Reject negative values for "b=AS".

webrtc/pc/webrtcsdp.cc

 /*
  *  Copyright 2011 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 2701 matching lines @@
     if (IsLineType(line, kLineTypeSessionBandwidth)) {
       std::string bandwidth;
       if (HasAttribute(line, kApplicationSpecificMaximum)) {
         if (!GetValue(line, kApplicationSpecificMaximum, &bandwidth, error)) {
           return false;
         } else {
           int b = 0;
           if (!GetValueFromString(line, bandwidth, &b, error)) {
             return false;
           }
+          if (b < 0) {
+            return ParseFailed(line, "b=AS value can't be negative.", error);
+          }
           // We should never use more than the default bandwidth for RTP-based
           // data channels. Don't allow SDP to set the bandwidth, because
           // that would give JS the opportunity to "break the Internet".
           // See: https://code.google.com/p/chromium/issues/detail?id=280726
           if (media_type == cricket::MEDIA_TYPE_DATA && IsRtp(protocol) &&
               b > cricket::kDataMaxBandwidth / 1000) {
             std::ostringstream description;
             description << "RTP-based data channels may not send more than "
                         << cricket::kDataMaxBandwidth / 1000 << "kbps.";
             return ParseFailed(line, description.str(), error);
@@ skipping 582 matching lines @@
     UpdateCodec<AudioContentDescription, cricket::AudioCodec>(
         media_desc, payload_type, feedback_param);
   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
     UpdateCodec<VideoContentDescription, cricket::VideoCodec>(
         media_desc, payload_type, feedback_param);
   }
   return true;
 }
 
 }  // namespace webrtc