From b10a96b7bce37081a65291b455d6617a382ac652 Mon Sep 17 00:00:00 2001
From: lichon <lichon@users.noreply.github.com>
Date: Sat, 15 Nov 2025 16:04:59 +0800
Subject: [PATCH] Apply suggestions from code review

Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
---
 examples/webrtc.rs       |  1 +
 examples/webrtc_dummy.rs |  2 +-
 src/webrtc.rs            | 10 +++++-----
 3 files changed, 7 insertions(+), 6 deletions(-)

diff --git a/examples/webrtc.rs b/examples/webrtc.rs
index 80e405a..df4a722 100644
--- a/examples/webrtc.rs
+++ b/examples/webrtc.rs
@@ -14,6 +14,7 @@ use bytes::Bytes;
 use clap::{Arg, Command};
 use tokio::time::Duration;
 
+#[cfg(feature = "webrtc")]
 use webrtc::peer_connection::math_rand_alpha;
 
 #[tokio::main]
diff --git a/examples/webrtc_dummy.rs b/examples/webrtc_dummy.rs
index e406990..c826d64 100644
--- a/examples/webrtc_dummy.rs
+++ b/examples/webrtc_dummy.rs
@@ -42,4 +42,4 @@ impl WebRTCStream {
     }
 }
 
-fn _main() {}
+
diff --git a/src/webrtc.rs b/src/webrtc.rs
index 5c0bc4c..f2b0dc2 100644
--- a/src/webrtc.rs
+++ b/src/webrtc.rs
@@ -32,7 +32,8 @@ pub struct WebRTCStream {
     send_timeout: u64,
 }
 
-/// message size limit for Chromium
+/// Standard maximum message size for WebRTC data channels (RFC 8831, 65535 bytes).
+/// Most browsers, including Chromium, enforce this protocol limit.
 const DATA_CHANNEL_BUFFER_SIZE: u16 = u16::MAX;
 
 lazy_static::lazy_static! {
@@ -222,9 +223,8 @@ impl WebRTCStream {
                             Ok(k) => {
                                 lock.remove(&k);
                                 log::debug!(
-                                    "WebRTC session removed key from cache: {} current len: {}",
-                                    k,
-                                    lock.len()
+                                    "WebRTC session removed key from cache: {}",
+                                    k
                                 );
                             }
                             Err(_e) => {}
@@ -274,7 +274,7 @@ impl WebRTCStream {
     #[inline]
     pub async fn get_local_endpoint(&self) -> ResultType<String> {
         if let Some(local_desc) = self.pc.local_description().await {
-            let sdp = serde_json::to_string(&local_desc).unwrap_or_default();
+            let sdp = serde_json::to_string(&local_desc)?;
             let endpoint = Self::sdp_to_endpoint(&sdp);
             Ok(endpoint)
         } else {