Bumped version to 0.13.3

Fix possible static de-initialization order issue in SctpTransport

CMakeLists.txt

@@ -1,6 +1,6 @@
 cmake_minimum_required(VERSION 3.7)
 project(libdatachannel
-	VERSION 0.12.1
+	VERSION 0.13.3
 	LANGUAGES CXX)
 set(PROJECT_DESCRIPTION "WebRTC Data Channels Library")
 
@@ -43,8 +43,7 @@ set(LIBDATACHANNEL_SOURCES
 	${CMAKE_CURRENT_SOURCE_DIR}/src/configuration.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/datachannel.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/description.cpp
-	${CMAKE_CURRENT_SOURCE_DIR}/src/init.cpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/global.cpp
-	${CMAKE_CURRENT_SOURCE_DIR}/src/log.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/message.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/peerconnection.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpreceivingsession.cpp
@@ -62,6 +61,7 @@ set(LIBDATACHANNEL_SOURCES
 	${CMAKE_CURRENT_SOURCE_DIR}/src/mediahandlerelement.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/mediahandlerrootelement.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpnackresponder.cpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/rtp.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/capi.cpp
 )
 
@@ -75,8 +75,7 @@ set(LIBDATACHANNEL_HEADERS
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/mediahandler.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcpreceivingsession.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/common.hpp
-	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/init.hpp
+	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/global.hpp
-	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/log.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/message.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/peerconnection.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/reliability.hpp
@@ -107,6 +106,7 @@ set(LIBDATACHANNEL_IMPL_SOURCES
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/dtlssrtptransport.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/dtlstransport.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/icetransport.cpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/init.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/peerconnection.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/logcounter.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/sctptransport.cpp
@@ -129,6 +129,8 @@ set(LIBDATACHANNEL_IMPL_HEADERS
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/dtlssrtptransport.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/dtlstransport.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/icetransport.hpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/init.hpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/internals.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/peerconnection.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/queue.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/impl/logcounter.hpp
@@ -213,17 +215,17 @@ set_target_properties(datachannel-static PROPERTIES
 	VERSION ${PROJECT_VERSION}
 	CXX_STANDARD 17)
 
-target_include_directories(datachannel PUBLIC ${CMAKE_CURRENT_SOURCE_DIR}/include)
+target_include_directories(datachannel PUBLIC $<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}/include> $<INSTALL_INTERFACE:include>)
 target_include_directories(datachannel PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}/include/rtc)
 target_include_directories(datachannel PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}/src)
-target_link_libraries(datachannel PUBLIC Threads::Threads plog::plog)
+target_link_libraries(datachannel PUBLIC Threads::Threads)
-target_link_libraries(datachannel PRIVATE Usrsctp::Usrsctp)
+target_link_libraries(datachannel PRIVATE Usrsctp::Usrsctp plog::plog)
 
 target_include_directories(datachannel-static PUBLIC ${CMAKE_CURRENT_SOURCE_DIR}/include)
 target_include_directories(datachannel-static PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}/include/rtc)
 target_include_directories(datachannel-static PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}/src)
-target_link_libraries(datachannel-static PUBLIC Threads::Threads plog::plog)
+target_link_libraries(datachannel-static PUBLIC Threads::Threads)
-target_link_libraries(datachannel-static PRIVATE Usrsctp::Usrsctp)
+target_link_libraries(datachannel-static PRIVATE Usrsctp::Usrsctp plog::plog)
 
 if(WIN32)
 	target_link_libraries(datachannel PUBLIC ws2_32) # winsock2
@@ -310,9 +312,6 @@ endif()
 add_library(LibDataChannel::LibDataChannel ALIAS datachannel)
 add_library(LibDataChannel::LibDataChannelStatic ALIAS datachannel-static)
 
-install(TARGETS datachannel LIBRARY DESTINATION lib)
-install(FILES ${LIBDATACHANNEL_HEADERS} DESTINATION include/rtc)
-
 if(NOT MSVC)
 	target_compile_options(datachannel PRIVATE -Wall -Wextra)
 	target_compile_options(datachannel-static PRIVATE -Wall -Wextra)
@@ -328,6 +327,22 @@ if(WARNINGS_AS_ERRORS)
 	endif()
 endif()
 
+install(TARGETS datachannel EXPORT libdatachannel-config
+	RUNTIME DESTINATION bin
+	LIBRARY DESTINATION lib
+	ARCHIVE DESTINATION lib
+)
+
+install(FILES ${LIBDATACHANNEL_HEADERS}
+	DESTINATION include/rtc
+)
+
+install(
+  EXPORT libdatachannel-config
+  NAMESPACE LibDatachannel::
+  DESTINATION share/cmake/libdatachannel
+)
+
 # Tests
 if(NOT NO_TESTS)
 	if(CMAKE_SYSTEM_NAME STREQUAL "WindowsStore")

Jamfile

@@ -12,9 +12,12 @@ feature.compose <gnutls>on
 lib libdatachannel
 	: # sources
 	[ glob ./src/*.cpp ]
+	[ glob ./src/impl/*.cpp ]
 	: # requirements
 	<cxxstd>17
+	<include>./include
 	<include>./include/rtc
+	<include>./src
 	<define>RTC_ENABLE_MEDIA=0
 	<define>RTC_ENABLE_WEBSOCKET=0
 	<define>USE_NICE=0
@@ -143,7 +146,7 @@ rule make_libjuice_openssl ( targets * : sources * : properties * )
 	{
 		# on macOS, default to pick up openssl from the homebrew installation
 		# brew install openssl
-		OPENSSL_INCLUDE = /usr/local/opt/openssl/include ;
+		OPENSSL_INCLUDE = /opt/homebrew/opt/openssl /usr/local/opt/openssl/include ;
 	}
 
 	if $(OPENSSL_INCLUDE) != ""
@@ -191,7 +194,7 @@ rule openssl-lib-path ( properties * )
     {
         # on macOS, default to pick up openssl from the homebrew installation
         # brew install openssl
-        OPENSSL_LIB = /usr/local/opt/openssl/lib ;
+        OPENSSL_LIB = /opt/homebrew/opt/openssl/lib /usr/local/opt/openssl/lib ;
     }
     else if <target-os>windows in $(properties) && $(OPENSSL_LIB) = ""
     {
@@ -217,7 +220,7 @@ rule openssl-include-path ( properties * )
     {
         # on macOS, default to pick up openssl from the homebrew installation
         # brew install openssl
-        OPENSSL_INCLUDE = /usr/local/opt/openssl/include ;
+        OPENSSL_INCLUDE = /opt/homebrew/opt/openssl/include /usr/local/opt/openssl/include ;
     }
     else if <target-os>windows in $(properties) && $(OPENSSL_INCLUDE) = ""
     {

examples/README.md

@@ -10,6 +10,7 @@ This directory contains different WebRTC clients and compatible WebSocket + JSON
 - [signaling-server-rust](signaling-server-rust) contains a similar signaling server in Rust (see [lerouxrgd/datachannel-rs](https://github.com/lerouxrgd/datachannel-rs) for Rust wrappers)
 
 - [media](media) is a copy/paste demo to send the webcam from your browser into gstreamer.
+- [sfu-media](sfu-media) is a copy/paste SFU demo to relay the webcam between browsers.
 - [streamer](streamer) streams h264 and opus samples to web browsers (signaling-server-python is required).
 
 Additionally, it contains two debugging tools for libdatachannel with copy-pasting as signaling:

examples/client-benchmark/README.md

@@ -1,8 +1,9 @@
 # libdatachannel - client-benchmark
 
-This directory contains a native client to open Data Channels with WebSocket signaling using libdatachannel and benchmark functionalities. It offers two functionalities;
+This directory contains a native client to open Data Channels with WebSocket signaling using libdatachannel and benchmark functionalities. It offers three functionalities;
 - Benchmark: Bi-directional data transfer benchmark (Also supports One-Way testing)
 - Constant Throughput Set: Send desired amount of data per second
+- Multiple Data Channel: Create desried count of data channel 
 
 ## Start Signaling Server
 - Start one of the signaling server from the examples folder. For example start  `signaling-server-nodejs` like;
@@ -12,7 +13,7 @@ This directory contains a native client to open Data Channels with WebSocket sig
 
 ## Start `client-benchmark` Applications
 
-Start 2 applications by using example calls below. Than copy one of the client's ID and paste to the other peer's screen to start offering process.
+Start 2 applications by using example calls below. Then copy one of the client's ID and paste to the other peer's screen to start offering process.
 
 ## Usage Examples
 
@@ -23,63 +24,90 @@ Start 2 applications by using example calls below. Than copy one of the client's
 Example Output (Offering Peer's Output);
 ```bash
 Stun server is stun:stun.l.google.com:19302
-The local ID is: EQmF
+The local ID is: H1E3
-Url is ws://localhost:8000/EQmF
+Url is ws://localhost:8000/H1E3
 Waiting for signaling to be connected...
-2021-03-25 14:21:58.045 INFO  [21386] [rtc::impl::TcpTransport::connect@159] Connected to localhost:8000
+2021-04-10 19:51:31.319 INFO  [16449] [rtc::impl::TcpTransport::connect@163] Connected to localhost:8000
-2021-03-25 14:21:58.045 INFO  [21386] [rtc::impl::TcpTransport::runLoop@327] TCP connected
+2021-04-10 19:51:31.319 INFO  [16449] [rtc::impl::TcpTransport::runLoop@331] TCP connected
-2021-03-25 14:21:58.046 INFO  [21386] [rtc::impl::WsTransport::incoming@118] WebSocket open
+2021-04-10 19:51:31.321 INFO  [16449] [rtc::impl::WsTransport::incoming@118] WebSocket open
 WebSocket connected, signaling ready
 Enter a remote ID to send an offer:
-0tDf
+n790
-Offering to 0tDf
+Offering to n790
-Creating DataChannel with label "benchmark"
+Creating DataChannel with label "DC-1"
-2021-03-25 14:22:07.972 INFO  [21379] [rtc::impl::IceTransport::IceTransport@106] Using STUN server "stun.l.google.com:19302"
+2021-04-10 19:51:32.464 INFO  [16442] [rtc::impl::IceTransport::IceTransport@106] Using STUN server "stun.l.google.com:19302"
-2021-03-25 14:22:07.973 INFO  [21379] [rtc::impl::PeerConnection::changeSignalingState@992] Changed signaling state to new
+2021-04-10 19:51:32.465 INFO  [16442] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to new
-2021-03-25 14:22:07.973 INFO  [21379] [rtc::impl::PeerConnection::changeGatheringState@981] Changed gathering state to in-progress
+2021-04-10 19:51:32.465 INFO  [16442] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to in-progress
 Gathering State: in-progress
-2021-03-25 14:22:07.974 INFO  [21379] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to gathering
+2021-04-10 19:51:32.465 INFO  [16442] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to gathering
-2021-03-25 14:22:07.974 WARN  [21379] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:239: Local description already has the maximum number of host candidates
 Benchmark will run for 300 seconds
-2021-03-25 14:22:07.976 INFO  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connecting
+2021-04-10 19:51:32.466 INFO  [16450] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connecting
-2021-03-25 14:22:07.976 INFO  [21396] [rtc::impl::PeerConnection::changeState@964] Changed state to connecting
+2021-04-10 19:51:32.466 INFO  [16450] [rtc::impl::PeerConnection::changeState@1016] Changed state to connecting
 State: connecting
-2021-03-25 14:22:08.055 INFO  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:610: Using STUN server stun.l.google.com:19302
+2021-04-10 19:51:32.489 INFO  [16450] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:610: Using STUN server stun.l.google.com:19302
-2021-03-25 14:22:08.055 INFO  [21386] [rtc::impl::PeerConnection::changeSignalingState@992] Changed signaling state to connecting
+2021-04-10 19:51:32.489 INFO  [16449] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to connecting
-2021-03-25 14:22:08.105 WARN  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:426: Send failed, errno=101
+2021-04-10 19:51:32.490 INFO  [16450] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connected
-2021-03-25 14:22:08.105 WARN  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1483: STUN message send failed, errno=101
+2021-04-10 19:51:32.491 INFO  [16453] [rtc::impl::DtlsTransport::runRecvLoop@503] DTLS handshake finished
-2021-03-25 14:22:08.105 INFO  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:843: STUN binding failed
+2021-04-10 19:51:32.497 INFO  [16443] [rtc::impl::SctpTransport::processNotification@713] SCTP connected
-2021-03-25 14:22:08.107 INFO  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1302: STUN server binding successful
+2021-04-10 19:51:32.497 INFO  [16443] [rtc::impl::PeerConnection::changeState@1016] Changed state to connected
-2021-03-25 14:22:08.107 INFO  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:2148: Candidate gathering done
-2021-03-25 14:22:08.107 INFO  [21396] [rtc::impl::PeerConnection::changeGatheringState@981] Changed gathering state to complete
-Gathering State: complete
-2021-03-25 14:22:08.155 INFO  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connected
-2021-03-25 14:22:08.206 INFO  [21396] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to completed
-#1 Received: 0 KB/s   Sent: 0 KB/s   BufferSize: 0
-2021-03-25 14:22:09.059 INFO  [21399] [rtc::impl::DtlsTransport::runRecvLoop@503] DTLS handshake finished
-2021-03-25 14:22:09.069 INFO  [21382] [rtc::impl::SctpTransport::processNotification@708] SCTP connected
-2021-03-25 14:22:09.069 INFO  [21382] [rtc::impl::PeerConnection::changeState@964] Changed state to connected
 State: connected
-DataChannel from 0tDf open
+DataChannel from n790 open
-#2 Received: 41488 KB/s   Sent: 42465 KB/s   BufferSize: 65535
+2021-04-10 19:51:32.542 INFO  [16450] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1328: STUN server binding successful
-#3 Received: 43925 KB/s   Sent: 43729 KB/s   BufferSize: 65535
+2021-04-10 19:51:32.589 INFO  [16450] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to completed
-#4 Received: 42491 KB/s   Sent: 42361 KB/s   BufferSize: 65535
+#1
-#5 Received: 45878 KB/s   Sent: 45682 KB/s   BufferSize: 65535
+      DC-1 Received: 40789 KB/s   Sent: 41180 KB/s   BufferSize: 65535
-Stats# Received Total: 174 MB   Sent Total: 175 MB   RTT: 17 ms
+      TOTL Received: 40789 KB/s   Sent: 41180 KB/s
+2021-04-10 19:51:34.039 INFO  [16450] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:843: STUN server binding failed (timeout)
+2021-04-10 19:51:34.039 INFO  [16450] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:2206: Candidate gathering done
+2021-04-10 19:51:34.039 INFO  [16450] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to complete
+Gathering State: complete
+#2
+      DC-1 Received: 41709 KB/s   Sent: 41774 KB/s   BufferSize: 65535
+      TOTL Received: 41709 KB/s   Sent: 41774 KB/s
+#3
+      DC-1 Received: 42165 KB/s   Sent: 42360 KB/s   BufferSize: 65535
+      TOTL Received: 42165 KB/s   Sent: 42360 KB/s
+#4
+      DC-1 Received: 42880 KB/s   Sent: 42750 KB/s   BufferSize: 65535
+      TOTL Received: 42880 KB/s   Sent: 42750 KB/s
+#5
+      DC-1 Received: 41771 KB/s   Sent: 42097 KB/s   BufferSize: 65535
+      TOTL Received: 41771 KB/s   Sent: 42097 KB/s
+Stats# Received Total: 210 MB   Sent Total: 211 MB   RTT: 20 ms
 
-#6 Received: 43665 KB/s   Sent: 43599 KB/s   BufferSize: 65535
+#6
-#7 Received: 45749 KB/s   Sent: 45488 KB/s   BufferSize: 65535
+      DC-1 Received: 46235 KB/s   Sent: 30433 KB/s   BufferSize: 65535
-#8 Received: 44055 KB/s   Sent: 44055 KB/s   BufferSize: 65535
+      TOTL Received: 46235 KB/s   Sent: 30433 KB/s
-#9 Received: 21572 KB/s   Sent: 58199 KB/s   BufferSize: 65535
+#7
-#10 Received: 22940 KB/s   Sent: 55005 KB/s   BufferSize: 65535
+      DC-1 Received: 47116 KB/s   Sent: 28413 KB/s   BufferSize: 65535
-Stats# Received Total: 353 MB   Sent Total: 422 MB   RTT: 15 ms
+      TOTL Received: 47116 KB/s   Sent: 28413 KB/s
+#8
+      DC-1 Received: 46923 KB/s   Sent: 32520 KB/s   BufferSize: 65535
+      TOTL Received: 46923 KB/s   Sent: 32520 KB/s
+#9
+      DC-1 Received: 44513 KB/s   Sent: 34020 KB/s   BufferSize: 65535
+      TOTL Received: 44513 KB/s   Sent: 34020 KB/s
+#10
+      DC-1 Received: 41966 KB/s   Sent: 36166 KB/s   BufferSize: 65535
+      TOTL Received: 41966 KB/s   Sent: 36166 KB/s
+Stats# Received Total: 438 MB   Sent Total: 373 MB   RTT: 19 ms
 
-#11 Received: 27501 KB/s   Sent: 53112 KB/s   BufferSize: 65535
+#11
-#12 Received: 29914 KB/s   Sent: 48162 KB/s   BufferSize: 65535
+      DC-1 Received: 42617 KB/s   Sent: 39619 KB/s   BufferSize: 65535
-#13 Received: 31869 KB/s   Sent: 45946 KB/s   BufferSize: 65535
+      TOTL Received: 42617 KB/s   Sent: 39619 KB/s
-#14 Received: 22484 KB/s   Sent: 53636 KB/s   BufferSize: 65535
+#12
-#15 Received: 16294 KB/s   Sent: 56833 KB/s   BufferSize: 65535
+      DC-1 Received: 43792 KB/s   Sent: 43338 KB/s   BufferSize: 65535
-Stats# Received Total: 482 MB   Sent Total: 682 MB   RTT: 13 ms
+      TOTL Received: 43792 KB/s   Sent: 43338 KB/s
+#13
+      DC-1 Received: 41715 KB/s   Sent: 41585 KB/s   BufferSize: 65535
+      TOTL Received: 41715 KB/s   Sent: 41585 KB/s
+#14
+      DC-1 Received: 39860 KB/s   Sent: 33822 KB/s   BufferSize: 65535
+      TOTL Received: 39860 KB/s   Sent: 33822 KB/s
+#15
+      DC-1 Received: 47576 KB/s   Sent: 25352 KB/s   BufferSize: 65535
+      TOTL Received: 47576 KB/s   Sent: 25352 KB/s
+Stats# Received Total: 655 MB   Sent Total: 558 MB   RTT: 13 ms
 ```
 
 ### Benchmark for 300 seconds (Only Send, One Way)
@@ -92,64 +120,92 @@ Start second peer as;
 
 Example Output (Offering Peer's Output);
 ```bash
+Not Sending data. (One way benchmark).
 Stun server is stun:stun.l.google.com:19302
-The local ID is: XLLn
+The local ID is: 7EaP
-Url is ws://localhost:8000/XLLn
+Url is ws://localhost:8000/7EaP
 Waiting for signaling to be connected...
-2021-03-25 14:34:24.479 INFO  [22332] [rtc::impl::TcpTransport::connect@159] Connected to localhost:8000
+2021-04-10 19:54:36.857 INFO  [16632] [rtc::impl::TcpTransport::connect@163] Connected to localhost:8000
-2021-03-25 14:34:24.479 INFO  [22332] [rtc::impl::TcpTransport::runLoop@327] TCP connected
+2021-04-10 19:54:36.857 INFO  [16632] [rtc::impl::TcpTransport::runLoop@331] TCP connected
-2021-03-25 14:34:24.479 INFO  [22332] [rtc::impl::WsTransport::incoming@118] WebSocket open
+2021-04-10 19:54:36.858 INFO  [16632] [rtc::impl::WsTransport::incoming@118] WebSocket open
 WebSocket connected, signaling ready
 Enter a remote ID to send an offer:
-Okt4
+UDL4
-Offering to Okt4
+Offering to UDL4
-Creating DataChannel with label "benchmark"
+Creating DataChannel with label "DC-1"
-2021-03-25 14:34:37.948 INFO  [22325] [rtc::impl::IceTransport::IceTransport@106] Using STUN server "stun.l.google.com:19302"
+2021-04-10 19:54:53.381 INFO  [16625] [rtc::impl::IceTransport::IceTransport@106] Using STUN server "stun.l.google.com:19302"
-2021-03-25 14:34:37.949 INFO  [22325] [rtc::impl::PeerConnection::changeSignalingState@992] Changed signaling state to new
+2021-04-10 19:54:53.382 INFO  [16625] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to new
-2021-03-25 14:34:37.949 INFO  [22325] [rtc::impl::PeerConnection::changeGatheringState@981] Changed gathering state to in-progress
+2021-04-10 19:54:53.382 INFO  [16625] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to in-progress
 Gathering State: in-progress
-2021-03-25 14:34:37.950 INFO  [22325] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to gathering
+2021-04-10 19:54:53.383 INFO  [16625] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to gathering
-2021-03-25 14:34:37.951 WARN  [22325] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:239: Local description already has the maximum number of host candidates
 Benchmark will run for 300 seconds
-2021-03-25 14:34:37.952 INFO  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connecting
+2021-04-10 19:54:53.384 INFO  [16646] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connecting
-2021-03-25 14:34:37.952 INFO  [22334] [rtc::impl::PeerConnection::changeState@964] Changed state to connecting
+2021-04-10 19:54:53.384 INFO  [16646] [rtc::impl::PeerConnection::changeState@1016] Changed state to connecting
 State: connecting
-2021-03-25 14:34:37.969 INFO  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:610: Using STUN server stun.l.google.com:19302
+2021-04-10 19:54:53.475 INFO  [16646] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:610: Using STUN server stun.l.google.com:19302
-2021-03-25 14:34:37.969 INFO  [22332] [rtc::impl::PeerConnection::changeSignalingState@992] Changed signaling state to connecting
+2021-04-10 19:54:53.475 INFO  [16632] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to connecting
-2021-03-25 14:34:37.970 INFO  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connected
+2021-04-10 19:54:53.527 INFO  [16646] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1328: STUN server binding successful
-2021-03-25 14:34:37.971 INFO  [22337] [rtc::impl::DtlsTransport::runRecvLoop@503] DTLS handshake finished
+2021-04-10 19:54:53.575 INFO  [16646] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connected
-2021-03-25 14:34:37.977 INFO  [22327] [rtc::impl::SctpTransport::processNotification@708] SCTP connected
+2021-04-10 19:54:53.625 INFO  [16646] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to completed
-2021-03-25 14:34:37.977 INFO  [22327] [rtc::impl::PeerConnection::changeState@964] Changed state to connected
+#1
+      DC-1 Received: 0 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 0 KB/s   Sent: 0 KB/s
+2021-04-10 19:54:54.481 INFO  [16653] [rtc::impl::DtlsTransport::runRecvLoop@503] DTLS handshake finished
+2021-04-10 19:54:54.491 INFO  [16627] [rtc::impl::SctpTransport::processNotification@713] SCTP connected
+2021-04-10 19:54:54.491 INFO  [16627] [rtc::impl::PeerConnection::changeState@1016] Changed state to connected
 State: connected
-DataChannel from Okt4 open
+DataChannel from UDL4 open
-2021-03-25 14:34:38.019 WARN  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:426: Send failed, errno=101
+#2
-2021-03-25 14:34:38.019 WARN  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1483: STUN message send failed, errno=101
+      DC-1 Received: 84326 KB/s   Sent: 0 KB/s   BufferSize: 0
-2021-03-25 14:34:38.019 INFO  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:843: STUN binding failed
+      TOTL Received: 84326 KB/s   Sent: 0 KB/s
-2021-03-25 14:34:38.022 INFO  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1302: STUN server binding successful
+#3
-2021-03-25 14:34:38.022 INFO  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:2148: Candidate gathering done
+      DC-1 Received: 99387 KB/s   Sent: 0 KB/s   BufferSize: 0
-2021-03-25 14:34:38.022 INFO  [22334] [rtc::impl::PeerConnection::changeGatheringState@981] Changed gathering state to complete
+      TOTL Received: 99387 KB/s   Sent: 0 KB/s
+2021-04-10 19:54:57.025 INFO  [16646] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:843: STUN server binding failed (timeout)
+2021-04-10 19:54:57.025 INFO  [16646] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:2206: Candidate gathering done
+2021-04-10 19:54:57.025 INFO  [16646] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to complete
 Gathering State: complete
-2021-03-25 14:34:38.069 INFO  [22334] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to completed
+#4
-#1 Received: 0 KB/s   Sent: 92223 KB/s   BufferSize: 65535
+      DC-1 Received: 94871 KB/s   Sent: 0 KB/s   BufferSize: 0
-#2 Received: 0 KB/s   Sent: 89291 KB/s   BufferSize: 65535
+      TOTL Received: 94871 KB/s   Sent: 0 KB/s
-#3 Received: 0 KB/s   Sent: 95087 KB/s   BufferSize: 65535
+#5
-#4 Received: 0 KB/s   Sent: 101050 KB/s   BufferSize: 65535
+      DC-1 Received: 96259 KB/s   Sent: 0 KB/s   BufferSize: 0
-#5 Received: 0 KB/s   Sent: 99778 KB/s   BufferSize: 0
+      TOTL Received: 96259 KB/s   Sent: 0 KB/s
-Stats# Received Total: 0 MB   Sent Total: 480 MB   RTT: 8 ms
+Stats# Received Total: 377 MB   Sent Total: 0 MB   RTT: 2 ms
 
-#6 Received: 0 KB/s   Sent: 100366 KB/s   BufferSize: 65535
+#6
-#7 Received: 0 KB/s   Sent: 101201 KB/s   BufferSize: 65535
+      DC-1 Received: 92873 KB/s   Sent: 0 KB/s   BufferSize: 0
-#8 Received: 0 KB/s   Sent: 100892 KB/s   BufferSize: 65535
+      TOTL Received: 92873 KB/s   Sent: 0 KB/s
-#9 Received: 0 KB/s   Sent: 101288 KB/s   BufferSize: 65535
+#7
-#10 Received: 0 KB/s   Sent: 95676 KB/s   BufferSize: 65535
+      DC-1 Received: 87724 KB/s   Sent: 0 KB/s   BufferSize: 0
-Stats# Received Total: 0 MB   Sent Total: 982 MB   RTT: 8 ms
+      TOTL Received: 87724 KB/s   Sent: 0 KB/s
+#8
+      DC-1 Received: 95123 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 95123 KB/s   Sent: 0 KB/s
+#9
+      DC-1 Received: 100022 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 100022 KB/s   Sent: 0 KB/s
+#10
+      DC-1 Received: 98124 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 98124 KB/s   Sent: 0 KB/s
+Stats# Received Total: 853 MB   Sent Total: 0 MB   RTT: 2 ms
 
-#11 Received: 0 KB/s   Sent: 96254 KB/s   BufferSize: 65535
+#11
-#12 Received: 0 KB/s   Sent: 105473 KB/s   BufferSize: 65535
+      DC-1 Received: 103628 KB/s   Sent: 0 KB/s   BufferSize: 0
-#13 Received: 0 KB/s   Sent: 95549 KB/s   BufferSize: 65535
+      TOTL Received: 103628 KB/s   Sent: 0 KB/s
-#14 Received: 0 KB/s   Sent: 100366 KB/s   BufferSize: 65535
+#12
-#15 Received: 0 KB/s   Sent: 101336 KB/s   BufferSize: 65535
+      DC-1 Received: 106166 KB/s   Sent: 0 KB/s   BufferSize: 0
-Stats# Received Total: 0 MB   Sent Total: 1484 MB   RTT: 8 ms
+      TOTL Received: 106166 KB/s   Sent: 0 KB/s
+#13
+      DC-1 Received: 98410 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 98410 KB/s   Sent: 0 KB/s
+#14
+      DC-1 Received: 99854 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 99854 KB/s   Sent: 0 KB/s
+#15
+      DC-1 Received: 98487 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 98487 KB/s   Sent: 0 KB/s
+Stats# Received Total: 1362 MB   Sent Total: 0 MB   RTT: 2 ms
 ```
 
