WebRTC: playback hanging and corrupted if slice-max-size x264 parameter larger than DTLS MTU

[rajkosto]

srs version: 4.0-release branch commit 4e93696 15.jun.2021
description: i am sending RTMP stream to srs using OBS and playing back with webrtc player, playback hangs every other keyframe and errors until next keyframe if i set slice-max-size=1400 (what i usually use) in advanced x264 settings in OBS. I noticed this makes every large DTLS packet over port 8000 to come in pairs, one sized 1400 and one sized 120, so i set slice-max-size=1280 and now every DTLS packet is "not fragmented", and playback is nearly perfect, no hanging, stuttering or errors except when there actually are network errors.
is this a bug with srs handling of packets RTMP->WEBRTC or just something to be wary of ?

[dongzy08]

In file srs_kernel_rtc_rtp.hpp,
change "const int kRtpPacketSize = 1500;" to "const int kRtpPacketSize = 1400;" , or 1300.
rebuild the srs, maybe help.

[chen-guanghua]

@rajkosto according to your comment, there are three points to be discussed.

1. The slice-max-size=1400 should not make the DTLS packet size large. The increase in DTLS packet size may be caused by the certificate used during DTLS negotiation. If you provide a Wireshark capture file, it will make it easier to check the issue.

2. The DTLS packet not being fragmented may be caused by SSL_set_mtu. The MTU size should be checked.

3. The default value of kRtpPacketSize is 1500. It should be adjusted according to the MTU and UDP packet header. We will commit a pull request to adjust kRtpPacketSize.

TRANS_BY_GPT3