**Problem**: Newly created sources (RTMP/SRT/RTC/RTSP) were being
immediately marked as "dead" and deleted by the cleanup timer before
publishers could connect, causing "new live source, dead=1" errors.
**Root Cause**: All source constructors initialized `stream_die_at_ =
0`, causing `stream_is_dead()` to return `true` immediately since
current time was always greater than `0 + 3 seconds`.
**Solution**: Changed all four source constructors to initialize
`stream_die_at_ = srs_time_now_cached()`, giving newly created sources a
proper 3-second grace period before cleanup.
This PR adds G.711 (PCMU/PCMA) audio codec support for WebRTC in SRS,
enabling relay-only streaming of G.711 audio between WebRTC clients via
WHIP/WHEP. G.711 is a widely-used, royalty-free audio codec with
excellent compatibility across VoIP systems, IP cameras, and legacy
telephony equipment.
Fixes#4075
Many IP cameras, VoIP systems, and IoT devices use G.711 (PCMU/PCMA) as
their default audio codec. Previously, SRS only supported Opus for
WebRTC audio, requiring transcoding or rejecting G.711 streams entirely.
This PR enables direct relay of G.711 audio streams in WebRTC, similar
to how VP9/AV1 video codecs are supported.
Enhanced WHIP/WHEP players with URL-based codec selection:
```
# Audio codec only
http://localhost:8080/players/whip.html?acodec=pcmuhttp://localhost:8080/players/whip.html?acodec=pcma
# Video + audio codecs
http://localhost:8080/players/whip.html?vcodec=vp9&acodec=pcmuhttp://localhost:8080/players/whep.html?vcodec=h264&acodec=pcma
# Backward compatible (codec = vcodec)
http://localhost:8080/players/whip.html?codec=vp9
```
Testing
```bash
# Build and run unit tests
cd trunk
make utest -j && ./objs/srs_utest
# Test with WHIP player
# 1. Start SRS server
./objs/srs -c conf/rtc.conf
# 2. Open WHIP publisher with PCMU audio
http://localhost:8080/players/whip.html?acodec=pcmu
# 3. Open WHEP player to receive stream
http://localhost:8080/players/whep.html
```
## Related Issues
- Fixes#4075 - WebRTC G.711A Audio Codec Support
- Related to #4548 - VP9 codec support (similar relay-only pattern)
1. It cannot retrieve codec information on `Firefox` by
`getSenders/getReceivers`
2. It can retrieve codec information on `Chrome` by `getReceivers`, but
incorrect, like this:
3. So, we retrieve codec information from `getStats`, and it works well.
4. The timer is used because sometimes the codec cannot be retrieved
when `iceGatheringState` is `complete`.
5. Testing has been completed on the browsers listed below.
- [x] Chrome
- [x] Edge
- [x] Safari
- [x] Firefox
---------
Co-authored-by: winlin <winlinvip@gmail.com>
## How to reproduce?
1. Refer this commit, which contains the web demo to capture screen as
video stream through RTC.
2. Copy the `trunk/research/players/whip.html` and
`trunk/research/players/js/srs.sdk.js` to replace the `develop` branch
source code.
3. `./configure && make`
4. `./objs/srs -c conf/rtc2rtmp.conf`
5. open `http://localhost:8080/players/whip.html?schema=http`
6. check `Screen` radio option.
7. click `publish`, then check the screen to share.
8. play the rtmp live stream: `rtmp://localhost/live/livestream`
9. check the video stuttering.
## Cause
When capture screen by the chrome web browser, which send RTP packet
with empty payload frequently, then all the cached RTP packets are
dropped before next key frame arrive in this case.
The OBS screen stream and camera stream do not have such problem.
## Add screen stream to WHIP demo
><img width="581" alt="Screenshot 2024-08-28 at 2 49 46 PM"
src="https://github.com/user-attachments/assets/9557dbd2-c799-4dfd-b336-5bbf2e4f8fb8">
---------
Co-authored-by: winlin <winlinvip@gmail.com>
