Asterisk audio dropout

oscarBravo

Over here in the brother's gaff in London, trying to set up an Asterisk server for conference calling. There will only ever be four people conferencing, so the idea is to use X-Lite softphones connecting directly to the server, which is behind a NAT router in the brother's house.

I have the Asterisk server set up, and conferencing is working just peachy as long as the extensions are on the LAN. Once I get someone to set up an extension on the outside, coming in through the NAT router, the problems start.

When calling between an extension on the LAN and one outside the router, the audio drops in and out on a regular cycle. It seems to be roughly once per second - half a second of audio, half a second of silence. This goes both ways.

Any ideas what could be causing this? I have the various NAT workarounds set up as suggested on the voip-info.org wiki.

daffy_duc

SIP Isn't very NAT friendly (UDP packets for RDP...).

Try setting up IAX instead. There are IAX softphones available.

sirlinux

post your relevant sip config bits and we can make some pointers, is reinvite turned off? do you have externip set, do you have your rtp port range forwarded?