SIP Trunk behind a firewall/NAT - Cisco Community
Dec 09, 2009 SIP Trunk behind a firewall/NAT - Cisco Community My NAT configuration translates the RTP incoming from carrier on port range from 20000 to 30000 and the SIP on port 5060. Everything leaving the router to the internet gets out through the firewall public IP address. Thanks a lot for your attention. *EDIT. Securing Your Asterisk VoIP Server with IPTables Disable unneeded Asterisk modules. By default, Asterisk listens on many TCP and UDP ports as can be shown by netstat -anput | grep asterisk. If you only use SIP but not IAX2, and have no VoIP hardware cards, you can disable some Asterisk modules and close those ports. This could increase security in case your firewall goes down. [Asterisk] TCP transport and NAT for Asterisk - VOIP Tech
Asterisk trying to send RTP packets to the internal
Jan 06, 2013 Solved: Disable NAT on SIP payload - breaks ICE - Check 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.
Curso Asterisk (VI): Lidiando con el NAT | The Infamous
Asterisk Guru Website. 1.1 Description of the problem:. Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange both call legs (incoming and outgoing audio). Asterisk / Firewall Rules : PFSENSE Jun 18, 2010