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WHIP: Fix bug for converting WHIP to RTMP/HLS. v5.0.208 (#3920)

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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>
This commit is contained in:
john 2024-02-06 14:22:33 +08:00
parent ba150beb0f
commit 732569f792
7 changed files with 96 additions and 31 deletions
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4
trunk/conf/full.conf
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@ -430,6 +430,10 @@ rtc_server {
# * Retrieve server IP automatically, from all network interfaces.
# $CANDIDATE Read the IP from ENV variable, use * if not set.
# x.x.x.x A specified IP address or DNS name, use * if 0.0.0.0.
# You can also set the candidate by the query string eip, note that you can also set the UDP port,
# for example:
# http://locahost:1985/rtc/v1/whip/?app=live&stream=livestream&eip=192.168.3.11
# http://locahost:1985/rtc/v1/whip/?app=live&stream=livestream&eip=192.168.3.11:18000
# @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
# Overwrite by env SRS_RTC_SERVER_CANDIDATE