fix: webrtc-rs#148

Author: univeroseroc

src/dtls_transport/dtls_fingerprint.rs

@@ -1,7 +1,7 @@
 use serde::{Deserialize, Serialize};
 
 /// DTLSFingerprint specifies the hash function algorithm and certificate
-/// fingerprint as described in https://tools.ietf.org/html/rfc4572.
+/// fingerprint as described in <https://tools.ietf.org/html/rfc4572>.
 #[derive(Default, Debug, Clone, Serialize, Deserialize)]
 pub struct RTCDtlsFingerprint {
     /// Algorithm specifies one of the the hash function algorithms defined in
@@ -10,6 +10,6 @@ pub struct RTCDtlsFingerprint {
 
     /// Value specifies the value of the certificate fingerprint in lowercase
     /// hex string as expressed utilizing the syntax of 'fingerprint' in
-    /// https://tools.ietf.org/html/rfc4572#section-5.
+    /// <https://tools.ietf.org/html/rfc4572#section-5>.
     pub value: String,
 }

src/dtls_transport/dtls_role.rs

@@ -24,7 +24,7 @@ pub enum DTLSRole {
     Server = 3,
 }
 
-/// https://tools.ietf.org/html/rfc5763
+/// <https://tools.ietf.org/html/rfc5763>
 /// The answerer MUST use either a
 /// setup attribute value of setup:active or setup:passive.  Note that
 /// if the answerer uses setup:passive, then the DTLS handshake will

src/ice_transport/ice_candidate.rs

@@ -142,7 +142,7 @@ impl RTCIceCandidate {
     }
 
     /// to_json returns an ICECandidateInit
-    /// as indicated by the spec https://w3c.github.io/webrtc-pc/#dom-rtcicecandidate-tojson
+    /// as indicated by the spec <https://w3c.github.io/webrtc-pc/#dom-rtcicecandidate-tojson>
     pub async fn to_json(&self) -> Result<RTCIceCandidateInit> {
         let candidate = self.to_ice().await?;
 

src/ice_transport/ice_candidate_type.rs

@@ -8,7 +8,7 @@ pub enum RTCIceCandidateType {
     Unspecified,
 
     /// ICECandidateTypeHost indicates that the candidate is of Host type as
-    /// described in https://tools.ietf.org/html/rfc8445#section-5.1.1.1. A
+    /// described in <https://tools.ietf.org/html/rfc8445#section-5.1.1.1>. A
     /// candidate obtained by binding to a specific port from an IP address on
     /// the host. This includes IP addresses on physical interfaces and logical
     /// ones, such as ones obtained through VPNs.
@@ -17,7 +17,7 @@ pub enum RTCIceCandidateType {
 
     /// ICECandidateTypeSrflx indicates the the candidate is of Server
     /// Reflexive type as described
-    /// https://tools.ietf.org/html/rfc8445#section-5.1.1.2. A candidate type
+    /// <https://tools.ietf.org/html/rfc8445#section-5.1.1.2>. A candidate type
     /// whose IP address and port are a binding allocated by a NAT for an ICE
     /// agent after it sends a packet through the NAT to a server, such as a
     /// STUN server.
@@ -32,7 +32,7 @@ pub enum RTCIceCandidateType {
     Prflx,
 
     /// ICECandidateTypeRelay indicates the the candidate is of Relay type as
-    /// described in https://tools.ietf.org/html/rfc8445#section-5.1.1.2. A
+    /// described in <https://tools.ietf.org/html/rfc8445#section-5.1.1.2>. A
     /// candidate type obtained from a relay server, such as a TURN server.
     #[serde(rename = "relay")]
     Relay,

src/ice_transport/ice_credential_type.rs

@@ -7,11 +7,11 @@ pub enum RTCIceCredentialType {
     Unspecified,
 
     /// ICECredential::Password describes username and password based
-    /// credentials as described in https://tools.ietf.org/html/rfc5389.
+    /// credentials as described in <https://tools.ietf.org/html/rfc5389>.
     Password,
 
