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Sip Client behind NAT (No audio until client unmutes ...
https://community.asterisk.org/t/sip-client-behind-nat-no-audio-until-client-unmutes/83751
When I call into the conference, my call gets connected but there is no audio because packets are sent to 172.18.0.6 address. ... The sip client is the one behind the NAT. ambiorixg12 April 24, 2020, 5:46am #4. creativeguitar: In my case is the other way around. The sip client is the one behind the NAT.
FreePBX behind NAT no audio at all - General Help ...
https://community.freepbx.org/t/freepbx-behind-nat-no-audio-at-all/68819
Hello, Everyone I am new to FreePBX and my box works fine with PSTN. Recently, I applied a sip.us trial account and tried to dial through sip but there is no audio and cdr show as answer. This box is behind NAT and all sip firewall rules applied, There is no sip connection blocked when I made sip call. Please give me some hint to solve this problem. Here is the log > …
Possible newbie issue - No audio on extensions behind NAT ...
https://community.freepbx.org/t/possible-newbie-issue-no-audio-on-extensions-behind-nat-server-tries-to-contact-private-ip/69403
Usually, proper operation of an extension behind NAT does require RTP Symmetric, Rewrite Contact and Force rport for the extension to all be Yes, though that’s not normally an issue because these are the defaults. On a call to an extension, Asterisk sends an INVITE requesting audio be sent to its external IP.
Using SIP Devices behind NAT – The Smartvox …
https://kb.smartvox.co.uk/voip-sip/sip-devices-nat/
When working with SIP devices behind NAT, the ports that you may need to set forwarding for are: 1. The main SIP connection port – usually this is port 5060. The protocol is nearly always UDP 2. The RTP media port or ports – often a range of higher port numbers. UDP protocol. You will need to find out which ports your IP phone uses for RTP media.
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