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ZLMediaKit
Commit 29c8c9bf by xia-chu
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Merge remote-tracking branch 'origin/dev' into dev

parents f939e0a8 753fe058
正在显示 包含 289 行增加 和 176 行删除
AUTHORS
...@@ -31,4 +31,5 @@ Luke <automan@easydarwin.org> ...@@ -31,4 +31,5 @@ Luke <automan@easydarwin.org>
大裤衩 <3503207480@qq.com> 大裤衩 <3503207480@qq.com>
droid.chow <droid.chow@gmail.com> droid.chow <droid.chow@gmail.com>
[陈晓林](https://github.com/musicwood) [陈晓林](https://github.com/musicwood)
[老衲不出家](https://github.com/monktan89) [老衲不出家](https://github.com/monktan89)
\ No newline at end of file [big panda](<2381267071@qq.com>)
\ No newline at end of file
conf/config.ini
...@@ -206,13 +206,21 @@ port=10000 ...@@ -206,13 +206,21 @@ port=10000
#rtp超时时间，单位秒 #rtp超时时间，单位秒
timeoutSec=15 timeoutSec=15
[rtc]
#rtc播放推流、播放超时时间
timeoutSec=15
#本机对rtc客户端的可见ip，作为服务器时一般为公网ip，置空时，会自动获取网卡ip
externIP=
[rtsp] [rtsp]
#rtsp专有鉴权方式是采用base64还是md5方式 #rtsp专有鉴权方式是采用base64还是md5方式
authBasic=0 authBasic=0
#rtsp拉流代理是否是直接代理模式 #rtsp拉流、推流代理是否是直接代理模式
#直接代理后支持任意编码格式，但是会导致GOP缓存无法定位到I帧，可能会导致开播花屏 #直接代理后支持任意编码格式，但是会导致GOP缓存无法定位到I帧，可能会导致开播花屏
#并且如果是tcp方式拉流，如果rtp大于mtu会导致无法使用udp方式代理 #并且如果是tcp方式拉流，如果rtp大于mtu会导致无法使用udp方式代理
#假定您的拉流源地址不是264或265或AAC，那么你可以使用直接代理的方式来支持rtsp代理 #假定您的拉流源地址不是264或265或AAC，那么你可以使用直接代理的方式来支持rtsp代理
#如果你是rtsp推拉流，但是rtc播放，也建议关闭直接代理模式，
#因为直接代理时，rtp中可能没有sps pps,会导致rtc无法播放
#默认开启rtsp直接代理，rtmp由于没有这些问题，是强制开启直接代理的 #默认开启rtsp直接代理，rtmp由于没有这些问题，是强制开启直接代理的
directProxy=1 directProxy=1
#rtsp必须在此时间内完成握手，否则服务器会断开链接，单位秒 #rtsp必须在此时间内完成握手，否则服务器会断开链接，单位秒
......
server/CMakeLists.txt
...@@ -48,4 +48,8 @@ else() ...@@ -48,4 +48,8 @@ else()
install(TARGETS MediaServer DESTINATION ${INSTALL_PATH_EXECUTABLE}) install(TARGETS MediaServer DESTINATION ${INSTALL_PATH_EXECUTABLE})
endif() endif()
target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST}) if (CMAKE_SYSTEM_NAME MATCHES "Linux")
target_link_libraries(MediaServer -Wl,--start-group jsoncpp ${LINK_LIB_LIST} -Wl,--end-group)
else ()
target_link_libraries(MediaServer jsoncpp ${LINK_LIB_LIST})
endif ()
server/main.cpp
...@@ -210,153 +210,6 @@ static void inline listen_shell_input(){ ...@@ -210,153 +210,6 @@ static void inline listen_shell_input(){
//全局变量，在WebApi中用于保存配置文件用 //全局变量，在WebApi中用于保存配置文件用
string g_ini_file; string g_ini_file;
void test_sdp(){
char str1[] = "v=0\n"
"o=- 380154348540553537 2 IN IP4 127.0.0.1\n"
"s=-\n"
"b=CT:1900\n"
"t=0 0\n"
"a=group:BUNDLE video\n"
"a=msid-semantic: WMS\n"
"m=video 9 RTP/SAVPF 96\n"
"c=IN IP4 0.0.0.0\n"
"a=rtcp:9 IN IP4 0.0.0.0\n"
"a=ice-ufrag:1ZFN\n"
"a=ice-pwd:70P3H0jPlGz1fiJl5XZfXMZH\n"
"a=ice-options:trickle\n"
"a=fingerprint:sha-256 3E:10:35:6B:9A:9E:B0:55:AC:2A:88:F5:74:C1:70:32:B5:8D:88:1D:37:B0:9C:69:A6:DD:07:10:73:27:1A:16\n"
"a=setup:active\n"
"a=mid:video\n"
"a=recvonly\n"
"a=rtcp-mux\n"
"a=rtpmap:96 H264/90000\n"
"a=fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f";
char str2[] = "v=0\n"
"o=- 2584450093346841581 2 IN IP4 127.0.0.1\n"
"s=-\n"
"t=0 0\n"
"a=group:BUNDLE audio video data\n"