     /// ICECredential::Oauth describes token based credential as described
-    /// in https://tools.ietf.org/html/rfc7635.
+    /// in <https://tools.ietf.org/html/rfc7635>.
     /// Not supported in WebRTC 1.0 spec
     Oauth,
 }

src/mux/mux_func.rs

@@ -18,7 +18,7 @@ pub fn match_range(lower: u8, upper: u8) -> MatchFunc {
 }
 
 /// MatchFuncs as described in RFC7983
-/// https://tools.ietf.org/html/rfc7983
+/// <https://tools.ietf.org/html/rfc7983>
 ///              +----------------+
 ///              |        [0..3] -+--> forward to STUN
 ///              |                |

src/peer_connection/mod.rs

@@ -209,7 +209,7 @@ impl RTCPeerConnection {
         let internal_rtcp_writer = Arc::clone(&internal) as Arc<dyn RTCPWriter + Send + Sync>;
         let interceptor_rtcp_writer = interceptor.bind_rtcp_writer(internal_rtcp_writer).await;
 
-        // https://w3c.github.io/webrtc-pc/#constructor (Step #2)
+        // <https://w3c.github.io/webrtc-pc/#constructor> (Step #2)
         // Some variables defined explicitly despite their implicit zero values to
         // allow better readability to understand what is happening.
         Ok(RTCPeerConnection {
@@ -241,7 +241,7 @@ impl RTCPeerConnection {
             }
         }
 
-        // https://www.w3.org/TR/webrtc/#constructor (step #3)
+        // <https://www.w3.org/TR/webrtc/#constructor> (step #3)
         if !configuration.certificates.is_empty() {
             let now = SystemTime::now();
             for cert in &configuration.certificates {
@@ -655,7 +655,7 @@ impl RTCPeerConnection {
     /// configuration of this PeerConnection object. The returned object is a
     /// copy and direct mutation on it will not take affect until set_configuration
     /// has been called with Configuration passed as its only argument.
-    /// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-getconfiguration
+    /// <https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-getconfiguration>
     pub fn get_configuration(&self) -> &RTCConfiguration {
         &self.configuration
     }
@@ -665,7 +665,7 @@ impl RTCPeerConnection {
     }
 
     /// create_offer starts the PeerConnection and generates the localDescription
-    /// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer
+    /// <https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer>
     pub async fn create_offer(
         &self,
         options: Option<RTCOfferOptions>,
@@ -806,7 +806,7 @@ impl RTCPeerConnection {
     }
 
     /// Update the PeerConnectionState given the state of relevant transports
-    /// https://www.w3.org/TR/webrtc/#rtcpeerconnectionstate-enum
+    /// <https://www.w3.org/TR/webrtc/#rtcpeerconnectionstate-enum>
     async fn update_connection_state(
         on_peer_connection_state_change_handler: &Arc<
             Mutex<Option<OnPeerConnectionStateChangeHdlrFn>>,
@@ -1229,7 +1229,7 @@ impl RTCPeerConnection {
     /// local_description returns PendingLocalDescription if it is not null and
     /// otherwise it returns CurrentLocalDescription. This property is used to
     /// determine if set_local_description has already been called.
-    /// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-localdescription
+    /// <https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-localdescription>
     pub async fn local_description(&self) -> Option<RTCSessionDescription> {
         if let Some(pending_local_description) = self.pending_local_description().await {
             return Some(pending_local_description);
@@ -1482,7 +1482,7 @@ impl RTCPeerConnection {
     /// remote_description returns pending_remote_description if it is not null and
     /// otherwise it returns current_remote_description. This property is used to
     /// determine if setRemoteDescription has already been called.
-    /// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-remotedescription
+    /// <https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-remotedescription>
     pub async fn remote_description(&self) -> Option<RTCSessionDescription> {
         self.internal.remote_description().await
     }
@@ -2049,7 +2049,7 @@ impl RTCPeerConnection {
     /// sctp returns the SCTPTransport for this PeerConnection
     ///
     /// The SCTP transport over which SCTP data is sent and received. If SCTP has not been negotiated, the value is nil.
-    /// https://www.w3.org/TR/webrtc/#attributes-15
+    /// <https://www.w3.org/TR/webrtc/#attributes-15>
     pub fn sctp(&self) -> Arc<RTCSctpTransport> {
         Arc::clone(&self.internal.sctp_transport)
     }

src/peer_connection/policy/sdp_semantics.rs

@@ -9,13 +9,13 @@ pub enum RTCSdpSemantics {
 
     /// UnifiedPlan uses unified-plan offers and answers
     /// (the default in Chrome since M72)
-    /// https://tools.ietf.org/html/draft-roach-mmusic-unified-plan-00
+    /// <https://tools.ietf.org/html/draft-roach-mmusic-unified-plan-00>
     #[serde(rename = "unified-plan")]
     UnifiedPlan = 1,
 
