After I failed to compile the webrtc source code, I tried to find other similar rtp transfer libraries, but unfortunately, they were either written too simply, without weak network adversarial mechanisms, or too strongly associated with their own projects. Finally, I decided to try writing my own.
I called my project litertp. As it's name "lite", this project only implement rtp network transport. Even without negotiation.
You can use this lib with your projects such as webrtc, rtsp, sip, h323 and others.
Litertp implements nack,fir,pli, not support fec,rtx,transport-cc.
The source code based on C++17.
This project uses the cmake build system.
LITERTP_SSL=ON/OFF # Build with openssl and srtp
LITERTP_SHARED=ON/OFF # Build with shared libaray
LITERTP_STATIC=ON/OFF # Build with static libaray
OPENSSL_INCLUDE=path/to/openssl/include # if LITERTP_SSL is ON, this must be set
OPENSSL_LIB_CRYPTO=path/to/libcrypto.lib # if LITERTP_SSL is ON, this must be set
OPENSSL_LIB_SSL=path/to/libssl.lib # if LITERTP_SSL is ON, this must be set
SRTP_INCLUDE=path/to/srtp/include # if LITERTP_SSL is ON, this must be set
SRTP_LIB=path/to/libsrtp2.lib # if LITERTP_SSL is ON, this must be set
set ABI=x64|win32
cmake . ^
-G "Visual Studio 17 2022" ^
-A %ABI% ^
-DLITERTP_SSL=ON ^
-DLITERTP_SHARED=ON ^
-DLITERTP_STATIC=ON ^
-DOPENSSL_INCLUDE="path/to/openssl/include" ^
-DOPENSSL_LIB_CRYPTO="path/to/libcrypto.lib" ^
-DOPENSSL_LIB_SSL="path/to/libssl.lib" ^
-DSRTP_INCLUDE="path/to/srtp/include" ^
-DSRTP_LIB="path/to/libsrtp2.lib"
ABI=MAC|MAC_ARM64|MAC_UNIVERSAL|OS64|OS64COMBINED
cmake . \
-G "Xcode"
-DCMAKE_CXX_FLAGS="--std=c++17" \
-DCMAKE_TOOLCHAIN_FILE=${TOOLCHAIN} \
-DPLATFORM=${ABI} \
-DLITERTP_SSL=ON \
-DLITERTP_SHARED=OFF \
-DOPENSSL_INCLUDE="path/to/openssl/include" \
-DOPENSSL_LIB_CRYPTO="path/to/libcrypto.lib" \
-DOPENSSL_LIB_SSL="path/to/libssl.lib" \
-DSRTP_INCLUDE="path/to/srtp/include" \
-DSRTP_LIB="path/to/libsrtp2.lib"
CMake for osx need a config file, I recommended use this project
https://github.com/leetal/ios-cmake
ABI=armeabi-v7a,arm64-v8a
cmake . \
-B "${BUILD_DIR}" -DCMAKE_BUILD_TYPE=release -DCMAKE_INSTALL_PREFIX="${INSTALL_DIR}" \
-DCMAKE_TOOLCHAIN_FILE=${TOOLCHAIN} \
-DCMAKE_CXX_FLAGS="--std=c++17" \
-DANDROID_ABI=${ABI} \
-DANDROID_PLATFORM=android-23 \
-DLITERTP_SSL=ON \
-DOPENSSL_INCLUDE="path/to/openssl/include" \
-DOPENSSL_LIB_CRYPTO="path/to/libcrypto.lib" \
-DOPENSSL_LIB_SSL="path/to/libssl.lib" \
-DSRTP_INCLUDE="path/to/srtp/include" \
-DSRTP_LIB="path/to/libsrtp2.lib"
I have prepared some scripts in source root path, you can edit slightly to build
./build-android.sh armeabie-v7a
./build-osx.sh MAC_UNIVERSAL
./build-win.bat win32
First of all, you must initialize the library
litertp_global_init()And if no longer need, cleanup the library
litertp_global_cleanup()//create session
litertp_session_t session=litertp_create_session(1); // paramter 1 indicates webrtc rtp
//set event callback
litertp_set_on_frame_eventhandler(session,onframe,ctx);
litertp_set_on_keyframe_required_eventhandler(session,on_keyframe_required,ctx);
//set audio capabilities
litertp_create_media_stream(session,media_type_audio,0,rtp_trans_mode_sendrecv,true,"0.0.0.0",50000,50000);
litertp_add_local_audio_track(session,codec_type_opus,111,48000,2);
litertp_add_local_audio_track(session,codec_type_pcma,8,8000,1);
//set video capabilities
litertp_create_media_stream(session,media_type_video,0,rtp_trans_mode_sendrecv,true,"0.0.0.0",50000,50000);
litertp_add_local_video_track(session,codec_type_h264,96,90000);
litertp_add_local_video_track(session,codec_type_vp8,97,90000);
litertp_add_local_attribute(session,media_type_video,96,"fmtp","level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f");
//if webrtc,you can add candidates, this only set into sdp, Not going to discover
litertp_add_local_candidate(session,media_type_audio,1,1,"192.168.0.11",50000,2122260223);
litertp_add_local_candidate(session,media_type_video,2,1,"192.168.0.11",50000,2122260223);
//generate sdp to send you remote end
char* local_sdp=nullptr;
int sdp_size=0;
litertp_create_offer(session,&local_sdp,&sdp_size);
litertp_free(&local_sdp)
After get remote sdp via your way.
