Use after free in WebRTC in Google Chrome prior to 97.0.4692.71 allowed a remote attacker who had compromised the renderer process to potentially perform a sandbox escape via a crafted HTML page. (Chromium security severity: High)

Insufficient control flow management in the Open WebRTC Toolkit before version 4.3.1 may allow an unauthenticated user to potentially enable escalation of privilege via network access.

An issue was discovered in janus-gateway (aka Janus WebRTC Server) through 0.10.0. janus_get_codec_from_pt in utils.c has a Buffer Overflow via long value in an SDP Offer packet.

An issue was discovered in janus-gateway (aka Janus WebRTC Server) through 0.10.0. janus_streaming_rtsp_parse_sdp in plugins/janus_streaming.c has a Buffer Overflow via a crafted RTSP server.

An issue was discovered in janus-gateway (aka Janus WebRTC Server) through 0.10.0. janus_sdp_merge in sdp.c has a stack-based buffer overflow.

A buffer overflow could occur when parsing and validating SCTP chunks in WebRTC. This could have led to memory corruption and a potentially exploitable crash. This vulnerability affects Firefox ESR < 68.8, Firefox < 76, and Thunderbird < 68.8.0.

The WebRTC component in the Signal Private Messenger application through 4.47.7 for Android processes videoconferencing RTP packets before a callee chooses to answer a call, which might make it easier for remote attackers to cause a denial of service or possibly have unspecified other impact via malformed packets. NOTE: the vendor plans to continue this behavior for performance reasons unless a WebRTC design change occurs

A use-after-free vulnerability can occur during WebRTC connections when interacting with the DTMF timers. This results in a potentially exploitable crash. This vulnerability affects Firefox ESR < 52.6 and Firefox < 58.

Use-after-free vulnerability in the mozilla::DataChannelConnection::Close function in Mozilla Firefox before 45.0 and Firefox ESR 38.x before 38.7 allows remote attackers to execute arbitrary code by leveraging mishandling of WebRTC data-channel connections.

Use-after-free vulnerability in browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.cc in the WebRTC Audio Private API implementation in Google Chrome before 49.0.2623.75 allows remote attackers to cause a denial of service or possibly have unspecified other impact by leveraging incorrect reliance on the resource context pointer.