"a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549\n"
"m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\n"
"c=IN IP4 0.0.0.0\n"
"a=rtcp:9 IN IP4 0.0.0.0\n"
"a=ice-ufrag:sXJ3\n"
"a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n"
"a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n"
"a=setup:actpass\n"
"a=mid:audio\n"
"a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level\n"
"a=sendrecv\n"
"a=rtcp-mux\n"
"a=rtpmap:111 opus/48000/2\n"
"a=rtcp-fb:111 transport-cc\n"
"a=fmtp:111 minptime=10;useinbandfec=1\n"
"a=rtpmap:103 ISAC/16000\n"
"a=rtpmap:104 ISAC/32000\n"
"a=rtpmap:9 G722/8000\n"
"a=rtpmap:0 PCMU/8000\n"
"a=rtpmap:8 PCMA/8000\n"
"a=rtpmap:106 CN/32000\n"
"a=rtpmap:105 CN/16000\n"
"a=rtpmap:13 CN/8000\n"
"a=rtpmap:110 telephone-event/48000\n"
"a=rtpmap:112 telephone-event/32000\n"
"a=rtpmap:113 telephone-event/16000\n"
"a=rtpmap:126 telephone-event/8000\n"
"a=ssrc:120276603 cname:iSkJ2vn5cYYubTve\n"
"a=ssrc:120276603 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 1da3d329-7399-4fe9-b20f-69606bebd363\n"
"a=ssrc:120276603 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n"
"a=ssrc:120276603 label:1da3d329-7399-4fe9-b20f-69606bebd363\n"
"m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125\n"
"c=IN IP4 0.0.0.0\n"
"a=rtcp:9 IN IP4 0.0.0.0\n"
"a=ice-ufrag:sXJ3\n"
"a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n"
"a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n"
"a=setup:actpass\n"
"a=mid:video\n"
"a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\n"
"a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\n"
"a=extmap:4 urn:3gpp:video-orientation\n"
"a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\n"
"a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\n"
"a=sendrecv\n"
"a=rtcp-mux\n"
"a=rtcp-rsize\n"
"a=rtpmap:96 VP8/90000\n"
"a=rtcp-fb:96 ccm fir\n"
"a=rtcp-fb:96 nack\n"
"a=rtcp-fb:96 nack pli\n"
"a=rtcp-fb:96 goog-remb\n"
"a=rtcp-fb:96 transport-cc\n"
"a=rtpmap:98 VP9/90000\n"
"a=rtcp-fb:98 ccm fir\n"
"a=rtcp-fb:98 nack\n"
"a=rtcp-fb:98 nack pli\n"
"a=rtcp-fb:98 goog-remb\n"
"a=rtcp-fb:98 transport-cc\n"
"a=rtpmap:100 H264/90000\n"
"a=rtcp-fb:100 ccm fir\n"
"a=rtcp-fb:100 nack\n"
"a=rtcp-fb:100 nack pli\n"
"a=rtcp-fb:100 goog-remb\n"
"a=rtcp-fb:100 transport-cc\n"
"a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\n"
"a=rtpmap:102 red/90000\n"
"a=rtpmap:127 ulpfec/90000\n"
"a=rtpmap:97 rtx/90000\n"
"a=fmtp:97 apt=96\n"
"a=rtpmap:99 rtx/90000\n"
"a=fmtp:99 apt=98\n"
"a=rtpmap:101 rtx/90000\n"
"a=fmtp:101 apt=100\n"
"a=rtpmap:125 rtx/90000\n"
"a=fmtp:125 apt=102\n"
"a=ssrc-group:FID 2580761338 611523443\n"
"a=ssrc:2580761338 cname:iSkJ2vn5cYYubTve\n"
"a=ssrc:2580761338 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n"
"a=ssrc:2580761338 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n"
"a=ssrc:2580761338 label:bf270496-a23e-47b5-b901-ef23096cd961\n"
"a=ssrc:611523443 cname:iSkJ2vn5cYYubTve\n"
"a=ssrc:611523443 msid:616cfbb1-33a3-4d8c-8275-a199d6005549 bf270496-a23e-47b5-b901-ef23096cd961\n"
"a=ssrc:611523443 mslabel:616cfbb1-33a3-4d8c-8275-a199d6005549\n"
