Nat Support For Sip

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IP Addressing: NAT Configuration Guide, Cisco IOS Release ...

    https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/ipaddr_nat/configuration/15-mt/nat-15-mt-book/nat-tcp-sip-alg.html
    Oct 16, 2018 · NAT TCP SIP ALG Support. The NAT TCP SIP ALG Support feature allows embedded messages of the Session Initiation Protocol (SIP) passing through a device that is configured with Network Address Translation (NAT) to be translated and encoded back to the packet.

NAT in VoIP - Cisco

    https://www.cisco.com/c/en/us/support/docs/ip/network-address-translation-nat/211269-NAT-in-VoIP.html
    May 23, 2017 · Network Address Translation is a commonly used technique to translate IP addresses on packets that flow between networks using different address spaces. The purpose of this document is not to review NAT. ... On Cisco routers, support for ALG SIP is enabled, by default, on the standard TCP port 5060. It is possible to configure ALG to support ...

NAT and Firewall Traversal Recommendation – OnSIP Support

    https://support.onsip.com/hc/en-us/articles/204691690-NAT-AND-FIREWALL-TRAVERSAL-RECOMMENDATION
    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 …

Using SIP Devices behind NAT – The Smartvox Knowledgebase

    https://kb.smartvox.co.uk/voip-sip/sip-devices-nat/
    SIP Devices behind NAT: What solutions are available? When an IP phone is installed behind NAT, problems can be created by the NAT device itself, by the phone’s inability to correctly understand its own networking environment or from a combination of the two.

How to setup your Asterisk PBX if you are behind a NAT ...

    https://support.gradwell.com/hc/en-gb/articles/215551423-How-to-setup-Asterisk-if-you-are-behind-a-NAT-firewall
    This configuration has been submitted by a Gradwell user, and is not supported by Gradwell support at this time. If your Asterisk PBX is behind a NAT firewall, i.e. the PBX has an IP such as 192.168.0.2 then you will need to perform additional configuration to allow Asterisk to route the SIP and RTP correctly.



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