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https://github.com/ossrs/srs.git
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ce2ce1542f
When debugging the RTMP protocol, we can capture packets using tcpdump and then replay the pcap file. For example: ```bash cd ~/git/srs/trunk/3rdparty/srs-bench/pcap tcpdump -i any -w t.pcap tcp port 1935 go run . -f ./t.pcap -s 127.0.0.1:1935 ``` However, sometimes due to poor network conditions between the server and the client, there may be many retransmitted packets. In such cases, setting up a transparent TCP proxy that listens on port 1935 and forwards to port 19350 can be a solution: ```bash ./objs/srs -c conf/origin.conf cd 3rdparty/srs-bench/tcpproxy/ && go run main.go tcpdump -i any -w t.pcap tcp port 19350 ``` This approach allows for the implementation of packet dumping, multipoint replication, or the provision of detailed timestamps and byte information at the proxy. It enables the collection of debugging information without the need to modify the server. --------- `TRANS_BY_GPT4` --------- Co-authored-by: john <hondaxiao@tencent.com> |
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| .. | ||
| ffmpeg-4-fit | ||
| gperftools-2-fit | ||
| gprof | ||
| gtest-fit | ||
| httpx-static | ||
| libsrtp-2-fit | ||
| openssl-1.1-fit | ||
| patches | ||
| signaling | ||
| srs-bench | ||
| srt-1-fit | ||
| st-srs | ||
| openssl-OpenSSL_1_0_2u.tar.gz | ||
| opus-1.3.1.tar.gz | ||
| README.md | ||
http-parser-2.1.zip
- for srs to support http callback.
- https://github.com/nodejs/http-parser
- https://github.com/ossrs/http-parser
- https://ossrs.net/lts/zh-cn/license#http-parser
nginx-1.5.7.zip
- http://nginx.org/
- for srs to support hls streaming.
srt-1-fit srt-1.5.3.tar.gz
openssl-1.1-fit openssl-1.1.1l.tar.gz
openssl-1.1.0e.zip openssl-OpenSSL_1_0_2u.tar.gz
- http://www.openssl.org/source/openssl-1.1.0e.tar.gz
- openssl for SRS(with-ssl) RTMP complex handshake to delivery h264+aac stream.
- SRTP depends on openssl 1.0.*, so we use both ssl versions.
- https://ossrs.net/lts/zh-cn/license#openssl
libsrtp-2.3.0.tar.gz
- For WebRTC, SRTP to encrypt and decrypt RTP.
- https://github.com/cisco/libsrtp/releases/tag/v2.3.0
ffmpeg-4.2.tar.gz opus-1.3.1.tar.gz
- http://ffmpeg.org/releases/ffmpeg-4.2.tar.gz
- https://github.com/xiph/opus/releases/tag/v1.3.1
- To support RTMP/WebRTC transcoding.
- https://ossrs.net/lts/zh-cn/license#ffmpeg
gtest-fit
- google test framework.
- https://github.com/google/googletest/releases/tag/release-1.11.0
gperftools-2-fit
- gperf tools for performance benchmark.
- https://github.com/gperftools/gperftools/releases/tag/gperftools-2.9.1
st-srs st-1.9.zip state-threads state-threads-1.9.1.tar.gz
- Patched ST from https://github.com/ossrs/state-threads
- https://ossrs.net/lts/zh-cn/license#state-threads
JSON
USRSCTP
links:
- state-threads: https://github.com/ossrs/state-threads
- x264: ftp://ftp.videolan.org/pub/videolan/x264/snapshots/x264-snapshot-20131129-2245-stable.tar.bz2
- lame: http://nchc.dl.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz
- yasm: http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz
- speex: http://downloads.xiph.org/releases/speex/speex-1.2rc1.tar.gz