 ### Constant Throughput Set 8000 byte, for 300 seconds, send buffer 10000 byte
@@ -159,61 +215,249 @@ Stats# Received Total: 0 MB   Sent Total: 1484 MB   RTT: 8 ms
 Example Output (Offering Peer's Output);
 ```bash
 Stun server is stun:stun.l.google.com:19302
-The local ID is: 1w9O
+The local ID is: 5zkC
-Url is ws://localhost:8000/1w9O
+Url is ws://localhost:8000/5zkC
 Waiting for signaling to be connected...
-2021-03-25 14:29:38.697 INFO  [21844] [rtc::impl::TcpTransport::connect@159] Connected to localhost:8000
+2021-04-10 19:52:49.788 INFO  [16530] [rtc::impl::TcpTransport::connect@163] Connected to localhost:8000
-2021-03-25 14:29:38.697 INFO  [21844] [rtc::impl::TcpTransport::runLoop@327] TCP connected
+2021-04-10 19:52:49.788 INFO  [16530] [rtc::impl::TcpTransport::runLoop@331] TCP connected
-2021-03-25 14:29:38.698 INFO  [21844] [rtc::impl::WsTransport::incoming@118] WebSocket open
+2021-04-10 19:52:49.789 INFO  [16530] [rtc::impl::WsTransport::incoming@118] WebSocket open
 WebSocket connected, signaling ready
 Enter a remote ID to send an offer:
-zi4B
+WawD
-Offering to zi4B
+Offering to WawD
-Creating DataChannel with label "benchmark"
+Creating DataChannel with label "DC-1"
-2021-03-25 14:29:48.729 INFO  [21837] [rtc::impl::IceTransport::IceTransport@106] Using STUN server "stun.l.google.com:19302"
+2021-04-10 19:52:57.720 INFO  [16523] [rtc::impl::IceTransport::IceTransport@106] Using STUN server "stun.l.google.com:19302"
-2021-03-25 14:29:48.729 INFO  [21837] [rtc::impl::PeerConnection::changeSignalingState@992] Changed signaling state to new
+2021-04-10 19:52:57.721 INFO  [16523] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to new
-2021-03-25 14:29:48.729 INFO  [21837] [rtc::impl::PeerConnection::changeGatheringState@981] Changed gathering state to in-progress
+2021-04-10 19:52:57.721 INFO  [16523] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to in-progress
 Gathering State: in-progress
-2021-03-25 14:29:48.729 INFO  [21837] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to gathering
+2021-04-10 19:52:57.722 INFO  [16523] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to gathering
-2021-03-25 14:29:48.730 WARN  [21837] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:239: Local description already has the maximum number of host candidates
 Benchmark will run for 300 seconds
-2021-03-25 14:29:48.730 INFO  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connecting
+2021-04-10 19:52:57.722 INFO  [16533] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connecting
-2021-03-25 14:29:48.731 INFO  [21866] [rtc::impl::PeerConnection::changeState@964] Changed state to connecting
+2021-04-10 19:52:57.722 INFO  [16533] [rtc::impl::PeerConnection::changeState@1016] Changed state to connecting
 State: connecting
-2021-03-25 14:29:48.732 INFO  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:610: Using STUN server stun.l.google.com:19302
+2021-04-10 19:52:57.725 INFO  [16533] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:610: Using STUN server stun.l.google.com:19302
-2021-03-25 14:29:48.732 INFO  [21844] [rtc::impl::PeerConnection::changeSignalingState@992] Changed signaling state to connecting
+2021-04-10 19:52:57.727 INFO  [16530] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to connecting
-2021-03-25 14:29:48.782 WARN  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:426: Send failed, errno=101
+2021-04-10 19:52:57.826 INFO  [16533] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connected
-2021-03-25 14:29:48.782 WARN  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1483: STUN message send failed, errno=101
+2021-04-10 19:52:57.828 INFO  [16533] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to completed
-2021-03-25 14:29:48.782 INFO  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:843: STUN binding failed
+2021-04-10 19:52:57.829 INFO  [16533] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1328: STUN server binding successful
-2021-03-25 14:29:48.787 INFO  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1302: STUN server binding successful
+2021-04-10 19:52:57.884 INFO  [16535] [rtc::impl::DtlsTransport::runRecvLoop@503] DTLS handshake finished
-2021-03-25 14:29:48.787 INFO  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:2148: Candidate gathering done
+2021-04-10 19:52:57.907 INFO  [16526] [rtc::impl::SctpTransport::processNotification@713] SCTP connected
-2021-03-25 14:29:48.787 INFO  [21866] [rtc::impl::PeerConnection::changeGatheringState@981] Changed gathering state to complete
+2021-04-10 19:52:57.907 INFO  [16526] [rtc::impl::PeerConnection::changeState@1016] Changed state to connected
-Gathering State: complete
-2021-03-25 14:29:48.832 INFO  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connected
-2021-03-25 14:29:48.882 INFO  [21866] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to completed
-2021-03-25 14:29:49.735 INFO  [21869] [rtc::impl::DtlsTransport::runRecvLoop@503] DTLS handshake finished
-2021-03-25 14:29:49.742 INFO  [21841] [rtc::impl::SctpTransport::processNotification@708] SCTP connected
-2021-03-25 14:29:49.742 INFO  [21841] [rtc::impl::PeerConnection::changeState@964] Changed state to connected
 State: connected
-DataChannel from zi4B open
+DataChannel from WawD open
-#1 Received: 0 KB/s   Sent: 78 KB/s   BufferSize: 0
+#1
-#2 Received: 8002 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-1 Received: 6515 KB/s   Sent: 6577 KB/s   BufferSize: 0
-#3 Received: 8002 KB/s   Sent: 7998 KB/s   BufferSize: 0
+      TOTL Received: 6515 KB/s   Sent: 6577 KB/s
-#4 Received: 7995 KB/s   Sent: 8000 KB/s   BufferSize: 0
+#2
-#5 Received: 8000 KB/s   Sent: 8001 KB/s   BufferSize: 0
+      DC-1 Received: 7998 KB/s   Sent: 7999 KB/s   BufferSize: 0
-Stats# Received Total: 33 MB   Sent Total: 33 MB   RTT: 0 ms
+      TOTL Received: 7998 KB/s   Sent: 7999 KB/s
+#3
+      DC-1 Received: 7933 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      TOTL Received: 7933 KB/s   Sent: 7999 KB/s
+2021-04-10 19:53:01.275 INFO  [16533] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:843: STUN server binding failed (timeout)
+2021-04-10 19:53:01.275 INFO  [16533] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:2206: Candidate gathering done
+2021-04-10 19:53:01.275 INFO  [16533] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to complete
+Gathering State: complete
+#4
+      DC-1 Received: 8070 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      TOTL Received: 8070 KB/s   Sent: 8000 KB/s
+#5
+      DC-1 Received: 7984 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      TOTL Received: 7984 KB/s   Sent: 8000 KB/s
+Stats# Received Total: 39 MB   Sent Total: 39 MB   RTT: 0 ms
 
-#6 Received: 8001 KB/s   Sent: 7999 KB/s   BufferSize: 0
+#6
-#7 Received: 7997 KB/s   Sent: 7998 KB/s   BufferSize: 0
+      DC-1 Received: 8004 KB/s   Sent: 7998 KB/s   BufferSize: 0
-#8 Received: 8001 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      TOTL Received: 8004 KB/s   Sent: 7998 KB/s
-#9 Received: 7998 KB/s   Sent: 8001 KB/s   BufferSize: 0
+#7
-#10 Received: 8003 KB/s   Sent: 7998 KB/s   BufferSize: 0
+      DC-1 Received: 7997 KB/s   Sent: 8000 KB/s   BufferSize: 0
-Stats# Received Total: 74 MB   Sent Total: 74 MB   RTT: 0 ms
+      TOTL Received: 7997 KB/s   Sent: 8000 KB/s
+#8
+      DC-1 Received: 8008 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      TOTL Received: 8008 KB/s   Sent: 8000 KB/s
+#9
+      DC-1 Received: 8007 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      TOTL Received: 8007 KB/s   Sent: 8000 KB/s
+#10
+      DC-1 Received: 7999 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      TOTL Received: 7999 KB/s   Sent: 7999 KB/s
+Stats# Received Total: 81 MB   Sent Total: 81 MB   RTT: 0 ms
 
-#11 Received: 7990 KB/s   Sent: 7998 KB/s   BufferSize: 0
+#11
-#12 Received: 7999 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      DC-1 Received: 7997 KB/s   Sent: 8001 KB/s   BufferSize: 0
-#13 Received: 8001 KB/s   Sent: 8002 KB/s   BufferSize: 0
+      TOTL Received: 7997 KB/s   Sent: 8001 KB/s
-#14 Received: 7998 KB/s   Sent: 7999 KB/s   BufferSize: 0
+#12
-#15 Received: 8000 KB/s   Sent: 7998 KB/s   BufferSize: 0
+      DC-1 Received: 7981 KB/s   Sent: 7997 KB/s   BufferSize: 0
-Stats# Received Total: 115 MB   Sent Total: 115 MB   RTT: 0 ms
+      TOTL Received: 7981 KB/s   Sent: 7997 KB/s
+#13
+      DC-1 Received: 8024 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      TOTL Received: 8024 KB/s   Sent: 8000 KB/s
+#14
+      DC-1 Received: 7990 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      TOTL Received: 7990 KB/s   Sent: 7999 KB/s
+#15
+      DC-1 Received: 8001 KB/s   Sent: 8002 KB/s   BufferSize: 0
+      TOTL Received: 8001 KB/s   Sent: 8002 KB/s
+Stats# Received Total: 122 MB   Sent Total: 122 MB   RTT: 0 ms
+```
+
+### Constant Throughput Set 8000 byte, for 300 seconds, send buffer 10000 byte, 5 Data Channel
+
+> `./client-benchmark -p -d 300 -r 8000 -b 10000 -c 5` 
+
+Example Output (Offering Peer's Output);
+```bash
+Stun server is stun:stun.l.google.com:19302
+The local ID is: QZ46
+Url is ws://localhost:8000/QZ46
+Waiting for signaling to be connected...
+2021-04-10 19:57:28.562 INFO  [17117] [rtc::impl::TcpTransport::connect@163] Connected to localhost:8000
+2021-04-10 19:57:28.562 INFO  [17117] [rtc::impl::TcpTransport::runLoop@331] TCP connected
+2021-04-10 19:57:28.563 INFO  [17117] [rtc::impl::WsTransport::incoming@118] WebSocket open
+WebSocket connected, signaling ready
+Enter a remote ID to send an offer:
+lTZA
+Offering to lTZA
+Creating DataChannel with label "DC-1"
+2021-04-10 19:57:37.371 INFO  [17110] [rtc::impl::IceTransport::IceTransport@106] Using STUN server "stun.l.google.com:19302"
+2021-04-10 19:57:37.372 INFO  [17110] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to new
+2021-04-10 19:57:37.373 INFO  [17110] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to in-progress
+2021-04-10 19:57:37.373 INFO  [17110] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to gathering
+Gathering State: in-progress
+Creating DataChannel with label "DC-2"
+2021-04-10 19:57:37.373 INFO  [17119] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connecting
+Creating DataChannel with label "DC-3"
+2021-04-10 19:57:37.374 INFO  [17119] [rtc::impl::PeerConnection::changeState@1016] Changed state to connecting
+Creating DataChannel with label "DC-4"
+Creating DataChannel with label "DC-5"
+State: Benchmark will run for connecting300 seconds
+
+2021-04-10 19:57:37.376 INFO  [17119] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:610: Using STUN server stun.l.google.com:19302
+2021-04-10 19:57:37.378 INFO  [17117] [rtc::impl::PeerConnection::changeSignalingState@1044] Changed signaling state to connecting
+2021-04-10 19:57:37.423 INFO  [17119] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:1328: STUN server binding successful
+2021-04-10 19:57:37.476 INFO  [17119] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to connected
+2021-04-10 19:57:37.478 INFO  [17119] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:787: Changing state to completed
+2021-04-10 19:57:38.383 INFO  [17122] [rtc::impl::DtlsTransport::runRecvLoop@503] DTLS handshake finished
+2021-04-10 19:57:38.392 INFO  [17113] [rtc::impl::SctpTransport::processNotification@713] SCTP connected
+2021-04-10 19:57:38.392 INFO  [17113] [rtc::impl::PeerConnection::changeState@1016] Changed state to connected
+State: connected
+DataChannel from lTZA open
+DataChannel from lTZA open
+#DataChannel from lTZA open
+DataChannel from lTZA open
+1
+      DC-5 Received: 0 KB/s   Sent: 79 KB/s   BufferSize: 0
+      DC-3 Received: 0 KB/s   Sent: 0 KB/s   BufferSize: 0
+      DC-4 Received: 0 KB/s   Sent: 79 KB/s   BufferSize: 0
+      DC-2 Received: 0 KB/s   Sent: 0 KB/s   BufferSize: 0
+      DC-1 Received: 0 KB/s   Sent: 0 KB/s   BufferSize: 0
+      TOTL Received: 0 KB/s   Sent: 158 KB/s
+DataChannel from lTZA open
+#2
+      DC-5 Received: 7960 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      DC-3 Received: 7804 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      DC-4 Received: 7883 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      DC-2 Received: 7882 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      DC-1 Received: 7804 KB/s   Sent: 8000 KB/s   BufferSize: 0
+      TOTL Received: 39333 KB/s   Sent: 40000 KB/s
+#3
+      DC-5 Received: 7966 KB/s   Sent: 7996 KB/s   BufferSize: 81504
+      DC-3 Received: 8047 KB/s   Sent: 7996 KB/s   BufferSize: 81504
+      DC-4 Received: 7958 KB/s   Sent: 7996 KB/s   BufferSize: 81504
+      DC-2 Received: 7958 KB/s   Sent: 7996 KB/s   BufferSize: 81504
+      DC-1 Received: 8067 KB/s   Sent: 7996 KB/s   BufferSize: 163597
+      TOTL Received: 39996 KB/s   Sent: 39980 KB/s
+2021-04-10 19:57:40.926 INFO  [17119] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:843: STUN server binding failed (timeout)
+2021-04-10 19:57:40.926 INFO  [17119] [rtc::impl::IceTransport::LogCallback@339] juice: agent.c:2206: Candidate gathering done
+2021-04-10 19:57:40.926 INFO  [17119] [rtc::impl::PeerConnection::changeGatheringState@1033] Changed gathering state to complete
+Gathering State: complete
+#4
+      DC-5 Received: 7970 KB/s   Sent: 8002 KB/s   BufferSize: 0
+      DC-3 Received: 7957 KB/s   Sent: 8002 KB/s   BufferSize: 0
+      DC-4 Received: 7910 KB/s   Sent: 8002 KB/s   BufferSize: 0
+      DC-2 Received: 7967 KB/s   Sent: 8002 KB/s   BufferSize: 0
+      DC-1 Received: 7957 KB/s   Sent: 8002 KB/s   BufferSize: 0
+      TOTL Received: 39761 KB/s   Sent: 40010 KB/s
+#5
+      DC-5 Received: 7996 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-3 Received: 8006 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-4 Received: 8078 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-2 Received: 8015 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-1 Received: 7928 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      TOTL Received: 40023 KB/s   Sent: 39995 KB/s
+Stats# Received Total: 165 MB   Sent Total: 166 MB   RTT: 1 ms
+
+#6
+      DC-5 Received: 7968 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-3 Received: 7962 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-4 Received: 7965 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-2 Received: 7970 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-1 Received: 8044 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      TOTL Received: 39909 KB/s   Sent: 39995 KB/s
+#7
+      DC-5 Received: 6658 KB/s   Sent: 8001 KB/s   BufferSize: 82228
+      DC-3 Received: 6584 KB/s   Sent: 8001 KB/s   BufferSize: 163596
+      DC-4 Received: 6572 KB/s   Sent: 8001 KB/s   BufferSize: 163596
+      DC-2 Received: 6571 KB/s   Sent: 8001 KB/s   BufferSize: 163596
+      DC-1 Received: 6492 KB/s   Sent: 8001 KB/s   BufferSize: 163596
+      TOTL Received: 32877 KB/s   Sent: 40005 KB/s
+#8
+      DC-5 Received: 5773 KB/s   Sent: 7997 KB/s   BufferSize: 0
+      DC-3 Received: 6555 KB/s   Sent: 7997 KB/s   BufferSize: 0
+      DC-4 Received: 6164 KB/s   Sent: 7997 KB/s   BufferSize: 0
+      DC-2 Received: 6241 KB/s   Sent: 7997 KB/s   BufferSize: 0
+      DC-1 Received: 5454 KB/s   Sent: 7997 KB/s   BufferSize: 0
+      TOTL Received: 30187 KB/s   Sent: 39985 KB/s
+#9
+      DC-5 Received: 7442 KB/s   Sent: 8002 KB/s   BufferSize: 326921
+      DC-3 Received: 7580 KB/s   Sent: 8002 KB/s   BufferSize: 326921
+      DC-4 Received: 7363 KB/s   Sent: 8002 KB/s   BufferSize: 326921
+      DC-2 Received: 7524 KB/s   Sent: 8002 KB/s   BufferSize: 326921
+      DC-1 Received: 7362 KB/s   Sent: 8002 KB/s   BufferSize: 408769
+      TOTL Received: 37271 KB/s   Sent: 40010 KB/s
+#10
+      DC-5 Received: 6134 KB/s   Sent: 7999 KB/s   BufferSize: 244963
+      DC-3 Received: 8032 KB/s   Sent: 7999 KB/s   BufferSize: 326286
+      DC-4 Received: 5897 KB/s   Sent: 7999 KB/s   BufferSize: 326286
+      DC-2 Received: 5657 KB/s   Sent: 7999 KB/s   BufferSize: 326286
+      DC-1 Received: 5581 KB/s   Sent: 7999 KB/s   BufferSize: 326286
+      TOTL Received: 31301 KB/s   Sent: 39995 KB/s
+Stats# Received Total: 343 MB   Sent Total: 372 MB   RTT: 16 ms
+
+#11
+      DC-5 Received: 6117 KB/s   Sent: 7998 KB/s   BufferSize: 570756
+      DC-3 Received: 6594 KB/s   Sent: 7998 KB/s   BufferSize: 570756
+      DC-4 Received: 6354 KB/s   Sent: 7998 KB/s   BufferSize: 570756
+      DC-2 Received: 6116 KB/s   Sent: 7998 KB/s   BufferSize: 570756
+      DC-1 Received: 5959 KB/s   Sent: 7998 KB/s   BufferSize: 570756
+      TOTL Received: 31140 KB/s   Sent: 39990 KB/s
+#12
+      DC-5 Received: 6840 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-3 Received: 7468 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-4 Received: 7472 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-2 Received: 7473 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      DC-1 Received: 7236 KB/s   Sent: 7999 KB/s   BufferSize: 0
+      TOTL Received: 36489 KB/s   Sent: 39995 KB/s
+#13
+      DC-5 Received: 8105 KB/s   Sent: 7989 KB/s   BufferSize: 0
+      DC-3 Received: 8020 KB/s   Sent: 7989 KB/s   BufferSize: 0
+      DC-4 Received: 8097 KB/s   Sent: 7989 KB/s   BufferSize: 0
+      DC-2 Received: 8106 KB/s   Sent: 7989 KB/s   BufferSize: 0
+      DC-1 Received: 8018 KB/s   Sent: 7989 KB/s   BufferSize: 0
+      TOTL Received: 40346 KB/s   Sent: 39945 KB/s
+#14
+      DC-5 Received: 8042 KB/s   Sent: 8007 KB/s   BufferSize: 0
+      DC-3 Received: 8029 KB/s   Sent: 8007 KB/s   BufferSize: 0
+      DC-4 Received: 8038 KB/s   Sent: 8007 KB/s   BufferSize: 0
+      DC-2 Received: 8035 KB/s   Sent: 8007 KB/s   BufferSize: 0
+      DC-1 Received: 8036 KB/s   Sent: 8007 KB/s   BufferSize: 0
+      TOTL Received: 40180 KB/s   Sent: 40035 KB/s
+#15
+      DC-5 Received: 7981 KB/s   Sent: 8001 KB/s   BufferSize: 0
+      DC-3 Received: 7987 KB/s   Sent: 8001 KB/s   BufferSize: 0
+      DC-4 Received: 7980 KB/s   Sent: 8001 KB/s   BufferSize: 0
+      DC-2 Received: 7974 KB/s   Sent: 8001 KB/s   BufferSize: 0
+      DC-1 Received: 7972 KB/s   Sent: 8001 KB/s   BufferSize: 82497
+      TOTL Received: 39894 KB/s   Sent: 40005 KB/s
+Stats# Received Total: 538 MB   Sent Total: 581 MB   RTT: 3 ms
 ```

examples/client-benchmark/main.cpp

@@ -31,6 +31,7 @@
 #include <atomic>
 #include <chrono>
 #include <future>
+#include <iomanip>
 #include <iostream>
 #include <memory>
 #include <random>
@@ -61,7 +62,8 @@ string randomId(size_t length);
 // Benchmark
 const size_t messageSize = 65535;
 binary messageData(messageSize);
-atomic<size_t> receivedSize = 0, sentSize = 0;
+unordered_map<string, atomic<size_t>> receivedSizeMap;
+unordered_map<string, atomic<size_t>> sentSizeMap;
 bool noSend = false;
 
 // Benchmark - enableThroughputSet params
@@ -69,7 +71,7 @@ bool enableThroughputSet;
 int throughtputSetAsKB;
 int bufferSize;
 const float STEP_COUNT_FOR_1_SEC = 100.0;
-const int stepDurationInMs = 1000 / STEP_COUNT_FOR_1_SEC;
+const int stepDurationInMs = int(1000 / STEP_COUNT_FOR_1_SEC);
 
 int main(int argc, char **argv) try {
 	Cmdline params(argc, argv);
@@ -188,63 +190,67 @@ int main(int argc, char **argv) try {
 	auto pc = createPeerConnection(config, ws, id);
 
 	// We are the offerer, so create a data channel to initiate the process
-	const string label = "benchmark";
+	for (int i = 1; i <= params.dataChannelCount(); i++) {
-	cout << "Creating DataChannel with label \"" << label << "\"" << endl;
+		const string label = "DC-" + std::to_string(i);
-	auto dc = pc->createDataChannel(label);
+		cout << "Creating DataChannel with label \"" << label << "\"" << endl;
+		auto dc = pc->createDataChannel(label);
+		receivedSizeMap.emplace(label, 0);
+		sentSizeMap.emplace(label, 0);
 
-	// Set Buffer Size
+		// Set Buffer Size
-	dc->setBufferedAmountLowThreshold(bufferSize);
+		dc->setBufferedAmountLowThreshold(bufferSize);
 
-	dc->onOpen([id, wdc = make_weak_ptr(dc)]() {
+		dc->onOpen([id, wdc = make_weak_ptr(dc), label]() {
-		cout << "DataChannel from " << id << " open" << endl;
+			cout << "DataChannel from " << id << " open" << endl;
-		if (noSend)
+			if (noSend)
-			return;
+				return;
 
-		if (enableThroughputSet)
+			if (enableThroughputSet)
-			return;
+				return;
 
-		if (auto dcLocked = wdc.lock()) {
+			if (auto dcLocked = wdc.lock()) {
+				try {
+					while (dcLocked->bufferedAmount() <= bufferSize) {
+						dcLocked->send(messageData);
+						sentSizeMap.at(label) += messageData.size();
+					}
+				} catch (const std::exception &e) {
+					std::cout << "Send failed: " << e.what() << std::endl;
+				}
+			}
+		});
+
+		dc->onBufferedAmountLow([wdc = make_weak_ptr(dc), label]() {
+			if (noSend)
+				return;
+
+			if (enableThroughputSet)
+				return;
+
+			auto dcLocked = wdc.lock();
+			if (!dcLocked)
+				return;
+
+			// Continue sending
 			try {
-				while (dcLocked->bufferedAmount() <= bufferSize) {
+				while (dcLocked->isOpen() && dcLocked->bufferedAmount() <= bufferSize) {
 					dcLocked->send(messageData);
-					sentSize += messageData.size();
+					sentSizeMap.at(label) += messageData.size();
 				}
 			} catch (const std::exception &e) {
 				std::cout << "Send failed: " << e.what() << std::endl;
 			}
-		}
+		});
-	});
 
-	dc->onBufferedAmountLow([wdc = make_weak_ptr(dc)]() {
+		dc->onClosed([id]() { cout << "DataChannel from " << id << " closed" << endl; });
-		if (noSend)
-			return;
 
-		if (enableThroughputSet)
+		dc->onMessage([id, wdc = make_weak_ptr(dc), label](variant<binary, string> data) {
-			return;
+			if (holds_alternative<binary>(data))
+				receivedSizeMap.at(label) += get<binary>(data).size();
+		});
 
-		auto dcLocked = wdc.lock();
+		dataChannelMap.emplace(label, dc);
-		if (!dcLocked)
+	}
-			return;
-
-		// Continue sending
-		try {
-			while (dcLocked->isOpen() && dcLocked->bufferedAmount() <= bufferSize) {
-				dcLocked->send(messageData);
-				sentSize += messageData.size();
-			}
-		} catch (const std::exception &e) {
-			std::cout << "Send failed: " << e.what() << std::endl;
-		}
-	});
-
-	dc->onClosed([id]() { cout << "DataChannel from " << id << " closed" << endl; });
-
-	dc->onMessage([id, wdc = make_weak_ptr(dc)](variant<binary, string> data) {
-		if (holds_alternative<binary>(data))
-			receivedSize += get<binary>(data).size();
-	});
-
-	dataChannelMap.emplace(id, dc);
 
 	const int duration = params.durationInSec() > 0 ? params.durationInSec() : INT32_MAX;
 	cout << "Benchmark will run for " << duration << " seconds" << endl;
@@ -271,25 +277,39 @@ int main(int argc, char **argv) try {
 			binary tempMessageData(byteToSendThisLoop);
 			fill(tempMessageData.begin(), tempMessageData.end(), std::byte(0xFF));
 
-			if (dc->isOpen() && dc->bufferedAmount() <= bufferSize * byteToSendOnEveryLoop) {
+			for (const auto &[label, dc] : dataChannelMap) {
-				dc->send(tempMessageData);
+				if (dc->isOpen() && dc->bufferedAmount() <= bufferSize * byteToSendOnEveryLoop) {
-				sentSize += tempMessageData.size();
+					dc->send(tempMessageData);
+					sentSizeMap.at(label) += tempMessageData.size();
+				}
 			}
 		}
 
 		if (printCounter >= STEP_COUNT_FOR_1_SEC) {
-			unsigned long _receivedSize = receivedSize.exchange(0);
-			unsigned long _sentSize = sentSize.exchange(0);
 			const double elapsedTimeInSecs =
 			    std::chrono::duration<double>(steady_clock::now() - printTime).count();
 			printTime = steady_clock::now();
 
-			cout << "#" << i / STEP_COUNT_FOR_1_SEC
+			unsigned long receiveSpeedTotal = 0;
-			     << " Received: " << static_cast<int>(_receivedSize / (elapsedTimeInSecs * 1000))
+			unsigned long sendSpeedTotal = 0;
-			     << " KB/s"
+			cout << "#" << i / STEP_COUNT_FOR_1_SEC << endl;
-			     << "   Sent: " << static_cast<int>(_sentSize / (elapsedTimeInSecs * 1000))
+			for (const auto &[label, dc] : dataChannelMap) {
-			     << " KB/s"
+				unsigned long channelReceiveSpeed = static_cast<int>(
-			     << "   BufferSize: " << dc->bufferedAmount() << endl;
+				    receivedSizeMap[label].exchange(0) / (elapsedTimeInSecs * 1000));
+				unsigned long channelSendSpeed =
+				    static_cast<int>(sentSizeMap[label].exchange(0) / (elapsedTimeInSecs * 1000));
+
+				cout << std::setw(10) << label << " Received: " << channelReceiveSpeed << " KB/s"
+				     << "   Sent: " << channelSendSpeed << " KB/s"
+				     << "   BufferSize: " << dc->bufferedAmount() << endl;
+
+				receiveSpeedTotal += channelReceiveSpeed;
+				sendSpeedTotal += channelSendSpeed;
+			}
+			cout << std::setw(10) << "TOTL"
+			     << " Received: " << receiveSpeedTotal << " KB/s"
+			     << "   Sent: " << sendSpeedTotal << " KB/s" << endl;
+
 			printStatCounter++;
 			printCounter = 0;
 		}
@@ -308,12 +328,16 @@ int main(int argc, char **argv) try {
 
 	dataChannelMap.clear();
 	peerConnectionMap.clear();
+	receivedSizeMap.clear();
+	sentSizeMap.clear();
 	return 0;
 
 } catch (const std::exception &e) {
 	std::cout << "Error: " << e.what() << std::endl;
 	dataChannelMap.clear();
 	peerConnectionMap.clear();
+	receivedSizeMap.clear();
+	sentSizeMap.clear();
 	return -1;
 }
 
@@ -346,13 +370,16 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 	});
 
 	pc->onDataChannel([id](shared_ptr<DataChannel> dc) {
-		cout << "DataChannel from " << id << " received with label \"" << dc->label() << "\""
+		const string label = dc->label();
-		     << endl;
+		cout << "DataChannel from " << id << " received with label \"" << label << "\"" << endl;
 
 		cout << "###########################################" << endl;
 		cout << "### Check other peer's screen for stats ###" << endl;
 		cout << "###########################################" << endl;
 
+		receivedSizeMap.emplace(dc->label(), 0);
+		sentSizeMap.emplace(dc->label(), 0);
+
 		// Set Buffer Size
 		dc->setBufferedAmountLowThreshold(bufferSize);
 
@@ -360,7 +387,7 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 			try {
 				while (dc->bufferedAmount() <= bufferSize) {
 					dc->send(messageData);
-					sentSize += messageData.size();
+					sentSizeMap.at(label) += messageData.size();
 				}
 			} catch (const std::exception &e) {
 				std::cout << "Send failed: " << e.what() << std::endl;
@@ -370,7 +397,7 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 		if (!noSend && enableThroughputSet) {
 			// Create Send Data Thread
 			// Thread will join when data channel destroyed or closed
-			std::thread([wdc = make_weak_ptr(dc)]() {
+			std::thread([wdc = make_weak_ptr(dc), label]() {
 				steady_clock::time_point stepTime = steady_clock::now();
 				// Byte count to send for every loop
 				int byteToSendOnEveryLoop = throughtputSetAsKB * stepDurationInMs;
@@ -399,7 +426,7 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 
 						if (dcLocked->bufferedAmount() <= bufferSize) {
 							dcLocked->send(tempMessageData);
-							sentSize += tempMessageData.size();
+							sentSizeMap.at(label) += tempMessageData.size();
 						}
 					} catch (const std::exception &e) {
 						std::cout << "Send failed: " << e.what() << std::endl;
@@ -409,7 +436,7 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 			}).detach();
 		}
 
-		dc->onBufferedAmountLow([wdc = make_weak_ptr(dc)]() {
+		dc->onBufferedAmountLow([wdc = make_weak_ptr(dc), label]() {
 			if (noSend)
 				return;
 
@@ -424,7 +451,7 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 			try {
 				while (dcLocked->isOpen() && dcLocked->bufferedAmount() <= bufferSize) {
 					dcLocked->send(messageData);
-					sentSize += messageData.size();
+					sentSizeMap.at(label) += messageData.size();
 				}
 			} catch (const std::exception &e) {
 				std::cout << "Send failed: " << e.what() << std::endl;
@@ -433,12 +460,12 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 
 		dc->onClosed([id]() { cout << "DataChannel from " << id << " closed" << endl; });
 
-		dc->onMessage([id, wdc = make_weak_ptr(dc)](variant<binary, string> data) {
+		dc->onMessage([id, wdc = make_weak_ptr(dc), label](variant<binary, string> data) {
 			if (holds_alternative<binary>(data))
-				receivedSize += get<binary>(data).size();
+				receivedSizeMap.at(label) += get<binary>(data).size();
 		});
 
-		dataChannelMap.emplace(id, dc);
+		dataChannelMap.emplace(label, dc);
 	});
 
 	peerConnectionMap.emplace(id, pc);

examples/client-benchmark/parse_cl.cpp

@@ -51,6 +51,7 @@ Cmdline::Cmdline(int argc, char *argv[]) // ISO C++17 not allowed: throw (std::s
 	                                       {"enableThroughputSet", no_argument, NULL, 'p'},
 	                                       {"throughtputSetAsKB", required_argument, NULL, 'r'},
 	                                       {"bufferSize", required_argument, NULL, 'b'},
+										   {"dataChannelCount", required_argument, NULL, 'c'},
 	                                       {"help", no_argument, NULL, 'h'},
 	                                       {NULL, 0, NULL, 0}};
 
