CVE-2013-6631

Use-after-free vulnerability in the Channel::SendRTCPPacket function in voiceengine/channel.cc in libjingle in WebRTC, as used in Google Chrome before 31.0.1650.48 and other products, allows remote attackers to cause a denial of service heap memory corruption or possibly have unspecified other...

CVE-2013-6631

Removed by vendor...

HTTP Client LAN IP Address Gather

This module retrieves a browser's network interface IP addresses using WebRTC. Module Options msf use auxiliary/gather/browser/lanipleak msf undefinedlanipleak show actions ...actions... msf undefinedlanipleak set ACTION msf undefinedlanipleak show options ...show and set options... msf...

CVE-2013-0843

content/renderer/media/webrtcaudiorenderer.cc in Google Chrome before 24.0.1312.56 on Mac OS X does not use an appropriate buffer size for the 96 kHz sampling rate, which allows remote attackers to cause a denial of service memory corruption and application crash or possibly have unspecified othe...

Memory corruption

content/renderer/media/webrtcaudiorenderer.cc in Google Chrome before 24.0.1312.56 on Mac OS X does not use an appropriate buffer size for the 96 kHz sampling rate, which allows remote attackers to cause a denial of service memory corruption and application crash or possibly have unspecified othe...

CVE-2013-0843

content/renderer/media/webrtcaudiorenderer.cc in Google Chrome before 24.0.1312.56 on Mac OS X does not use an appropriate buffer size for the 96 kHz sampling rate, which allows remote attackers to cause a denial of service memory corruption and application crash or possibly have unspecified othe...

CVE-2013-0843

content/renderer/media/webrtcaudiorenderer.cc in Google Chrome before 24.0.1312.56 on Mac OS X does not use an appropriate buffer size for the 96 kHz sampling rate, which allows remote attackers to cause a denial of service memory corruption and application crash or possibly have unspecified othe...

CVE-2013-0843

CVE-2013-0843 affects Google Chrome on macOS, where the renderer code in webrtc_audio_renderer.cc mishandles the buffer size for a 96 kHz WebRTC audio stream. This can lead to memory corruption and a resulting crash (DoS) or potentially other unspecified impact when processing audio from a web si...