_CANDIDATE_ERROR,function(e)
{// ICE 协商出错
console.log('ICE 协商出错')
});
player.on(ZLMRTCClient.Events.WEBRTC_ON_REMOTE_STREAMS,function(e)
{//获取到了远端流，可以播放
console.log('播放成功',e.streams)
});
player.on(ZLMRTCClient.Events.WEBRTC_OFFER_ANSWER_EXCHANGE_FAILED,function(e)
{// offer answer 交换失败
console.log('offer answer 交换失败',e)
stop();
});
player.on(ZLMRTCClient.Events.WEBRTC_ON_LOCAL_STREAM,function(s)
{// 获取到了本地流
document.getElementById('selfVideo').srcObject=s;
document.getElementById('selfVideo').muted = true;
//console.log('offer answer 交换失败',e)
});
player.on(ZLMRTCClient.Events.CAPTURE_STREAM_FAILED,function(s)
{// 获取本地流失败
console.log('获取本地流失败')
});
}
function start()
{
stop();
let elr = document.getElementById("resolution");
let res = elr.options[elr.selectedIndex].text.match(/\d+/g);
let h = parseInt(res.pop());
let w = parseInt(res.pop());
if(document.getElementById('useCamera').checked && !recvOnly)
{
ZLMRTCClient.isSupportResolution(w,h).then(e=>{
start_play()
}).catch(e=>{
alert("not support resolution")
});
}else{
start_play()
}
}
function stop()
{
if(player)
{
player.close();
player = null;
var local = document.getElementById('selfVideo');
local.removeAttribute('srcObject');
local.load();
}
}
</script>
</body>
<script>
</script>
</html>
\ No newline at end of file
www/webrtc/readme.txt 0 → 100644
感谢[big panda](<2381267071@qq.com>) 开发并贡献此webrtc js测试客户端,
其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js
\ No newline at end of file
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