When usrsctp is used with a custom transport, an address must be provided to usrsctp_conninput be used as the source and destination address of the incoming packet. WebRTC uses the address of the SctpTransport instance for this value. Unfortunately, this value is often transmitted to the peer, for example to validate signing of the cookie. This could allow an attacker access to the location in memory of the SctpTransport of a peer, bypassing ASLR.
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