I have an installation with a single public IP address that uses an Asterisk PBX connected to a Twilio SIP Trunk. The provider does not offer additional IP addresses.
Right now, in order for the SIP audio to work, I need to forward UDP ports 10000-20000 to the PBX since Twilio says media can come on any of those ports. However, this breaks the ability of other users on that connection to use WebRTC media because WebRTC uses that same port range for media. The only real information that I have found discussed in the past is about using sipproxd in the case of having multiple SIP devices inside the firewall to allow all of them to use port 5060 (SIP signaling) and have the firewall rewrite the SIP traffic for each one. However, I can't seem to find any information about my use-case of a single SIP device and not having to forward the ports for the media. Can sipproxd help me with that? Any other ideas? Thanks, Moshe -- Moshe Katz -- kohenk...@gmail.com -- +1(301)867-3732 _______________________________________________ pfSense mailing list https://lists.pfsense.org/mailman/listinfo/list Support the project with Gold! https://pfsense.org/gold