Commit Graph

4 Commits

Author SHA1 Message Date
OSSRS-AI
6e93dd73b5
AI: WebRTC: Support optional msid attribute per RFC 8830. v7.0.126 (#4570) (#4572)
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
2025-11-11 10:22:31 -05:00
OSSRS-AI
bfb91f9b82
AI: WebRTC: Support G.711 (PCMU/PCMA) audio codec for WebRTC. v7.0.124 (#4075) (#4568)
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=pcmu
http://localhost:8080/players/whip.html?acodec=pcma

# Video + audio codecs
http://localhost:8080/players/whip.html?vcodec=vp9&acodec=pcmu
http://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)
2025-11-09 12:08:03 -05:00
OSSRS-AI
7fcd406a63
AI: WebRTC: Support VP9 codec for WebRTC-to-WebRTC streaming. v7.0.123 (#4548) (#4565)
VP9 is a similar codec to HEVC, but for WebRTC, VP9 works better than
AVC/HEVC in some special cases. However, SRS only support VP9 for
WebRTC, doesn't support converting it to RTMP, for RTMP only support
AVC/HEVC/AV1 and SRS cannot support transcoding.

Usage:
* Publish with VP9:
[http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&codec=vp9](http://localhost:8080/players/whip.html?codec=vp9)
* Play with VP9:
[http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream&codec=vp9](http://localhost:8080/players/whep.html?codec=vp9)
2025-11-08 12:47:31 -05:00
OSSRS-AI
2fb216e86d AI: Refine utest file rules. 2025-10-23 09:44:28 -04:00