rustrtc 0.3.49

A high-performance implementation of WebRTC
Documentation 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232

use anyhow::Result;
use bytes::Bytes;
use rustrtc::media::track::MediaStreamTrack;
use rustrtc::{MediaKind, PeerConnection, RtcConfiguration, TransceiverDirection};
use std::sync::Arc;
use std::time::Duration;
use tokio::time::timeout;
use webrtc::api::APIBuilder;
use webrtc::api::media_engine::MediaEngine;
use webrtc::interceptor::registry::Registry;
use webrtc::peer_connection::configuration::RTCConfiguration as WebrtcConfiguration;
use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
use webrtc::rtp::header::{Extension, Header};
use webrtc::rtp::packet::Packet;
use webrtc::rtp_transceiver::RTCRtpEncodingParameters;
use webrtc::rtp_transceiver::RTCRtpTransceiverInit;
use webrtc::rtp_transceiver::rtp_codec::RTCRtpHeaderExtensionCapability;
use webrtc::rtp_transceiver::rtp_codec::{RTCRtpCodecCapability, RTPCodecType};
use webrtc::rtp_transceiver::rtp_transceiver_direction::RTCRtpTransceiverDirection;
use webrtc::track::track_local::TrackLocalWriter;
use webrtc::track::track_local::track_local_static_rtp::TrackLocalStaticRTP;

#[tokio::test]
async fn test_simulcast_ingest_and_switch() -> Result<()> {
    rustls::crypto::CryptoProvider::install_default(rustls::crypto::ring::default_provider()).ok();

    let _ = env_logger::builder().is_test(true).try_init();

    // 1. Create RustRTC PeerConnection (SFU)
    let rust_config = RtcConfiguration::default();
    let rust_pc = PeerConnection::new(rust_config);

    // Add a transceiver to receive Video (Simulcast)
    let transceiver = rust_pc.add_transceiver(MediaKind::Video, TransceiverDirection::RecvOnly);

    // 2. Create WebRTC PeerConnection (Client)
    let mut m = MediaEngine::default();
    m.register_default_codecs()?;
    m.register_header_extension(
        RTCRtpHeaderExtensionCapability {
            uri: "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id".to_owned(),
        },
        RTPCodecType::Video,
        Some(RTCRtpTransceiverDirection::Sendrecv),
    )?;
    m.register_header_extension(
        RTCRtpHeaderExtensionCapability {
            uri: "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id".to_owned(),
        },
        RTPCodecType::Video,
        Some(RTCRtpTransceiverDirection::Sendrecv),
    )?;
    m.register_header_extension(
        RTCRtpHeaderExtensionCapability {
            uri: "urn:ietf:params:rtp-hdrext:sdes:mid".to_owned(),
        },
        RTPCodecType::Video,
        Some(RTCRtpTransceiverDirection::Sendrecv),
    )?;
    let registry = Registry::new();
    // registry = webrtc::api::interceptor_registry::register_default_interceptors(registry, &mut m)?;
    let api = APIBuilder::new()
        .with_media_engine(m)
        .with_interceptor_registry(registry)
        .build();

    let webrtc_config = WebrtcConfiguration::default();
    let webrtc_pc = api.new_peer_connection(webrtc_config).await?;

    // Create a video track using TrackLocalStaticRTP
    let codec = RTCRtpCodecCapability {
        mime_type: webrtc::api::media_engine::MIME_TYPE_VP8.to_owned(),
        clock_rate: 90000,
        channels: 0,
        ..Default::default()
    };
    let video_track = Arc::new(TrackLocalStaticRTP::new(
        codec,
        "video".to_string(),
        "webrtc_stream".to_string(),
    ));

    let _webrtc_transceiver = webrtc_pc
        .add_transceiver_from_track(
            Arc::clone(&video_track)
                as Arc<dyn webrtc::track::track_local::TrackLocal + Send + Sync>,
            Some(RTCRtpTransceiverInit {
                direction: RTCRtpTransceiverDirection::Sendonly,
                send_encodings: vec![
                    RTCRtpEncodingParameters {
                        rid: "hi".into(),
                        ..Default::default()
                    },
                    RTCRtpEncodingParameters {
                        rid: "mid".into(),
                        ..Default::default()
                    },
                    RTCRtpEncodingParameters {
                        rid: "lo".into(),
                        ..Default::default()
                    },
                ],
            }),
        )
        .await?;

