diff --git a/CHANGELOG.md b/CHANGELOG.md
index e17d854ba..fbeba27cb 100644
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -20,6 +20,7 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* v4.0, 2021-07-08, For [#2403](https://github.com/ossrs/srs/issues/2403), fix padding packets for RTMP2RTC. 4.0.140
 * v4.0, 2021-07-04, For [#2424](https://github.com/ossrs/srs/issues/2424), use srandom/random to generate. 4.0.139
 * v4.0, 2021-07-01, Merge [#2452](https://github.com/ossrs/srs/pull/2452), fix FFmpeg bug by updating channel_layout. 4.0.138
 * v4.0, 2021-06-30, Merge [#2440](https://github.com/ossrs/srs/pull/2440), fix [#2390](https://github.com/ossrs/srs/issues/2390), SRT bug for zerolatency. 4.0.137
diff --git a/README.md b/README.md
index f1d593a7b..18234e7f4 100755
--- a/README.md
+++ b/README.md
@@ -22,7 +22,7 @@ please set the CANDIDATE ([CN][v4_CN_WebRTC#config-candidate],[EN][v4_EN_WebRTC#
 ```bash
 docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \
     --env CANDIDATE=$(ifconfig en0 inet| grep 'inet '|awk '{print $2}') -p 8000:8000/udp \
-    ossrs/srs:v4.0.117 ./objs/srs -c conf/srs.conf
+    ossrs/srs:v4.0.139 ./objs/srs -c conf/srs.conf
 ```
 
 <a name="usage-source"></a>
diff --git a/trunk/research/players/js/srs.sdk.js b/trunk/research/players/js/srs.sdk.js
index 5151aebb4..1bfff4d1e 100644
--- a/trunk/research/players/js/srs.sdk.js
+++ b/trunk/research/players/js/srs.sdk.js
@@ -33,8 +33,8 @@ function SrsRtcPublisherAsync() {
     //      webrtc://r.ossrs.net:11985/live/mystream
     // or set the api server to myapi.domain.com:
     //      webrtc://myapi.domain.com/live/livestream
-    // or set the candidate(ip) of answer:
-    //      webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
+    // or set the candidate(eip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
     // or force to access https API:
     //      webrtc://r.ossrs.net/live/livestream?schema=https
     // or use plaintext, without SRTP:
@@ -272,8 +272,8 @@ function SrsRtcPlayerAsync() {
     //      webrtc://r.ossrs.net:11985/live/mystream
     // or set the api server to myapi.domain.com:
     //      webrtc://myapi.domain.com/live/livestream
-    // or set the candidate(ip) of answer:
-    //      webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
+    // or set the candidate(eip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
     // or force to access https API:
     //      webrtc://r.ossrs.net/live/livestream?schema=https
     // or use plaintext, without SRTP:
diff --git a/trunk/src/app/srs_app_rtc_api.cpp b/trunk/src/app/srs_app_rtc_api.cpp
index 16d6a7b4f..6cba0888c 100644
--- a/trunk/src/app/srs_app_rtc_api.cpp
+++ b/trunk/src/app/srs_app_rtc_api.cpp
@@ -118,8 +118,11 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
         srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
     }
 
-    // For client to specifies the EIP of server.
+    // For client to specifies the candidate(EIP) of server.
     string eip = r->query_get("eip");
+    if (eip.empty()) {
+        eip = r->query_get("candidate");
+    }
     string codec = r->query_get("codec");
     // For client to specifies whether encrypt by SRTP.
     string srtp = r->query_get("encrypt");
@@ -338,8 +341,11 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
         srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
     }
 
-    // For client to specifies the EIP of server.
+    // For client to specifies the candidate(EIP) of server.
     string eip = r->query_get("eip");
+    if (eip.empty()) {
+        eip = r->query_get("candidate");
+    }
     string codec = r->query_get("codec");
 
     srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
diff --git a/trunk/src/app/srs_app_rtc_sdp.cpp b/trunk/src/app/srs_app_rtc_sdp.cpp
index 15af0699e..d671bbaa1 100644
--- a/trunk/src/app/srs_app_rtc_sdp.cpp
+++ b/trunk/src/app/srs_app_rtc_sdp.cpp
@@ -747,6 +747,9 @@ srs_error_t SrsSdp::parse(const std::string& sdp_str)
             line.erase(line.size()-1, 1);
         }
 
+        // Strip the space of line, for pion WebRTC client.
+        line = srs_string_trim_end(line, " ");
+
         if ((err = parse_line(line)) != srs_success) {
             return srs_error_wrap(err, "parse sdp line failed");
         }
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index dfb188e12..46bfd26be 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    139
+#define VERSION_REVISION    140
 
 #endif