"a=ssrc:611523443 label:bf270496-a23e-47b5-b901-ef23096cd961\n"
"a=candidate:3575467457 1 udp 2113937151 10.15.83.23 57857 typ host generation 0 ufrag 6R0z network-cost 999\n"
"m=application 9 DTLS/SCTP 5000\n"
"c=IN IP4 0.0.0.0\n"
"a=ice-ufrag:sXJ3\n"
"a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV\n"
"a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79\n"
"a=setup:actpass\n"
"a=mid:data\n"
"a=sctpmap:5000 webrtc-datachannel 1024\n"
"a=sctp-port:5000";
RtcSessionSdp sdp1;
sdp1.parse(str1);
RtcSessionSdp sdp2;
sdp2.parse(str2);
for (auto media : sdp1.medias) {
InfoL << getRtpDirectionString(media.getDirection());
}
for (auto media : sdp2.medias) {
InfoL << getRtpDirectionString(media.getDirection());
}
InfoL << sdp1.toString();
InfoL << sdp2.toString();
RtcSession session1;
session1.loadFrom(str1);
RtcSession session2;
session2.loadFrom(str2);
DebugL << session1.toString();
DebugL << session2.toString();
}
int start_main(int argc,char *argv[]) { int start_main(int argc,char *argv[]) {
{ {
CMD_main cmd_main; CMD_main cmd_main;
...@@ -414,7 +267,6 @@ int start_main(int argc,char *argv[]) { ...@@ -414,7 +267,6 @@ int start_main(int argc,char *argv[]) {
}); });
} }
// test_sdp();
uint16_t shellPort = mINI::Instance()[Shell::kPort]; uint16_t shellPort = mINI::Instance()[Shell::kPort];
uint16_t rtspPort = mINI::Instance()[Rtsp::kPort]; uint16_t rtspPort = mINI::Instance()[Rtsp::kPort];
uint16_t rtspsPort = mINI::Instance()[Rtsp::kSSLPort]; uint16_t rtspsPort = mINI::Instance()[Rtsp::kSSLPort];
......
src/Extension/CommonRtp.cpp
...@@ -69,7 +69,7 @@ CommonRtpEncoder::CommonRtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_si ...@@ -69,7 +69,7 @@ CommonRtpEncoder::CommonRtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_si
} }
void CommonRtpEncoder::inputFrame(const Frame::Ptr &frame){ void CommonRtpEncoder::inputFrame(const Frame::Ptr &frame){
auto stamp = frame->dts(); auto stamp = frame->pts();
auto ptr = frame->data() + frame->prefixSize(); auto ptr = frame->data() + frame->prefixSize();
auto len = frame->size() - frame->prefixSize(); auto len = frame->size() - frame->prefixSize();
auto remain_size = len; auto remain_size = len;
......
src/Rtsp/Rtsp.cpp
...@@ -521,6 +521,10 @@ RtpHeader* RtpPacket::getHeader(){ ...@@ -521,6 +521,10 @@ RtpHeader* RtpPacket::getHeader(){
return (RtpHeader*)(data() + RtpPacket::kRtpTcpHeaderSize); return (RtpHeader*)(data() + RtpPacket::kRtpTcpHeaderSize);
} }
string RtpPacket::dumpString() const{
return ((RtpPacket *) this)->getHeader()->dumpString(size() - RtpPacket::kRtpTcpHeaderSize);
}
uint16_t RtpPacket::getSeq(){ uint16_t RtpPacket::getSeq(){
return ntohs(getHeader()->seq); return ntohs(getHeader()->seq);
} }
......
src/Rtsp/Rtsp.h
...@@ -150,7 +150,11 @@ public: ...@@ -150,7 +150,11 @@ public:
kRtpTcpHeaderSize = 4 kRtpTcpHeaderSize = 4
}; };
//获取rtp头
RtpHeader* getHeader(); RtpHeader* getHeader();
//打印调试信息
string dumpString() const;
//主机字节序的seq //主机字节序的seq
uint16_t getSeq(); uint16_t getSeq();
//主机字节序的时间戳，已经转换为毫秒 //主机字节序的时间戳，已经转换为毫秒
......
