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>
In an SDK that supports RTC Opus stereo, the parameter "stereo=1" may
appear. SRS (Spatial Reference System) needs to handle this correctly
and return an answer to enable WebRTC stereo support.
---------
`TRANS_BY_GPT4`
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>
1. Ignore SDP GROUP LS.
2. Support ice in global session info.
3. Support audio codec "OPUS" or "opus".
---------
Co-authored-by: Johnny <hellojinqiang@gmail.com>
* Remove extern SrsPps* duplicate declarations
* fix(rtmp2rtc): fix video payload type for rtmp to rtc bridge (#3041)
* Revert changes not belongs to this PR.
* Fix naming issue, follow SRS style.
* Use srs_assert instead of assert.
* Fix firefox no audio issue.
Co-authored-by: winlin <winlin@vip.126.com>
1. Cache IDR frame's rtp timestamp instead of avsync timestamp.
2. Cache clock rate calculate by sender report.
3. Using srs_rtp_seq_distance instead of direct minus.
4. Add utest of av timestamp sync when duplicated sender report.
