PJSIP: TLS server identity (hostname) verification bypass via embedded NUL in certificate SubjectAltName (OpenSSL and GnuTLS backends)
PJSIP is a free and open source multimedia communication library written in C. In 2.17 and earlier, the OpenSSL and GnuTLS backends in pjlib/src/pj/ssl_sock_ossl.c and pjlib/src/pj/ssl_sock_gtls.c copy DNS SubjectAltName values with string functions that recalculate their length and truncate an embedded NUL byte. With server verification enabled through --tls-verify-server for the PJSIP TLS/SIPS transport, a certificate containing a DNS SubjectAltName formed from the target hostname prefix followed by an embedded NUL and an attacker-controlled suffix can therefore be accepted for the prefix hostname. An attacker who possesses such a certifica
PJSIP is a free and open source multimedia communication library written in C. In 2.17 and earlier, the PJSIP AVI parser in pjmedia/src/pjmedia/avi_player.c uses an input-file video chunk length as the number of bytes copied into a frame buffer whose capacity is derived from the declared media dimensions. A crafted AVI file can therefore cause an attacker-controlled out-of-bounds write past the heap allocation when an application plays the file or pulls its frames. The existing size assertion does not protect production release builds, where assertions are disabled. Typical local playback can crash the process, while applications that accept
PJSIP: Pre-authentication overflow in the telnet CLI error
PJSIP is a free and open source multimedia communication library written in C. Prior to commit 4472a31, a stack buffer overflow exists in the PJLIB-UTIL telnet CLI front-end when rendering feedback for an entered command line. Several command-line handling paths write an attacker-influenced amount of data into fixed-size buffers without sufficient bounds checking, so a long command line can overflow them. This affects only applications that enable the telnet CLI front-end (e.g. pj_cli_telnet_create() / --cli-telnet-port). The telnet CLI is an interactive administration interface with no authentication, so any client able to reach it can alrea
PJSIP: Pre-authentication overflow in the telnet CLI history
PJSIP is a free and open source multimedia communication library written in C. Prior to commit 628b716, a stack buffer overflow exists in the PJLIB-UTIL telnet CLI front-end when redrawing the command line during history recall (handle_up_down() in cli_telnet.c). This affects only applications that enable the telnet CLI front-end (same gating as the related CLI issue). The line-redraw sequence for a recalled history entry can accumulate more data than a fixed-size stack buffer holds, which may lead to application termination. Exploitation requires access to the unauthenticated telnet CLI, which already permits arbitrary CLI commands, so the a
PJSIP is a free and open source multimedia communication library written in C. Prior to commit 8d5956a, a heap buffer overflow exists in the PJLIB-UTIL HTTP client (http_client.c) when buffering an HTTP response body. This affects applications that use the PJLIB-UTIL HTTP client to receive a whole response body at once (a completion callback with no incremental on_data_read callback). When growing the response buffer, an incorrect size calculation based on the server-supplied Content-Length can leave the buffer too small, causing response data to be written past the end of the allocation. A malicious or man-in-the-middle HTTP server can trigg
PJSIP: Stack overflow parsing a TLS peer certificate's SubjectAltName in GnuTLS backend
PJSIP is a free and open source multimedia communication library written in C. Prior to commit c4a151a, a stack buffer overflow exists in the GnuTLS TLS backend when parsing the Subject Alternative Name extension of a peer certificate (tls_cert_get_info() in ssl_sock_gtls.c). Only GnuTLS builds are affected (--with-gnutls); OpenSSL and Apple SecureTransport/Network.framework builds are not affected. While extracting certificate information after a TLS handshake, an incorrect buffer-size value can cause an oversized SubjectAltName entry to be written past the end of a fixed-size stack buffer. A network-positioned attacker presenting a crafted
PJSIP is a free and open source multimedia communication library written in C. Prior to commit a1b707c, a stack buffer overflow exists in the SRTP/SDES media transport when processing a=crypto attributes during SDP offer/answer (sdes_encode_sdp() in transport_srtp_sdes.c). This affects applications with SRTP enabled (use_srtp optional or mandatory, using SDES keying). During media negotiation, the crypto attributes from the remote SDP are collected into a fixed-size array without bounding their number; a remote peer that includes an excessive number of a=crypto attributes in a single media description can write past the end of that array on t
PJSIP: Stack overflow handling Service-Route headers in a registration response
PJSIP is a free and open source multimedia communication library written in C. Prior to commit acc03b5, a stack buffer overflow exists in PJSUA when processing Service-Route headers in a registration response (update_service_route() in pjsua_acc.c). This affects applications that register using the PJSUA/PJSUA2 account API (the default registration path). The Service-Route URIs from a 2xx response to REGISTER are stored into a fixed-size array without bounding the number of headers; a registrar that returns an excessive number of Service-Route headers can write past the end of the array on the stack. The values written are internal pointers r
PJSIP is a free and open source multimedia communication library written in C. Prior to commit d6a0e7f, a buffer overflow can occur in pjsip_generic_array_hdr_print() in pjsip/src/pjsip/sip_msg.c, the function that serializes generic array headers (such as Allow, Require, Supported, and Unsupported). Under certain output-buffer boundary conditions the function can write one byte past the end of the buffer. This is reachable mainly in applications that parse and re-serialize incoming SIP requests — for example a proxy, SBC, or B2BUA — where a remote peer can influence the serialized message. The out-of-bounds write is a single fixed byte; code
PJSIP: SDP parser out-of-bounds write in remote payload-type map maintenance
PJSIP is a free and open source multimedia communication library written in C. Prior to commit 673b978, a remote out-of-bounds read and write can occur in the SDP negotiator when the remote payload-type map maintenance feature is enabled. assign_pt_and_update_map() in pjmedia/src/pjmedia/sdp_neg.c uses payload-type numbers taken from a remote SDP offer or answer to index fixed-size internal tables without sufficient bounds validation, so a crafted remote SDP can cause memory access outside those tables. The practical impact is memory corruption and denial of service; code execution is not demonstrated. This path is only reached when PJMEDIA_S
GnuTLS backend silently skips certificate chain verification when verify_peer is false
PJSIP is a free and open source multimedia communication library written in C. Prior to version 2.17, on GnuTLS builds, the SIP TLS transport (sip_transport_tls) can accept connections with invalid or untrusted certificates even when the application explicitly enables certificate verification via verify_server = PJ_TRUE or verify_client = PJ_TRUE. This issue has been patched in version 2.17.
