PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In versions 2.11.1 and prior, parsing an incoming SIP message that contains a malformed multipart can potentially cause out-of-bound read access. This issue affects all PJSIP users that accept SIP multipart. The patch is available as commit in the `master` branch. There are no known workarounds.
Potential out-of-bound read during RTP/RTCP parsing in PJSIP
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In version 2.11.1 and prior, there are various cases where it is possible that certain incoming RTP/RTCP packets can potentially cause out-of-bound read access. This issue affects all users that use PJMEDIA and accept incoming RTP/RTCP. A patch is available as a commit in the `master` branch. There are no known workarounds.
PJSIP is a free and open source multimedia communication library. In version 2.11.1 and prior, if incoming RTCP XR message contain block, the data field is not checked against the received packet size, potentially resulting in an out-of-bound read access. This affects all users that use PJMEDIA and RTCP XR. A malicious actor can send a RTCP XR message with an invalid packet size.
Out-of-bounds read when parsing RTCP BYE message in PJSIP
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In affected versions if the incoming RTCP BYE message contains a reason's length, this declared length is not checked against the actual received packet size, potentially resulting in an out-of-bound read access. This issue affects all users that use PJMEDIA and RTCP. A malicious actor can send a RTCP BYE message with an invalid reason length. Users are advised to upgrade as soon as possible. There are no known workarounds.
Potential integer underflow upon receiving STUN message in PJSIP
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In affected versions if the incoming STUN message contains an ERROR-CODE attribute, the header length is not checked before performing a subtraction operation, potentially resulting in an integer underflow scenario. This issue affects all users that use STUN. A malicious actor located within the victim’s network may forge and send a specially crafted UDP (STUN) message that could remotely execute arbitrary code on the victim’s machine. Users are advised to upgrade as soon as p
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In PJSIP before version 2.11.1, there are a couple of issues found in the SSL socket. First, a race condition between callback and destroy, due to the accepted socket having no group lock. Second, the SSL socket parent/listener may get destroyed during handshake. Both issues were reported to happen intermittently in heavy load TLS connections. They cause a crash, resulting in a denial of service. These are fixed in version 2.11.1.
Crash in receiving updated SDP answer after initial SDP negotiation failed
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In PJSIP version 2.10 and earlier, after an initial INVITE has been sent, when two 183 responses are received, with the first one causing negotiation failure, a crash will occur. This results in a denial of service.
Existing TLS connections can be reused without checking remote hostname
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In version 2.10 and earlier, PJSIP transport can be reused if they have the same IP address + port + protocol. However, this is insufficient for secure transport since it lacks remote hostname authentication. Suppose we have created a TLS connection to `sip.foo.com`, which has an IP address `100.1.1.1`. If we want to create a TLS connection to another hostname, say `sip.bar.com`, which has the same IP address, then it will reuse that existing connection, even though `100.1.1.1