Fix issue #4570 by supporting optional `msid` attribute in WebRTC SDP
negotiation, enabling compatibility with libdatachannel and other
clients that don't include msid information.
SRS failed to negotiate WebRTC connections from libdatachannel clients
because:
- libdatachannel SDP lacks `a=ssrc:XX msid:stream_id track_id`
attributes
- SRS required msid information to create track descriptions
- According to RFC 8830, the msid attribute and its appdata (track_id)
are **optional**
If diligently look at the SDP generated by libdatachannel:
```
a=ssrc:42 cname:video-send
a=ssrc:43 cname:audio-send
```
It's deliberately missing the `a=ssrc:XX msid:stream_id track_id` line,
comparing that with this one:
```
a=ssrc:42 cname:video-send
a=ssrc:42 msid:stream_id video_track_id
a=ssrc:43 cname:audio-send
a=ssrc:43 msid:stream_id audio_track_id
```
In such a situation, to keep compatible with libdatachannel, if no msid
line in sdp, SRS comprehensively and consistently uses:
* app/stream as stream_id, such as live/livestream
* type=video|audio, cname, and ssrc as track_id, such as
track-video-video-send-43
for issue #4418, #4151, #4076 .DVR Missing First Few Seconds of
Audio/Video
### Root Cause
When recording WebRTC streams to FLV files using DVR, the first 4-6
seconds of audio/video are missing. This occurs because:
1. **Packets are discarded before A/V sync is available**: The
RTC-to-RTMP conversion pipeline actively discards all RTP packets when
avsync_time <= 0.
2. **Original algorithm requires 2 RTCP SR packets**: The previous
implementation needed to receive two RTCP Sender Report (SR) packets
before it could calculate the rate for audio/video synchronization
timestamp conversion.
3. **Delay causes packet loss**: Since RTCP SR packets typically arrive
every 2-3 seconds, waiting for 2 SRs means 4-6 seconds of packets are
discarded before A/V sync becomes available.
4. **Audio SR arrives slower than video SR**: As reported in the issue,
video RTCP SR packets arrive much faster than audio SR packets. This
asymmetry causes audio packets to be discarded for a longer period,
resulting in the audio loss observed in DVR recordings.
### Solution
1. **Initialize rate from SDP**: Use the sample rate from SDP (Session
Description Protocol) to calculate the initial rate immediately when the
track is created.
Audio (Opus): 48000 Hz → rate = 48 (RTP units per millisecond)
Video (H.264/H.265): 90000 Hz → rate = 90 (RTP units per millisecond)
2. **Enable immediate A/V sync:** With the SDP rate available,
cal_avsync_time() can calculate valid timestamps from the very first RTP
packet, eliminating packet loss.
3. **Smooth transition to precise rate**: After receiving the 2nd RTCP
SR, update to the precisely calculated rate based on actual RTP/NTP
timestamp mapping.
## Configuration
Added new configuration option `init_rate_from_sdp` in the RTC vhost
section:
```nginx
vhost rtc.vhost.srs.com {
rtc {
# Whether initialize RTP rate from SDP sample rate for immediate A/V sync.
# When enabled, the RTP rate (units per millisecond) is initialized from the SDP
# sample rate (e.g., 90 for video 90kHz, 48 for audio 48kHz) before receiving
# 2 RTCP SR packets. This allows immediate audio/video synchronization.
# The rate will be updated to a more precise value after receiving the 2nd SR.
# Overwrite by env SRS_VHOST_RTC_INIT_RATE_FROM_SDP for all vhosts.
# Default: off
init_rate_from_sdp off;
}
}
```
**⚠️ Important Note**: This config defaults to **off** because:
- ✅ When **enabled**: Fixes the audio loss problem (no missing first 4-6
seconds)
- ❌ When **enabled**: VLC on macOS cannot play the video properly
- ✅ Other platforms work fine (Windows, Linux)
- ✅ FFplay works fine on all platforms
Users experiencing audio loss in DVR recordings can enable this option
if they don't need VLC macOS compatibility. We're investigating the VLC
macOS issue to make this feature safe to enable by default in the
future.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