PJSIP: Asymmetric ptime integer overflow in Media Stream
PJSIP is a free and open source multimedia communication library written in C. In 2.16 and earlier, there is an integer overflow in media stream buffer size calculation when processing SDP with asymmetric ptime configuration. The overflow may result in an undersized buffer allocation, which can lead to unexpected application termination or memory corruption This vulnerability is fixed in 2.17.
PJSIP is a free and open source multimedia communication library written in C. In 2.16 and earlier, there is an out-of-bounds read when parsing a malformed Content-ID URI in SIP multipart message body. Insufficient length validation can cause reads beyond the intended buffer bounds. This vulnerability is fixed in 2.17.
PJSIP: Stack buffer overflow in pjsip_auth_create_digest2()
PJSIP is a free and open source multimedia communication library written in C. In 2.16 and earlier, a stack buffer overflow exists in pjsip_auth_create_digest2() in PJSIP when using pre-computed digest credentials (PJSIP_CRED_DATA_DIGEST). The function copies credential data using cred_info->data.slen as the length without an upper-bound check, which can overflow the fixed-size ha1 stack buffer (128 bytes) if data.slen exceeds the expected digest string length.
PJSIP: Heap buffer overflow in Opus codec decoding
PJSIP is a free and open source multimedia communication library written in C. In 2.16 and earlier, there is a buffer overflow when decoding Opus audio frames due to insufficient buffer size validation in the Opus codec decode path. The FEC decode buffers (dec_frame[].buf) were allocated based on a PCM-derived formula: (sample_rate/1000) * 60 * channel_cnt * 2. At 8 kHz mono this yields only 960 bytes, but codec_parse() can output encoded frames up to MAX_ENCODED_PACKET_SIZE (1280) bytes via opus_repacketizer_out_range(). The three pj_memcpy() calls in codec_decode() copied input->size bytes without bounds checking, causing a heap buffer over
ast_coredumper running as root sources ast_debug_tools.conf from /etc/asterisk; potentially leading to privilege escalation
Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, the asterisk/contrib/scripts/ast_coredumper runs as root, as noted by the NOTES tag on line 689 of the ast_coredumper file. The script will source the contents of /etc/asterisk/ast_debug_tools.conf, which resides in a folder that is writeable by the asterisk user:group. Due to the /etc/asterisk/ast_debug_tools.conf file following bash semantics and it being loaded; an attacker with write permissions may add or modify the file such that when the root ast_coredumper is run; it would source and thereby exe
Asterisk vulnerable to potential privilege escalation
Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, when ast_coredumper writes its gdb init and output files to a directory that is world-writable (for example /tmp), an attacker with write permission(which is all users on a linux system) to that directory can cause root to execute arbitrary commands or overwrite arbitrary files by controlling the gdb init file and output paths. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2.
Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, the ast_xml_open() function in xml.c parses XML documents using libxml with unsafe parsing options that enable entity expansion and XInclude processing. Specifically, it invokes xmlReadFile() with the XML_PARSE_NOENT flag and later processes XIncludes via xmlXIncludeProcess().If any untrusted or user-supplied XML file is passed to this function, it can allow an attacker to trigger XML External Entity (XXE) or XInclude-based local file disclosure, potentially exposing sensitive files from the host system
The Asterisk embedded web server 's /httpstatus page echos user supplied values(cookie and query string) without sanitization
Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, user supplied/control values for Cookies and any GET variable query Parameter are directly interpolated into the HTML of the page using ast_str_append. The endpoint at GET /httpstatus is the potential vulnerable endpoint relating to asterisk/main /http.c. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2.
Asterisk can crash from a specifically malformed Authorization header in an incoming SIP request
Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.15.2, 21.10.2, and 22.5.2, if a SIP request is received with an Authorization header that contains a realm that wasn't in a previous 401 response's WWW-Authenticate header, or an Authorization header with an incorrect realm was received without a previous 401 response being sent, the get_authorization_header() function in res_pjsip_authenticator_digest will return a NULL. This wasn't being checked before attempting to get the digest algorithm from the header which causes a SEGV. This issue has been patched in versions 20.15.2, 21.10.2, and 22.5.2. T