RTP Packets Sent&Received but no audio over remoteip phone


I am trying to use a ip phone to connect to an asterisk box behind nat. I can get it to register and make calls but when the call connects, there is no audio either way. I used to have a problem with it connecting and then the call dropping but after configuring externip and localnet and all that good stuff, that problem went away. also, i have the rtp ports defined and open in the router that the asterisk box is sitting behind.

I have also set the peer to have nat=yes and canreinvite=no, etc.

When I do an rtp debug in asterisk, it shows that packets are being sent and received but no audio still. Any ideas?


You haven’t set the primary NAT options (externip/eternaddr/stunaddr). Also canreinvite is deprecated from 1.8 (use directmedia) and nat=yes is deprecated from 11 (use the indivual options).

I have set externip as my previous post indicated. Or are you referring to something else? I will change canreinvite to directmedia (still =no though right?). In regards to the individual options instead of nat=yes, what sorts of options should i be using?


I’m not convinced you need any nat options in simple cases, but the options would depend on the reason for the problem. I believe the comedia one is the most likely one to make a difference.

Having the SIP INVITEs, responses and corresponding SDP would make it much easier to work out what is happening. Do not obfuscate in a way that makes different IP addresses look the same or private ones look like public ones.