@@ -68,9 +69,10 @@ Cmdline::Cmdline(int argc, char *argv[]) // ISO C++17 not allowed: throw (std::s
 	_p = false;
 	_r = 300;
 	_b = 0;
+	_c = 1;
 
 	optind = 0;
-	while ((c = getopt_long(argc, argv, "s:t:w:x:d:r:b:enhvop", long_options, &optind)) != -1) {
+	while ((c = getopt_long(argc, argv, "s:t:w:x:d:r:b:c:enhvop", long_options, &optind)) != -1) {
 		switch (c) {
 		case 'n':
 			_n = true;
@@ -147,6 +149,15 @@ Cmdline::Cmdline(int argc, char *argv[]) // ISO C++17 not allowed: throw (std::s
 			}
 			break;
 
+		case 'c':
+			_c = atoi(optarg);
+			if (_c <= 0) {
+				std::string err;
+				err += "parameter range error: c must be > 0";
+				throw(std::range_error(err));
+			}
+			break;
+
 		case 'h':
 			_h = true;
 			this->usage(EXIT_SUCCESS);
@@ -196,6 +207,8 @@ libdatachannel client implementing WebRTC Data Channels with WebSocket signaling
           Send a constant data per second (KB). See throughtputSetAsKB params.\n\
    [ -r ] [ --throughtputSetAsKB ] (type=INTEGER, range>0...INT_MAX, default=300)\n\
           Send constant data per second (KB).\n\
+   [ -c ] [ --dataChannelCount ] (type=INTEGER, range>0...INT_MAX, default=1)\n\
+          Dat Channel count to create.\n\
    [ -h ] [ --help ] (type=FLAG)\n\
           Display this help and exit.\n";
 	}

examples/client-benchmark/parse_cl.h

@@ -45,6 +45,7 @@ private:
   bool _p;
   int _r;
   int _b;
+  int _c;
 
   /* other stuff to keep track of */
   std::string _program_name;
@@ -69,9 +70,10 @@ public:
   bool h () const { return _h; }
   int durationInSec () const { return _d; }
   bool noSend () const { return _o; }
-  int bufferSize() const { return _b;}
+  int bufferSize() const { return _b; }
   bool enableThroughputSet () const { return _p; }
-  int throughtputSetAsKB() const { return _r;}  
+  int throughtputSetAsKB() const { return _r; }  
+  int dataChannelCount() const { return _c; }
 };
 
 #endif

examples/client/main.cpp

@@ -222,9 +222,14 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 		cout << "DataChannel from " << id << " received with label \"" << dc->label() << "\""
 		     << endl;
 
+		dc->onOpen([wdc = make_weak_ptr(dc)]() {
+			if (auto dc = wdc.lock())
+				dc->send("Hello from " + localId);
+		});
+
 		dc->onClosed([id]() { cout << "DataChannel from " << id << " closed" << endl; });
 
-		dc->onMessage([id, wdc = make_weak_ptr(dc)](variant<binary, string> data) {
+		dc->onMessage([id](variant<binary, string> data) {
 			if (holds_alternative<string>(data))
 				cout << "Message from " << id << " received: " << get<string>(data) << endl;
 			else
@@ -232,8 +237,6 @@ shared_ptr<PeerConnection> createPeerConnection(const Configuration &config,
 				     << " received, size=" << get<binary>(data).size() << endl;
 		});
 
-		dc->send("Hello from " + localId);
-
 		dataChannelMap.emplace(id, dc);
 	});
 
@@ -251,4 +254,3 @@ string randomId(size_t length) {
 	generate(id.begin(), id.end(), [&]() { return characters.at(dist(rng)); });
 	return id;
 }
-

examples/signaling-server-nodejs/package-lock.json

@@ -9,7 +9,7 @@
       "version": "0.1.0",
       "license": "GPL-2.0",
       "dependencies": {
-        "websocket": "^1.0.33"
+        "websocket": "^1.0.34"
       }
     },
     "node_modules/bufferutil": {
@@ -128,9 +128,9 @@
       }
     },
     "node_modules/websocket": {
-      "version": "1.0.33",
+      "version": "1.0.34",
-      "resolved": "https://registry.npmjs.org/websocket/-/websocket-1.0.33.tgz",
+      "resolved": "https://registry.npmjs.org/websocket/-/websocket-1.0.34.tgz",
-      "integrity": "sha512-XwNqM2rN5eh3G2CUQE3OHZj+0xfdH42+OFK6LdC2yqiC0YU8e5UK0nYre220T0IyyN031V/XOvtHvXozvJYFWA==",
+      "integrity": "sha512-PRDso2sGwF6kM75QykIesBijKSVceR6jL2G8NGYyq2XrItNC2P5/qL5XeR056GhA+Ly7JMFvJb9I312mJfmqnQ==",
       "dependencies": {
         "bufferutil": "^4.0.1",
         "debug": "^2.2.0",
@@ -264,9 +264,9 @@
       }
     },
     "websocket": {
-      "version": "1.0.33",
+      "version": "1.0.34",
-      "resolved": "https://registry.npmjs.org/websocket/-/websocket-1.0.33.tgz",
+      "resolved": "https://registry.npmjs.org/websocket/-/websocket-1.0.34.tgz",
-      "integrity": "sha512-XwNqM2rN5eh3G2CUQE3OHZj+0xfdH42+OFK6LdC2yqiC0YU8e5UK0nYre220T0IyyN031V/XOvtHvXozvJYFWA==",
+      "integrity": "sha512-PRDso2sGwF6kM75QykIesBijKSVceR6jL2G8NGYyq2XrItNC2P5/qL5XeR056GhA+Ly7JMFvJb9I312mJfmqnQ==",
       "requires": {
         "bufferutil": "^4.0.1",
         "debug": "^2.2.0",

examples/signaling-server-nodejs/package.json

@@ -2,7 +2,7 @@
   "name": "libdatachannel-signaling-server",
   "version": "0.1.0",
   "description": "Signaling server example for libdatachannel",
-  "main": "server.js",
+  "main": "signaling-server.js",
   "scripts": {
     "start": "node signaling-server.js",
     "test": "echo \"Error: no test specified\" && exit 1"
@@ -18,6 +18,6 @@
   },
   "homepage": "https://github.com/paullouisageneau/libdatachannel#readme",
   "dependencies": {
-    "websocket": "^1.0.33"
+    "websocket": "^1.0.34"
   }
 }

examples/signaling-server-nodejs/signaling-server.js

@@ -17,7 +17,6 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-const fs = require('fs');
 const http = require('http');
 const websocket = require('websocket');
 

examples/streamer/h264fileparser.cpp

@@ -17,9 +17,16 @@
  */
 
 #include "h264fileparser.hpp"
-#include <fstream>
 #include "rtc/rtc.hpp"
 
+#include <fstream>
+
+#ifdef _WIN32
+#include <winsock2.h>
+#else
+#include <arpa/inet.h>
+#endif
+
 using namespace std;
 
 H264FileParser::H264FileParser(string directory, uint32_t fps, bool loop): FileParser(directory, ".h264", fps, loop) { }

examples/streamer/helpers.cpp

@@ -19,10 +19,10 @@
 #include "helpers.hpp"
 #include <ctime>
 
-#if _WIN32
+#ifdef _WIN32
 // taken from https://stackoverflow.com/questions/10905892/equivalent-of-gettimeday-for-windows
-
+#include <windows.h>
-#include <Windows.h>
+#include <winsock2.h> // for struct timeval
 
 struct timezone {
     int tz_minuteswest;
@@ -55,6 +55,8 @@ int gettimeofday(struct timeval *tv, struct timezone *tz)
     }
     return 0;
 }
+#else
+#include <sys/time.h>
 #endif
 
 using namespace std;

examples/streamer/main.cpp

@@ -218,11 +218,11 @@ shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const
     video.addSSRC(ssrc, cname, msid, cname);
     auto track = pc->addTrack(video);
     // create RTP configuration
-    auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate));
+    auto rtpConfig = make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate);
     // create packetizer
-    auto packetizer = shared_ptr<H264RtpPacketizer>(new H264RtpPacketizer(H264RtpPacketizer::Separator::Length, rtpConfig));
+    auto packetizer = make_shared<H264RtpPacketizer>(H264RtpPacketizer::Separator::Length, rtpConfig);
     // create H264 handler
-    shared_ptr<H264PacketizationHandler> h264Handler(new H264PacketizationHandler(packetizer));
+    auto h264Handler = make_shared<H264PacketizationHandler>(packetizer);
     // add RTCP SR handler
     auto srReporter = make_shared<RtcpSrReporter>(rtpConfig);
     h264Handler->addToChain(srReporter);
@@ -242,7 +242,7 @@ shared_ptr<ClientTrackData> addAudio(const shared_ptr<PeerConnection> pc, const
     audio.addSSRC(ssrc, cname, msid, cname);
     auto track = pc->addTrack(audio);
     // create RTP configuration
-    auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate));
+    auto rtpConfig = make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate);
     // create packetizer
     auto packetizer = make_shared<OpusRtpPacketizer>(rtpConfig);
     // create opus handler
@@ -265,7 +265,7 @@ shared_ptr<Client> createPeerConnection(const Configuration &config,
                                                 weak_ptr<WebSocket> wws,
                                                 string id) {
     auto pc = make_shared<PeerConnection>(config);
-    shared_ptr<Client> client(new Client(pc));
+    auto client = make_shared<Client>(pc);
 
     pc->onStateChange([id](PeerConnection::State state) {
         cout << "State: " << state << endl;

examples/streamer/stream.cpp

@@ -19,7 +19,7 @@
 #include "stream.hpp"
 #include "helpers.hpp"
 
-#if _WIN32
+#ifdef _WIN32
 // taken from https://stackoverflow.com/questions/5801813/c-usleep-is-obsolete-workarounds-for-windows-mingw
 #include <windows.h>
 
@@ -35,6 +35,8 @@ void usleep(__int64 usec)
     WaitForSingleObject(timer, INFINITE);
     CloseHandle(timer);
 }
+#else
+#include <unistd.h>
 #endif
 
 void StreamSource::stop() {

examples/web/script.js

@@ -118,7 +118,7 @@ function createPeerConnection(ws, id) {
   pc.onconnectionstatechange = () => console.log(`Connection state: ${pc.connectionState}`);
   pc.onicegatheringstatechange = () => console.log(`Gathering state: ${pc.iceGatheringState}`);
   pc.onicecandidate = (e) => {
-    if (e.candidate) {
+    if (e.candidate && e.candidate.candidate) {
       // Send candidate
       sendLocalCandidate(ws, id, e.candidate);
     }

include/rtc/common.hpp

@@ -41,7 +41,6 @@
 
 #include "rtc.h" // for C API defines
 
-#include "log.hpp"
 #include "utils.hpp"
 
 #include <cstddef>
@@ -70,6 +69,7 @@ using binary = std::vector<byte>;
 using binary_ptr = std::shared_ptr<binary>;
 
 using std::size_t;
+using std::ptrdiff_t;
 using std::uint16_t;
 using std::uint32_t;
 using std::uint64_t;

include/rtc/description.hpp

@@ -42,7 +42,14 @@ class RTC_CPP_EXPORT Description {
 public:
 	enum class Type { Unspec, Offer, Answer, Pranswer, Rollback };
 	enum class Role { ActPass, Passive, Active };
-	enum class Direction { SendOnly, RecvOnly, SendRecv, Inactive, Unknown };
+
+	enum class Direction {
+		SendOnly = RTC_DIRECTION_SENDONLY,
+		RecvOnly = RTC_DIRECTION_RECVONLY,
+		SendRecv = RTC_DIRECTION_SENDRECV,
+		Inactive = RTC_DIRECTION_INACTIVE,
+		Unknown = RTC_DIRECTION_UNKNOWN
+	};
 
 	Description(const string &sdp, Type type = Type::Unspec, Role role = Role::ActPass);
 	Description(const string &sdp, string typeString);

include/rtc/global.hpp

@@ -1,5 +1,5 @@
 /**
- * Copyright (c) 2019 Paul-Louis Ageneau
+ * Copyright (c) 2020-2021 Paul-Louis Ageneau
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
@@ -16,27 +16,14 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef RTC_LOG_H
+#ifndef RTC_GLOBAL_H
-#define RTC_LOG_H
+#define RTC_GLOBAL_H
-
-// Disable warnings before including plog
-#if defined(__GNUC__) || defined(__clang__)
-#pragma GCC diagnostic push
-#pragma GCC diagnostic ignored "-Wall"
-#elif defined(_MSC_VER)
-#pragma warning(push, 0)
-#endif
-
-#include "plog/Log.h"
-
-#if defined(__GNUC__) || defined(__clang__)
-#pragma GCC diagnostic pop
-#elif defined(_MSC_VER)
-#pragma warning(pop)
-#endif
 
 #include "common.hpp"
 
+#include <chrono>
+#include <iostream>
+
 namespace rtc {
 
 enum class LogLevel { // Don't change, it must match plog severity
@@ -49,8 +36,31 @@ enum class LogLevel { // Don't change, it must match plog severity
 	Verbose = 6
 };
 
-RTC_CPP_EXPORT void InitLogger(LogLevel level);
+typedef std::function<void(LogLevel level, string message)> LogCallback;
+
+RTC_CPP_EXPORT void InitLogger(LogLevel level, LogCallback callback = nullptr);
+#ifdef PLOG
 RTC_CPP_EXPORT void InitLogger(plog::Severity severity, plog::IAppender *appender = nullptr);
+#endif
+
+RTC_CPP_EXPORT void Preload();
+RTC_CPP_EXPORT void Cleanup();
+
+struct SctpSettings {
+	// For the following settings, not set means optimized default
+	optional<size_t> recvBufferSize;                // in bytes
+	optional<size_t> sendBufferSize;                // in bytes
+	optional<size_t> maxChunksOnQueue;              // in chunks
+	optional<size_t> initialCongestionWindow;       // in MTUs
+	optional<size_t> maxBurst;                      // in MTUs
+	optional<unsigned int> congestionControlModule; // 0: RFC2581, 1: HSTCP, 2: H-TCP, 3: RTCC
+	optional<std::chrono::milliseconds> delayedSackTime;
+};
+
+RTC_CPP_EXPORT void SetSctpSettings(SctpSettings s);
+
 } // namespace rtc
 
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, rtc::LogLevel level);
+
 #endif

include/rtc/h264packetizationhandler.hpp

@@ -1,23 +1,23 @@
-/*
+/**
- * libdatachannel client example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef H264_PACKETIZATION_HANDLER_H
+#ifndef RTC_H264_PACKETIZATION_HANDLER_H
-#define H264_PACKETIZATION_HANDLER_H
+#define RTC_H264_PACKETIZATION_HANDLER_H
 
 #if RTC_ENABLE_MEDIA
 
@@ -28,7 +28,7 @@
 namespace rtc {
 
 /// Handler for H264 packetization
-class RTC_CPP_EXPORT H264PacketizationHandler : public MediaChainableHandler {
+class RTC_CPP_EXPORT H264PacketizationHandler final : public MediaChainableHandler {
 public:
 	/// Construct handler for H264 packetization.
 	/// @param packetizer RTP packetizer for h264
@@ -39,4 +39,4 @@ public:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* H264_PACKETIZATION_HANDLER_H */
+#endif /* RTC_H264_PACKETIZATION_HANDLER_H */

include/rtc/h264rtppacketizer.hpp

@@ -1,23 +1,23 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef H264_RTP_PACKETIZER_H
+#ifndef RTC_H264_RTP_PACKETIZER_H
-#define H264_RTP_PACKETIZER_H
+#define RTC_H264_RTP_PACKETIZER_H
 
 #if RTC_ENABLE_MEDIA
 
@@ -28,13 +28,13 @@
 namespace rtc {
 
 /// RTP packetization of h264 payload
-class RTC_CPP_EXPORT H264RtpPacketizer : public RtpPacketizer, public MediaHandlerRootElement {
+class RTC_CPP_EXPORT H264RtpPacketizer final : public RtpPacketizer, public MediaHandlerRootElement {
 	shared_ptr<NalUnits> splitMessage(binary_ptr message);
 	const uint16_t maximumFragmentSize;
 
 public:
 	/// Default clock rate for H264 in RTP
-	static const auto defaultClockRate = 90 * 1000;
+	inline static const uint32_t defaultClockRate = 90 * 1000;
 
 	/// Nalunit separator
 	enum class Separator {
@@ -64,4 +64,4 @@ private:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* H264_RTP_PACKETIZER_H */
+#endif /* RTC_H264_RTP_PACKETIZER_H */

include/rtc/mediachainablehandler.hpp

@@ -1,18 +1,19 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_MEDIA_CHAINABLE_HANDLER_H

include/rtc/mediahandlerelement.hpp

@@ -1,18 +1,19 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_MEDIA_HANDLER_ELEMENT_H

include/rtc/mediahandlerrootelement.hpp

@@ -1,22 +1,23 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef RTCP_MEDIA_HANDLER_ROOT_ELEMENT_H
+#ifndef RTC_MEDIA_HANDLER_ROOT_ELEMENT_H
-#define RTCP_MEDIA_HANDLER_ROOT_ELEMENT_H
+#define RTC_MEDIA_HANDLER_ROOT_ELEMENT_H
 
 #if RTC_ENABLE_MEDIA
 
@@ -42,4 +43,4 @@ public:
 
 #endif // RTC_ENABLE_MEDIA
 
-#endif // RTCP_MEDIA_HANDLER_ROOT_ELEMENT_H
+#endif // RTC_MEDIA_HANDLER_ROOT_ELEMENT_H

include/rtc/nalunit.hpp

@@ -1,34 +1,38 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef NAL_UNIT_H
+#ifndef RTC_NAL_UNIT_H
-#define NAL_UNIT_H
+#define RTC_NAL_UNIT_H
 
 #if RTC_ENABLE_MEDIA
 
 #include "common.hpp"
 
+#include <cassert>
+
 namespace rtc {
 
 #pragma pack(push, 1)
 
 /// Nalu header
 struct RTC_CPP_EXPORT NalUnitHeader {
+	uint8_t _first = 0;
+
 	bool forbiddenBit() { return _first >> 7; }
 	uint8_t nri() { return _first >> 5 & 0x03; }
 	uint8_t unitType() { return _first & 0x1F; }
@@ -36,13 +40,12 @@ struct RTC_CPP_EXPORT NalUnitHeader {
 	void setForbiddenBit(bool isSet) { _first = (_first & 0x7F) | (isSet << 7); }
 	void setNRI(uint8_t nri) { _first = (_first & 0x9F) | ((nri & 0x03) << 5); }
 	void setUnitType(uint8_t type) { _first = (_first & 0xE0) | (type & 0x1F); }
-
-private:
-	uint8_t _first = 0;
 };
 
 /// Nalu fragment header
 struct RTC_CPP_EXPORT NalUnitFragmentHeader {
+	uint8_t _first = 0;
+
 	bool isStart() { return _first >> 7; }
 	bool reservedBit6() { return (_first >> 6) & 0x01; }
 	bool isEnd() { return (_first >> 5) & 0x01; }
@@ -52,9 +55,6 @@ struct RTC_CPP_EXPORT NalUnitFragmentHeader {
 	void setEnd(bool isSet) { _first = (_first & 0xDF) | (isSet << 6); }
 	void setReservedBit6(bool isSet) { _first = (_first & 0xBF) | (isSet << 5); }
 	void setUnitType(uint8_t type) { _first = (_first & 0xE0) | (type & 0x1F); }
-
-private:
-	uint8_t _first = 0;
 };
 
 #pragma pack(pop)
@@ -152,4 +152,4 @@ public:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* NAL_UNIT_H */
+#endif /* RTC_NAL_UNIT_H */

include/rtc/opuspacketizationhandler.hpp

@@ -1,19 +1,19 @@
-/*
+/**
- * libdatachannel client example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_OPUS_PACKETIZATION_HANDLER_H
@@ -27,7 +27,7 @@
 namespace rtc {
 
 /// Handler for opus packetization
-class RTC_CPP_EXPORT OpusPacketizationHandler : public MediaChainableHandler {
+class RTC_CPP_EXPORT OpusPacketizationHandler final : public MediaChainableHandler {
 
 public:
 	/// Construct handler for opus packetization.

include/rtc/opusrtppacketizer.hpp

@@ -1,19 +1,19 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_OPUS_RTP_PACKETIZER_H
@@ -27,10 +27,10 @@
 namespace rtc {
 
 /// RTP packetizer for opus
-class RTC_CPP_EXPORT OpusRtpPacketizer : public RtpPacketizer, public MediaHandlerRootElement {
+class RTC_CPP_EXPORT OpusRtpPacketizer final : public RtpPacketizer, public MediaHandlerRootElement {
 public:
 	/// default clock rate used in opus RTP communication
-	static const uint32_t defaultClockRate = 48 * 1000;
+	inline static const uint32_t defaultClockRate = 48 * 1000;
 
 	/// Constructs opus packetizer with given RTP configuration.
 	/// @note RTP configuration is used in packetization process which may change some configuration

include/rtc/peerconnection.hpp

@@ -24,7 +24,6 @@
 #include "datachannel.hpp"
 #include "description.hpp"
 #include "common.hpp"
-#include "init.hpp"
 #include "message.hpp"
 #include "reliability.hpp"
 #include "track.hpp"
@@ -82,8 +81,6 @@ public:
 	State state() const;
 	GatheringState gatheringState() const;
 	SignalingState signalingState() const;
-	bool hasLocalDescription() const;
-	bool hasRemoteDescription() const;
 	bool hasMedia() const;
 	optional<Description> localDescription() const;
 	optional<Description> remoteDescription() const;

include/rtc/rtc.h

@@ -46,7 +46,8 @@ extern "C" {
 #define RTC_DEFAULT_MTU 1280 // IPv6 minimum guaranteed MTU
 
 #if RTC_ENABLE_MEDIA
-#define RTC_DEFAULT_MAXIMUM_FRAGMENT_SIZE ((uint16_t)(RTC_DEFAULT_MTU - 12 - 8 - 40)) // SRTP/UDP/IPv6
+#define RTC_DEFAULT_MAXIMUM_FRAGMENT_SIZE                                                          \
+	((uint16_t)(RTC_DEFAULT_MTU - 12 - 8 - 40)) // SRTP/UDP/IPv6
 #define RTC_DEFAULT_MAXIMUM_PACKET_COUNT_FOR_NACK_CACHE ((unsigned)512)
 #endif
 
@@ -94,61 +95,30 @@ typedef enum {
 	RTC_CERTIFICATE_RSA = 2,
 } rtcCertificateType;
 
-#if RTC_ENABLE_MEDIA
-
 typedef enum {
-    // video
+	// video
-    RTC_CODEC_H264 = 0,
+	RTC_CODEC_H264 = 0,
-    RTC_CODEC_VP8 = 1,
+	RTC_CODEC_VP8 = 1,
-    RTC_CODEC_VP9 = 2,
+	RTC_CODEC_VP9 = 2,
 
-    // audio
+	// audio
-    RTC_CODEC_OPUS = 128
+	RTC_CODEC_OPUS = 128
 } rtcCodec;
 
 typedef enum {
-    RTC_DIRECTION_UNKNOWN = 0,
+	RTC_DIRECTION_UNKNOWN = 0,
-    RTC_DIRECTION_SENDONLY = 1,
+	RTC_DIRECTION_SENDONLY = 1,
-    RTC_DIRECTION_RECVONLY = 2,
+	RTC_DIRECTION_RECVONLY = 2,
-    RTC_DIRECTION_SENDRECV = 3,
+	RTC_DIRECTION_SENDRECV = 3,
-    RTC_DIRECTION_INACTIVE = 4
+	RTC_DIRECTION_INACTIVE = 4
 } rtcDirection;
 
-#endif // RTC_ENABLE_MEDIA
-
 #define RTC_ERR_SUCCESS 0
 #define RTC_ERR_INVALID -1   // invalid argument
 #define RTC_ERR_FAILURE -2   // runtime error
 #define RTC_ERR_NOT_AVAIL -3 // element not available
 #define RTC_ERR_TOO_SMALL -4 // buffer too small
 
-typedef struct {
-	const char **iceServers;
-	int iceServersCount;
-	rtcCertificateType certificateType;
-	bool enableIceTcp;
-	bool disableAutoNegotiation;
-	uint16_t portRangeBegin;
-	uint16_t portRangeEnd;
-	int mtu; // <= 0 means automatic
-	int maxMessageSize; // <= 0 means default
-} rtcConfiguration;
-
-typedef struct {
-	bool unordered;
-	bool unreliable;
-	int maxPacketLifeTime; // ignored if reliable
-	int maxRetransmits;    // ignored if reliable
-} rtcReliability;
-
-typedef struct {
-	rtcReliability reliability;
-	const char *protocol; // empty string if NULL
-	bool negotiated;
-	bool manualStream;
-	uint16_t stream; // numeric ID 0-65534, ignored if manualStream is false
-} rtcDataChannelInit;
-
 typedef void(RTC_API *rtcLogCallbackFunc)(rtcLogLevel level, const char *message);
 typedef void(RTC_API *rtcDescriptionCallbackFunc)(int pc, const char *sdp, const char *type,
                                                   void *ptr);
@@ -167,14 +137,28 @@ typedef void(RTC_API *rtcBufferedAmountLowCallbackFunc)(int id, void *ptr);
 typedef void(RTC_API *rtcAvailableCallbackFunc)(int id, void *ptr);
 
 // Log
+
 // NULL cb on the first call will log to stdout
 RTC_EXPORT void rtcInitLogger(rtcLogLevel level, rtcLogCallbackFunc cb);
 
 // User pointer
 RTC_EXPORT void rtcSetUserPointer(int id, void *ptr);
-RTC_EXPORT void * rtcGetUserPointer(int i);
+RTC_EXPORT void *rtcGetUserPointer(int i);
 
 // PeerConnection
+
+typedef struct {
+	const char **iceServers;
+	int iceServersCount;
+	rtcCertificateType certificateType;
+	bool enableIceTcp;
+	bool disableAutoNegotiation;
+	uint16_t portRangeBegin;
+	uint16_t portRangeEnd;
+	int mtu;            // <= 0 means automatic
+	int maxMessageSize; // <= 0 means default
+} rtcConfiguration;
+
 RTC_EXPORT int rtcCreatePeerConnection(const rtcConfiguration *config); // returns pc id
 RTC_EXPORT int rtcDeletePeerConnection(int pc);
 
@@ -201,6 +185,22 @@ RTC_EXPORT int rtcGetSelectedCandidatePair(int pc, char *local, int localSize, c
                                            int remoteSize);
 
 // DataChannel
+
+typedef struct {
+	bool unordered;
+	bool unreliable;
+	int maxPacketLifeTime; // ignored if reliable
+	int maxRetransmits;    // ignored if reliable
+} rtcReliability;
+
+typedef struct {
+	rtcReliability reliability;
+	const char *protocol; // empty string if NULL
+	bool negotiated;
+	bool manualStream;
+	uint16_t stream; // numeric ID 0-65534, ignored if manualStream is false
+} rtcDataChannelInit;
+
 RTC_EXPORT int rtcSetDataChannelCallback(int pc, rtcDataChannelCallbackFunc cb);
 RTC_EXPORT int rtcCreateDataChannel(int pc, const char *label); // returns dc id
 RTC_EXPORT int rtcCreateDataChannelEx(int pc, const char *label,
@@ -213,108 +213,96 @@ RTC_EXPORT int rtcGetDataChannelProtocol(int dc, char *buffer, int size);
 RTC_EXPORT int rtcGetDataChannelReliability(int dc, rtcReliability *reliability);
 
 // Track
+
+typedef struct {
+	rtcDirection direction;
+	rtcCodec codec;
+	int payloadType;
+	uint32_t ssrc;
+	const char *mid;
+	const char *name;    // optional
+	const char *msid;    // optional
+	const char *trackId; // optional, track ID used in MSID
+} rtcTrackInit;
+
 RTC_EXPORT int rtcSetTrackCallback(int pc, rtcTrackCallbackFunc cb);
 RTC_EXPORT int rtcAddTrack(int pc, const char *mediaDescriptionSdp); // returns tr id
+RTC_EXPORT int rtcAddTrackEx(int pc, const rtcTrackInit *init);      // returns tr id
 RTC_EXPORT int rtcDeleteTrack(int tr);
 
 RTC_EXPORT int rtcGetTrackDescription(int tr, char *buffer, int size);
 
-// Media
 #if RTC_ENABLE_MEDIA
 
-/// Add track
+// Media
-/// @param pc Peer connection id
-/// @param codec Codec
-/// @param payloadType Payload type
-/// @param ssrc SSRC
-/// @param _mid MID
-/// @param _direction Direction
-/// @param _name Name (optional)
-/// @param _msid MSID (optional)
-/// @param _trackID Track ID used in MSID (optional)
-/// @returns Track id
-RTC_EXPORT int rtcAddTrackEx(int pc, rtcCodec codec, int payloadType, uint32_t ssrc, const char *_mid, rtcDirection direction, const char *_name, const char *_msid, const char *_trackID);
 
-/// Set H264PacketizationHandler for track
+typedef struct {
-/// @param tr Track id
+	uint32_t ssrc;
-/// @param ssrc SSRC
+	const char *cname;
-/// @param cname CName
+	uint8_t payloadType;
-/// @param payloadType Payload Type
+	uint32_t clockRate;
-/// @param clockRate Clock rate
+	uint16_t maxFragmentSize; // Maximum NALU fragment size
-/// @param maxFragmentSize Maximum NALU fragment size
+	uint16_t sequenceNumber;
-/// @param sequenceNumber Sequence number
+	uint32_t timestamp;
-/// @param timestamp Timestamp
+} rtcPacketizationHandlerInit;
-RTC_EXPORT int rtcSetH264PacketizationHandler(int tr, uint32_t ssrc, const char * cname, uint8_t payloadType, uint32_t clockRate, uint16_t maxFragmentSize, uint16_t sequenceNumber, uint32_t timestamp);
 
-/// Set OpusPacketizationHandler for track
+typedef struct {
-/// @param tr Track id
+	double seconds;     // Start time in seconds
-/// @param ssrc SSRC
+	bool since1970;     // true if seconds since 1970
-/// @param cname CName
+	                    // false if seconds since 1900
-/// @param payloadType Payload Type
+	uint32_t timestamp; // Start timestamp
-/// @param clockRate Clock rate
+} rtcStartTime;
-/// @param _sequenceNumber Sequence number
-/// @param _timestamp Timestamp
-RTC_EXPORT int rtcSetOpusPacketizationHandler(int tr, uint32_t ssrc, const char * cname, uint8_t payloadType, uint32_t clockRate, uint16_t _sequenceNumber, uint32_t _timestamp);
 
-/// Chain RtcpSrReporter to handler chain for given track
+// Set H264PacketizationHandler for track
-/// @param tr Track id
+RTC_EXPORT int rtcSetH264PacketizationHandler(int tr, const rtcPacketizationHandlerInit *init);
-int rtcChainRtcpSrReporter(int tr);
 
