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Pfsense and Asterisk….one-way audio problem with SIP ...
https://forum.netgate.com/topic/465/pfsense-and-asterisk-one-way-audio-problem-with-sip
This fixes the 'one-way audio' problem on SIP. I now have audio both in and out using a SIP voip provider. I've tested this with Broadvoice, Sipgate and Gossiptel. My config in the advanced outbound NAT screen was as follows… Iface Src …
PFSense Firewall Settings for VoIP – OnSIP Support
https://support.onsip.com/hc/en-us/articles/204029430-PFSense-Firewall-Settings-for-VoIP
Disable source port rewriting - by default, pfSense rewrites the source port on all outbound traffic. This is necessary for proper NAT in some circumstances such as having multiple SIP phones behind a single public IP registering to a single external PBX. With a minority of providers, rewriting the source port of RTP can cause one way audio.
pfSense Configuration Recipes — Configuring NAT for …
https://docs.netgate.com/pfsense/en/latest/recipes/nat-voip-phones.html
By default pfSense® software rewrites the source port on all outbound traffic. This is necessary for proper NAT in some circumstances such as having multiple SIP phones behind a single public IP registering to a single external PBX. With a minority of providers, rewriting the source port of RTP can cause one way audio.
One-way audio behind pfsense firewall | Page 2 | The VoIP ...
https://www.voip-info.org/forum/threads/one-way-audio-behind-pfsense-firewall.10304/page-2
Well, a lot of people have voip providers that cause one-way audio if you don't use static mode, and the argument about TCP stacks not randomizing ports well is BS - maybe 10 years ago that was true. Now, everything behind my firewall that goes outside is either a recent linux stack, freebsd stack, or windows stack.
One way SIP or dropped SIP after 30 or so seconds : …
https://www.reddit.com/r/PFSENSE/comments/6s6rgg/one_way_sip_or_dropped_sip_after_30_or_so_seconds/
One way SIP or dropped SIP after 30 or so seconds. Hey guys, I've installed a new set of pfSense (v2.3.4) firewalls with an IPSEC VPN between. SIP calls seem to work for about 30 seconds before call drops. Or caller can't hear our end of the call. All of the normal goto settings for VOIP don't seem to help.
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