Prevent loss of initial packet(s) in RTP streams (#387)

The method by which we detected non-signaled SSRCs would silently drop a
few packets. This is particularly problematic for a video stream since
these initial packets usually contain the first part of a keyframe to
start the stream.

Author: lookback-hugotunius

webrtc/CHANGELOG.md

@@ -5,6 +5,8 @@
 * Added support for insecure/deprecated signature verification algorithms, opt in via `SettingsEngine::allow_insecure_verification_algorithm` [#342](https://github.com/webrtc-rs/webrtc/pull/342).
 * Make RTCRtpCodecCapability::payloader_for_codec public API [#349](https://github.com/webrtc-rs/webrtc/pull/349).
 * Fixed a panic in `calculate_rtt_ms` [#350](https://github.com/webrtc-rs/webrtc/pull/350).
+* Fixed `TrackRemote` missing at least the first, sometimes more, RTP packet during probing. [#387](https://github.com/webrtc-rs/webrtc/pull/387)
+
 ### Breaking changes
 
 * Change `RTCPeerConnection::on_track` callback signature to `|track: Arc<TrackRemote>, receiver: Arc<RTCRtpReceiver>, transceiver: Arc<RTCRtpTransceiver>|` [#355](https://github.com/webrtc-rs/webrtc/pull/355).

webrtc/src/dtls_transport/mod.rs

@@ -582,10 +582,10 @@ impl RTCDtlsTransport {
         stream_info: &StreamInfo,
         interceptor: &Arc<dyn Interceptor + Send + Sync>,
     ) -> Result<(
-        Option<Arc<srtp::stream::Stream>>,
-        Option<Arc<dyn RTPReader + Send + Sync>>,
-        Option<Arc<srtp::stream::Stream>>,
-        Option<Arc<dyn RTCPReader + Send + Sync>>,
+        Arc<srtp::stream::Stream>,
+        Arc<dyn RTPReader + Send + Sync>,
+        Arc<srtp::stream::Stream>,
+        Arc<dyn RTCPReader + Send + Sync>,
     )> {
         let srtp_session = self
             .get_srtp_session()
@@ -608,10 +608,10 @@ impl RTCDtlsTransport {
         let rtcp_interceptor = interceptor.bind_rtcp_reader(rtcp_stream_reader).await;
 
         Ok((
-            Some(rtp_read_stream),
-            Some(rtp_interceptor),
-            Some(rtcp_read_stream),
-            Some(rtcp_interceptor),
+            rtp_read_stream,
+            rtp_interceptor,
+            rtcp_read_stream,
+            rtcp_interceptor,
         ))
     }
 }