-/// Chain RtcpNackResponder to handler chain for given track
+// Set OpusPacketizationHandler for track
-/// @param tr Track id
+RTC_EXPORT int rtcSetOpusPacketizationHandler(int tr, const rtcPacketizationHandlerInit *init);
-/// @param maxStoredPacketsCount Maximum stored packet count
+
-int rtcChainRtcpNackResponder(int tr, unsigned maxStoredPacketsCount);
+// Chain RtcpSrReporter to handler chain for given track
+RTC_EXPORT int rtcChainRtcpSrReporter(int tr);
+
+// Chain RtcpNackResponder to handler chain for given track
+RTC_EXPORT int rtcChainRtcpNackResponder(int tr, unsigned int maxStoredPacketsCount);
 
 /// Set start time for RTP stream
-/// @param startTime_s Start time in seconds
+RTC_EXPORT int rtcSetRtpConfigurationStartTime(int id, const rtcStartTime *startTime);
-/// @param timeIntervalSince1970 Set true if `startTime_s` is time interval since 1970, false if `startTime_s` is time interval since 1900
-/// @param _timestamp Start timestamp
-int rtcSetRtpConfigurationStartTime(int id, double startTime_s, bool timeIntervalSince1970, uint32_t _timestamp);
 
-/// Start stats recording for RTCP Sender Reporter
+// Start stats recording for RTCP Sender Reporter
-/// @param id Track identifier
+RTC_EXPORT int rtcStartRtcpSenderReporterRecording(int id);
-int rtcStartRtcpSenderReporterRecording(int id);
 
-/// Transform seconds to timestamp using track's clock rate
+// Transform seconds to timestamp using track's clock rate
-/// @param id Track id
+// Result is written to timestamp
-/// @param seconds Seconds
+RTC_EXPORT int rtcTransformSecondsToTimestamp(int id, double seconds, uint32_t *timestamp);
-/// @param timestamp Pointer to result
-int rtcTransformSecondsToTimestamp(int id, double seconds, uint32_t * timestamp);
 
-/// Transform timestamp to seconds using track's clock rate
+// Transform timestamp to seconds using track's clock rate
-/// @param id Track id
+// Result is written to seconds
-/// @param timestamp Timestamp
+RTC_EXPORT int rtcTransformTimestampToSeconds(int id, uint32_t timestamp, double *seconds);
-/// @param seconds Pointer for result
-int rtcTransformTimestampToSeconds(int id, uint32_t timestamp, double * seconds);
 
-/// Get current timestamp
+// Get current timestamp
-/// @param id Track id
+// Result is written to timestamp
-/// @param timestamp Pointer for result
+RTC_EXPORT int rtcGetCurrentTrackTimestamp(int id, uint32_t *timestamp);
-int rtcGetCurrentTrackTimestamp(int id, uint32_t * timestamp);
 
-/// Get start timestamp for track identified by given id
+// Get start timestamp for track identified by given id
-/// @param id Track id
+// Result is written to timestamp
-/// @param timestamp Pointer for result
+RTC_EXPORT int rtcGetTrackStartTimestamp(int id, uint32_t *timestamp);
-int rtcGetTrackStartTimestamp(int id, uint32_t * timestamp);
 
-/// Set RTP timestamp for track identified by given id
+// Set RTP timestamp for track identified by given id
-/// @param id Track id
+RTC_EXPORT int rtcSetTrackRtpTimestamp(int id, uint32_t timestamp);
-/// @param timestamp New timestamp
-int rtcSetTrackRTPTimestamp(int id, uint32_t timestamp);
 
-/// Get timestamp of previous RTCP SR
+// Get timestamp of previous RTCP SR
-/// @param id Track id
+// Result is written to timestamp
-/// @param timestamp Pointer for result
+RTC_EXPORT int rtcGetPreviousTrackSenderReportTimestamp(int id, uint32_t *timestamp);
-int rtcGetPreviousTrackSenderReportTimestamp(int id, uint32_t * timestamp);
 
-/// Set `NeedsToReport` flag in RtcpSrReporter handler identified by given track id
+// Set NeedsToReport flag in RtcpSrReporter handler identified by given track id
-/// @param id Track id
+RTC_EXPORT int rtcSetNeedsToSendRtcpSr(int id);
-int rtcSetNeedsToSendRtcpSr(int id);
 
 #endif // RTC_ENABLE_MEDIA
 
-// WebSocket
 #if RTC_ENABLE_WEBSOCKET
+
+// WebSocket
+
 typedef struct {
 	bool disableTlsVerification; // if true, don't verify the TLS certificate
 } rtcWsConfiguration;
@@ -322,9 +310,11 @@ typedef struct {
 RTC_EXPORT int rtcCreateWebSocket(const char *url); // returns ws id
 RTC_EXPORT int rtcCreateWebSocketEx(const char *url, const rtcWsConfiguration *config);
 RTC_EXPORT int rtcDeleteWebsocket(int ws);
+
 #endif
 
 // DataChannel, Track, and WebSocket common API
+
 RTC_EXPORT int rtcSetOpenCallback(int id, rtcOpenCallbackFunc cb);
 RTC_EXPORT int rtcSetClosedCallback(int id, rtcClosedCallbackFunc cb);
 RTC_EXPORT int rtcSetErrorCallback(int id, rtcErrorCallbackFunc cb);
@@ -336,14 +326,31 @@ RTC_EXPORT int rtcSetBufferedAmountLowThreshold(int id, int amount);
 RTC_EXPORT int rtcSetBufferedAmountLowCallback(int id, rtcBufferedAmountLowCallbackFunc cb);
 
 // DataChannel, Track, and WebSocket common extended API
+
 RTC_EXPORT int rtcGetAvailableAmount(int id); // total size available to receive
 RTC_EXPORT int rtcSetAvailableCallback(int id, rtcAvailableCallbackFunc cb);
 RTC_EXPORT int rtcReceiveMessage(int id, char *buffer, int *size);
 
-// Optional preload and cleanup
+// Optional global preload and cleanup
+
 RTC_EXPORT void rtcPreload(void);
 RTC_EXPORT void rtcCleanup(void);
 
+// SCTP global settings
+
+typedef struct {
+	int recvBufferSize;          // in bytes, <= 0 means optimized default
+	int sendBufferSize;          // in bytes, <= 0 means optimized default
+	int maxChunksOnQueue;        // in chunks, <= 0 means optimized default
+	int initialCongestionWindow; // in MTUs, <= 0 means optimized default
+	int maxBurst;                // in MTUs, 0 means optimized default, < 0 means disabled
+	int congestionControlModule; // 0: RFC2581 (default), 1: HSTCP, 2: H-TCP, 3: RTCC
+	int delayedSackTimeMs;       // in msecs, <= 0 means optimized default
+} rtcSctpSettings;
+
+// Note: SCTP settings apply to newly-created PeerConnections only
+RTC_EXPORT int rtcSetSctpSettings(const rtcSctpSettings *settings);
+
 #ifdef __cplusplus
 } // extern "C"
 #endif

include/rtc/rtc.hpp

@@ -21,12 +21,11 @@
 
 // C++ API
 #include "common.hpp"
-#include "init.hpp" // for rtc::Cleanup()
+#include "global.hpp"
-#include "log.hpp"
 //
 #include "datachannel.hpp"
-#include "track.hpp"
 #include "peerconnection.hpp"
+#include "track.hpp"
 
 #if RTC_ENABLE_WEBSOCKET
 
@@ -38,14 +37,13 @@
 #if RTC_ENABLE_MEDIA
 
 // Media handling
-#include "rtcpreceivingsession.hpp"
 #include "mediachainablehandler.hpp"
-#include "rtcpsrreporter.hpp"
 #include "rtcpnackresponder.hpp"
+#include "rtcpreceivingsession.hpp"
+#include "rtcpsrreporter.hpp"
 
 // Opus/h264 streaming
 #include "h264packetizationhandler.hpp"
 #include "opuspacketizationhandler.hpp"
 
 #endif // RTC_ENABLE_MEDIA
-

include/rtc/rtcpnackresponder.hpp

@@ -1,18 +1,19 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_RTCP_NACK_RESPONDER_H
@@ -27,11 +28,11 @@
 
 namespace rtc {
 
-class RTC_CPP_EXPORT RtcpNackResponder: public MediaHandlerElement {
+class RTC_CPP_EXPORT RtcpNackResponder final: public MediaHandlerElement {
 
 	/// Packet storage
 	class RTC_CPP_EXPORT Storage {
-		
+
 		/// Packet storage element
 		struct RTC_CPP_EXPORT Element {
 			Element(binary_ptr packet, uint16_t sequenceNumber, shared_ptr<Element> next = nullptr);

include/rtc/rtcpsrreporter.hpp

@@ -1,18 +1,19 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_RTCP_SENDER_REPORTABLE_H
@@ -26,7 +27,7 @@
 
 namespace rtc {
 
-class RTC_CPP_EXPORT RtcpSrReporter: public MediaHandlerElement {
+class RTC_CPP_EXPORT RtcpSrReporter final: public MediaHandlerElement {
 
 	bool needsToReport = false;
 

include/rtc/rtp.hpp

@@ -21,23 +21,9 @@
 #ifndef RTC_RTP_HPP
 #define RTC_RTP_HPP
 
-#include "log.hpp"
+#include "common.hpp"
 
-#include <cmath>
+#include <vector>
-
-#ifdef _WIN32
-#include <winsock2.h>
-#else
-#include <arpa/inet.h>
-#endif
-
-#ifndef htonll
-#define htonll(x)                                                                                  \
-	((uint64_t)(((uint64_t)htonl((uint32_t)(x))) << 32) | (uint64_t)htonl((uint32_t)((x) >> 32)))
-#endif
-#ifndef ntohll
-#define ntohll(x) htonll(x)
-#endif
 
 namespace rtc {
 
@@ -45,63 +31,40 @@ typedef uint32_t SSRC;
 
 #pragma pack(push, 1)
 
-struct RTP {
+struct RTC_CPP_EXPORT RTP {
-private:
 	uint8_t _first;
 	uint8_t _payloadType;
 	uint16_t _seqNumber;
 	uint32_t _timestamp;
 	SSRC _ssrc;
+	SSRC _csrc[16];
 
-public:
+	[[nodiscard]] uint8_t version() const;
-	SSRC csrc[16];
+	[[nodiscard]] bool padding() const;
+	[[nodiscard]] bool extension() const;
+	[[nodiscard]] uint8_t csrcCount() const;
+	[[nodiscard]] uint8_t marker() const;
+	[[nodiscard]] uint8_t payloadType() const;
+	[[nodiscard]] uint16_t seqNumber() const;
+	[[nodiscard]] uint32_t timestamp() const;
+	[[nodiscard]] uint32_t ssrc() const;
 
-	inline uint8_t version() const { return _first >> 6; }
+	[[nodiscard]] size_t getSize() const;
-	inline bool padding() const { return (_first >> 5) & 0x01; }
+	[[nodiscard]] const char *getBody() const;
-	inline bool extension() const { return (_first >> 4) & 0x01; }
+	[[nodiscard]] char *getBody();
-	inline uint8_t csrcCount() const { return _first & 0x0F; }
-	inline uint8_t marker() const { return _payloadType & 0b10000000; }
-	inline uint8_t payloadType() const { return _payloadType & 0b01111111; }
-	inline uint16_t seqNumber() const { return ntohs(_seqNumber); }
-	inline uint32_t timestamp() const { return ntohl(_timestamp); }
-	inline uint32_t ssrc() const { return ntohl(_ssrc); }
 
-	inline size_t getSize() const {
+	void log() const;
-		return reinterpret_cast<const char *>(&csrc) - reinterpret_cast<const char *>(this) +
-		       sizeof(SSRC) * csrcCount();
-	}
 
-	[[nodiscard]] char *getBody() {
+	void preparePacket();
-		return reinterpret_cast<char *>(&csrc) + sizeof(SSRC) * csrcCount();
+	void setSeqNumber(uint16_t newSeqNo);
-	}
+	void setPayloadType(uint8_t newPayloadType);
-
+	void setSsrc(uint32_t in_ssrc);
-	[[nodiscard]] const char *getBody() const {
+	void setMarker(bool marker);
-		return reinterpret_cast<const char *>(&csrc) + sizeof(SSRC) * csrcCount();
+	void setTimestamp(uint32_t i);
-	}
-
-	inline void preparePacket() { _first |= (1 << 7); }
-
-	inline void setSeqNumber(uint16_t newSeqNo) { _seqNumber = htons(newSeqNo); }
-	inline void setPayloadType(uint8_t newPayloadType) {
-		_payloadType = (_payloadType & 0b10000000u) | (0b01111111u & newPayloadType);
-	}
-	inline void setSsrc(uint32_t in_ssrc) { _ssrc = htonl(in_ssrc); }
-	inline void setMarker(bool marker) { _payloadType = (_payloadType & 0x7F) | (marker << 7); };
-
-	void setTimestamp(uint32_t i) { _timestamp = htonl(i); }
-
-	void log() {
-		PLOG_VERBOSE << "RTP V: " << (int)version() << " P: " << (padding() ? "P" : " ")
-		             << " X: " << (extension() ? "X" : " ") << " CC: " << (int)csrcCount()
-		             << " M: " << (marker() ? "M" : " ") << " PT: " << (int)payloadType()
-		             << " SEQNO: " << seqNumber() << " TS: " << timestamp();
-	}
 };
 
-struct RTCP_ReportBlock {
+struct RTC_CPP_EXPORT RTCP_ReportBlock {
-	SSRC ssrc;
+	SSRC _ssrc;
-
-private:
 	uint32_t _fractionLostAndPacketsLost; // fraction lost is 8-bit, packets lost is 24-bit
 	uint16_t _seqNoCycles;
 	uint16_t _highestSeqNo;
@@ -109,136 +72,68 @@ private:
 	uint32_t _lastReport;
 	uint32_t _delaySinceLastReport;
 
-public:
+	[[nodiscard]] uint16_t seqNoCycles() const;
-	inline void preparePacket(SSRC in_ssrc, [[maybe_unused]] unsigned int packetsLost,
+	[[nodiscard]] uint16_t highestSeqNo() const;
-	                          [[maybe_unused]] unsigned int totalPackets, uint16_t highestSeqNo,
+	[[nodiscard]] uint32_t jitter() const;
-	                          uint16_t seqNoCycles, uint32_t jitter, uint64_t lastSR_NTP,
+	[[nodiscard]] uint32_t delaySinceSR() const;
-	                          uint64_t lastSR_DELAY) {
-		setSeqNo(highestSeqNo, seqNoCycles);
-		setJitter(jitter);
-		setSSRC(in_ssrc);
 
-		// Middle 32 bits of NTP Timestamp
+	[[nodiscard]] SSRC getSSRC() const;
-		//		  this->lastReport = lastSR_NTP >> 16u;
+	[[nodiscard]] uint32_t getNTPOfSR() const;
-		setNTPOfSR(uint64_t(lastSR_NTP));
+	[[nodiscard]] unsigned int getLossPercentage() const;
-		setDelaySinceSR(uint32_t(lastSR_DELAY));
+	[[nodiscard]] unsigned int getPacketLostCount() const;
 
-		// The delay, expressed in units of 1/65536 seconds
+	void preparePacket(SSRC in_ssrc, unsigned int packetsLost, unsigned int totalPackets,
-		// this->delaySinceLastReport = lastSR_DELAY;
+	                   uint16_t highestSeqNo, uint16_t seqNoCycles, uint32_t jitter,
-	}
+	                   uint64_t lastSR_NTP, uint64_t lastSR_DELAY);
+	void setSSRC(SSRC in_ssrc);
+	void setPacketsLost(unsigned int packetsLost, unsigned int totalPackets);
+	void setSeqNo(uint16_t highestSeqNo, uint16_t seqNoCycles);
+	void setJitter(uint32_t jitter);
+	void setNTPOfSR(uint64_t ntp);
+	void setDelaySinceSR(uint32_t sr);
 
-	inline void setSSRC(SSRC in_ssrc) { this->ssrc = htonl(in_ssrc); }
+	void log() const;
-	[[nodiscard]] inline SSRC getSSRC() const { return ntohl(ssrc); }
-
-	inline void setPacketsLost([[maybe_unused]] unsigned int packetsLost,
-	                           [[maybe_unused]] unsigned int totalPackets) {
-		// TODO Implement loss percentages.
-		_fractionLostAndPacketsLost = 0;
-	}
-
-	[[nodiscard]] inline unsigned int getLossPercentage() const {
-		// TODO Implement loss percentages.
-		return 0;
-	}
-	[[nodiscard]] inline unsigned int getPacketLostCount() const {
-		// TODO Implement total packets lost.
-		return 0;
-	}
-
-	inline uint16_t seqNoCycles() const { return ntohs(_seqNoCycles); }
-	inline uint16_t highestSeqNo() const { return ntohs(_highestSeqNo); }
-	inline uint32_t jitter() const { return ntohl(_jitter); }
-
-	inline void setSeqNo(uint16_t highestSeqNo, uint16_t seqNoCycles) {
-		_highestSeqNo = htons(highestSeqNo);
-		_seqNoCycles = htons(seqNoCycles);
-	}
-
-	inline void setJitter(uint32_t jitter) { _jitter = htonl(jitter); }
-
-	inline void setNTPOfSR(uint64_t ntp) { _lastReport = htonll(ntp >> 16u); }
-	[[nodiscard]] inline uint32_t getNTPOfSR() const { return ntohl(_lastReport) << 16u; }
-
-	inline void setDelaySinceSR(uint32_t sr) {
-		// The delay, expressed in units of 1/65536 seconds
-		_delaySinceLastReport = htonl(sr);
-	}
-	[[nodiscard]] inline uint32_t getDelaySinceSR() const { return ntohl(_delaySinceLastReport); }
-
-	inline void log() const {
-		PLOG_VERBOSE << "RTCP report block: "
-		             << "ssrc="
-		             << ntohl(ssrc)
-		             // TODO: Implement these reports
-		             //	<< ", fractionLost=" << fractionLost
-		             //	<< ", packetsLost=" << packetsLost
-		             << ", highestSeqNo=" << highestSeqNo() << ", seqNoCycles=" << seqNoCycles()
-		             << ", jitter=" << jitter() << ", lastSR=" << getNTPOfSR()
-		             << ", lastSRDelay=" << getDelaySinceSR();
-	}
 };
 
-struct RTCP_HEADER {
+struct RTC_CPP_EXPORT RTCP_HEADER {
-private:
 	uint8_t _first;
 	uint8_t _payloadType;
 	uint16_t _length;
 
-public:
+	[[nodiscard]] uint8_t version() const;
-	inline uint8_t version() const { return _first >> 6; }
+	[[nodiscard]] bool padding() const;
-	inline bool padding() const { return (_first >> 5) & 0x01; }
+	[[nodiscard]] uint8_t reportCount() const;
-	inline uint8_t reportCount() const { return _first & 0x0F; }
+	[[nodiscard]] uint8_t payloadType() const;
-	inline uint8_t payloadType() const { return _payloadType; }
+	[[nodiscard]] uint16_t length() const;
-	inline uint16_t length() const { return ntohs(_length); }
+	[[nodiscard]] size_t lengthInBytes() const;
-	inline size_t lengthInBytes() const { return (1 + length()) * 4; }
 
-	inline void setPayloadType(uint8_t type) { _payloadType = type; }
+	void prepareHeader(uint8_t payloadType, uint8_t reportCount, uint16_t length);
-	inline void setReportCount(uint8_t count) {
+	void setPayloadType(uint8_t type);
-		_first = (_first & 0b11100000u) | (count & 0b00011111u);
+	void setReportCount(uint8_t count);
-	}
+	void setLength(uint16_t length);
-	inline void setLength(uint16_t length) { _length = htons(length); }
 
-	inline void prepareHeader(uint8_t payloadType, uint8_t reportCount, uint16_t length) {
+	void log() const;
-		_first = 0b10000000; // version 2, no padding
-		setReportCount(reportCount);
-		setPayloadType(payloadType);
-		setLength(length);
-	}
-
-	inline void log() const {
-		PLOG_VERBOSE << "RTCP header: "
-		             << "version=" << unsigned(version()) << ", padding=" << padding()
-		             << ", reportCount=" << unsigned(reportCount())
-		             << ", payloadType=" << unsigned(payloadType()) << ", length=" << length();
-	}
 };
 
-struct RTCP_FB_HEADER {
+struct RTC_CPP_EXPORT RTCP_FB_HEADER {
 	RTCP_HEADER header;
-	SSRC packetSender;
-	SSRC mediaSource;
 
-	[[nodiscard]] SSRC getPacketSenderSSRC() const { return ntohl(packetSender); }
+	SSRC _packetSender;
+	SSRC _mediaSource;
 
-	[[nodiscard]] SSRC getMediaSourceSSRC() const { return ntohl(mediaSource); }
+	[[nodiscard]] SSRC packetSenderSSRC() const;
+	[[nodiscard]] SSRC mediaSourceSSRC() const;
 
-	void setPacketSenderSSRC(SSRC ssrc) { this->packetSender = htonl(ssrc); }
+	void setPacketSenderSSRC(SSRC ssrc);
+	void setMediaSourceSSRC(SSRC ssrc);
 
-	void setMediaSourceSSRC(SSRC ssrc) { this->mediaSource = htonl(ssrc); }
+	void log() const;
-
-	void log() {
-		header.log();
-		PLOG_VERBOSE << "FB: "
-		             << " packet sender: " << getPacketSenderSSRC()
-		             << " media source: " << getMediaSourceSSRC();
-	}
 };
 
-struct RTCP_SR {
+struct RTC_CPP_EXPORT RTCP_SR {
 	RTCP_HEADER header;
+
 	SSRC _senderSSRC;
-
-private:
 	uint64_t _ntpTimestamp;
 	uint32_t _rtpTimestamp;
 	uint32_t _packetCount;
@@ -246,418 +141,175 @@ private:
 
 	RTCP_ReportBlock _reportBlocks;
 
-public:
+	[[nodiscard]] static unsigned int Size(unsigned int reportCount);
-	inline void preparePacket(SSRC senderSSRC, uint8_t reportCount) {
-		unsigned int length =
-		    ((sizeof(header) + 24 + reportCount * sizeof(RTCP_ReportBlock)) / 4) - 1;
-		header.prepareHeader(200, reportCount, uint16_t(length));
-		this->_senderSSRC = htonl(senderSSRC);
-	}
 
-	[[nodiscard]] inline RTCP_ReportBlock *getReportBlock(int num) { return &_reportBlocks + num; }
+	[[nodiscard]] uint64_t ntpTimestamp() const;
-	[[nodiscard]] inline const RTCP_ReportBlock *getReportBlock(int num) const {
+	[[nodiscard]] uint32_t rtpTimestamp() const;
-		return &_reportBlocks + num;
+	[[nodiscard]] uint32_t packetCount() const;
-	}
+	[[nodiscard]] uint32_t octetCount() const;
+	[[nodiscard]] uint32_t senderSSRC() const;
 
-	[[nodiscard]] static unsigned int size(unsigned int reportCount) {
+	[[nodiscard]] const RTCP_ReportBlock *getReportBlock(int num) const;
-		return sizeof(RTCP_HEADER) + 24 + reportCount * sizeof(RTCP_ReportBlock);
+	[[nodiscard]] RTCP_ReportBlock *getReportBlock(int num);
-	}
+	[[nodiscard]] unsigned int size(unsigned int reportCount);
+	[[nodiscard]] size_t getSize() const;
 
-	[[nodiscard]] inline size_t getSize() const {
+	void preparePacket(SSRC senderSSRC, uint8_t reportCount);
-		// "length" in packet is one less than the number of 32 bit words in the packet.
+	void setNtpTimestamp(uint64_t ts);
-		return sizeof(uint32_t) * (1 + size_t(header.length()));
+	void setRtpTimestamp(uint32_t ts);
-	}
+	void setOctetCount(uint32_t ts);
+	void setPacketCount(uint32_t ts);
 
-	inline uint64_t ntpTimestamp() const { return ntohll(_ntpTimestamp); }
+	void log() const;
-	inline uint32_t rtpTimestamp() const { return ntohl(_rtpTimestamp); }
-	inline uint32_t packetCount() const { return ntohl(_packetCount); }
-	inline uint32_t octetCount() const { return ntohl(_octetCount); }
-	inline uint32_t senderSSRC() const { return ntohl(_senderSSRC); }
-
-	inline void setNtpTimestamp(uint64_t ts) { _ntpTimestamp = htonll(ts); }
-	inline void setRtpTimestamp(uint32_t ts) { _rtpTimestamp = htonl(ts); }
-	inline void setOctetCount(uint32_t ts) { _octetCount = htonl(ts); }
-	inline void setPacketCount(uint32_t ts) { _packetCount = htonl(ts); }
-
-	inline void log() const {
-		header.log();
-		PLOG_VERBOSE << "RTCP SR: "
-		             << " SSRC=" << senderSSRC() << ", NTP_TS=" << ntpTimestamp()
-		             << ", RTP_TS=" << rtpTimestamp() << ", packetCount=" << packetCount()
-		             << ", octetCount=" << octetCount();
-
-		for (unsigned i = 0; i < unsigned(header.reportCount()); i++) {
-			getReportBlock(i)->log();
-		}
-	}
 };
 
-struct RTCP_SDES_ITEM {
+struct RTC_CPP_EXPORT RTCP_SDES_ITEM {
-public:
 	uint8_t type;
 
-private:
 	uint8_t _length;
 	char _text[1];
 
-public:
+	[[nodiscard]] static unsigned int Size(uint8_t textLength);
-	inline std::string text() const { return std::string(_text, _length); }
-	inline void setText(std::string text) {
-		if(text.size() > 0xFF)
-			throw std::invalid_argument("text is too long");
 
-		_length = uint8_t(text.size());
+	[[nodiscard]] string text() const;
-		memcpy(_text, text.data(), text.size());
+	[[nodiscard]] uint8_t length() const;
-	}
 
-	inline uint8_t length() { return _length; }
+	void setText(string text);
-
-	[[nodiscard]] static unsigned int size(uint8_t textLength) { return textLength + 2; }
 };
 
 struct RTCP_SDES_CHUNK {
-private:
 	SSRC _ssrc;
 	RTCP_SDES_ITEM _items;
 
-public:
+	[[nodiscard]] static unsigned int Size(const std::vector<uint8_t> textLengths);
-	inline SSRC ssrc() const { return ntohl(_ssrc); }
-	inline void setSSRC(SSRC ssrc) { _ssrc = htonl(ssrc); }
 
-	/// Get item at given index
+	[[nodiscard]] SSRC ssrc() const;
-	/// @note All items with index < `num` must be valid, otherwise this function has undefined
-	/// behaviour (use `safelyCountChunkSize` to check if chunk is valid)
-	/// @param num Index of item to return
-	inline RTCP_SDES_ITEM *getItem(int num) {
-		auto base = &_items;
-		while (num-- > 0) {
-			auto itemSize = RTCP_SDES_ITEM::size(base->length());
-			base = reinterpret_cast<RTCP_SDES_ITEM *>(reinterpret_cast<uint8_t *>(base) + itemSize);
-		}
-		return reinterpret_cast<RTCP_SDES_ITEM *>(base);
-	}
 
-	long safelyCountChunkSize(size_t maxChunkSize) {
+	void setSSRC(SSRC ssrc);
-		if (maxChunkSize < RTCP_SDES_CHUNK::size({})) {
-			// chunk is truncated
-			return -1;
-		} else {
-			size_t size = sizeof(SSRC);
-			unsigned int i = 0;
-			// We can always access first 4 bytes of first item (in case of no items there will be 4
-			// null bytes)
-			auto item = getItem(i);
-			std::vector<uint8_t> textsLength{};
-			while (item->type != 0) {
-				if (size + RTCP_SDES_ITEM::size(0) > maxChunkSize) {
-					// item is too short
-					return -1;
-				}
-				auto itemLength = item->length();
-				if (size + RTCP_SDES_ITEM::size(itemLength) >= maxChunkSize) {
-					// item is too large (it can't be equal to chunk size because after item there
-					// must be 1-4 null bytes as padding)
-					return -1;
-				}
-				textsLength.push_back(itemLength);
-				// safely to access next item
-				item = getItem(++i);
-			}
-			auto realSize = RTCP_SDES_CHUNK::size(textsLength);
-			if (realSize > maxChunkSize) {
-				// Chunk is too large
-				return -1;
-			}
-			return realSize;
-		}
-	}
 
-	[[nodiscard]] static unsigned int size(const std::vector<uint8_t> textLengths) {
+	// Get item at given index
-		unsigned int itemsSize = 0;
+	// All items with index < num must be valid, otherwise this function has undefined behaviour
-		for (auto length : textLengths) {
+	// (use safelyCountChunkSize() to check if chunk is valid).
-			itemsSize += RTCP_SDES_ITEM::size(length);
+	[[nodiscard]] const RTCP_SDES_ITEM *getItem(int num) const;
-		}
+	[[nodiscard]] RTCP_SDES_ITEM *getItem(int num);
-		auto nullTerminatedItemsSize = itemsSize + 1;
-		auto words = uint8_t(std::ceil(double(nullTerminatedItemsSize) / 4)) + 1;
-		return words * 4;
-	}
 
-	/// Get size of chunk
+	// Get size of chunk
-	/// @note All  items must be valid, otherwise this function has undefined behaviour (use
+	// All items must be valid, otherwise this function has undefined behaviour (use
-	/// `safelyCountChunkSize` to check if chunk is valid)
+	// safelyCountChunkSize() to check if chunk is valid)
-	[[nodiscard]] unsigned int getSize() {
+	[[nodiscard]] unsigned int getSize() const;
-		std::vector<uint8_t> textLengths{};
+
-		unsigned int i = 0;
+	long safelyCountChunkSize(size_t maxChunkSize) const;
-		auto item = getItem(i);
-		while (item->type != 0) {
-			textLengths.push_back(item->length());
-			item = getItem(++i);
-		}
-		return size(textLengths);
-	}
 };
 
-struct RTCP_SDES {
+struct RTC_CPP_EXPORT RTCP_SDES {
 	RTCP_HEADER header;
-
-private:
 	RTCP_SDES_CHUNK _chunks;
 
-public:
+	[[nodiscard]] static unsigned int Size(const std::vector<std::vector<uint8_t>> lengths);
-	inline void preparePacket(uint8_t chunkCount) {
-		unsigned int chunkSize = 0;
-		for (uint8_t i = 0; i < chunkCount; i++) {
-			auto chunk = getChunk(i);
-			chunkSize += chunk->getSize();
-		}
-		uint16_t length = uint16_t((sizeof(header) + chunkSize) / 4 - 1);
-		header.prepareHeader(202, chunkCount, length);
-	}
 