     /// PlanB uses plan-b offers and answers
     /// NB: This format should be considered deprecated
-    /// https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00
+    /// <https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00>
     #[serde(rename = "plan-b")]
     PlanB = 2,
 

src/peer_connection/sdp/mod.rs

@@ -905,7 +905,7 @@ pub(crate) fn rtp_extensions_from_media_description(
 /// update_sdp_origin saves sdp.Origin in PeerConnection when creating 1st local SDP;
 /// for subsequent calling, it updates Origin for SessionDescription from saved one
 /// and increments session version by one.
-/// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-25#section-5.2.2
+/// <https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-25#section-5.2.2>
 pub(crate) fn update_sdp_origin(origin: &mut Origin, d: &mut SessionDescription) {
     //TODO: if atomic.CompareAndSwapUint64(&origin.SessionVersion, 0, d.Origin.SessionVersion)
     if origin.session_version == 0 {

src/rtp_transceiver/mod.rs

@@ -31,14 +31,14 @@ pub(crate) mod srtp_writer_future;
 /// A synchronization source is a randomly chosen
 /// value meant to be globally unique within a particular
 /// RTP session. Used to identify a single stream of media.
-/// https://tools.ietf.org/html/rfc3550#section-3
+/// <https://tools.ietf.org/html/rfc3550#section-3>
 #[allow(clippy::upper_case_acronyms)]
 pub type SSRC = u32;
 
 /// PayloadType identifies the format of the RTP payload and determines
 /// its interpretation by the application. Each codec in a RTP Session
 /// will have a different PayloadType
-/// https://tools.ietf.org/html/rfc3550#section-3
+/// <https://tools.ietf.org/html/rfc3550#section-3>
 pub type PayloadType = u8;
 
 /// TYPE_RTCP_FBT_RANSPORT_CC ..
@@ -57,11 +57,11 @@ pub const TYPE_RTCP_FB_CCM: &str = "ccm";
 pub const TYPE_RTCP_FB_NACK: &str = "nack";
 
 /// rtcpfeedback signals the connection to use additional RTCP packet types.
-/// https://draft.ortc.org/#dom-rtcrtcpfeedback
+/// <https://draft.ortc.org/#dom-rtcrtcpfeedback>
 #[derive(Default, Debug, Clone, PartialEq)]
 pub struct RTCPFeedback {
     /// Type is the type of feedback.
-    /// see: https://draft.ortc.org/#dom-rtcrtcpfeedback
+    /// see: <https://draft.ortc.org/#dom-rtcrtcpfeedback>
     /// valid: ack, ccm, nack, goog-remb, transport-cc
     pub typ: String,
 
@@ -71,23 +71,23 @@ pub struct RTCPFeedback {
 }
 
 /// RTPCapabilities represents the capabilities of a transceiver
-/// https://w3c.github.io/webrtc-pc/#rtcrtpcapabilities
+/// <https://w3c.github.io/webrtc-pc/#rtcrtpcapabilities>
 #[derive(Default, Debug, Clone)]
 pub struct RTCRtpCapabilities {
     pub codecs: Vec<RTCRtpCodecCapability>,
     pub header_extensions: Vec<RTCRtpHeaderExtensionCapability>,
 }
 
 /// RTPRtxParameters dictionary contains information relating to retransmission (RTX) settings.
-/// https://draft.ortc.org/#dom-rtcrtprtxparameters
+/// <https://draft.ortc.org/#dom-rtcrtprtxparameters>
 #[derive(Default, Debug, Clone, Serialize, Deserialize)]
 pub struct RTCRtpRtxParameters {
     pub ssrc: SSRC,
 }
 