src/Rtsp/RtspMediaSourceImp.h
...@@ -56,7 +56,11 @@ public: ...@@ -56,7 +56,11 @@ public:
//需要解复用rtp //需要解复用rtp
key_pos = _demuxer->inputRtp(rtp); key_pos = _demuxer->inputRtp(rtp);
} }
RtspMediaSource::onWrite(std::move(rtp), key_pos); GET_CONFIG(bool, directProxy, Rtsp::kDirectProxy);
if (directProxy) {
//直接代理模式才直接使用原始rtp
RtspMediaSource::onWrite(std::move(rtp), key_pos);
}
} }
/** /**
...@@ -72,8 +76,10 @@ public: ...@@ -72,8 +76,10 @@ public:
* @param enableMP4 是否mp4录制 * @param enableMP4 是否mp4录制
*/ */
void setProtocolTranslation(bool enableHls,bool enableMP4){ void setProtocolTranslation(bool enableHls,bool enableMP4){
//不重复生成rtsp GET_CONFIG(bool, directProxy, Rtsp::kDirectProxy);
_muxer = std::make_shared<MultiMediaSourceMuxer>(getVhost(), getApp(), getId(), _demuxer->getDuration(), false, true, enableHls, enableMP4); //开启直接代理模式时，rtsp直接代理，不重复产生；但是有些rtsp推流端，由于sdp中已有sps pps，rtp中就不再包括sps pps,
//导致rtc无法播放，所以在rtsp推流rtc播放时，建议关闭直接代理模式
_muxer = std::make_shared<MultiMediaSourceMuxer>(getVhost(), getApp(), getId(), _demuxer->getDuration(), !directProxy, true, enableHls, enableMP4);
_muxer->setMediaListener(getListener()); _muxer->setMediaListener(getListener());
_muxer->setTrackListener(static_pointer_cast<RtspMediaSourceImp>(shared_from_this())); _muxer->setTrackListener(static_pointer_cast<RtspMediaSourceImp>(shared_from_this()));
//让_muxer对象拦截一部分事件(比如说录像相关事件) //让_muxer对象拦截一部分事件(比如说录像相关事件)
......
webrtc/DtlsTransport.cpp
#define MS_CLASS "RTC::DtlsTransport" /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#define MS_CLASS "RTC::DtlsTransport"
// #define MS_LOG_DEV_LEVEL 3 // #define MS_LOG_DEV_LEVEL 3
#include "DtlsTransport.hpp" #include "DtlsTransport.hpp"
......
webrtc/DtlsTransport.hpp
#ifndef MS_RTC_DTLS_TRANSPORT_HPP /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_DTLS_TRANSPORT_HPP
#define MS_RTC_DTLS_TRANSPORT_HPP #define MS_RTC_DTLS_TRANSPORT_HPP
#include "SrtpSession.hpp" #include "SrtpSession.hpp"
......
webrtc/IceServer.cpp
#define MS_CLASS "RTC::IceServer" /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#define MS_CLASS "RTC::IceServer"
// #define MS_LOG_DEV_LEVEL 3 // #define MS_LOG_DEV_LEVEL 3
#include <utility> #include <utility>
......
webrtc/IceServer.hpp
#ifndef MS_RTC_ICE_SERVER_HPP /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_ICE_SERVER_HPP
#define MS_RTC_ICE_SERVER_HPP #define MS_RTC_ICE_SERVER_HPP
#include "StunPacket.hpp" #include "StunPacket.hpp"
......
webrtc/Sdp.cpp
// /*
// Created by xzl on 2021/3/27. * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
// *
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#include "Sdp.h" #include "Sdp.h"
#include "Common/Parser.h" #include "Common/Parser.h"
......
webrtc/Sdp.h
// /*
// Created by xzl on 2021/3/27. * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
// *
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#ifndef ZLMEDIAKIT_SDP_H #ifndef ZLMEDIAKIT_SDP_H
#define ZLMEDIAKIT_SDP_H #define ZLMEDIAKIT_SDP_H
......
webrtc/SrtpSession.cpp
#define MS_CLASS "RTC::SrtpSession" /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#define MS_CLASS "RTC::SrtpSession"
// #define MS_LOG_DEV_LEVEL 3 // #define MS_LOG_DEV_LEVEL 3
#include "SrtpSession.hpp" #include "SrtpSession.hpp"
...@@ -226,7 +244,7 @@ namespace RTC ...@@ -226,7 +244,7 @@ namespace RTC
} }
std::memcpy(EncryptBuffer, *data, *len); std::memcpy(EncryptBuffer, *data, *len);
EncryptBuffer[1] |= (pt & 0x7F); EncryptBuffer[1] = (pt & 0x7F) | (EncryptBuffer[1] & 0x80);
srtp_err_status_t err = srtp_err_status_t err =
srtp_protect(this->session, static_cast<void*>(EncryptBuffer), reinterpret_cast<int*>(len)); srtp_protect(this->session, static_cast<void*>(EncryptBuffer), reinterpret_cast<int*>(len));
......