-	bool isValid() {
+	bool isValid() const;
-		auto chunksSize = header.lengthInBytes() - sizeof(header);
-		if (chunksSize == 0) {
-			return true;
-		} else {
-			// there is at least one chunk
-			unsigned int i = 0;
-			unsigned int size = 0;
-			while (size < chunksSize) {
-				if (chunksSize < size + RTCP_SDES_CHUNK::size({})) {
-					// chunk is truncated
-					return false;
-				}
-				auto chunk = getChunk(i++);
-				auto chunkSize = chunk->safelyCountChunkSize(chunksSize - size);
-				if (chunkSize < 0) {
-					// chunk is invalid
-					return false;
-				}
-				size += chunkSize;
-			}
-			return size == chunksSize;
-		}
-	}
 
-	/// Returns number of chunks in this packet
+	// Returns number of chunks in this packet
-	/// @note Returns 0 if packet is invalid
+	// Returns 0 if packet is invalid
-	inline unsigned int chunksCount() {
+	unsigned int chunksCount() const;
-		if (!isValid()) {
-			return 0;
-		}
-		uint16_t chunksSize = 4 * (header.length() + 1) - sizeof(header);
-		unsigned int size = 0;
-		unsigned int i = 0;
-		while (size < chunksSize) {
-			size += getChunk(i++)->getSize();
-		}
-		return i;
-	}
 
-	/// Get chunk at given index
+	// Get chunk at given index
-	/// @note All chunks (and their items) with index < `num` must be valid, otherwise this function
+	// All chunks (and their items) with index < `num` must be valid, otherwise this function has
-	/// has undefined behaviour (use `isValid` to check if chunk is valid)
+	// undefined behaviour (use `isValid` to check if chunk is valid).
-	/// @param num Index of chunk to return
+	const RTCP_SDES_CHUNK *getChunk(int num) const;
-	inline RTCP_SDES_CHUNK *getChunk(int num) {
+	RTCP_SDES_CHUNK *getChunk(int num);
-		auto base = &_chunks;
-		while (num-- > 0) {
-			auto chunkSize = base->getSize();
-			base =
-			    reinterpret_cast<RTCP_SDES_CHUNK *>(reinterpret_cast<uint8_t *>(base) + chunkSize);
-		}
-		return reinterpret_cast<RTCP_SDES_CHUNK *>(base);
-	}
 
-	[[nodiscard]] static unsigned int size(const std::vector<std::vector<uint8_t>> lengths) {
+	void preparePacket(uint8_t chunkCount);
-		unsigned int chunks_size = 0;
-		for (auto length : lengths) {
-			chunks_size += RTCP_SDES_CHUNK::size(length);
-		}
-		return 4 + chunks_size;
-	}
 };
 
-struct RTCP_RR {
+struct RTC_CPP_EXPORT RTCP_RR {
 	RTCP_HEADER header;
-	SSRC _senderSSRC;
 
-private:
+	SSRC _senderSSRC;
 	RTCP_ReportBlock _reportBlocks;
 
-public:
+	[[nodiscard]] static size_t SizeWithReportBlocks(uint8_t reportCount);
-	[[nodiscard]] inline RTCP_ReportBlock *getReportBlock(int num) { return &_reportBlocks + num; }
-	[[nodiscard]] inline const RTCP_ReportBlock *getReportBlock(int num) const {
-		return &_reportBlocks + num;
-	}
 
-	inline SSRC senderSSRC() const { return ntohl(_senderSSRC); }
+	SSRC senderSSRC() const;
-	inline void setSenderSSRC(SSRC ssrc) { this->_senderSSRC = htonl(ssrc); }
+	bool isSenderReport();
+	bool isReceiverReport();
 
-	[[nodiscard]] inline size_t getSize() const {
+	[[nodiscard]] RTCP_ReportBlock *getReportBlock(int num);
-		// "length" in packet is one less than the number of 32 bit words in the packet.
+	[[nodiscard]] const RTCP_ReportBlock *getReportBlock(int num) const;
-		return sizeof(uint32_t) * (1 + size_t(header.length()));
+	[[nodiscard]] size_t getSize() const;
-	}
 
-	inline void preparePacket(SSRC senderSSRC, uint8_t reportCount) {
+	void preparePacket(SSRC senderSSRC, uint8_t reportCount);
-		// "length" in packet is one less than the number of 32 bit words in the packet.
+	void setSenderSSRC(SSRC ssrc);
-		size_t length = (sizeWithReportBlocks(reportCount) / 4) - 1;
-		header.prepareHeader(201, reportCount, uint16_t(length));
-		this->_senderSSRC = htonl(senderSSRC);
-	}
 
-	inline static size_t sizeWithReportBlocks(uint8_t reportCount) {
+	void log() const;
-		return sizeof(header) + 4 + size_t(reportCount) * sizeof(RTCP_ReportBlock);
-	}
-
-	inline bool isSenderReport() { return header.payloadType() == 200; }
-
-	inline bool isReceiverReport() { return header.payloadType() == 201; }
-
-	inline void log() const {
-		header.log();
-		PLOG_VERBOSE << "RTCP RR: "
-		             << " SSRC=" << ntohl(_senderSSRC);
-
-		for (unsigned i = 0; i < unsigned(header.reportCount()); i++) {
-			getReportBlock(i)->log();
-		}
-	}
 };
 
-struct RTCP_REMB {
+struct RTC_CPP_EXPORT RTCP_REMB {
 	RTCP_FB_HEADER header;
 
-	/*! \brief Unique identifier ('R' 'E' 'M' 'B') */
+	char _id[4];       // Unique identifier ('R' 'E' 'M' 'B')
-	char id[4];
+	uint32_t _bitrate; // Num SSRC, Br Exp, Br Mantissa (bit mask)
+	SSRC _ssrc[1];
 
-	/*! \brief Num SSRC, Br Exp, Br Mantissa (bit mask) */
+	[[nodiscard]] static size_t SizeWithSSRCs(int count);
-	uint32_t bitrate;
 
-	SSRC ssrc[1];
+	[[nodiscard]] unsigned int getSize() const;
 
-	[[nodiscard]] unsigned int getSize() const {
+	void preparePacket(SSRC senderSSRC, unsigned int numSSRC, unsigned int in_bitrate);
-		// "length" in packet is one less than the number of 32 bit words in the packet.
+	void setBitrate(unsigned int numSSRC, unsigned int in_bitrate);
-		return sizeof(uint32_t) * (1 + header.header.length());
+	void setSsrc(int iterator, SSRC newSssrc);
-	}
-
-	void preparePacket(SSRC senderSSRC, unsigned int numSSRC, unsigned int in_bitrate) {
-
-		// Report Count becomes the format here.
-		header.header.prepareHeader(206, 15, 0);
-
-		// Always zero.
-		header.setMediaSourceSSRC(0);
-
-		header.setPacketSenderSSRC(senderSSRC);
-
-		id[0] = 'R';
-		id[1] = 'E';
-		id[2] = 'M';
-		id[3] = 'B';
-
-		setBitrate(numSSRC, in_bitrate);
-	}
-
-	void setBitrate(unsigned int numSSRC, unsigned int in_bitrate) {
-		unsigned int exp = 0;
-		while (in_bitrate > pow(2, 18) - 1) {
-			exp++;
-			in_bitrate /= 2;
-		}
-
-		// "length" in packet is one less than the number of 32 bit words in the packet.
-		header.header.setLength(
-		    uint16_t((offsetof(RTCP_REMB, ssrc) / sizeof(uint32_t)) - 1 + numSSRC));
-
-		this->bitrate = htonl((numSSRC << (32u - 8u)) | (exp << (32u - 8u - 6u)) | in_bitrate);
-	}
-
-	void setSsrc(int iterator, SSRC newSssrc) { ssrc[iterator] = htonl(newSssrc); }
-
-	size_t static inline sizeWithSSRCs(int count) {
-		return sizeof(RTCP_REMB) + (count - 1) * sizeof(SSRC);
-	}
 };
 
-struct RTCP_PLI {
+struct RTC_CPP_EXPORT RTCP_PLI {
 	RTCP_FB_HEADER header;
 
-	void preparePacket(SSRC messageSSRC) {
+	[[nodiscard]] static unsigned int Size();
-		header.header.prepareHeader(206, 1, 2);
-		header.setPacketSenderSSRC(messageSSRC);
-		header.setMediaSourceSSRC(messageSSRC);
-	}
 
-	void print() { header.log(); }
+	void preparePacket(SSRC messageSSRC);
 
-	[[nodiscard]] static unsigned int size() { return sizeof(RTCP_FB_HEADER); }
+	void log() const;
 };
 
-struct RTCP_FIR_PART {
+struct RTC_CPP_EXPORT RTCP_FIR_PART {
 	uint32_t ssrc;
 	uint8_t seqNo;
 	uint8_t dummy1;
 	uint16_t dummy2;
 };
 
-struct RTCP_FIR {
+struct RTC_CPP_EXPORT RTCP_FIR {
 	RTCP_FB_HEADER header;
 	RTCP_FIR_PART parts[1];
 
-	void preparePacket(SSRC messageSSRC, uint8_t seqNo) {
+	static unsigned int Size();
-		header.header.prepareHeader(206, 4, 2 + 2 * 1);
-		header.setPacketSenderSSRC(messageSSRC);
-		header.setMediaSourceSSRC(messageSSRC);
-		parts[0].ssrc = htonl(messageSSRC);
-		parts[0].seqNo = seqNo;
-	}
 
-	void print() { header.log(); }
+	void preparePacket(SSRC messageSSRC, uint8_t seqNo);
 
-	[[nodiscard]] static unsigned int size() {
+	void log() const;
-		return sizeof(RTCP_FB_HEADER) + sizeof(RTCP_FIR_PART);
-	}
 };
 
-struct RTCP_NACK_PART {
+struct RTC_CPP_EXPORT RTCP_NACK_PART {
 	uint16_t _pid;
 	uint16_t _blp;
 
-	uint16_t getPID() { return ntohs(_pid); }
+	uint16_t pid();
-	uint16_t getBLP() { return ntohs(_blp); }
+	uint16_t blp();
 
-	void setPID(uint16_t pid) { _pid = htons(pid); }
+	void setPid(uint16_t pid);
-	void setBLP(uint16_t blp) { _blp = htons(blp); }
+	void setBlp(uint16_t blp);
 
-	std::vector<uint16_t> getSequenceNumbers() {
+	std::vector<uint16_t> getSequenceNumbers();
-		std::vector<uint16_t> result{};
-		result.reserve(17);
-		uint16_t pid = getPID();
-		result.push_back(pid);
-		uint16_t bitmask = getBLP();
-		uint16_t i = pid + 1;
-		while (bitmask > 0) {
-			if (bitmask & 0x1) {
-				result.push_back(i);
-			}
-			i += 1;
-			bitmask >>= 1;
-		}
-		return result;
-	}
 };
 
-class RTCP_NACK {
+struct RTC_CPP_EXPORT RTCP_NACK {
-public:
 	RTCP_FB_HEADER header;
 	RTCP_NACK_PART parts[1];
 
-public:
+	[[nodiscard]] static unsigned int Size(unsigned int discreteSeqNoCount);
-	void preparePacket(SSRC ssrc, unsigned int discreteSeqNoCount) {
+
-		header.header.prepareHeader(205, 1, 2 + uint16_t(discreteSeqNoCount));
+	[[nodiscard]] unsigned int getSeqNoCount();
-		header.setMediaSourceSSRC(ssrc);
+
-		header.setPacketSenderSSRC(ssrc);
+	void preparePacket(SSRC ssrc, unsigned int discreteSeqNoCount);
-	}
 
 	/**
 	 * Add a packet to the list of missing packets.
@@ -668,71 +320,22 @@ public:
 	 * @param missingPacket The seq no of the missing packet. This will be added to the queue.
 	 * @return true if the packet has grown, false otherwise.
 	 */
-	bool addMissingPacket(unsigned int *fciCount, uint16_t *fciPID, uint16_t missingPacket) {
+	bool addMissingPacket(unsigned int *fciCount, uint16_t *fciPID, uint16_t missingPacket);
-		if (*fciCount == 0 || missingPacket < *fciPID || missingPacket > (*fciPID + 16)) {
-			parts[*fciCount].setPID(missingPacket);
-			parts[*fciCount].setBLP(0);
-			*fciPID = missingPacket;
-			(*fciCount)++;
-			return true;
-		} else {
-			// TODO SPEED!
-			uint16_t blp = parts[(*fciCount) - 1].getBLP();
-			uint16_t newBit = uint16_t(1u << (missingPacket - (1 + *fciPID)));
-			parts[(*fciCount) - 1].setBLP(blp | newBit);
-			return false;
-		}
-	}
-
-	[[nodiscard]] static unsigned int getSize(unsigned int discreteSeqNoCount) {
-		return offsetof(RTCP_NACK, parts) + sizeof(RTCP_NACK_PART) * discreteSeqNoCount;
-	}
-
-	[[nodiscard]] unsigned int getSeqNoCount() { return header.header.length() - 2; }
 };
 
-class RTP_RTX {
+struct RTC_CPP_EXPORT RTP_RTX {
-private:
 	RTP header;
 
-public:
+	[[nodiscard]] const char *getBody() const;
-	size_t copyTo(RTP *dest, size_t totalSize, uint8_t originalPayloadType) {
+	[[nodiscard]] char *getBody();
-		memmove((char *)dest, (char *)this, header.getSize());
+	[[nodiscard]] size_t getBodySize(size_t totalSize) const;
-		dest->setSeqNumber(getOriginalSeqNo());
+	[[nodiscard]] size_t getSize() const;
-		dest->setPayloadType(originalPayloadType);
+	[[nodiscard]] uint16_t getOriginalSeqNo() const;
-		memmove(dest->getBody(), getBody(), getBodySize(totalSize));
-		return totalSize;
-	}
 
-	[[nodiscard]] uint16_t getOriginalSeqNo() const {
+	// Returns the new size of the packet
-		return ntohs(*(uint16_t *)(header.getBody()));
+	size_t normalizePacket(size_t totalSize, SSRC originalSSRC, uint8_t originalPayloadType);
-	}
 
-	[[nodiscard]] char *getBody() { return header.getBody() + sizeof(uint16_t); }
+	size_t copyTo(RTP *dest, size_t totalSize, uint8_t originalPayloadType);
-
-	[[nodiscard]] const char *getBody() const { return header.getBody() + sizeof(uint16_t); }
-
-	[[nodiscard]] size_t getBodySize(size_t totalSize) {
-		return totalSize - (getBody() - reinterpret_cast<char *>(this));
-	}
-
-	[[nodiscard]] size_t getSize()  const{
-		return header.getSize() + sizeof(uint16_t);
-	}
-
-	[[nodiscard]] RTP &getHeader() { return header; }
-
-	/*
-	 * Returns the new size of the packet
-	 */
-	size_t normalizePacket(size_t totalSize, SSRC originalSSRC, uint8_t originalPayloadType) {
-		header.setSeqNumber(getOriginalSeqNo());
-		header.setSsrc(originalSSRC);
-		header.setPayloadType(originalPayloadType);
-		// TODO, the -12 is the size of the header (which is variable!)
-		memmove(header.getBody(), getBody(), totalSize - getSize());
-		return totalSize - 2;
-	}
 };
 
 #pragma pack(pop)

include/rtc/rtppacketizationconfig.hpp

@@ -1,19 +1,19 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_RTP_PACKETIZATION_CONFIG_H

include/rtc/rtppacketizer.hpp

@@ -1,19 +1,19 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef RTC_RTP_PACKETIZER_H

include/rtc/utils.hpp

@@ -22,6 +22,9 @@
 #include <functional>
 #include <memory>
 #include <mutex>
+#include <optional>
+#include <tuple>
+#include <utility>
 
 namespace rtc {
 
@@ -32,11 +35,10 @@ template <class... Ts> overloaded(Ts...) -> overloaded<Ts...>;
 // weak_ptr bind helper
 template <typename F, typename T, typename... Args> auto weak_bind(F &&f, T *t, Args &&..._args) {
 	return [bound = std::bind(f, t, _args...), weak_this = t->weak_from_this()](auto &&...args) {
-		using result_type = typename decltype(bound)::result_type;
 		if (auto shared_this = weak_this.lock())
 			return bound(args...);
 		else
-			return static_cast<result_type>(false);
+			return static_cast<decltype(bound(args...))>(false);
 	};
 }
 
@@ -58,40 +60,35 @@ private:
 };
 
 // callback with built-in synchronization
-template <typename... Args> class synchronized_callback final {
+template <typename... Args> class synchronized_callback {
 public:
 	synchronized_callback() = default;
 	synchronized_callback(synchronized_callback &&cb) { *this = std::move(cb); }
 	synchronized_callback(const synchronized_callback &cb) { *this = cb; }
 	synchronized_callback(std::function<void(Args...)> func) { *this = std::move(func); }
-	~synchronized_callback() { *this = nullptr; }
+	virtual ~synchronized_callback() { *this = nullptr; }
 
 	synchronized_callback &operator=(synchronized_callback &&cb) {
 		std::scoped_lock lock(mutex, cb.mutex);
-		callback = std::move(cb.callback);
+		set(std::exchange(cb.callback, nullptr));
-		cb.callback = nullptr;
 		return *this;
 	}
 
 	synchronized_callback &operator=(const synchronized_callback &cb) {
 		std::scoped_lock lock(mutex, cb.mutex);
-		callback = cb.callback;
+		set(cb.callback);
 		return *this;
 	}
 
 	synchronized_callback &operator=(std::function<void(Args...)> func) {
 		std::lock_guard lock(mutex);
-		callback = std::move(func);
+		set(std::move(func));
 		return *this;
 	}
 
 	bool operator()(Args... args) const {
 		std::lock_guard lock(mutex);
-		if (!callback)
+		return call(std::move(args)...);
-			return false;
-
-		callback(std::move(args)...);
-		return true;
 	}
 
 	operator bool() const {
@@ -103,11 +100,48 @@ public:
 		return [this](Args... args) { (*this)(std::move(args)...); };
 	}
 
-private:
+protected:
+	virtual void set(std::function<void(Args...)> func) { callback = std::move(func); }
+	virtual bool call(Args... args) const {
+		if (!callback)
+			return false;
+
+		callback(std::move(args)...);
+		return true;
+	}
+
 	std::function<void(Args...)> callback;
 	mutable std::recursive_mutex mutex;
 };
 
+// callback with built-in synchronization and replay of the last missed call
+template <typename... Args>
+class synchronized_stored_callback final : public synchronized_callback<Args...> {
+public:
+	template <typename... CArgs>
+	synchronized_stored_callback(CArgs &&...cargs)
+	    : synchronized_callback<Args...>(std::forward<CArgs>(cargs)...) {}
+	~synchronized_stored_callback() {}
+
+private:
+	void set(std::function<void(Args...)> func) {
+		synchronized_callback<Args...>::set(func);
+		if (func && stored) {
+			std::apply(func, std::move(*stored));
+			stored.reset();
+		}
+	}
+
+	bool call(Args... args) const {
+		if (!synchronized_callback<Args...>::call(args...))
+			stored.emplace(std::move(args)...);
+
+		return true;
+	}
+
+	mutable std::optional<std::tuple<Args...>> stored;
+};
+
 // pimpl base class
 template <typename T> using impl_ptr = std::shared_ptr<T>;
 template <typename T> class CheshireCat {

src/candidate.cpp

@@ -17,7 +17,8 @@
  */
 
 #include "candidate.hpp"
-#include "globals.hpp"
+
+#include "impl/internals.hpp"
 
 #include <algorithm>
 #include <array>
@@ -92,7 +93,7 @@ void Candidate::parse(string candidate) {
 	                                        {"so", TransportType::TcpSo}};
 
 	const std::array prefixes{"a=", "candidate:"};
-	for (const string &prefix : prefixes)
+	for (string prefix : prefixes)
 		if (match_prefix(candidate, prefix))
 			candidate.erase(0, prefix.size());
 

src/capi.cpp

@@ -17,10 +17,9 @@
  */
 
 #include "rtc.h"
-
 #include "rtc.hpp"
 
-#include "plog/Formatters/FuncMessageFormatter.h"
+#include "impl/internals.hpp"
 
 #include <chrono>
 #include <exception>
@@ -29,11 +28,6 @@
 #include <unordered_map>
 #include <utility>
 
-#ifdef _WIN32
-#include <codecvt>
-#include <locale>
-#endif
-
 using namespace rtc;
 using std::chrono::milliseconds;
 
@@ -116,14 +110,14 @@ int emplaceTrack(shared_ptr<Track> ptr) {
 void erasePeerConnection(int pc) {
 	std::lock_guard lock(mutex);
 	if (peerConnectionMap.erase(pc) == 0)
-		throw std::invalid_argument("PeerConnection ID does not exist");
+		throw std::invalid_argument("Peer Connection ID does not exist");
 	userPointerMap.erase(pc);
 }
 
 void eraseDataChannel(int dc) {
 	std::lock_guard lock(mutex);
 	if (dataChannelMap.erase(dc) == 0)
-		throw std::invalid_argument("DataChannel ID does not exist");
+		throw std::invalid_argument("Data Channel ID does not exist");
 	userPointerMap.erase(dc);
 }
 
@@ -146,7 +140,7 @@ shared_ptr<RtcpSrReporter> getRtcpSrReporter(int id) {
 	if (auto it = rtcpSrReporterMap.find(id); it != rtcpSrReporterMap.end()) {
 		return it->second;
 	} else {
-		throw std::invalid_argument("RtcpSRReporter ID does not exist");
+		throw std::invalid_argument("RTCP SR reporter ID does not exist");
 	}
 }
 
@@ -160,7 +154,7 @@ shared_ptr<MediaChainableHandler> getMediaChainableHandler(int id) {
 	if (auto it = rtcpChainableHandlerMap.find(id); it != rtcpChainableHandlerMap.end()) {
 		return it->second;
 	} else {
-		throw std::invalid_argument("RtcpChainableHandler ID does not exist");
+		throw std::invalid_argument("RTCP chainable handler ID does not exist");
 	}
 }
 
@@ -169,44 +163,31 @@ void emplaceMediaChainableHandler(shared_ptr<MediaChainableHandler> ptr, int tr)
 	rtcpChainableHandlerMap.emplace(std::make_pair(tr, ptr));
 }
 
-shared_ptr<RtpPacketizationConfig> getRTPConfig(int id) {
+shared_ptr<RtpPacketizationConfig> getRtpConfig(int id) {
 	std::lock_guard lock(mutex);
 	if (auto it = rtpConfigMap.find(id); it != rtpConfigMap.end()) {
 		return it->second;
 	} else {
-		throw std::invalid_argument("RTPConfiguration ID does not exist");
+		throw std::invalid_argument("RTP configuration ID does not exist");
 	}
 }
 
-void emplaceRTPConfig(shared_ptr<RtpPacketizationConfig> ptr, int tr) {
+void emplaceRtpConfig(shared_ptr<RtpPacketizationConfig> ptr, int tr) {
 	std::lock_guard lock(mutex);
 	rtpConfigMap.emplace(std::make_pair(tr, ptr));
 }
 
-Description::Direction rtcDirectionToDirection(rtcDirection direction) {
-	switch (direction) {
-	case RTC_DIRECTION_SENDONLY:
-		return Description::Direction::SendOnly;
-	case RTC_DIRECTION_RECVONLY:
-		return Description::Direction::RecvOnly;
-	case RTC_DIRECTION_SENDRECV:
-		return Description::Direction::SendRecv;
-	case RTC_DIRECTION_INACTIVE:
-		return Description::Direction::Inactive;
-	default:
-		return Description::Direction::Unknown;
-	}
-}
-
 shared_ptr<RtpPacketizationConfig>
-getNewRtpPacketizationConfig(uint32_t ssrc, const char *cname, uint8_t payloadType,
+createRtpPacketizationConfig(const rtcPacketizationHandlerInit *init) {
-                             uint32_t clockRate, uint16_t sequenceNumber, uint32_t timestamp) {
+	if (!init)
-	if (!cname) {
+		throw std::invalid_argument("Unexpected null pointer for packetization handler init");
-		throw std::invalid_argument("Unexpected null pointer for cname");
-	}
 
-	return std::make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, clockRate,
+	if (!init->cname)
-	                                                sequenceNumber, timestamp);
+		throw std::invalid_argument("Unexpected null pointer for cname");
+
+	return std::make_shared<RtpPacketizationConfig>(init->ssrc, init->cname, init->payloadType,
+	                                                init->clockRate, init->sequenceNumber,
+	                                                init->timestamp);
 }
 
 #endif // RTC_ENABLE_MEDIA
@@ -287,58 +268,16 @@ int copyAndReturn(binary b, char *buffer, int size) {
 	return int(b.size());
 }
 
-class plogAppender : public plog::IAppender {
-public:
-	plogAppender(rtcLogCallbackFunc cb = nullptr) { setCallback(cb); }
-
-	plogAppender(plogAppender &&appender) : callback(nullptr) {
-		std::lock_guard lock(appender.callbackMutex);
-		std::swap(appender.callback, callback);
-	}
-
-	void setCallback(rtcLogCallbackFunc cb) {
-		std::lock_guard lock(callbackMutex);
-		callback = cb;
-	}
-
-	void write(const plog::Record &record) override {
-		plog::Severity severity = record.getSeverity();
-		auto formatted = plog::FuncMessageFormatter::format(record);
-		formatted.pop_back(); // remove newline
-#ifdef _WIN32
-		using convert_type = std::codecvt_utf8<wchar_t>;
-		std::wstring_convert<convert_type, wchar_t> converter;
-		std::string str = converter.to_bytes(formatted);
-#else
-		std::string str = formatted;
-#endif
-		std::lock_guard lock(callbackMutex);
-		if (callback)
-			callback(static_cast<rtcLogLevel>(record.getSeverity()), str.c_str());
-		else
-			std::cout << plog::severityToString(severity) << " " << str << std::endl;
-	}
-
-private:
-	rtcLogCallbackFunc callback;
-	std::mutex callbackMutex;
-};
-
 } // namespace
 
 void rtcInitLogger(rtcLogLevel level, rtcLogCallbackFunc cb) {
-	static optional<plogAppender> appender;
+	LogCallback callback = nullptr;
-	const auto severity = static_cast<plog::Severity>(level);
+	if (cb)
-	std::lock_guard lock(mutex);
+		callback = [cb](LogLevel level, string message) {
-	if (appender) {
+			cb(static_cast<rtcLogLevel>(level), message.c_str());
-		appender->setCallback(cb);
+		};
-		InitLogger(severity, nullptr); // change the severity
+
-	} else if (cb) {
+	InitLogger(static_cast<LogLevel>(level), callback);
-		appender.emplace(plogAppender(cb));
-		InitLogger(severity, &appender.value());
-	} else {
-		InitLogger(severity, nullptr); // log to stdout
-	}
 }
 
 void rtcSetUserPointer(int i, void *ptr) { setUserPointer(i, ptr); }
@@ -438,69 +377,65 @@ int rtcDeleteDataChannel(int dc) {
 	});
 }
 
-#if RTC_ENABLE_MEDIA
+int rtcAddTrack(int pc, const char *mediaDescriptionSdp) {
+	return wrap([&] {
+		if (!mediaDescriptionSdp)
+			throw std::invalid_argument("Unexpected null pointer for track media description");
 
-void setSSRC(Description::Media *description, uint32_t ssrc, const char *_name, const char *_msid,
+		auto peerConnection = getPeerConnection(pc);
-             const char *_trackID) {
+		Description::Media media{string(mediaDescriptionSdp)};
+		int tr = emplaceTrack(peerConnection->addTrack(std::move(media)));
+		if (auto ptr = getUserPointer(pc))
+			rtcSetUserPointer(tr, *ptr);
 
-	optional<string> name = nullopt;
+		return tr;
-	if (_name) {
+	});
-		name = string(_name);
-	}
-
-	optional<string> msid = nullopt;
-	if (_msid) {
-		msid = string(_msid);
-	}
-
-	optional<string> trackID = nullopt;
-	if (_trackID) {
-		trackID = string(_trackID);
-	}
-
-	description->addSSRC(ssrc, name, msid, trackID);
 }
 
-int rtcAddTrackEx(int pc, rtcCodec codec, int payloadType, uint32_t ssrc, const char *_mid,
+int rtcAddTrackEx(int pc, const rtcTrackInit *init) {
-                  rtcDirection _direction, const char *_name, const char *_msid,
-                  const char *_trackID) {
 	return wrap([&] {
 		auto peerConnection = getPeerConnection(pc);
 
-		auto direction = rtcDirectionToDirection(_direction);
+		if (!init)
+			throw std::invalid_argument("Unexpected null pointer for track init");
 
-		string mid = "video";
+		auto direction = static_cast<Description::Direction>(init->direction);
-		switch (codec) {
-		case RTC_CODEC_H264:
-		case RTC_CODEC_VP8:
-		case RTC_CODEC_VP9:
-			mid = "video";
-			break;
-		case RTC_CODEC_OPUS:
-			mid = "audio";
-			break;
-		}
 
-		if (_mid) {
+		string mid;
-			mid = string(_mid);
+		if (init->mid) {
+			mid = string(init->mid);
+		} else {
+			switch (init->codec) {
+			case RTC_CODEC_H264:
+			case RTC_CODEC_VP8:
+			case RTC_CODEC_VP9:
+				mid = "video";
+				break;
+			case RTC_CODEC_OPUS:
+				mid = "audio";
+				break;
+			default:
+				mid = "video";
+				break;
+			}
 		}
 
 		optional<Description::Media> optDescription = nullopt;
 
-		switch (codec) {
+		switch (init->codec) {
 		case RTC_CODEC_H264:
 		case RTC_CODEC_VP8:
 		case RTC_CODEC_VP9: {
 			auto desc = Description::Video(mid, direction);
-			switch (codec) {
+			switch (init->codec) {
 			case RTC_CODEC_H264:
-				desc.addH264Codec(payloadType);
+				desc.addH264Codec(init->payloadType);
 				break;
 			case RTC_CODEC_VP8:
-				desc.addVP8Codec(payloadType);
+				desc.addVP8Codec(init->payloadType);
 				break;
 			case RTC_CODEC_VP9:
-				desc.addVP8Codec(payloadType);
+				desc.addVP8Codec(init->payloadType);
 				break;
 			default:
 				break;
@@ -510,9 +445,9 @@ int rtcAddTrackEx(int pc, rtcCodec codec, int payloadType, uint32_t ssrc, const
 		}
 		case RTC_CODEC_OPUS: {
 			auto desc = Description::Audio(mid, direction);
-			switch (codec) {
+			switch (init->codec) {
 			case RTC_CODEC_OPUS:
-				desc.addOpusCodec(payloadType);
+				desc.addOpusCodec(init->payloadType);
 				break;
 			default:
 				break;
@@ -524,168 +459,16 @@ int rtcAddTrackEx(int pc, rtcCodec codec, int payloadType, uint32_t ssrc, const
 			break;
 		}
 
