The SIP registrar server configured in this and the registrar host field is the real registrar. Or the values entered in those fields map to the home proxy address and home proxy port of the SIP NAT with external proxy address and external proxy port values that correspond to the real registrar. The default value is 0. The valid range is:
For NAT, you need to set NAT=yes if the machine is actually behind NAT. You also need to forward the ports to the server from the NAT router. Lastly, make sure that you define all local address spaces that do NOT have a NAT router between them and the Asterisk box (ie: the local LAN, another subnet connected via a non-NAT router, and subnets connected via IPSec). This will allow SIP signaling and RTP media to successfully traverse a NAT without requiring any configuration changes on the NAT. STUN presents a working solution for most NATs that are not symmetric NAT, e.g., most of the SOHO routers have non-symmetric NAT and in this case, it is OK to use STUN. Mar 01, 2007 · Network Address Translation (NAT) is a common practice used in networks, and it doesn't play well with VoIP. Solving this problem requires an understanding of NAT, VoIP and your VoIP setup. This article focuses on the SIP protocol for VoIP and the Asterisk VoIP software, but the problems and solutions are applicable to most other situations. With the introduction of the Asterisk SIP Settings module, most SIP settings are made available in the GUI. This includes the all important NAT, External IP, Local Network, Enabled Codecs and Codec order. Logging In. From the top menu click Settings; From the drop down click Asterisk Sip Settings; Settings. Allow Anonymous inbound SIP Calls I using only sip_any service on any to any rule. Make sure that in the "Advanced" properties of the service, the "Accept Replies" option is checked. Also I activated "Hide NAT changes source port for sip over udp" option from "Inspection Settings > SIP General>Default Inspection>Advanced" If you using multiple network. It is complicated. If you are behind NAT and your Trunk is showing "Registered" at SIP.US, but it is registered to a private IP Address you will need to navigate to "PBX" ---> "SIP Settings" ---> "- NAT" and input your external IP Address in the "External IP Address" field. You must also put your local network address in the "Local Network Address" field. sip_nat_detected . sip_nat_detected is set to true when NAT is detected. Use it in your dialplan to handle NATted devices differently.
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.
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
May 03, 2010 · The host part of the SIP URI in the Contact Header in the REGISTER request is truly your NAT device’s public IP (and port). Port forwarding on the NAT device is indeed set correctly. The VoIP provider’s response to a REGISTER request is usually a challenge (407).
vSRX,SRX Series. Understanding the SIP ALG, Understanding SIP ALG Hold Resources, Understanding the SIP ALG and NAT, Example: Setting SIP ALG Call Duration and Timeouts, Example: Configuring SIP ALG DoS Attack Protection, Example: Allowing Unknown SIP ALG Message Types, Example: Configuring Interface Source NAT for Incoming SIP Calls, Example: Decreasing Network Complexity by Configuring a 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. actually i did copy the relevant config and that is not the original issue. The router is not license for CUBE or any other VoIP functionality (besides nat sip service and sip-sbc) and its the one provided by the ISP to all the other customers where it is working fine with NATing. 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.