 /// RTPCodingParameters provides information relating to both encoding and decoding.
 /// This is a subset of the RFC since Pion WebRTC doesn't implement encoding/decoding itself
-/// http://draft.ortc.org/#dom-rtcrtpcodingparameters
+/// <http://draft.ortc.org/#dom-rtcrtpcodingparameters>
 #[derive(Default, Debug, Clone, Serialize, Deserialize)]
 pub struct RTCRtpCodingParameters {
     pub rid: String,
@@ -98,12 +98,12 @@ pub struct RTCRtpCodingParameters {
 
 /// RTPDecodingParameters provides information relating to both encoding and decoding.
 /// This is a subset of the RFC since Pion WebRTC doesn't implement decoding itself
-/// http://draft.ortc.org/#dom-rtcrtpdecodingparameters
+/// <http://draft.ortc.org/#dom-rtcrtpdecodingparameters>
 pub type RTCRtpDecodingParameters = RTCRtpCodingParameters;
 
 /// RTPEncodingParameters provides information relating to both encoding and decoding.
 /// This is a subset of the RFC since Pion WebRTC doesn't implement encoding itself
-/// http://draft.ortc.org/#dom-rtcrtpencodingparameters
+/// <http://draft.ortc.org/#dom-rtcrtpencodingparameters>
 pub type RTCRtpEncodingParameters = RTCRtpCodingParameters;
 
 /// RTPReceiveParameters contains the RTP stack settings used by receivers

src/rtp_transceiver/rtp_codec.rs

@@ -55,7 +55,7 @@ impl fmt::Display for RTPCodecType {
 }
 
 /// RTPCodecCapability provides information about codec capabilities.
-/// https://w3c.github.io/webrtc-pc/#dictionary-rtcrtpcodeccapability-members
+/// <https://w3c.github.io/webrtc-pc/#dictionary-rtcrtpcodeccapability-members>
 #[derive(Default, Debug, Clone, PartialEq)]
 pub struct RTCRtpCodecCapability {
     pub mime_type: String,
@@ -92,14 +92,14 @@ impl RTCRtpCodecCapability {
 }
 
 /// RTPHeaderExtensionCapability is used to define a RFC5285 RTP header extension supported by the codec.
-/// https://w3c.github.io/webrtc-pc/#dom-rtcrtpcapabilities-headerextensions
+/// <https://w3c.github.io/webrtc-pc/#dom-rtcrtpcapabilities-headerextensions>
 #[derive(Default, Debug, Clone)]
 pub struct RTCRtpHeaderExtensionCapability {
     pub uri: String,
 }
 
 /// RTPHeaderExtensionParameter represents a negotiated RFC5285 RTP header extension.
-/// https://w3c.github.io/webrtc-pc/#dictionary-rtcrtpheaderextensionparameters-members
+/// <https://w3c.github.io/webrtc-pc/#dictionary-rtcrtpheaderextensionparameters-members>
 #[derive(Default, Debug, Clone, PartialEq)]
 pub struct RTCRtpHeaderExtensionParameters {
     pub uri: String,
@@ -109,7 +109,7 @@ pub struct RTCRtpHeaderExtensionParameters {
 /// RTPCodecParameters is a sequence containing the media codecs that an RtpSender
 /// will choose from, as well as entries for RTX, RED and FEC mechanisms. This also
 /// includes the PayloadType that has been negotiated
-/// https://w3c.github.io/webrtc-pc/#rtcrtpcodecparameters
+/// <https://w3c.github.io/webrtc-pc/#rtcrtpcodecparameters>
 #[derive(Default, Debug, Clone, PartialEq)]
 pub struct RTCRtpCodecParameters {
     pub capability: RTCRtpCodecCapability,
@@ -118,7 +118,7 @@ pub struct RTCRtpCodecParameters {
 }
 
 /// RTPParameters is a list of negotiated codecs and header extensions
-/// https://w3c.github.io/webrtc-pc/#dictionary-rtcrtpparameters-members
+/// <https://w3c.github.io/webrtc-pc/#dictionary-rtcrtpparameters-members>
 #[derive(Default, Debug, Clone)]
 pub struct RTCRtpParameters {
     pub header_extensions: Vec<RTCRtpHeaderExtensionParameters>,