-		if (!optDescription.has_value()) {
+		if (!optDescription)
 			throw std::invalid_argument("Unexpected codec");
-		} else {
-			auto description = optDescription.value();
-			setSSRC(&description, ssrc, _name, _msid, _trackID);
 
-			int tr = emplaceTrack(peerConnection->addTrack(std::move(description)));
+		auto desc = std::move(*optDescription);
-			if (auto ptr = getUserPointer(pc)) {
+		desc.addSSRC(init->ssrc, init->name ? std::make_optional(string(init->name)) : nullopt,
-				rtcSetUserPointer(tr, *ptr);
+		             init->msid ? std::make_optional(string(init->msid)) : nullopt,
-			}
+		             init->trackId ? std::make_optional(string(init->trackId)) : nullopt);
-			return tr;
-		}
-	});
-}
 
-int rtcSetH264PacketizationHandler(int tr, uint32_t ssrc, const char *cname, uint8_t payloadType,
+		int tr = emplaceTrack(peerConnection->addTrack(std::move(desc)));
-                                   uint32_t clockRate, uint16_t maxFragmentSize,
-                                   uint16_t sequenceNumber, uint32_t timestamp) {
-	return wrap([&] {
-		auto track = getTrack(tr);
-		// create RTP configuration
-		auto rtpConfig = getNewRtpPacketizationConfig(ssrc, cname, payloadType, clockRate,
-		                                              sequenceNumber, timestamp);
-		// create packetizer
-		auto packetizer = std::make_shared<H264RtpPacketizer>(rtpConfig, maxFragmentSize);
-		// create H264 handler
-		auto h264Handler = std::make_shared<H264PacketizationHandler>(packetizer);
-		emplaceMediaChainableHandler(h264Handler, tr);
-		emplaceRTPConfig(rtpConfig, tr);
-		// set handler
-		track->setMediaHandler(h264Handler);
-		return RTC_ERR_SUCCESS;
-	});
-}
 
-int rtcSetOpusPacketizationHandler(int tr, uint32_t ssrc, const char *cname, uint8_t payloadType,
-                                   uint32_t clockRate, uint16_t sequenceNumber,
-                                   uint32_t timestamp) {
-	return wrap([&] {
-		auto track = getTrack(tr);
-		// create RTP configuration
-		auto rtpConfig = getNewRtpPacketizationConfig(ssrc, cname, payloadType, clockRate,
-		                                              sequenceNumber, timestamp);
-		// create packetizer
-		auto packetizer = std::make_shared<OpusRtpPacketizer>(rtpConfig);
-		// create Opus handler
-		auto opusHandler = std::make_shared<OpusPacketizationHandler>(packetizer);
-		emplaceMediaChainableHandler(opusHandler, tr);
-		emplaceRTPConfig(rtpConfig, tr);
-		// set handler
-		track->setMediaHandler(opusHandler);
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcChainRtcpSrReporter(int tr) {
-	return wrap([tr] {
-		auto config = getRTPConfig(tr);
-		auto reporter = std::make_shared<RtcpSrReporter>(config);
-		emplaceRtcpSrReporter(reporter, tr);
-		auto chainableHandler = getMediaChainableHandler(tr);
-		chainableHandler->addToChain(reporter);
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcChainRtcpNackResponder(int tr, unsigned maxStoredPacketsCount) {
-	return wrap([tr, maxStoredPacketsCount] {
-		auto responder = std::make_shared<RtcpNackResponder>(maxStoredPacketsCount);
-		auto chainableHandler = getMediaChainableHandler(tr);
-		chainableHandler->addToChain(responder);
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcSetRtpConfigurationStartTime(int id, double startTime_s, bool timeIntervalSince1970,
-                                    uint32_t timestamp) {
-	return wrap([&] {
-		auto config = getRTPConfig(id);
-		auto epoch = RtpPacketizationConfig::EpochStart::T1900;
-		if (timeIntervalSince1970) {
-			epoch = RtpPacketizationConfig::EpochStart::T1970;
-		}
-		config->setStartTime(startTime_s, epoch, timestamp);
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcStartRtcpSenderReporterRecording(int id) {
-	return wrap([id] {
-		auto sender = getRtcpSrReporter(id);
-		sender->startRecording();
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcTransformSecondsToTimestamp(int id, double seconds, uint32_t *timestamp) {
-	return wrap([&] {
-		auto config = getRTPConfig(id);
-		*timestamp = config->secondsToTimestamp(seconds);
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcTransformTimestampToSeconds(int id, uint32_t timestamp, double *seconds) {
-	return wrap([&] {
-		auto config = getRTPConfig(id);
-		*seconds = config->timestampToSeconds(timestamp);
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcGetCurrentTrackTimestamp(int id, uint32_t *timestamp) {
-	return wrap([&] {
-		auto config = getRTPConfig(id);
-		*timestamp = config->timestamp;
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcGetTrackStartTimestamp(int id, uint32_t *timestamp) {
-	return wrap([&] {
-		auto config = getRTPConfig(id);
-		*timestamp = config->startTimestamp;
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcSetTrackRTPTimestamp(int id, uint32_t timestamp) {
-	return wrap([&] {
-		auto config = getRTPConfig(id);
-		config->timestamp = timestamp;
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcGetPreviousTrackSenderReportTimestamp(int id, uint32_t *timestamp) {
-	return wrap([&] {
-		auto sender = getRtcpSrReporter(id);
-		*timestamp = sender->previousReportedTimestamp;
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-int rtcSetNeedsToSendRtcpSr(int id) {
-	return wrap([id] {
-		auto sender = getRtcpSrReporter(id);
-		sender->setNeedsToReport();
-		return RTC_ERR_SUCCESS;
-	});
-}
-
-#endif // RTC_ENABLE_MEDIA
-
-int rtcAddTrack(int pc, const char *mediaDescriptionSdp) {
-	return wrap([&] {
-		if (!mediaDescriptionSdp)
-			throw std::invalid_argument("Unexpected null pointer for track media description");
-
-		auto peerConnection = getPeerConnection(pc);
-		Description::Media media{string(mediaDescriptionSdp)};
-		int tr = emplaceTrack(peerConnection->addTrack(std::move(media)));
 		if (auto ptr = getUserPointer(pc))
 			rtcSetUserPointer(tr, *ptr);
 
@@ -715,7 +498,141 @@ int rtcGetTrackDescription(int tr, char *buffer, int size) {
 	});
 }
 
+#if RTC_ENABLE_MEDIA
+
+int rtcSetH264PacketizationHandler(int tr, const rtcPacketizationHandlerInit *init) {
+	return wrap([&] {
+		auto track = getTrack(tr);
+		// create RTP configuration
+		auto rtpConfig = createRtpPacketizationConfig(init);
+		// create packetizer
+		auto maxFragmentSize = init && init->maxFragmentSize ? init->maxFragmentSize
+		                                                     : RTC_DEFAULT_MAXIMUM_FRAGMENT_SIZE;
+		auto packetizer = std::make_shared<H264RtpPacketizer>(rtpConfig, maxFragmentSize);
+		// create H264 handler
+		auto h264Handler = std::make_shared<H264PacketizationHandler>(packetizer);
+		emplaceMediaChainableHandler(h264Handler, tr);
+		emplaceRtpConfig(rtpConfig, tr);
+		// set handler
+		track->setMediaHandler(h264Handler);
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcSetOpusPacketizationHandler(int tr, const rtcPacketizationHandlerInit *init) {
+	return wrap([&] {
+		auto track = getTrack(tr);
+		// create RTP configuration
+		auto rtpConfig = createRtpPacketizationConfig(init);
+		// create packetizer
+		auto packetizer = std::make_shared<OpusRtpPacketizer>(rtpConfig);
+		// create Opus handler
+		auto opusHandler = std::make_shared<OpusPacketizationHandler>(packetizer);
+		emplaceMediaChainableHandler(opusHandler, tr);
+		emplaceRtpConfig(rtpConfig, tr);
+		// set handler
+		track->setMediaHandler(opusHandler);
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcChainRtcpSrReporter(int tr) {
+	return wrap([tr] {
+		auto config = getRtpConfig(tr);
+		auto reporter = std::make_shared<RtcpSrReporter>(config);
+		emplaceRtcpSrReporter(reporter, tr);
+		auto chainableHandler = getMediaChainableHandler(tr);
+		chainableHandler->addToChain(reporter);
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcChainRtcpNackResponder(int tr, unsigned int maxStoredPacketsCount) {
+	return wrap([tr, maxStoredPacketsCount] {
+		auto responder = std::make_shared<RtcpNackResponder>(maxStoredPacketsCount);
+		auto chainableHandler = getMediaChainableHandler(tr);
+		chainableHandler->addToChain(responder);
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcSetRtpConfigurationStartTime(int id, const rtcStartTime *startTime) {
+	return wrap([&] {
+		auto config = getRtpConfig(id);
+		auto epoch = startTime->since1970 ? RtpPacketizationConfig::EpochStart::T1970
+		                                  : RtpPacketizationConfig::EpochStart::T1900;
+		config->setStartTime(startTime->seconds, epoch, startTime->timestamp);
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcStartRtcpSenderReporterRecording(int id) {
+	return wrap([id] {
+		auto sender = getRtcpSrReporter(id);
+		sender->startRecording();
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcTransformSecondsToTimestamp(int id, double seconds, uint32_t *timestamp) {
+	return wrap([&] {
+		auto config = getRtpConfig(id);
+		*timestamp = config->secondsToTimestamp(seconds);
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcTransformTimestampToSeconds(int id, uint32_t timestamp, double *seconds) {
+	return wrap([&] {
+		auto config = getRtpConfig(id);
+		*seconds = config->timestampToSeconds(timestamp);
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcGetCurrentTrackTimestamp(int id, uint32_t *timestamp) {
+	return wrap([&] {
+		auto config = getRtpConfig(id);
+		*timestamp = config->timestamp;
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcGetTrackStartTimestamp(int id, uint32_t *timestamp) {
+	return wrap([&] {
+		auto config = getRtpConfig(id);
+		*timestamp = config->startTimestamp;
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcSetTrackRtpTimestamp(int id, uint32_t timestamp) {
+	return wrap([&] {
+		auto config = getRtpConfig(id);
+		config->timestamp = timestamp;
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcGetPreviousTrackSenderReportTimestamp(int id, uint32_t *timestamp) {
+	return wrap([&] {
+		auto sender = getRtcpSrReporter(id);
+		*timestamp = sender->previousReportedTimestamp;
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+int rtcSetNeedsToSendRtcpSr(int id) {
+	return wrap([id] {
+		auto sender = getRtcpSrReporter(id);
+		sender->setNeedsToReport();
+		return RTC_ERR_SUCCESS;
+	});
+}
+
+#endif // RTC_ENABLE_MEDIA
 #if RTC_ENABLE_WEBSOCKET
+
 int rtcCreateWebSocket(const char *url) {
 	return wrap([&] {
 		auto ws = std::make_shared<WebSocket>();
@@ -748,6 +665,7 @@ int rtcDeleteWebsocket(int ws) {
 		return RTC_ERR_SUCCESS;
 	});
 }
+
 #endif
 
 int rtcSetLocalDescriptionCallback(int pc, rtcDescriptionCallbackFunc cb) {
@@ -1189,4 +1107,37 @@ int rtcReceiveMessage(int id, char *buffer, int *size) {
 }
 
 void rtcPreload() { rtc::Preload(); }
+
 void rtcCleanup() { rtc::Cleanup(); }
+
+int rtcSetSctpSettings(const rtcSctpSettings *settings) {
+	return wrap([&] {
+		SctpSettings s = {};
+
+		if (settings->recvBufferSize > 0)
+			s.recvBufferSize = size_t(settings->recvBufferSize);
+
+		if (settings->sendBufferSize > 0)
+			s.sendBufferSize = size_t(settings->sendBufferSize);
+
+		if (settings->maxChunksOnQueue > 0)
+			s.maxChunksOnQueue = size_t(settings->maxChunksOnQueue);
+
+		if (settings->initialCongestionWindow > 0)
+			s.initialCongestionWindow = size_t(settings->initialCongestionWindow);
+
+		if (settings->maxBurst > 0)
+			s.maxBurst = size_t(settings->maxBurst);
+		else if (settings->maxBurst < 0)
+			s.maxBurst = size_t(0); // setting to 0 disables, not setting chooses optimized default
+
+		if (settings->congestionControlModule >= 0)
+			s.congestionControlModule = unsigned(settings->congestionControlModule);
+
+		if (settings->delayedSackTimeMs > 0)
+			s.delayedSackTime = std::chrono::milliseconds(settings->delayedSackTimeMs);
+
+		SetSctpSettings(std::move(s));
+		return RTC_ERR_SUCCESS;
+	});
+}

src/channel.cpp

@@ -17,8 +17,8 @@
  */
 
 #include "channel.hpp"
-#include "globals.hpp"
 
+#include "impl/internals.hpp"
 #include "impl/channel.hpp"
 
 namespace rtc {
@@ -43,10 +43,7 @@ void Channel::onError(std::function<void(string error)> callback) {
 
 void Channel::onMessage(std::function<void(message_variant data)> callback) {
 	impl()->messageCallback = callback;
-
+	impl()->flushPendingMessages();
-	// Pass pending messages
-	while (auto message = receive())
-		impl()->messageCallback(*message);
 }
 
 void Channel::onMessage(std::function<void(binary data)> binaryCallback,

src/datachannel.cpp

@@ -17,11 +17,11 @@
  */
 
 #include "datachannel.hpp"
-#include "globals.hpp"
 #include "common.hpp"
 #include "peerconnection.hpp"
 
 #include "impl/datachannel.hpp"
+#include "impl/internals.hpp"
 #include "impl/peerconnection.hpp"
 
 #ifdef _WIN32

src/description.cpp

@@ -19,6 +19,8 @@
 
 #include "description.hpp"
 
+#include "impl/internals.hpp"
+
 #include <algorithm>
 #include <array>
 #include <cctype>
@@ -978,18 +980,17 @@ std::ostream &operator<<(std::ostream &out, rtc::Description::Type type) {
 
 std::ostream &operator<<(std::ostream &out, rtc::Description::Role role) {
 	using Role = rtc::Description::Role;
-	const char *str;
 	// Used for SDP generation, do not change
 	switch (role) {
 	case Role::Active:
-		str = "active";
+		out << "active";
 		break;
 	case Role::Passive:
-		str = "passive";
+		out << "passive";
 		break;
 	default:
-		str = "actpass";
+		out << "actpass";
 		break;
 	}
-	return out << str;
+	return out;
 }

src/global.cpp

@@ -0,0 +1,123 @@
+/**
+ * Copyright (c) 2020-2021 Paul-Louis Ageneau
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "plog/Appenders/ColorConsoleAppender.h"
+#include "plog/Formatters/FuncMessageFormatter.h"
+#include "plog/Formatters/TxtFormatter.h"
+#include "plog/Init.h"
+#include "plog/Log.h"
+#include "plog/Logger.h"
+//
+#include "global.hpp"
+
+#include "impl/init.hpp"
+
+#include <mutex>
+
+#ifdef _WIN32
+#include <codecvt>
+#include <locale>
+#endif
+
+namespace rtc {
+
+struct LogAppender : public plog::IAppender {
+	synchronized_callback<LogLevel, string> callback;
+
+	void write(const plog::Record &record) override {
+		const auto severity = record.getSeverity();
+		auto formatted = plog::FuncMessageFormatter::format(record);
+		formatted.pop_back(); // remove newline
+
+#ifdef _WIN32
+		using convert_type = std::codecvt_utf8<wchar_t>;
+		std::wstring_convert<convert_type, wchar_t> converter;
+		std::string str = converter.to_bytes(formatted);
+#else
+		std::string str = formatted;
+#endif
+
+		if (!callback(static_cast<LogLevel>(severity), str))
+			std::cout << plog::severityToString(severity) << " " << str << std::endl;
+	}
+};
+
+void InitLogger(LogLevel level, LogCallback callback) {
+	static unique_ptr<LogAppender> appender;
+	const auto severity = static_cast<plog::Severity>(level);
+	if (appender) {
+		appender->callback = std::move(callback);
+		InitLogger(severity, nullptr); // change the severity
+	} else if (callback) {
+		appender = std::make_unique<LogAppender>();
+		appender->callback = std::move(callback);
+		InitLogger(severity, appender.get());
+	} else {
+		InitLogger(severity, nullptr); // log to cout
+	}
+}
+
+void InitLogger(plog::Severity severity, plog::IAppender *appender) {
+	static plog::ColorConsoleAppender<plog::TxtFormatter> consoleAppender;
+	static plog::Logger<0> *logger = nullptr;
+	static std::mutex mutex;
+	std::lock_guard lock(mutex);
+	if (!logger) {
+		logger = &plog::init(severity, appender ? appender : &consoleAppender);
+		PLOG_DEBUG << "Logger initialized";
+	} else {
+		logger->setMaxSeverity(severity);
+		if (appender)
+			logger->addAppender(appender);
+	}
+}
+
+void Preload() { Init::Preload(); }
+void Cleanup() { Init::Cleanup(); }
+
+void SetSctpSettings(SctpSettings s) { Init::SetSctpSettings(std::move(s)); }
+
+} // namespace rtc
+
+RTC_CPP_EXPORT std::ostream &operator<<(std::ostream &out, rtc::LogLevel level) {
+	switch (level) {
+	case rtc::LogLevel::Fatal:
+		out << "fatal";
+		break;
+	case rtc::LogLevel::Error:
+		out << "error";
+		break;
+	case rtc::LogLevel::Warning:
+		out << "warning";
+		break;
+	case rtc::LogLevel::Info:
+		out << "info";
+		break;
+	case rtc::LogLevel::Debug:
+		out << "debug";
+		break;
+	case rtc::LogLevel::Verbose:
+		out << "verbose";
+		break;
+	default:
+		out << "none";
+		break;
+	}
+	return out;
+}
+

src/h264packetizationhandler.cpp

@@ -1,19 +1,19 @@
-/*
+/**
- * libdatachannel client example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA

src/h264rtppacketizer.cpp

@@ -1,25 +1,35 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "h264rtppacketizer.hpp"
 
+#include "impl/internals.hpp"
+
+#include <cassert>
+
+#ifdef _WIN32
+#include <winsock2.h>
+#else
+#include <arpa/inet.h>
+#endif
+
 namespace rtc {
 
 typedef enum {

src/impl/channel.cpp

@@ -20,22 +20,21 @@
 
 namespace rtc::impl {
 
-void Channel::triggerOpen() { openCallback(); }
+void Channel::triggerOpen() {
+	mOpenTriggered = true;
+	openCallback();
+	flushPendingMessages();
+}
 
 void Channel::triggerClosed() { closedCallback(); }
 
-void Channel::triggerError(string error) { errorCallback(error); }
+void Channel::triggerError(string error) { errorCallback(std::move(error)); }
 
 void Channel::triggerAvailable(size_t count) {
 	if (count == 1)
 		availableCallback();
 
-	while (messageCallback && count--) {
+	flushPendingMessages();
-		auto message = receive();
-		if (!message)
-			break;
-		messageCallback(*message);
-	}
 }
 
 void Channel::triggerBufferedAmount(size_t amount) {
@@ -45,13 +44,32 @@ void Channel::triggerBufferedAmount(size_t amount) {
 		bufferedAmountLowCallback();
 }
 
+void Channel::flushPendingMessages() {
+	if (!mOpenTriggered)
+		return;
+
+	while (messageCallback) {
+		auto next = receive();
+		if (!next)
+			break;
+
+		messageCallback(*next);
+	}
+}
+
+void Channel::resetOpenCallback() {
+	mOpenTriggered = false;
+	openCallback = nullptr;
+}
+
 void Channel::resetCallbacks() {
+	mOpenTriggered = false;
 	openCallback = nullptr;
 	closedCallback = nullptr;
 	errorCallback = nullptr;
-	messageCallback = nullptr;
 	availableCallback = nullptr;
 	bufferedAmountLowCallback = nullptr;
+	messageCallback = nullptr;
 }
 
 } // namespace rtc::impl

src/impl/channel.hpp

@@ -38,17 +38,23 @@ struct Channel {
 	virtual void triggerAvailable(size_t count);
 	virtual void triggerBufferedAmount(size_t amount);
 
-	virtual void resetCallbacks();
+	void flushPendingMessages();
+	void resetOpenCallback();
+	void resetCallbacks();
+
+	synchronized_stored_callback<> openCallback;
+	synchronized_stored_callback<> closedCallback;
+	synchronized_stored_callback<string> errorCallback;
+	synchronized_stored_callback<> availableCallback;
+	synchronized_stored_callback<> bufferedAmountLowCallback;
 
-	synchronized_callback<> openCallback;
-	synchronized_callback<> closedCallback;
-	synchronized_callback<string> errorCallback;
 	synchronized_callback<message_variant> messageCallback;
-	synchronized_callback<> availableCallback;
-	synchronized_callback<> bufferedAmountLowCallback;
 
 	std::atomic<size_t> bufferedAmount = 0;
 	std::atomic<size_t> bufferedAmountLowThreshold = 0;
+
+private:
+	std::atomic<bool> mOpenTriggered = false;
 };
 
 } // namespace rtc::impl

src/impl/datachannel.cpp

@@ -18,7 +18,7 @@
 
 #include "datachannel.hpp"
 #include "common.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 #include "logcounter.hpp"
 #include "peerconnection.hpp"
 #include "sctptransport.hpp"

src/impl/datachannel.hpp

@@ -36,7 +36,7 @@ struct PeerConnection;
 struct DataChannel : Channel, std::enable_shared_from_this<DataChannel> {
 	DataChannel(weak_ptr<PeerConnection> pc, uint16_t stream, string label, string protocol,
 	            Reliability reliability);
-	~DataChannel();
+	virtual ~DataChannel();
 
 	void close();
 	void remoteClose();

src/impl/dtlstransport.cpp

@@ -17,7 +17,7 @@
  */
 
 #include "dtlstransport.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 #include "icetransport.hpp"
 
 #include <chrono>

src/impl/icetransport.cpp

@@ -18,7 +18,7 @@
 
 #include "icetransport.hpp"
 #include "configuration.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 #include "transport.hpp"
 
 #include <iostream>

src/impl/init.cpp

@@ -17,7 +17,7 @@
  */
 
 #include "init.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 
 #include "impl/certificate.hpp"
 #include "impl/dtlstransport.hpp"
@@ -42,7 +42,6 @@ namespace rtc {
 namespace {
 
 void doInit() {
-	PLOG_DEBUG << "Global initialization";
 
 #ifdef _WIN32
 	WSADATA wsaData;
@@ -69,8 +68,6 @@ void doInit() {
 }
 
 void doCleanup() {
-	PLOG_DEBUG << "Global cleanup";
-
 	impl::ThreadPool::Instance().join();
 
 	impl::SctpTransport::Cleanup();
@@ -92,6 +89,7 @@ void doCleanup() {
 weak_ptr<void> Init::Weak;
 shared_ptr<void> *Init::Global = nullptr;
 bool Init::Initialized = false;
+SctpSettings Init::CurrentSctpSettings = {};
 std::recursive_mutex Init::Mutex;
 
 init_token Init::Token() {
@@ -118,10 +116,20 @@ void Init::Cleanup() {
 	Global = nullptr;
 }
 
+void Init::SetSctpSettings(SctpSettings s) {
+	auto token = Token();
+	std::unique_lock lock(Mutex);
+	impl::SctpTransport::SetSettings(s);
+	CurrentSctpSettings = std::move(s); // store for next init
+}
+
 Init::Init() {
 	// Mutex is locked by Token() here
-	if (!std::exchange(Initialized, true))
+	if (!std::exchange(Initialized, true)) {
+		PLOG_DEBUG << "Global initialization";
 		doInit();
+		impl::SctpTransport::SetSettings(CurrentSctpSettings);
+	}
 }
 
 Init::~Init() {
@@ -130,8 +138,11 @@ Init::~Init() {
 		std::unique_lock lock(Mutex);
 		if (Global)
 			return;
-		if (std::exchange(Initialized, false))
+
+		if (std::exchange(Initialized, false)) {
+			PLOG_DEBUG << "Global cleanup";
 			doCleanup();
+		}
 	});
 	t.detach();
 }

src/impl/init.hpp

@@ -16,22 +16,25 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef RTC_INIT_H
+#ifndef RTC_IMPL_INIT_H
-#define RTC_INIT_H
+#define RTC_IMPL_INIT_H
 
 #include "common.hpp"
+#include "global.hpp" // for SctpSettings
 
+#include <chrono>
 #include <mutex>
 
 namespace rtc {
 
 using init_token = shared_ptr<void>;
 
-class RTC_CPP_EXPORT Init {
+class Init {
 public:
 	static init_token Token();
 	static void Preload();
 	static void Cleanup();
+	static void SetSctpSettings(SctpSettings s);
 
 	~Init();
 
@@ -41,12 +44,10 @@ private:
 	static weak_ptr<void> Weak;
 	static shared_ptr<void> *Global;
 	static bool Initialized;
+	static SctpSettings CurrentSctpSettings;
 	static std::recursive_mutex Mutex;
 };
 
-inline void Preload() { Init::Preload(); }
-inline void Cleanup() { Init::Cleanup(); }
-
 } // namespace rtc
 
 #endif

src/impl/internals.hpp

@@ -16,17 +16,33 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef RTC_GLOBALS_H
+#ifndef RTC_IMPL_INTERNALS_H
-#define RTC_GLOBALS_H
+#define RTC_IMPL_INTERNALS_H
 
 #include "common.hpp"
 
+// Disable warnings before including plog
+#if defined(__GNUC__) || defined(__clang__)
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wall"
+#elif defined(_MSC_VER)
+#pragma warning(push, 0)
+#endif
+
+#include "plog/Log.h"
+
+#if defined(__GNUC__) || defined(__clang__)
+#pragma GCC diagnostic pop
+#elif defined(_MSC_VER)
+#pragma warning(pop)
+#endif
+
 namespace rtc {
 
 const size_t MAX_NUMERICNODE_LEN = 48; // Max IPv6 string representation length
 const size_t MAX_NUMERICSERV_LEN = 6;  // Max port string representation length
 
-const uint16_t DEFAULT_SCTP_PORT = 5000;                  // SCTP port to use by default
+const uint16_t DEFAULT_SCTP_PORT = 5000; // SCTP port to use by default
 
 const size_t DEFAULT_LOCAL_MAX_MESSAGE_SIZE = 256 * 1024; // Default local max message size
 const size_t DEFAULT_MAX_MESSAGE_SIZE = 65536; // Remote max message size if not specified in SDP

src/impl/peerconnection.cpp

@@ -21,7 +21,7 @@
 #include "certificate.hpp"
 #include "common.hpp"
 #include "dtlstransport.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 #include "icetransport.hpp"
 #include "logcounter.hpp"
 #include "peerconnection.hpp"
@@ -189,7 +189,7 @@ shared_ptr<DtlsTransport> PeerConnection::initDtlsTransport() {
 		auto certificate = mCertificate.get();
 		auto lower = std::atomic_load(&mIceTransport);
 		auto verifierCallback = weak_bind(&PeerConnection::checkFingerprint, this, _1);
-		auto stateChangeCallback =
+		auto dtlsStateChangeCallback =
 		    [this, weak_this = weak_from_this()](DtlsTransport::State transportState) {
 			    auto shared_this = weak_this.lock();
 			    if (!shared_this)
@@ -224,7 +224,7 @@ shared_ptr<DtlsTransport> PeerConnection::initDtlsTransport() {
 			// DTLS-SRTP
 			transport = std::make_shared<DtlsSrtpTransport>(
 			    lower, certificate, config.mtu, verifierCallback,
-			    weak_bind(&PeerConnection::forwardMedia, this, _1), stateChangeCallback);
+			    weak_bind(&PeerConnection::forwardMedia, this, _1), dtlsStateChangeCallback);
 #else
 			PLOG_WARNING << "Ignoring media support (not compiled with media support)";
 #endif
@@ -233,7 +233,7 @@ shared_ptr<DtlsTransport> PeerConnection::initDtlsTransport() {
 		if (!transport) {
 			// DTLS only
 			transport = std::make_shared<DtlsTransport>(lower, certificate, config.mtu,
-			                                            verifierCallback, stateChangeCallback);
+			                                            verifierCallback, dtlsStateChangeCallback);
 		}
 
 		std::atomic_store(&mDtlsTransport, transport);
@@ -440,8 +440,8 @@ void PeerConnection::forwardMedia(message_ptr message) {
 			offset += header->lengthInBytes();
 			if (header->payloadType() == 205 || header->payloadType() == 206) {
 				auto rtcpfb = reinterpret_cast<RTCP_FB_HEADER *>(header);
-				ssrcs.insert(rtcpfb->getPacketSenderSSRC());
+				ssrcs.insert(rtcpfb->packetSenderSSRC());
-				ssrcs.insert(rtcpfb->getMediaSourceSSRC());
+				ssrcs.insert(rtcpfb->mediaSourceSSRC());
 
 			} else if (header->payloadType() == 200 || header->payloadType() == 201) {
 				auto rtcpsr = reinterpret_cast<RTCP_SR *>(header);
@@ -738,7 +738,7 @@ void PeerConnection::validateRemoteDescription(const Description &description) {
 }
 
 void PeerConnection::processLocalDescription(Description description) {
-	const size_t localSctpPort = DEFAULT_SCTP_PORT;
+	const uint16_t localSctpPort = DEFAULT_SCTP_PORT;
 	const size_t localMaxMessageSize =
 	    config.maxMessageSize.value_or(DEFAULT_LOCAL_MAX_MESSAGE_SIZE);
 
@@ -973,33 +973,54 @@ string PeerConnection::localBundleMid() const {
 
 void PeerConnection::triggerDataChannel(weak_ptr<DataChannel> weakDataChannel) {
 	auto dataChannel = weakDataChannel.lock();
-	if (dataChannel)
+	if (dataChannel) {
+		dataChannel->resetOpenCallback(); // might be set internally
 		mPendingDataChannels.push(std::move(dataChannel));
+	}
+	triggerPendingDataChannels();
+}
 
+void PeerConnection::triggerTrack(weak_ptr<Track> weakTrack) {
+	auto track = weakTrack.lock();
+	if (track) {
+		track->resetOpenCallback(); // might be set internally
+		mPendingTracks.push(std::move(track));
+	}
+	triggerPendingTracks();
+}
+
+void PeerConnection::triggerPendingDataChannels() {
 	while (dataChannelCallback) {
 		auto next = mPendingDataChannels.tryPop();
 		if (!next)
 			break;
 