webrtc/SrtpSession.hpp
#ifndef MS_RTC_SRTP_SESSION_HPP /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_SRTP_SESSION_HPP
#define MS_RTC_SRTP_SESSION_HPP #define MS_RTC_SRTP_SESSION_HPP
#include "Utils.hpp" #include "Utils.hpp"
......
webrtc/StunPacket.cpp
#define MS_CLASS "RTC::StunPacket" /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#define MS_CLASS "RTC::StunPacket"
// #define MS_LOG_DEV_LEVEL 3 // #define MS_LOG_DEV_LEVEL 3
#include "StunPacket.hpp" #include "StunPacket.hpp"
......
webrtc/StunPacket.hpp
#ifndef MS_RTC_STUN_PACKET_HPP /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_RTC_STUN_PACKET_HPP
#define MS_RTC_STUN_PACKET_HPP #define MS_RTC_STUN_PACKET_HPP
......
webrtc/Utils.hpp
#ifndef MS_UTILS_HPP /**
ISC License
Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef MS_UTILS_HPP
#define MS_UTILS_HPP #define MS_UTILS_HPP
#if defined(_WIN32) #if defined(_WIN32)
......
webrtc/WebRtcTransport.cpp
#include "WebRtcTransport.h" /*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#include "WebRtcTransport.h"
#include <iostream> #include <iostream>
#include "Rtcp/Rtcp.h" #include "Rtcp/Rtcp.h"
#include "Rtsp/RtpReceiver.h" #include "Rtsp/RtpReceiver.h"
...@@ -583,7 +593,7 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp ...@@ -583,7 +593,7 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
auto &pt = _send_rtp_pt[rtp->type]; auto &pt = _send_rtp_pt[rtp->type];
if (!pt) { if (pt == 0xFF) {
//忽略，对方不支持该编码类型 //忽略，对方不支持该编码类型
return; return;
} }
......
webrtc/WebRtcTransport.h
#pragma once /*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#pragma once
#include <memory> #include <memory>
#include <string> #include <string>
...@@ -202,7 +212,7 @@ private: ...@@ -202,7 +212,7 @@ private:
//rtc rtp推流的视频ssrc //rtc rtp推流的视频ssrc
uint32_t _recv_video_ssrc; uint32_t _recv_video_ssrc;
//记录协商的rtp的pt类型 //记录协商的rtp的pt类型
uint8_t _send_rtp_pt[2] = {0, 0}; uint8_t _send_rtp_pt[2] = {0xFF, 0xFF};
//复合udp端口，接收一切rtp与rtcp //复合udp端口，接收一切rtp与rtcp
Socket::Ptr _socket; Socket::Ptr _socket;
//推流的rtsp源 //推流的rtsp源
......
webrtc/readme.md 0 → 100644
# 致谢与声明
本文件夹下部分文件提取自[MediaSoup](https://github.com/versatica/mediasoup) ，分别为：
- ice相关功能：
- IceServer.cpp
- IceServer.hpp
- StunPacket.cpp
- StunPacket.hpp
- Utils.hpp
- dtls相关功能：
- DtlsTransport.cpp
- DtlsTransport.hpp
- srtp相关功能：
- SrtpSession.cpp
- SrtpSession.hpp
以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者，
用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。
同时,在此也感谢开源项目[easy_webrtc_server](https://github.com/Mihawk086/easy_webrtc_server) 及作者，
在集成MediaSoup相关代码前期，主要参考这个项目。
另外，感谢[big panda](<2381267071@qq.com>) 开发并贡献的webrtc js测试客户端(www/webrtc目录下文件)，
其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js
# 现状与规划
ZLMediaKit的WebRTC相关功能目前仅供测试与开发，现在还不成熟，后续主要工作有：
- 1、完善webrtc rtcp相关功能，包括丢包重传、带宽检测等功能。
- 2、实现rtp重传等相关功能。
- 3、实现simulecast相关功能。
- 4、fec、rtp扩展等其他功能。
- 5、如果精力允许，逐步替换MediaSoup相关代码，改用自有版权代码。
www/webrtc/readme.txt 0 → 100644
感谢[big panda](<2381267071@qq.com>) 开发并贡献此webrtc js测试客户端,
其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js
\ No newline at end of file
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