1. When converting RTC to RTMP, it is necessary to synchronize the audio
and video timestamps. When the synchronization status changes, whether
it is unsynchronized or synchronized, print logs to facilitate
troubleshooting of such issues.
2. Chrome uses the STAP-A packet, which means a single RTP packet
contains SPS/PPS information. OBS WHIP, on the other hand, sends SPS and
PPS in separate RTP packets. Therefore, SPS and PPS are in two
independent RTP packets, and SRS needs to cache these two packets.
---------
Co-authored-by: john <hondaxiao@tencent.com>
When accessing the SRS Stack, you should log in and use a token for each
request, or utilize the HTTP API with a secret Bearer token included in
every request. The SRS Stack HTTP API proxies both /api/v1 and /rtc/v1
to the SRS HTTP API while ensuring secure authentication. Additionally,
there is a console in the SRS Stack that requires the same token to
request the SRS Stack HTTP API, which is then proxied to the SRS HTTP
API.
The SRS Stack runs SRS with the HTTP API listening at 127.0.0.1:1985 on
the local loopback interface, allowing only the SRS Stack to access it
without authentication. All other users must login and access the SRS
Stack through its interface, rather than directly accessing the SRS HTTP
API within the SRS Stack.
---------
Co-authored-by: panda <542638787@qq.com>
* fix annotation spell failed
* RTC to RTMP using SenderReport to sync av timestamp
* update pion/webrtc versio from v3.0.4 -> v3.0.13, auto config sender/receiver report
* Add rtc push flv play regression test
* Add unit test of ntp and av sync time
* Take flag CXX to makefile of utest
* Add annotation about rtc unit test
* Fix compiler error in C++98
* Add FFmpeg log callback funciton.
1. TWCC should not be passed from end to end.
2. Publisher TWCC information, should be ignore when pass to player
3. Player should regenerate its own TWCC.