-		mProcessor->enqueue(dataChannelCallback.wrap(),
+		auto impl = std::move(*next);
-		                    std::make_shared<rtc::DataChannel>(std::move(*next)));
+		dataChannelCallback(std::make_shared<rtc::DataChannel>(impl));
+		impl->triggerOpen();
 	}
 }
 
-void PeerConnection::triggerTrack(weak_ptr<Track> weakTrack) {
+void PeerConnection::triggerPendingTracks() {
-	auto track = weakTrack.lock();
-	if (track)
-		mPendingTracks.push(std::move(track));
-
 	while (trackCallback) {
 		auto next = mPendingTracks.tryPop();
 		if (!next)
 			break;
 
-		mProcessor->enqueue(trackCallback.wrap(), std::make_shared<rtc::Track>(std::move(*next)));
+		auto impl = std::move(*next);
+		trackCallback(std::make_shared<rtc::Track>(impl));
+		impl->triggerOpen();
 	}
 }
 
+void PeerConnection::flushPendingDataChannels() {
+	mProcessor->enqueue(std::bind(&PeerConnection::triggerPendingDataChannels, this));
+}
+
+void PeerConnection::flushPendingTracks() {
+	mProcessor->enqueue(std::bind(&PeerConnection::triggerPendingTracks, this));
+}
+
 bool PeerConnection::changeState(State newState) {
 	State current;
 	do {

src/impl/peerconnection.hpp

@@ -84,8 +84,14 @@ struct PeerConnection : std::enable_shared_from_this<PeerConnection> {
 	void processRemoteCandidate(Candidate candidate);
 	string localBundleMid() const;
 
-	void triggerDataChannel(weak_ptr<DataChannel> weakDataChannel = {});
+	void triggerDataChannel(weak_ptr<DataChannel> weakDataChannel);
-	void triggerTrack(weak_ptr<Track> weakTrack = {});
+	void triggerTrack(weak_ptr<Track> weakTrack);
+
+	void triggerPendingDataChannels();
+	void triggerPendingTracks();
+
+	void flushPendingDataChannels();
+	void flushPendingTracks();
 
 	bool changeState(State newState);
 	bool changeGatheringState(GatheringState newState);

src/impl/sctptransport.cpp

@@ -18,12 +18,13 @@
 
 #include "sctptransport.hpp"
 #include "dtlstransport.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 #include "logcounter.hpp"
 
 #include <chrono>
 #include <exception>
 #include <iostream>
+#include <limits>
 #include <thread>
 #include <vector>
 
@@ -54,6 +55,26 @@
 using namespace std::chrono_literals;
 using namespace std::chrono;
 
+namespace {
+
+template <typename T> uint16_t to_uint16(T i) {
+	if (i >= 0 && static_cast<typename std::make_unsigned<T>::type>(i) <=
+	                  std::numeric_limits<uint16_t>::max())
+		return static_cast<uint16_t>(i);
+	else
+		throw std::invalid_argument("Integer out of range");
+}
+
+template <typename T> uint32_t to_uint32(T i) {
+	if (i >= 0 && static_cast<typename std::make_unsigned<T>::type>(i) <=
+	                  std::numeric_limits<uint32_t>::max())
+		return static_cast<uint32_t>(i);
+	else
+		throw std::invalid_argument("Integer out of range");
+}
+
+} // namespace
+
 namespace rtc::impl {
 
 static LogCounter COUNTER_UNKNOWN_PPID(plog::warning,
@@ -64,12 +85,35 @@ static LogCounter
 static LogCounter COUNTER_BAD_SCTP_STATUS(plog::warning,
                                           "Number of SCTP packets received with a bad status");
 
-std::unordered_set<SctpTransport *> SctpTransport::Instances;
+class SctpTransport::InstancesSet {
-std::shared_mutex SctpTransport::InstancesMutex;
+public:
+	void insert(SctpTransport *instance) {
+		std::unique_lock lock(mMutex);
+		mSet.insert(instance);
+	}
+
+	void erase(SctpTransport *instance) {
+		std::unique_lock lock(mMutex);
+		mSet.erase(instance);
+	}
+
+	using shared_lock = std::shared_lock<std::shared_mutex>;
+	optional<shared_lock> lock(SctpTransport *instance) {
+		shared_lock lock(mMutex);
+		return mSet.find(instance) != mSet.end() ? std::make_optional(std::move(lock)) : nullopt;
+	}
+
+private:
+	std::unordered_set<SctpTransport *> mSet;
+	std::shared_mutex mMutex;
+};
+
+SctpTransport::InstancesSet *SctpTransport::Instances = new InstancesSet;
 
 void SctpTransport::Init() {
 	usrsctp_init(0, &SctpTransport::WriteCallback, nullptr);
-	usrsctp_sysctl_set_sctp_ecn_enable(0);
+	usrsctp_sysctl_set_sctp_pr_enable(1);  // Enable Partial Reliability Extension (RFC 3758)
+	usrsctp_sysctl_set_sctp_ecn_enable(0); // Disable Explicit Congestion Notification
 	usrsctp_sysctl_set_sctp_init_rtx_max_default(5);
 	usrsctp_sysctl_set_sctp_path_rtx_max_default(5);
 	usrsctp_sysctl_set_sctp_assoc_rtx_max_default(5);              // single path
@@ -78,21 +122,31 @@ void SctpTransport::Init() {
 	usrsctp_sysctl_set_sctp_rto_initial_default(1 * 1000);         // ms
 	usrsctp_sysctl_set_sctp_init_rto_max_default(10 * 1000);       // ms
 	usrsctp_sysctl_set_sctp_heartbeat_interval_default(10 * 1000); // ms
+}
 
-	usrsctp_sysctl_set_sctp_max_chunks_on_queue(10 * 1024);
+void SctpTransport::SetSettings(const SctpSettings &s) {
+	// The send and receive window size of usrsctp is 256KiB, which is too small for realistic RTTs,
+	// therefore we increase it to 1MiB by default for better performance.
+	// See https://bugzilla.mozilla.org/show_bug.cgi?id=1051685
+	usrsctp_sysctl_set_sctp_recvspace(to_uint32(s.recvBufferSize.value_or(1024 * 1024)));
+	usrsctp_sysctl_set_sctp_sendspace(to_uint32(s.sendBufferSize.value_or(1024 * 1024)));
 
-	// Use default congestion control (RFC 4960)
+	// Increase maximum chunks number on queue to 10K by default
+	usrsctp_sysctl_set_sctp_max_chunks_on_queue(to_uint32(s.maxChunksOnQueue.value_or(10 * 1024)));
+
+	// Increase initial congestion window size to 10 MTUs (RFC 6928) by default
+	usrsctp_sysctl_set_sctp_initial_cwnd(to_uint32(s.initialCongestionWindow.value_or(10)));
+
+	// Set max burst to 10 MTUs by default (max burst is initially 0, meaning disabled)
+	usrsctp_sysctl_set_sctp_max_burst_default(to_uint32(s.maxBurst.value_or(10)));
+
+	// Use standard SCTP congestion control (RFC 4960) by default
 	// See https://github.com/paullouisageneau/libdatachannel/issues/354
-	usrsctp_sysctl_set_sctp_default_cc_module(0);
+	usrsctp_sysctl_set_sctp_default_cc_module(to_uint32(s.congestionControlModule.value_or(0)));
 
-	// Enable Partial Reliability Extension (RFC 3758)
+	// Reduce SACK delay to 20ms by default
-	usrsctp_sysctl_set_sctp_pr_enable(1);
+	usrsctp_sysctl_set_sctp_delayed_sack_time_default(
-
+	    to_uint32(s.delayedSackTime.value_or(20ms).count()));
-	// Increase the initial window size to 10 MTUs (RFC 6928)
-	usrsctp_sysctl_set_sctp_initial_cwnd(10);
-
-	// Reduce SACK delay from the default 200ms to 20ms
-	usrsctp_sysctl_set_sctp_delayed_sack_time_default(20); // ms
 }
 
 void SctpTransport::Cleanup() {
@@ -111,10 +165,7 @@ SctpTransport::SctpTransport(shared_ptr<Transport> lower, const Configuration &c
 	PLOG_DEBUG << "Initializing SCTP transport";
 
 	usrsctp_register_address(this);
-	{
+	Instances->insert(this);
-		std::unique_lock lock(InstancesMutex);
-		Instances.insert(this);
-	}
 
 	mSock = usrsctp_socket(AF_CONN, SOCK_STREAM, IPPROTO_SCTP, nullptr, nullptr, 0, nullptr);
 	if (!mSock)
@@ -195,7 +246,7 @@ SctpTransport::SctpTransport(shared_ptr<Transport> lower, const Configuration &c
 		// The MTU value provided specifies the space available for chunks in the
 		// packet, so we also subtract the SCTP header size.
 		size_t pmtu = config.mtu.value_or(DEFAULT_MTU) - 12 - 37 - 8 - 40; // SCTP/DTLS/UDP/IPv6
-		spp.spp_pathmtu = uint32_t(pmtu);
+		spp.spp_pathmtu = to_uint32(pmtu);
 		PLOG_VERBOSE << "Path MTU discovery disabled, SCTP MTU set to " << pmtu;
 	}
 
@@ -222,18 +273,28 @@ SctpTransport::SctpTransport(shared_ptr<Transport> lower, const Configuration &c
 		throw std::runtime_error("Could not disable SCTP fragmented interleave, errno=" +
 		                         std::to_string(errno));
 
-	// The default send and receive window size of usrsctp is 256KiB, which is too small for
+	int rcvBuf = 0;
-	// realistic RTTs, therefore we increase it to at least 1MiB for better performance.
+	socklen_t rcvBufLen = sizeof(rcvBuf);
-	// See https://bugzilla.mozilla.org/show_bug.cgi?id=1051685
+	if (usrsctp_getsockopt(mSock, SOL_SOCKET, SO_RCVBUF, &rcvBuf, &rcvBufLen))
-	const size_t minBufferSize = 1024 * 1024;
+		throw std::runtime_error("Could not get SCTP recv buffer size, errno=" +
+		                         std::to_string(errno));
+	int sndBuf = 0;
+	socklen_t sndBufLen = sizeof(sndBuf);
+	if (usrsctp_getsockopt(mSock, SOL_SOCKET, SO_SNDBUF, &sndBuf, &sndBufLen))
+		throw std::runtime_error("Could not get SCTP send buffer size, errno=" +
+		                         std::to_string(errno));
 
 	// Ensure the buffer is also large enough to accomodate the largest messages
 	const size_t maxMessageSize = config.maxMessageSize.value_or(DEFAULT_LOCAL_MAX_MESSAGE_SIZE);
-	const int bufferSize = int(std::max(minBufferSize, maxMessageSize * 2)); // usrsctp reads as int
+	const int minBuf = int(std::min(maxMessageSize, size_t(std::numeric_limits<int>::max())));
-	if (usrsctp_setsockopt(mSock, SOL_SOCKET, SO_RCVBUF, &bufferSize, sizeof(bufferSize)))
+	rcvBuf = std::max(rcvBuf, minBuf);
+	sndBuf = std::max(sndBuf, minBuf);
+
+	if (usrsctp_setsockopt(mSock, SOL_SOCKET, SO_RCVBUF, &rcvBuf, sizeof(rcvBuf)))
 		throw std::runtime_error("Could not set SCTP recv buffer size, errno=" +
 		                         std::to_string(errno));
-	if (usrsctp_setsockopt(mSock, SOL_SOCKET, SO_SNDBUF, &bufferSize, sizeof(bufferSize)))
+
+	if (usrsctp_setsockopt(mSock, SOL_SOCKET, SO_SNDBUF, &sndBuf, sizeof(sndBuf)))
 		throw std::runtime_error("Could not set SCTP send buffer size, errno=" +
 		                         std::to_string(errno));
 }
@@ -243,10 +304,7 @@ SctpTransport::~SctpTransport() {
 	close();
 
 	usrsctp_deregister_address(this);
-	{
+	Instances->erase(this);
-		std::unique_lock lock(InstancesMutex);
-		Instances.erase(this);
-	}
 }
 
 void SctpTransport::start() {
@@ -336,7 +394,7 @@ bool SctpTransport::send(message_ptr message) {
 		return true;
 
 	mSendQueue.push(message);
-	updateBufferedAmount(uint16_t(message->stream), long(message_size_func(message)));
+	updateBufferedAmount(to_uint16(message->stream), ptrdiff_t(message_size_func(message)));
 	return false;
 }
 
@@ -353,7 +411,7 @@ bool SctpTransport::flush() {
 }
 
 void SctpTransport::closeStream(unsigned int stream) {
-	send(make_message(0, Message::Reset, uint16_t(stream)));
+	send(make_message(0, Message::Reset, to_uint16(stream)));
 }
 
 void SctpTransport::incoming(message_ptr message) {
@@ -374,9 +432,6 @@ void SctpTransport::incoming(message_ptr message) {
 
 	PLOG_VERBOSE << "Incoming size=" << message->size();
 
-	// TODO: There seems to be a possible data race between usrsctp_sendv() and usrsctp_conninput()
-	// As a mitigation, lock the send mutex before calling usrsctp_conninput()
-	std::lock_guard lock(mSendMutex);
 	usrsctp_conninput(this, message->data(), message->size(), 0);
 }
 
@@ -458,7 +513,7 @@ bool SctpTransport::trySendQueue() {
 		if (!trySendMessage(message))
 			return false;
 		mSendQueue.pop();
-		updateBufferedAmount(uint16_t(message->stream), -long(message_size_func(message)));
+		updateBufferedAmount(to_uint16(message->stream), -ptrdiff_t(message_size_func(message)));
 	}
 	return true;
 }
@@ -511,12 +566,12 @@ bool SctpTransport::trySendMessage(message_ptr message) {
 	case Reliability::Type::Rexmit:
 		spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
 		spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX;
-		spa.sendv_prinfo.pr_value = uint32_t(std::get<int>(reliability.rexmit));
+		spa.sendv_prinfo.pr_value = to_uint32(std::get<int>(reliability.rexmit));
 		break;
 	case Reliability::Type::Timed:
 		spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
 		spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL;
-		spa.sendv_prinfo.pr_value = uint32_t(std::get<milliseconds>(reliability.rexmit).count());
+		spa.sendv_prinfo.pr_value = to_uint32(std::get<milliseconds>(reliability.rexmit).count());
 		break;
 	default:
 		spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_NONE;
@@ -548,10 +603,10 @@ bool SctpTransport::trySendMessage(message_ptr message) {
 	return true;
 }
 
-void SctpTransport::updateBufferedAmount(uint16_t streamId, long delta) {
+void SctpTransport::updateBufferedAmount(uint16_t streamId, ptrdiff_t delta) {
 	// Requires mSendMutex to be locked
 	auto it = mBufferedAmount.insert(std::make_pair(streamId, 0)).first;
-	size_t amount = size_t(std::max(long(it->second) + delta, long(0)));
+	size_t amount = size_t(std::max(ptrdiff_t(it->second) + delta, ptrdiff_t(0)));
 	if (amount == 0)
 		mBufferedAmount.erase(it);
 	else
@@ -809,11 +864,8 @@ optional<milliseconds> SctpTransport::rtt() {
 void SctpTransport::UpcallCallback(struct socket *, void *arg, int /* flags */) {
 	auto *transport = static_cast<SctpTransport *>(arg);
 
-	std::shared_lock lock(InstancesMutex);
+	if(auto lock = Instances->lock(transport))
-	if (Instances.find(transport) == Instances.end())
+		transport->handleUpcall();
-		return;
-
-	transport->handleUpcall();
 }
 
 int SctpTransport::WriteCallback(void *ptr, void *data, size_t len, uint8_t tos, uint8_t set_df) {
@@ -821,11 +873,10 @@ int SctpTransport::WriteCallback(void *ptr, void *data, size_t len, uint8_t tos,
 
 	// Workaround for sctplab/usrsctp#405: Send callback is invoked on already closed socket
 	// https://github.com/sctplab/usrsctp/issues/405
-	std::shared_lock lock(InstancesMutex);
+	if(auto lock = Instances->lock(transport))
-	if (Instances.find(transport) == Instances.end())
+		return transport->handleWrite(static_cast<byte *>(data), len, tos, set_df);
+	else
 		return -1;
-
-	return transport->handleWrite(static_cast<byte *>(data), len, tos, set_df);
 }
 
 } // namespace rtc::impl

src/impl/sctptransport.hpp

@@ -39,6 +39,7 @@ namespace rtc::impl {
 class SctpTransport final : public Transport {
 public:
 	static void Init();
+	static void SetSettings(const SctpSettings &s);
 	static void Cleanup();
 
 	using amount_callback = std::function<void(uint16_t streamId, size_t amount)>;
@@ -83,7 +84,7 @@ private:
 	void doFlush();
 	bool trySendQueue();
 	bool trySendMessage(message_ptr message);
-	void updateBufferedAmount(uint16_t streamId, long delta);
+	void updateBufferedAmount(uint16_t streamId, ptrdiff_t delta);
 	void triggerBufferedAmount(uint16_t streamId, size_t amount);
 	void sendReset(uint16_t streamId);
 
@@ -119,8 +120,8 @@ private:
 	static void UpcallCallback(struct socket *sock, void *arg, int flags);
 	static int WriteCallback(void *sctp_ptr, void *data, size_t len, uint8_t tos, uint8_t set_df);
 
-	static std::unordered_set<SctpTransport *> Instances;
+	class InstancesSet;
-	static std::shared_mutex InstancesMutex;
+	static InstancesSet *Instances;
 };
 
 } // namespace rtc::impl

src/impl/tcptransport.cpp

@@ -17,7 +17,7 @@
  */
 
 #include "tcptransport.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 
 #if RTC_ENABLE_WEBSOCKET
 

src/impl/threadpool.hpp

@@ -21,6 +21,7 @@
 
 #include "common.hpp"
 #include "init.hpp"
+#include "internals.hpp"
 
 #include <chrono>
 #include <condition_variable>

src/impl/tls.cpp

@@ -18,6 +18,8 @@
 
 #include "tls.hpp"
 
+#include "internals.hpp"
+
 #if USE_GNUTLS
 
 namespace rtc::gnutls {

src/impl/track.cpp

@@ -17,7 +17,7 @@
  */
 
 #include "track.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 #include "logcounter.hpp"
 #include "peerconnection.hpp"
 

src/impl/transport.hpp

@@ -20,6 +20,7 @@
 #define RTC_IMPL_TRANSPORT_H
 
 #include "common.hpp"
+#include "internals.hpp"
 #include "message.hpp"
 
 #include <atomic>

src/impl/websocket.cpp

@@ -19,7 +19,7 @@
 #if RTC_ENABLE_WEBSOCKET
 
 #include "websocket.hpp"
-#include "globals.hpp"
+#include "internals.hpp"
 #include "common.hpp"
 #include "threadpool.hpp"
 

src/log.cpp

@@ -1,47 +0,0 @@
-/**
- * Copyright (c) 2019-2020 Paul-Louis Ageneau
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * This library 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "log.hpp"
-
-#include "plog/Appenders/ColorConsoleAppender.h"
-#include "plog/Formatters/TxtFormatter.h"
-#include "plog/Init.h"
-#include "plog/Log.h"
-#include "plog/Logger.h"
-
-#include <mutex>
-
-namespace rtc {
-
-void InitLogger(LogLevel level) { InitLogger(static_cast<plog::Severity>(level)); }
-
-void InitLogger(plog::Severity severity, plog::IAppender *appender) {
-	static plog::ColorConsoleAppender<plog::TxtFormatter> consoleAppender;
-	static plog::Logger<0> *logger = nullptr;
-	static std::mutex mutex;
-	std::lock_guard lock(mutex);
-	if (!logger) {
-		logger = &plog::init(severity, appender ? appender : &consoleAppender);
-		PLOG_DEBUG << "Logger initialized";
-	} else {
-		logger->setMaxSeverity(severity);
-		if (appender)
-			logger->addAppender(appender);
-	}
-}
-} // namespace rtc

src/mediachainablehandler.cpp

@@ -1,24 +1,29 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "mediachainablehandler.hpp"
 
+#include "impl/internals.hpp"
+
+#include <cassert>
+
 namespace rtc {
 
 MediaChainableHandler::MediaChainableHandler(shared_ptr<MediaHandlerRootElement> root): MediaHandler(), root(root), leaf(root) { }

src/mediahandlerelement.cpp

@@ -1,24 +1,29 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "mediahandlerelement.hpp"
 
+#include "impl/internals.hpp"
+
+#include <cassert>
+
 namespace rtc {
 
 ChainedMessagesProduct make_chained_messages_product() {

src/mediahandlerrootelement.cpp

@@ -1,18 +1,19 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA

src/nalunit.cpp

@@ -1,25 +1,26 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "nalunit.hpp"
-#include "globals.hpp"
+
+#include "impl/internals.hpp"
 
 #include <cmath>
 

src/opuspacketizationhandler.cpp

@@ -1,19 +1,19 @@
-/*
+/**
- * libdatachannel client example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA

src/opusrtppacketizer.cpp

@@ -1,25 +1,27 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "opusrtppacketizer.hpp"
 
+#include <cassert>
+
 namespace rtc {
 
 OpusRtpPacketizer::OpusRtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig)

src/peerconnection.cpp

@@ -24,6 +24,7 @@
 #include "impl/certificate.hpp"
 #include "impl/dtlstransport.hpp"
 #include "impl/icetransport.hpp"
+#include "impl/internals.hpp"
 #include "impl/peerconnection.hpp"
 #include "impl/processor.hpp"
 #include "impl/sctptransport.hpp"
@@ -266,7 +267,7 @@ shared_ptr<DataChannel> PeerConnection::createDataChannel(string label, DataChan
 void PeerConnection::onDataChannel(
     std::function<void(shared_ptr<DataChannel> dataChannel)> callback) {
 	impl()->dataChannelCallback = callback;
-	impl()->triggerDataChannel(); // trigger pending DataChannels
+	impl()->flushPendingDataChannels();
 }
 
 std::shared_ptr<Track> PeerConnection::addTrack(Description::Media description) {
@@ -281,7 +282,7 @@ std::shared_ptr<Track> PeerConnection::addTrack(Description::Media description)
 
 void PeerConnection::onTrack(std::function<void(std::shared_ptr<Track>)> callback) {
 	impl()->trackCallback = callback;
-	impl()->triggerTrack(); // trigger pending tracks
+	impl()->flushPendingTracks();
 }
 
 void PeerConnection::onLocalDescription(std::function<void(Description description)> callback) {

src/rtcpnackresponder.cpp

@@ -1,24 +1,29 @@
 /**
- * libdatachannel streamer example
+ * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "rtcpnackresponder.hpp"
 
+#include "impl/internals.hpp"
+
+#include <cassert>
+
 namespace rtc {
 
 RtcpNackResponder::Storage::Element::Element(binary_ptr packet, uint16_t sequenceNumber, shared_ptr<Element> next)
@@ -80,7 +85,7 @@ ChainedIncomingControlProduct RtcpNackResponder::processIncomingControlMessage(m
 		if (nack->header.header.payloadType() != 205 || nack->header.header.reportCount() != 1) {
 			continue;
 		}
-		
+
 		auto fieldsCount = nack->getSeqNoCount();
 
 		std::vector<uint16_t> missingSequenceNumbers{};

src/rtcpreceivingsession.cpp

@@ -97,7 +97,7 @@ void RtcpReceivingSession::requestBitrate(unsigned int newBitrate) {
 }
 
 void RtcpReceivingSession::pushREMB(unsigned int bitrate) {
-	message_ptr msg = make_message(RTCP_REMB::sizeWithSSRCs(1), Message::Type::Control);
+	message_ptr msg = make_message(RTCP_REMB::SizeWithSSRCs(1), Message::Type::Control);
 	auto remb = reinterpret_cast<RTCP_REMB *>(msg->data());
 	remb->preparePacket(mSsrc, 1, bitrate);
 	remb->setSsrc(0, mSsrc);
@@ -106,7 +106,7 @@ void RtcpReceivingSession::pushREMB(unsigned int bitrate) {
 }
 
 void RtcpReceivingSession::pushRR(unsigned int lastSR_delay) {
-	auto msg = make_message(RTCP_RR::sizeWithReportBlocks(1), Message::Type::Control);
+	auto msg = make_message(RTCP_RR::SizeWithReportBlocks(1), Message::Type::Control);
 	auto rr = reinterpret_cast<RTCP_RR *>(msg->data());
 	rr->preparePacket(mSsrc, 1);
 	rr->getReportBlock(0)->preparePacket(mSsrc, 0, 0, uint16_t(mGreatestSeqNo), 0, 0, mSyncNTPTS,
@@ -132,7 +132,7 @@ bool RtcpReceivingSession::requestKeyframe() {
 }
 
 void RtcpReceivingSession::pushPLI() {
-	auto msg = make_message(RTCP_PLI::size(), Message::Type::Control);
+	auto msg = make_message(RTCP_PLI::Size(), Message::Type::Control);
 	auto *pli = reinterpret_cast<RTCP_PLI *>(msg->data());
 	pli->preparePacket(mSsrc);
 	send(msg);

src/rtcpsrreporter.cpp

@@ -1,24 +1,28 @@
 /**
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "rtcpsrreporter.hpp"
 
+#include <cmath>
+#include <cassert>
+
 namespace rtc {
 
 ChainedOutgoingProduct RtcpSrReporter::processOutgoingBinaryMessage(ChainedMessagesProduct messages, message_ptr control) {
@@ -60,8 +64,8 @@ uint64_t RtcpSrReporter::secondsToNTP(double seconds) {
 void RtcpSrReporter::setNeedsToReport() { needsToReport = true; }
 
 message_ptr RtcpSrReporter::getSenderReport(uint32_t timestamp) {
-	auto srSize = RTCP_SR::size(0);
+	auto srSize = RTCP_SR::Size(0);
-	auto msg = make_message(srSize + RTCP_SDES::size({{uint8_t(rtpConfig->cname.size())}}),
+	auto msg = make_message(srSize + RTCP_SDES::Size({{uint8_t(rtpConfig->cname.size())}}),
 							Message::Type::Control);
 	auto sr = reinterpret_cast<RTCP_SR *>(msg->data());
 	auto timestamp_s = rtpConfig->timestampToSeconds(timestamp);