    // 3. WebRTC creates Offer
    let offer = webrtc_pc.create_offer(None).await?;
    let mut gather_complete = webrtc_pc.gathering_complete_promise().await;
    webrtc_pc.set_local_description(offer.clone()).await?;
    let _ = gather_complete.recv().await;

    let offer = webrtc_pc.local_description().await.unwrap();
    println!("WebRTC Offer SDP:\n{}", offer.sdp);

    // Convert WebRTC SDP to RustRTC SDP
    let mut offer_sdp = offer.sdp;
    if offer_sdp.contains("m=video") {
        offer_sdp = offer_sdp.replace("a=sendonly", "a=sendonly\r\na=rid:hi send\r\na=rid:mid send\r\na=rid:lo send\r\na=simulcast:send hi;mid;lo");
    }
    let rust_offer = rustrtc::SessionDescription::parse(rustrtc::SdpType::Offer, &offer_sdp)?;

    // 4. RustRTC sets Remote Description
    rust_pc.set_remote_description(rust_offer).await?;

    // 5. RustRTC creates Answer
    let answer = rust_pc.create_answer().await?;
    rust_pc.set_local_description(answer.clone())?;

    // Convert RustRTC SDP to WebRTC SDP
    let answer_sdp = answer.to_sdp_string();
    println!("RustRTC Answer SDP:\n{}", answer_sdp);
    let webrtc_answer = RTCSessionDescription::answer(answer_sdp)?;

    // 6. WebRTC sets Remote Description
    webrtc_pc.set_remote_description(webrtc_answer).await?;

    // 7. Wait for connection
    rust_pc.wait_for_connected().await?;

    // 8. Start sending data from WebRTC
    let video_track_clone = video_track.clone();
    tokio::spawn(async move {
        let mut sequence_number = 0u16;
        loop {
            tokio::time::sleep(Duration::from_millis(33)).await;

            // Send packets for each RID
            for rid in &["hi", "mid", "lo"] {
                let packet = Packet {
                    header: Header {
                        version: 2,
                        payload_type: 96,
                        sequence_number,
                        timestamp: sequence_number as u32 * 3000,
                        ssrc: 0,
                        extensions: vec![Extension {
                            id: 1, // Negotiated ID for urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
                            payload: Bytes::from(rid.as_bytes().to_vec()),
                        }],
                        ..Default::default()
                    },
                    payload: vec![0u8; 100].into(),
                };

                if let Err(e) = video_track_clone.write_rtp(&packet).await {
                    println!("Failed to write RTP: {}", e);
                    return;
                }
            }
            sequence_number = sequence_number.wrapping_add(1);
            // println!("Wrote RTP packets"); // Reduce spam
        }
    });

    // 9. Verify RustRTC receives Simulcast tracks
    let receiver = transceiver.receiver().unwrap();

    // Wait a bit for packets to flow and tracks to be created
    tokio::time::sleep(Duration::from_secs(2)).await;

    let simulcast_rids = receiver.get_simulcast_rids();
    println!("Received Simulcast RIDs: {:?}", simulcast_rids);

    assert_eq!(simulcast_rids.len(), 3);
    assert!(simulcast_rids.contains(&"hi".to_string()));
    assert!(simulcast_rids.contains(&"mid".to_string()));
    assert!(simulcast_rids.contains(&"lo".to_string()));

    // 10. Create SelectorTrack and switch
    let hi_track = receiver.simulcast_track("hi").expect("hi track not found");
    let mid_track = receiver
        .simulcast_track("mid")
        .expect("mid track not found");
    let lo_track = receiver.simulcast_track("lo").expect("lo track not found");

    println!(
        "Track IDs: hi={}, mid={}, lo={}",
        hi_track.id(),
        mid_track.id(),
        lo_track.id()
    );

    let selector = rustrtc::media::track::SelectorTrack::new(lo_track.clone());

    // Verify we can switch
    println!("Switching to mid");
    selector.switch_to(mid_track).await.unwrap();

    println!("Waiting for packet on mid...");
    match timeout(Duration::from_secs(2), selector.recv()).await {
        Ok(Ok(_)) => println!("Received packet on mid"),
        Ok(Err(e)) => println!("Error receiving on mid: {}", e),
        Err(_) => println!("Timeout receiving on mid"),
    }

    println!("Switching to hi");
    selector.switch_to(hi_track).await.unwrap();

    println!("Waiting for packet on hi...");
    match timeout(Duration::from_secs(2), selector.recv()).await {
        Ok(Ok(_)) => println!("Received packet on hi"),
        Ok(Err(e)) => println!("Error receiving on hi: {}", e),
        Err(_) => println!("Timeout receiving on hi"),
    }

    // Cleanup
    rust_pc.close();
    webrtc_pc.close().await?;

    Ok(())
}