Sep 24, 2014 · The vulnerability is due to how Session Initiation Protocol (SIP) messages that require network address translation (NAT) are processed on an affected device. An attacker could exploit this vulnerability by sending crafted SIP messages to be processed and translated by an affected device.

A NAT router with a built-in SIP ALG can re-write information within the SIP messages (SIP headers and SDP body) making signaling and audio traffic between the client behind NAT and the SIP endpoint possible. Jul 03, 2019 · SIP ALG helps for outgoing calls but it’s not the best for incoming calls. Endpoints registered under the SIP proxy still have to maintain a connection. They’re called “keep-alives” and only function with a NATed endpoint. This forces the SIP ALG to rewrite the request, causing the NAT to go undetected. The SIP ALG could also break SIP ip nat sip-sbc proxy 200.200.200.10 5060 15.3.33.22 5060 protocol udp call-id-pool call-id-pool session-timeout 300 mode allow-flow-around override port -> Without the sip phone registering to Asterisk or the ip of the NAT device in SIP.conf, the asterisk server has no idea where to look for the phone, thus the call will never go through. (This is the same for all NAT devices). Network Address Translation (NAT) and Router Ports Each device (computers, business phones , cellphones, and tablets) on a local network is assigned an internal IP address. In addition to having IP addresses, devices are also assigned what's known as a "port", or a channel to the internet. Consistent NAT enhances standard NAT policy to provide greater compatibility with peer-to-peer applications that require a consistent IP address to connect to, such as VoIP. Consistent NAT uses an MD5 hashing method to consistently assign the same mapped public IP address and UDP Port pair to each internal private IP address and port pair.

The nat-port-range variable is used to specify a port range in the VoIP profile to restrict the NAT port range for real-time transport protocol/real-time transport control protocol (RTP/RTCP) packets in a session initiation protocol (SIP) call session that is handled by the SIP application layer gateway (ALG) in a FortiGate device.

Feb 19, 2006 · SIP announces the RTP address and port, but if the client is behind NAT, it announces the client's RTP port, which can be different from the port the NAT allocates externally. Even if a lot of SIP implementations and carriers are based on the fact that NAT will always try to allocate the same port, that assumption is false. Basically a NAT with a built-in ALG can rewrite information within the SIP messages and can hold address bindings until the session terminates. A SIP ALG will also handle SDP in the body of SIP messages (which is used ubiquitously in VoIP to set up media endpoints), since SDP also contains literal IP addresses and ports that must be translated.

Aug 27, 2010 · Firewalls, NAT devices, Session Border Controllers and SIP Proxys are in the signalling path and they will affect the call. SIP Call setup - INVITE-200 OK - ACK. To set up a SIP call, there's an INVITE transaction. The SIP software that initiates the call sends an INVITE, then wait to get a reply. When a reply arrives, the caller sends an ACK.

How do we disable NAT on SIP and SDP payloads, when using NAT? The ATRG: VoIP documentation states the following: We're running Asterisk with ICE (Interactive Connectivity Establishment), which essentially provides multiple candidates in INVITE or SDP negotiation messages, where each is an IP and port combination.