src/rtp.cpp

@@ -0,0 +1,600 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "rtp.hpp"
+
+#include "impl/internals.hpp"
+
+#include <cmath>
+#include <cstring>
+
+#ifdef _WIN32
+#include <winsock2.h>
+#else
+#include <arpa/inet.h>
+#endif
+
+#ifndef htonll
+#define htonll(x)                                                                                  \
+	((uint64_t)(((uint64_t)htonl((uint32_t)(x))) << 32) | (uint64_t)htonl((uint32_t)((x) >> 32)))
+#endif
+#ifndef ntohll
+#define ntohll(x) htonll(x)
+#endif
+
+namespace rtc {
+
+uint8_t RTP::version() const { return _first >> 6; }
+bool RTP::padding() const { return (_first >> 5) & 0x01; }
+bool RTP::extension() const { return (_first >> 4) & 0x01; }
+uint8_t RTP::csrcCount() const { return _first & 0x0F; }
+uint8_t RTP::marker() const { return _payloadType & 0b10000000; }
+uint8_t RTP::payloadType() const { return _payloadType & 0b01111111; }
+uint16_t RTP::seqNumber() const { return ntohs(_seqNumber); }
+uint32_t RTP::timestamp() const { return ntohl(_timestamp); }
+uint32_t RTP::ssrc() const { return ntohl(_ssrc); }
+
+size_t RTP::getSize() const {
+	return reinterpret_cast<const char *>(&_csrc) - reinterpret_cast<const char *>(this) +
+	       sizeof(SSRC) * csrcCount();
+}
+
+const char *RTP::getBody() const {
+	return reinterpret_cast<const char *>(&_csrc) + sizeof(SSRC) * csrcCount();
+}
+
+char *RTP::getBody() { return reinterpret_cast<char *>(&_csrc) + sizeof(SSRC) * csrcCount(); }
+
+void RTP::preparePacket() { _first |= (1 << 7); }
+
+void RTP::setSeqNumber(uint16_t newSeqNo) { _seqNumber = htons(newSeqNo); }
+
+void RTP::setPayloadType(uint8_t newPayloadType) {
+	_payloadType = (_payloadType & 0b10000000u) | (0b01111111u & newPayloadType);
+}
+
+void RTP::setSsrc(uint32_t in_ssrc) { _ssrc = htonl(in_ssrc); }
+
+void RTP::setMarker(bool marker) { _payloadType = (_payloadType & 0x7F) | (marker << 7); };
+
+void RTP::setTimestamp(uint32_t i) { _timestamp = htonl(i); }
+
+void RTP::log() const {
+	PLOG_VERBOSE << "RTP V: " << (int)version() << " P: " << (padding() ? "P" : " ")
+	             << " X: " << (extension() ? "X" : " ") << " CC: " << (int)csrcCount()
+	             << " M: " << (marker() ? "M" : " ") << " PT: " << (int)payloadType()
+	             << " SEQNO: " << seqNumber() << " TS: " << timestamp();
+}
+
+SSRC RTCP_ReportBlock::getSSRC() const { return ntohl(_ssrc); }
+
+void RTCP_ReportBlock::preparePacket(SSRC in_ssrc, [[maybe_unused]] unsigned int packetsLost,
+                                     [[maybe_unused]] unsigned int totalPackets,
+                                     uint16_t highestSeqNo, uint16_t seqNoCycles, uint32_t jitter,
+                                     uint64_t lastSR_NTP, uint64_t lastSR_DELAY) {
+	setSeqNo(highestSeqNo, seqNoCycles);
+	setJitter(jitter);
+	setSSRC(in_ssrc);
+
+	// Middle 32 bits of NTP Timestamp
+	// _lastReport = lastSR_NTP >> 16u;
+	setNTPOfSR(uint64_t(lastSR_NTP));
+	setDelaySinceSR(uint32_t(lastSR_DELAY));
+
+	// The delay, expressed in units of 1/65536 seconds
+	// _delaySinceLastReport = lastSR_DELAY;
+}
+
+void RTCP_ReportBlock::setSSRC(SSRC in_ssrc) { _ssrc = htonl(in_ssrc); }
+
+void RTCP_ReportBlock::setPacketsLost([[maybe_unused]] unsigned int packetsLost,
+                                      [[maybe_unused]] unsigned int totalPackets) {
+	// TODO Implement loss percentages.
+	_fractionLostAndPacketsLost = 0;
+}
+
+unsigned int RTCP_ReportBlock::getLossPercentage() const {
+	// TODO Implement loss percentages.
+	return 0;
+}
+
+unsigned int RTCP_ReportBlock::getPacketLostCount() const {
+	// TODO Implement total packets lost.
+	return 0;
+}
+
+uint16_t RTCP_ReportBlock::seqNoCycles() const { return ntohs(_seqNoCycles); }
+
+uint16_t RTCP_ReportBlock::highestSeqNo() const { return ntohs(_highestSeqNo); }
+
+uint32_t RTCP_ReportBlock::jitter() const { return ntohl(_jitter); }
+
+uint32_t RTCP_ReportBlock::delaySinceSR() const { return ntohl(_delaySinceLastReport); }
+
+void RTCP_ReportBlock::setSeqNo(uint16_t highestSeqNo, uint16_t seqNoCycles) {
+	_highestSeqNo = htons(highestSeqNo);
+	_seqNoCycles = htons(seqNoCycles);
+}
+
+void RTCP_ReportBlock::setJitter(uint32_t jitter) { _jitter = htonl(jitter); }
+
+void RTCP_ReportBlock::setNTPOfSR(uint64_t ntp) { _lastReport = htonll(ntp >> 16u); }
+
+uint32_t RTCP_ReportBlock::getNTPOfSR() const { return ntohl(_lastReport) << 16u; }
+
+void RTCP_ReportBlock::setDelaySinceSR(uint32_t sr) {
+	// The delay, expressed in units of 1/65536 seconds
+	_delaySinceLastReport = htonl(sr);
+}
+
+void RTCP_ReportBlock::log() const {
+	PLOG_VERBOSE << "RTCP report block: "
+	             << "ssrc="
+	             << ntohl(_ssrc)
+	             // TODO: Implement these reports
+	             //	<< ", fractionLost=" << fractionLost
+	             //	<< ", packetsLost=" << packetsLost
+	             << ", highestSeqNo=" << highestSeqNo() << ", seqNoCycles=" << seqNoCycles()
+	             << ", jitter=" << jitter() << ", lastSR=" << getNTPOfSR()
+	             << ", lastSRDelay=" << delaySinceSR();
+}
+
+uint8_t RTCP_HEADER::version() const { return _first >> 6; }
+
+bool RTCP_HEADER::padding() const { return (_first >> 5) & 0x01; }
+
+uint8_t RTCP_HEADER::reportCount() const { return _first & 0x0F; }
+
+uint8_t RTCP_HEADER::payloadType() const { return _payloadType; }
+
+uint16_t RTCP_HEADER::length() const { return ntohs(_length); }
+
+size_t RTCP_HEADER::lengthInBytes() const { return (1 + length()) * 4; }
+
+void RTCP_HEADER::setPayloadType(uint8_t type) { _payloadType = type; }
+
+void RTCP_HEADER::setReportCount(uint8_t count) {
+	_first = (_first & 0b11100000u) | (count & 0b00011111u);
+}
+
+void RTCP_HEADER::setLength(uint16_t length) { _length = htons(length); }
+
+void RTCP_HEADER::prepareHeader(uint8_t payloadType, uint8_t reportCount, uint16_t length) {
+	_first = 0b10000000; // version 2, no padding
+	setReportCount(reportCount);
+	setPayloadType(payloadType);
+	setLength(length);
+}
+
+void RTCP_HEADER::log() const {
+	PLOG_VERBOSE << "RTCP header: "
+	             << "version=" << unsigned(version()) << ", padding=" << padding()
+	             << ", reportCount=" << unsigned(reportCount())
+	             << ", payloadType=" << unsigned(payloadType()) << ", length=" << length();
+}
+
+SSRC RTCP_FB_HEADER::packetSenderSSRC() const { return ntohl(_packetSender); }
+
+SSRC RTCP_FB_HEADER::mediaSourceSSRC() const { return ntohl(_mediaSource); }
+
+void RTCP_FB_HEADER::setPacketSenderSSRC(SSRC ssrc) { _packetSender = htonl(ssrc); }
+
+void RTCP_FB_HEADER::setMediaSourceSSRC(SSRC ssrc) { _mediaSource = htonl(ssrc); }
+
+void RTCP_FB_HEADER::log() const {
+	header.log();
+	PLOG_VERBOSE << "FB: "
+	             << " packet sender: " << packetSenderSSRC()
+	             << " media source: " << mediaSourceSSRC();
+}
+
+unsigned int RTCP_SR::Size(unsigned int reportCount) {
+	return sizeof(RTCP_HEADER) + 24 + reportCount * sizeof(RTCP_ReportBlock);
+}
+
+void RTCP_SR::preparePacket(SSRC senderSSRC, uint8_t reportCount) {
+	unsigned int length = ((sizeof(header) + 24 + reportCount * sizeof(RTCP_ReportBlock)) / 4) - 1;
+	header.prepareHeader(200, reportCount, uint16_t(length));
+	this->_senderSSRC = htonl(senderSSRC);
+}
+
+const RTCP_ReportBlock *RTCP_SR::getReportBlock(int num) const { return &_reportBlocks + num; }
+
+RTCP_ReportBlock *RTCP_SR::getReportBlock(int num) { return &_reportBlocks + num; }
+
+size_t RTCP_SR::getSize() const {
+	// "length" in packet is one less than the number of 32 bit words in the packet.
+	return sizeof(uint32_t) * (1 + size_t(header.length()));
+}
+
+uint64_t RTCP_SR::ntpTimestamp() const { return ntohll(_ntpTimestamp); }
+uint32_t RTCP_SR::rtpTimestamp() const { return ntohl(_rtpTimestamp); }
+uint32_t RTCP_SR::packetCount() const { return ntohl(_packetCount); }
+uint32_t RTCP_SR::octetCount() const { return ntohl(_octetCount); }
+uint32_t RTCP_SR::senderSSRC() const { return ntohl(_senderSSRC); }
+
+void RTCP_SR::setNtpTimestamp(uint64_t ts) { _ntpTimestamp = htonll(ts); }
+void RTCP_SR::setRtpTimestamp(uint32_t ts) { _rtpTimestamp = htonl(ts); }
+void RTCP_SR::setOctetCount(uint32_t ts) { _octetCount = htonl(ts); }
+void RTCP_SR::setPacketCount(uint32_t ts) { _packetCount = htonl(ts); }
+
+void RTCP_SR::log() const {
+	header.log();
+	PLOG_VERBOSE << "RTCP SR: "
+	             << " SSRC=" << senderSSRC() << ", NTP_TS=" << ntpTimestamp()
+	             << ", RTP_TS=" << rtpTimestamp() << ", packetCount=" << packetCount()
+	             << ", octetCount=" << octetCount();
+
+	for (unsigned i = 0; i < unsigned(header.reportCount()); i++) {
+		getReportBlock(i)->log();
+	}
+}
+
+unsigned int RTCP_SDES_ITEM::Size(uint8_t textLength) { return textLength + 2; }
+
+std::string RTCP_SDES_ITEM::text() const { return std::string(_text, _length); }
+
+void RTCP_SDES_ITEM::setText(std::string text) {
+	if (text.size() > 0xFF)
+		throw std::invalid_argument("text is too long");
+
+	_length = uint8_t(text.size());
+	memcpy(_text, text.data(), text.size());
+}
+
+uint8_t RTCP_SDES_ITEM::length() const { return _length; }
+
+unsigned int RTCP_SDES_CHUNK::Size(const std::vector<uint8_t> textLengths) {
+	unsigned int itemsSize = 0;
+	for (auto length : textLengths) {
+		itemsSize += RTCP_SDES_ITEM::Size(length);
+	}
+	auto nullTerminatedItemsSize = itemsSize + 1;
+	auto words = uint8_t(std::ceil(double(nullTerminatedItemsSize) / 4)) + 1;
+	return words * 4;
+}
+
+SSRC RTCP_SDES_CHUNK::ssrc() const { return ntohl(_ssrc); }
+
+void RTCP_SDES_CHUNK::setSSRC(SSRC ssrc) { _ssrc = htonl(ssrc); }
+
+const RTCP_SDES_ITEM *RTCP_SDES_CHUNK::getItem(int num) const {
+	auto base = &_items;
+	while (num-- > 0) {
+		auto itemSize = RTCP_SDES_ITEM::Size(base->length());
+		base = reinterpret_cast<const RTCP_SDES_ITEM *>(reinterpret_cast<const uint8_t *>(base) +
+		                                                itemSize);
+	}
+	return reinterpret_cast<const RTCP_SDES_ITEM *>(base);
+}
+
+RTCP_SDES_ITEM *RTCP_SDES_CHUNK::getItem(int num) {
+	auto base = &_items;
+	while (num-- > 0) {
+		auto itemSize = RTCP_SDES_ITEM::Size(base->length());
+		base = reinterpret_cast<RTCP_SDES_ITEM *>(reinterpret_cast<uint8_t *>(base) + itemSize);
+	}
+	return reinterpret_cast<RTCP_SDES_ITEM *>(base);
+}
+
+unsigned int RTCP_SDES_CHUNK::getSize() const {
+	std::vector<uint8_t> textLengths{};
+	unsigned int i = 0;
+	auto item = getItem(i);
+	while (item->type != 0) {
+		textLengths.push_back(item->length());
+		item = getItem(++i);
+	}
+	return Size(textLengths);
+}
+
+long RTCP_SDES_CHUNK::safelyCountChunkSize(size_t maxChunkSize) const {
+	if (maxChunkSize < RTCP_SDES_CHUNK::Size({})) {
+		// chunk is truncated
+		return -1;
+	}
+
+	size_t size = sizeof(SSRC);
+	unsigned int i = 0;
+	// We can always access first 4 bytes of first item (in case of no items there will be 4
+	// null bytes)
+	auto item = getItem(i);
+	std::vector<uint8_t> textsLength{};
+	while (item->type != 0) {
+		if (size + RTCP_SDES_ITEM::Size(0) > maxChunkSize) {
+			// item is too short
+			return -1;
+		}
+		auto itemLength = item->length();
+		if (size + RTCP_SDES_ITEM::Size(itemLength) >= maxChunkSize) {
+			// item is too large (it can't be equal to chunk size because after item there
+			// must be 1-4 null bytes as padding)
+			return -1;
+		}
+		textsLength.push_back(itemLength);
+		// safely to access next item
+		item = getItem(++i);
+	}
+	auto realSize = RTCP_SDES_CHUNK::Size(textsLength);
+	if (realSize > maxChunkSize) {
+		// Chunk is too large
+		return -1;
+	}
+	return realSize;
+}
+
+unsigned int RTCP_SDES::Size(const std::vector<std::vector<uint8_t>> lengths) {
+	unsigned int chunks_size = 0;
+	for (auto length : lengths)
+		chunks_size += RTCP_SDES_CHUNK::Size(length);
+
+	return 4 + chunks_size;
+}
+
+bool RTCP_SDES::isValid() const {
+	auto chunksSize = header.lengthInBytes() - sizeof(header);
+	if (chunksSize == 0) {
+		return true;
+	}
+	// there is at least one chunk
+	unsigned int i = 0;
+	unsigned int size = 0;
+	while (size < chunksSize) {
+		if (chunksSize < size + RTCP_SDES_CHUNK::Size({})) {
+			// chunk is truncated
+			return false;
+		}
+		auto chunk = getChunk(i++);
+		auto chunkSize = chunk->safelyCountChunkSize(chunksSize - size);
+		if (chunkSize < 0) {
+			// chunk is invalid
+			return false;
+		}
+		size += chunkSize;
+	}
+	return size == chunksSize;
+}
+
+unsigned int RTCP_SDES::chunksCount() const {
+	if (!isValid()) {
+		return 0;
+	}
+	uint16_t chunksSize = 4 * (header.length() + 1) - sizeof(header);
+	unsigned int size = 0;
+	unsigned int i = 0;
+	while (size < chunksSize) {
+		size += getChunk(i++)->getSize();
+	}
+	return i;
+}
+
+const RTCP_SDES_CHUNK *RTCP_SDES::getChunk(int num) const {
+	auto base = &_chunks;
+	while (num-- > 0) {
+		auto chunkSize = base->getSize();
+		base = reinterpret_cast<const RTCP_SDES_CHUNK *>(reinterpret_cast<const uint8_t *>(base) +
+		                                                 chunkSize);
+	}
+	return reinterpret_cast<const RTCP_SDES_CHUNK *>(base);
+}
+
+RTCP_SDES_CHUNK *RTCP_SDES::getChunk(int num) {
+	auto base = &_chunks;
+	while (num-- > 0) {
+		auto chunkSize = base->getSize();
+		base = reinterpret_cast<RTCP_SDES_CHUNK *>(reinterpret_cast<uint8_t *>(base) + chunkSize);
+	}
+	return reinterpret_cast<RTCP_SDES_CHUNK *>(base);
+}
+
+void RTCP_SDES::preparePacket(uint8_t chunkCount) {
+	unsigned int chunkSize = 0;
+	for (uint8_t i = 0; i < chunkCount; i++) {
+		auto chunk = getChunk(i);
+		chunkSize += chunk->getSize();
+	}
+	uint16_t length = uint16_t((sizeof(header) + chunkSize) / 4 - 1);
+	header.prepareHeader(202, chunkCount, length);
+}
+
+const RTCP_ReportBlock *RTCP_RR::getReportBlock(int num) const { return &_reportBlocks + num; }
+
+RTCP_ReportBlock *RTCP_RR::getReportBlock(int num) { return &_reportBlocks + num; }
+
+size_t RTCP_RR::SizeWithReportBlocks(uint8_t reportCount) {
+	return sizeof(header) + 4 + size_t(reportCount) * sizeof(RTCP_ReportBlock);
+}
+
+SSRC RTCP_RR::senderSSRC() const { return ntohl(_senderSSRC); }
+
+bool RTCP_RR::isSenderReport() { return header.payloadType() == 200; }
+
+bool RTCP_RR::isReceiverReport() { return header.payloadType() == 201; }
+
+size_t RTCP_RR::getSize() const {
+	// "length" in packet is one less than the number of 32 bit words in the packet.
+	return sizeof(uint32_t) * (1 + size_t(header.length()));
+}
+
+void RTCP_RR::preparePacket(SSRC senderSSRC, uint8_t reportCount) {
+	// "length" in packet is one less than the number of 32 bit words in the packet.
+	size_t length = (SizeWithReportBlocks(reportCount) / 4) - 1;
+	header.prepareHeader(201, reportCount, uint16_t(length));
+	this->_senderSSRC = htonl(senderSSRC);
+}
+
+void RTCP_RR::setSenderSSRC(SSRC ssrc) { this->_senderSSRC = htonl(ssrc); }
+
+void RTCP_RR::log() const {
+	header.log();
+	PLOG_VERBOSE << "RTCP RR: "
+	             << " SSRC=" << ntohl(_senderSSRC);
+
+	for (unsigned i = 0; i < unsigned(header.reportCount()); i++) {
+		getReportBlock(i)->log();
+	}
+}
+
+size_t RTCP_REMB::SizeWithSSRCs(int count) {
+	return sizeof(RTCP_REMB) + (count - 1) * sizeof(SSRC);
+}
+
+unsigned int RTCP_REMB::getSize() const {
+	// "length" in packet is one less than the number of 32 bit words in the packet.
+	return sizeof(uint32_t) * (1 + header.header.length());
+}
+
+void RTCP_REMB::preparePacket(SSRC senderSSRC, unsigned int numSSRC, unsigned int in_bitrate) {
+
+	// Report Count becomes the format here.
+	header.header.prepareHeader(206, 15, 0);
+
+	// Always zero.
+	header.setMediaSourceSSRC(0);
+
+	header.setPacketSenderSSRC(senderSSRC);
+
+	_id[0] = 'R';
+	_id[1] = 'E';
+	_id[2] = 'M';
+	_id[3] = 'B';
+
+	setBitrate(numSSRC, in_bitrate);
+}
+
+void RTCP_REMB::setBitrate(unsigned int numSSRC, unsigned int in_bitrate) {
+	unsigned int exp = 0;
+	while (in_bitrate > pow(2, 18) - 1) {
+		exp++;
+		in_bitrate /= 2;
+	}
+
+	// "length" in packet is one less than the number of 32 bit words in the packet.
+	header.header.setLength(
+	    uint16_t((offsetof(RTCP_REMB, _ssrc) / sizeof(uint32_t)) - 1 + numSSRC));
+
+	_bitrate = htonl((numSSRC << (32u - 8u)) | (exp << (32u - 8u - 6u)) | in_bitrate);
+}
+
+void RTCP_REMB::setSsrc(int iterator, SSRC newSssrc) { _ssrc[iterator] = htonl(newSssrc); }
+
+unsigned int RTCP_PLI::Size() { return sizeof(RTCP_FB_HEADER); }
+
+void RTCP_PLI::preparePacket(SSRC messageSSRC) {
+	header.header.prepareHeader(206, 1, 2);
+	header.setPacketSenderSSRC(messageSSRC);
+	header.setMediaSourceSSRC(messageSSRC);
+}
+
+void RTCP_PLI::log() const { header.log(); }
+
+unsigned int RTCP_FIR::Size() { return sizeof(RTCP_FB_HEADER) + sizeof(RTCP_FIR_PART); }
+
+void RTCP_FIR::preparePacket(SSRC messageSSRC, uint8_t seqNo) {
+	header.header.prepareHeader(206, 4, 2 + 2 * 1);
+	header.setPacketSenderSSRC(messageSSRC);
+	header.setMediaSourceSSRC(messageSSRC);
+	parts[0].ssrc = htonl(messageSSRC);
+	parts[0].seqNo = seqNo;
+}
+
+void RTCP_FIR::log() const { header.log(); }
+
+uint16_t RTCP_NACK_PART::pid() { return ntohs(_pid); }
+uint16_t RTCP_NACK_PART::blp() { return ntohs(_blp); }
+
+void RTCP_NACK_PART::setPid(uint16_t pid) { _pid = htons(pid); }
+void RTCP_NACK_PART::setBlp(uint16_t blp) { _blp = htons(blp); }
+
+std::vector<uint16_t> RTCP_NACK_PART::getSequenceNumbers() {
+	std::vector<uint16_t> result{};
+	result.reserve(17);
+	uint16_t p = pid();
+	result.push_back(p);
+	uint16_t bitmask = blp();
+	uint16_t i = p + 1;
+	while (bitmask > 0) {
+		if (bitmask & 0x1) {
+			result.push_back(i);
+		}
+		i += 1;
+		bitmask >>= 1;
+	}
+	return result;
+}
+
+unsigned int RTCP_NACK::Size(unsigned int discreteSeqNoCount) {
+	return offsetof(RTCP_NACK, parts) + sizeof(RTCP_NACK_PART) * discreteSeqNoCount;
+}
+
+unsigned int RTCP_NACK::getSeqNoCount() { return header.header.length() - 2; }
+
+void RTCP_NACK::preparePacket(SSRC ssrc, unsigned int discreteSeqNoCount) {
+	header.header.prepareHeader(205, 1, 2 + uint16_t(discreteSeqNoCount));
+	header.setMediaSourceSSRC(ssrc);
+	header.setPacketSenderSSRC(ssrc);
+}
+
+bool RTCP_NACK::addMissingPacket(unsigned int *fciCount, uint16_t *fciPID, uint16_t missingPacket) {
+	if (*fciCount == 0 || missingPacket < *fciPID || missingPacket > (*fciPID + 16)) {
+		parts[*fciCount].setPid(missingPacket);
+		parts[*fciCount].setBlp(0);
+		*fciPID = missingPacket;
+		(*fciCount)++;
+		return true;
+	} else {
+		// TODO SPEED!
+		uint16_t blp = parts[(*fciCount) - 1].blp();
+		uint16_t newBit = uint16_t(1u << (missingPacket - (1 + *fciPID)));
+		parts[(*fciCount) - 1].setBlp(blp | newBit);
+		return false;
+	}
+}
+
+uint16_t RTP_RTX::getOriginalSeqNo() const { return ntohs(*(uint16_t *)(header.getBody())); }
+
+const char *RTP_RTX::getBody() const { return header.getBody() + sizeof(uint16_t); }
+
+char *RTP_RTX::getBody() { return header.getBody() + sizeof(uint16_t); }
+
+size_t RTP_RTX::getBodySize(size_t totalSize) const {
+	return totalSize - (getBody() - reinterpret_cast<const char *>(this));
+}
+
+size_t RTP_RTX::getSize() const { return header.getSize() + sizeof(uint16_t); }
+
+size_t RTP_RTX::normalizePacket(size_t totalSize, SSRC originalSSRC, uint8_t originalPayloadType) {
+	header.setSeqNumber(getOriginalSeqNo());
+	header.setSsrc(originalSSRC);
+	header.setPayloadType(originalPayloadType);
+	// TODO, the -12 is the size of the header (which is variable!)
+	memmove(header.getBody(), getBody(), totalSize - getSize());
+	return totalSize - 2;
+}
+
+size_t RTP_RTX::copyTo(RTP *dest, size_t totalSize, uint8_t originalPayloadType) {
+	memmove((char *)dest, (char *)this, header.getSize());
+	dest->setSeqNumber(getOriginalSeqNo());
+	dest->setPayloadType(originalPayloadType);
+	memmove(dest->getBody(), getBody(), getBodySize(totalSize));
+	return totalSize;
+}
+
+}; // namespace rtc

src/rtppacketizationconfig.cpp

@@ -1,25 +1,27 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "rtppacketizationconfig.hpp"
 
+#include <cassert>
+
 namespace rtc {
 
 RtpPacketizationConfig::RtpPacketizationConfig(SSRC ssrc, string cname, uint8_t payloadType,

src/rtppacketizer.cpp

@@ -1,25 +1,27 @@
-/*
+/**
- * libdatachannel streamer example
  * Copyright (c) 2020 Filip Klembara (in2core)
  *
- * This program is free software; you can redistribute it and/or
+ * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
+ * modify it under the terms of the GNU Lesser General Public
- * as published by the Free Software Foundation; either version 2
+ * License as published by the Free Software Foundation; either
- * of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * GNU General Public License for more details.
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
+ * You should have received a copy of the GNU Lesser General Public
- * along with this program; If not, see <http://www.gnu.org/licenses/>.
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #if RTC_ENABLE_MEDIA
 
 #include "rtppacketizer.hpp"
 
+#include <cstring>
+
 namespace rtc {
 
 RtpPacketizer::RtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig)
@@ -37,7 +39,7 @@ binary_ptr RtpPacketizer::packetize(shared_ptr<binary> payload, bool setMark) {
 		rtp->setMarker(true);
 	}
 	rtp->preparePacket();
-	memcpy(msg->data() + rtpHeaderSize, payload->data(), payload->size());
+	std::memcpy(msg->data() + rtpHeaderSize, payload->data(), payload->size());
 	return msg;
 }
 

src/track.cpp

@@ -17,8 +17,8 @@
  */
 
 #include "track.hpp"
-#include "globals.hpp"
 
+#include "impl/internals.hpp"
 #include "impl/track.hpp"
 
 namespace rtc {
@@ -46,9 +46,7 @@ bool Track::isOpen(void) const { return impl()->isOpen(); }
 
 bool Track::isClosed(void) const { return impl()->isClosed(); }
 
-size_t Track::maxMessageSize() const {
+size_t Track::maxMessageSize() const { return impl()->maxMessageSize(); }
-	return impl()->maxMessageSize();
-}
 
 void Track::setMediaHandler(shared_ptr<MediaHandler> handler) {
 	impl()->setMediaHandler(std::move(handler));

src/websocket.cpp

@@ -19,10 +19,10 @@
 #if RTC_ENABLE_WEBSOCKET
 
 #include "websocket.hpp"
-#include "globals.hpp"
 #include "common.hpp"
 
 #include "impl/websocket.hpp"
+#include "impl/internals.hpp"
 
 #include <regex>
 

test/benchmark.cpp

@@ -121,12 +121,11 @@ size_t benchmark(milliseconds duration) {
 			}
 		} catch (const std::exception &e) {
 			std::cout << "Send failed: " << e.what() << std::endl;
-		}
+		}		
-
-		// When sent data is buffered in the DataChannel,
-		// wait for onBufferedAmountLow callback to continue
 	});
 
+	// When sent data is buffered in the DataChannel,
+	// wait for onBufferedAmountLow callback to continue
 	dc1->onBufferedAmountLow([wdc1 = make_weak_ptr(dc1), &messageData]() {
 		auto dc1 = wdc1.lock();
 		if (!dc1)

test/capi_connectivity.cpp

@@ -137,11 +137,11 @@ static void RTC_API dataChannelCallback(int pc, int dc, void *ptr) {
 		return;
 	}
 
+	rtcSetOpenCallback(dc, openCallback);
 	rtcSetClosedCallback(dc, closedCallback);
 	rtcSetMessageCallback(dc, messageCallback);
 
 	peer->dc = dc;
-	peer->connected = true;
 
 	const char *message = peer == peer1 ? "Hello from 1" : "Hello from 2";
 	rtcSendMessage(peer->dc, message, -1); // negative size indicates a null-terminated string

test/capi_track.cpp

@@ -83,7 +83,7 @@ static void RTC_API closedCallback(int id, void *ptr) {
 static void RTC_API trackCallback(int pc, int tr, void *ptr) {
 	Peer *peer = (Peer *)ptr;
 	peer->tr = tr;
-	peer->connected = true;
+	rtcSetOpenCallback(tr, openCallback);
 	rtcSetClosedCallback(tr, closedCallback);
 
 	char buffer[1024];

test/connectivity.cpp

@@ -24,6 +24,8 @@
 #include <memory>
 #include <thread>
 
+#define CUSTOM_MAX_MESSAGE_SIZE 1048576
+
 using namespace rtc;
 using namespace std;
 
@@ -39,7 +41,7 @@ void test_connectivity() {
 	// Custom MTU example
 	config1.mtu = 1500;
 	// Custom max message size
-	config1.maxMessageSize = 1048576;
+	config1.maxMessageSize = CUSTOM_MAX_MESSAGE_SIZE;
 
 	PeerConnection pc1(config1);
 
@@ -50,7 +52,7 @@ void test_connectivity() {
 	// Custom MTU example
 	config2.mtu = 1500;
 	// Custom max message size
-	config2.maxMessageSize = 1048576;
+	config2.maxMessageSize = CUSTOM_MAX_MESSAGE_SIZE;
 	// Port range example
 	config2.portRangeBegin = 5000;
 	config2.portRangeEnd = 6000;
@@ -105,26 +107,29 @@ void test_connectivity() {
 			return;
 		}
 
+		dc->onOpen([wdc = make_weak_ptr(dc)]() {
+			if (auto dc = wdc.lock())
+				dc->send("Hello from 2");
+		});
+
 		dc->onMessage([](variant<binary, string> message) {
 			if (holds_alternative<string>(message)) {
 				cout << "Message 2: " << get<string>(message) << endl;
 			}
 		});
 
-		dc->send("Hello from 2");
-
 		std::atomic_store(&dc2, dc);
 	});
 
 	auto dc1 = pc1.createDataChannel("test");
-	dc1->onOpen([wdc1 = make_weak_ptr(dc1)]() {
-		auto dc1 = wdc1.lock();
-		if (!dc1)
-			return;
 
-		cout << "DataChannel 1: Open" << endl;
+	dc1->onOpen([wdc1 = make_weak_ptr(dc1)]() {
-		dc1->send("Hello from 1");
+		if (auto dc1 = wdc1.lock()) {
+			cout << "DataChannel 1: Open" << endl;
+			dc1->send("Hello from 1");
+		}
 	});
+
 	dc1->onMessage([](const variant<binary, string> &message) {
 		if (holds_alternative<string>(message)) {
 			cout << "Message 1: " << get<string>(message) << endl;
@@ -144,7 +149,7 @@ void test_connectivity() {
 	if (!adc2 || !adc2->isOpen() || !dc1->isOpen())
 		throw runtime_error("DataChannel is not open");
 
-	if (dc1->maxMessageSize() != 1048576 || dc2->maxMessageSize() != 1048576)
+	if (dc1->maxMessageSize() != CUSTOM_MAX_MESSAGE_SIZE || dc2->maxMessageSize() != CUSTOM_MAX_MESSAGE_SIZE)
 		throw runtime_error("DataChannel max message size is incorrect");
 
 	if (auto addr = pc1.localAddress())
@@ -175,25 +180,26 @@ void test_connectivity() {
 			return;
 		}
 
+		dc->onOpen([wdc = make_weak_ptr(dc)]() {
+			if (auto dc = wdc.lock())
+				dc->send("Second hello from 2");
+		});
+
 		dc->onMessage([](variant<binary, string> message) {
 			if (holds_alternative<string>(message)) {
 				cout << "Second Message 2: " << get<string>(message) << endl;
 			}
 		});
 
-		dc->send("Send hello from 2");
-
 		std::atomic_store(&second2, dc);
 	});
 
 	auto second1 = pc1.createDataChannel("second");
 	second1->onOpen([wsecond1 = make_weak_ptr(dc1)]() {
-		auto second1 = wsecond1.lock();
+		if (auto second1 = wsecond1.lock()) {
-		if (!second1)
+			cout << "Second DataChannel 1: Open" << endl;
-			return;
+			second1->send("Second hello from 1");
-
+		}
-		cout << "Second DataChannel 1: Open" << endl;
-		second1->send("Second hello from 1");
 	});
 	dc1->onMessage([](const variant<binary, string> &message) {
 		if (holds_alternative<string>(message)) {

test/turn_connectivity.cpp

@@ -108,14 +108,17 @@ void test_turn_connectivity() {
 			return;
 		}
 
+		dc->onOpen([wdc = make_weak_ptr(dc)]() {
+			if (auto dc = wdc.lock())
+				dc->send("Hello from 2");
+		});
+
 		dc->onMessage([](variant<binary, string> message) {
 			if (holds_alternative<string>(message)) {
 				cout << "Message 2: " << get<string>(message) << endl;
 			}
 		});
 
-		dc->send("Hello from 2");
-
 		std::atomic_store(&dc2, dc);
 	});
 
@@ -175,14 +178,17 @@ void test_turn_connectivity() {
 			return;
 		}
 
+		dc->onOpen([wdc = make_weak_ptr(dc)]() {
+			if (auto dc = wdc.lock())
+				dc->send("Second hello from 2");
+		});
+
 		dc->onMessage([](variant<binary, string> message) {
 			if (holds_alternative<string>(message)) {
 				cout << "Second Message 2: " << get<string>(message) << endl;
 			}
 		});
 
-		dc->send("Send hello from 2");
-
 		std::atomic_store(&second2, dc);
 	});