litertp_set_remote_sdp(session,sdp,sdp_type_answer);
//create session
litertp_session_t session=litertp_create_session(1); // paramter 1 indicates webrtc rtp
//set event callback
litertp_set_on_frame_eventhandler(session,onframe,ctx);
litertp_set_on_keyframe_required_eventhandler(session,on_keyframe_required,ctx);
//set audio capabilities
litertp_create_media_stream(session,media_type_audio,0,rtp_trans_mode_sendrecv,true,"0.0.0.0",50000,50000);
litertp_add_local_audio_track(session,codec_type_opus,111,48000,2);
litertp_add_local_audio_track(session,codec_type_pcma,8,8000,1);
//set video capabilities
litertp_create_media_stream(session,media_type_video,0,rtp_trans_mode_sendrecv,true,"0.0.0.0",50000,50000);
litertp_add_local_video_track(session,codec_type_h264,96,90000);
litertp_add_local_video_track(session,codec_type_vp8,97,90000);
litertp_add_local_attribute(session,media_type_video,96,"fmtp","level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f");
//if webrtc,you can add candidates, this only set into sdp, Not going to discover
litertp_add_local_candidate(session,media_type_audio,1,1,"192.168.0.11",50000,2122260223);
litertp_add_local_candidate(session,media_type_video,2,1,"192.168.0.11",50000,2122260223);
//set remote offer sdp to negotiate
litertp_set_remote_sdp(session,sdp,sdp_type_offer);
//create answer
char* local_sdp=nullptr;
int sdp_size=0;
litertp_create_answer(session,&local_sdp,&sdp_size);
litertp_free(&local_sdp)
You also can set local sdp for initialize local streams.
litertp_set_local_sdp(session,sdp)This is equivalent to
//set audio capabilities
litertp_create_media_stream(session,media_type_audio,0,rtp_trans_mode_sendrecv,true,"0.0.0.0",50000,50000);
litertp_add_local_audio_track(session,codec_type_opus,111,48000,2);
litertp_add_local_audio_track(session,codec_type_pcma,8,8000,1);
//set video capabilities
litertp_create_media_stream(session,media_type_video,0,rtp_trans_mode_sendrecv,true,"0.0.0.0",50000,50000);
litertp_add_local_video_track(session,codec_type_h264,96,90000);
litertp_add_local_video_track(session,codec_type_vp8,97,90000);
litertp_add_local_attribute(session,media_type_video,96,"fmtp","level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f");
//if webrtc,you can add candidates, this only set into sdp, Not going to discover
litertp_add_local_candidate(session,media_type_audio,1,1,"192.168.0.11",50000,2122260223);
litertp_add_local_candidate(session,media_type_video,2,1,"192.168.0.11",50000,2122260223);And some transport protocol like h.323, you won't get remote sdp, so you can not call litertp_set_remote_sdp
in this case you need to manually construct remote streams
//set audio capabilities
litertp_create_media_stream
litertp_add_remote_audio_track
litertp_add_remote_audio_track
//set video capabilities
litertp_create_media_stream
litertp_add_remote_video_track
litertp_add_remote_video_track
litertp_add_remote_attribute
litertp_set_remote_ssrc
litertp_set_remote_trans_mode
litertp_set_remote_mid
litertp_set_remote_setup
litertp_set_remote_rtp_endpoint
litertp_set_remote_rtcp_endpoint
This is equivalent to
litertp_set_remote_sdp
Example for supporting rtsp over tcp
//create session
litertp_session_t session=litertp_create_session(0); // paramter 0 indicates normal rtp
//set event callback
litertp_set_on_frame_eventhandler(session,onframe,ctx);
litertp_set_on_keyframe_required_eventhandler(session,on_keyframe_required,ctx);
//set audio capabilities
litertp_create_media_stream_custom_transport(session,media_type_audio,0,rtp_trans_mode_sendrecv,false,
0,on_send_packet,nullptr);
litertp_add_local_audio_track(session,codec_type_opus,111,48000,2);
litertp_add_local_audio_track(session,codec_type_pcma,8,8000,1);
//set video capabilities
litertp_create_media_stream_custom_transport(session,media_type_video,0,rtp_trans_mode_sendrecv,false,
1,on_send_packet,nullptr);
litertp_add_local_video_track(session,codec_type_h264,96,90000);
litertp_add_local_video_track(session,codec_type_vp8,97,90000);
litertp_add_local_attribute(session,media_type_video,96,"fmtp","level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f");The difference of litertp_create_media_stream_custom_transport function is on parameters port, on_send_packet and ctx . Custom_transport can not send packet inside, so when it needs to send rtp/rtcp packet, on_send_packet will be called back, you need to send by youself
void on_send_packet(void* ctx, int port, int channel, const uint8_t* data, int size)
{
// magic $
char magic = '$';
socket_->send(&magic, 1);
// channel
if (port == 0) //audio
{
if (channel == 0) //rtp
{
socket_->send(&audio_rtp_channel_, 1);
}
else if (channel == 1) //rtcp
{
socket_->send(&audio_rtcp_channel_, 1);
}
}
else if (port == 1) //video
{
if (channel == 0)//rtp
{
socket_->send(&video_rtp_channel_, 1);
}
else if (channel == 1)//rtcp
{
socket_->send(&video_rtcp_channel_, 1);
}
}
// length
uint16_t len = (uint16_t)size;
len=sys::socket::hton16(len);
socket_->send((const char*)&len, 2);
// data
socket_->send((const char*)data, size);
}For receiving data, you need to receive data and separate out packet bytes by youself, Then call
litertp_receive_rtp_packet(session_,port,rtp_data,size)
litertp_receive_rtcp_packet(session_,port,rtcp_data,size)
This will trigger onframe and other rtcp events like normal transport receive data.
Now you can send and receive media frame.
The received frames will raised from litertp_on_frame set by litertp_set_on_frame_eventhandler
To send frame call litertp_send_frame
You can timing call litertp_get_stats to get rtcp stats info, Details to see struct rtp_stats_t