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Voip no audio | Netgate Forum
https://forum.netgate.com/topic/154886/voip-no-audio#:~:text=VoIP%20without%20audio%20is%20a%20NAT%20issue.%20If,is%20no%20audio%20then%20RTP%20is%20not%20working.
Voip no audio | Netgate Forum
https://forum.netgate.com/topic/154886/voip-no-audio
Jun 29, 2020, 10:57 AM. VoIP without audio is a NAT issue. If they connect then SIP signaling is working, generally on port 5060 or 5061. If there is no audio then RTP is not working. Check to be sure that SIP ALG is turned off in your modem.
Asterisk/FreePBX behind pfSense – no audio in/out – iTecTec
https://itectec.com/superuser/asterisk-freepbx-behind-pfsense-no-audio-in-out/
Asterisk/FreePBX behind pfSense – no audio in/out. asterisk freepbx nat; pfsense port-forwarding. I have following setup: ISP modem in bridge mode -> pfSense firewall -> HP2920 switch -> asterisk | VoIP phones. I finally got inbound and outbound calls working but I …
PFSense Firewall Settings for VoIP - OnSIP Support
https://support.onsip.com/hc/en-us/articles/204029430-PFSense-Firewall-Settings-for-VoIP
If your VoIP deployment is not working properly, try the following: 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 …
r/PFSENSE - No or delayed audio on VoIP connections on ...
https://www.reddit.com/r/PFSENSE/comments/3pd14v/no_or_delayed_audio_on_voip_connections_on/
No or delayed audio on VoIP connections on Android phone. Just recently got around to re-implementing pfSense in my network. So far everything is running smoothly except VoIP services on my Android device either have no audio or delayed audio. Specifically I have a T-Mobile device and while the Wifi calling feature will connection through pfSense without an issue, I cannot …
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.
SIP Phone no audio. Rule set to allow all traffic from IP ...
https://www.reddit.com/r/PFSENSE/comments/8r2xj8/sip_phone_no_audio_rule_set_to_allow_all_traffic/
The phone system is hosted in a data centre which is fronted by pfSense. There are lots of users connecting to this phone system without any issue and they all get two way audio. My local network is protected by a local installation of pfSense. Within the LAN I have a local test phone system which has SIP trunks connected and they are working fine.
[SOLVED] Intermittent one way audio with VoIP - double NAT ...
https://community.spiceworks.com/topic/1469500-intermittent-one-way-audio-with-voip-double-nat-issue-possibly
If you have a firewall problem, then the entire RTP stream will not make it through, period - but if you have RTP packets with no media payload - That just sounds like the media isn't getting captured by the VOIP Provider - in which case, that PCAP showing empty RTP packets, should be enough... or if the PCAP shows a payload, but it's static, that would also indicate …
Firewall Best Practices for VoIP on pfSense - YouTube
https://www.youtube.com/watch?v=C0JgrzxXIBY
Monthly pfSense Hangout videos are brought to you by Netgate. Visit https://www.netgate.com/videos for a complete list of available video resources.- Slides:...
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