Asterisk SIP/SDP Issue With RTP - One-Way Audio

Asterisk Asterisk SIP
1

All,

Does anyone know how to configure Asterisk 13.16.0 so that I can hear the RTP audio stream from a PBX server that sits behind NAT. I’ve reviewed several tickets with respect to this issue. What setting should be configured to allow a softphone client to reach the PBX server with the correct IP address when RTP communication is established. In my debugging, I found that the private IP address of the PBX server is being sent during the SIP session rather than the public IP address. Which NAT setting in 13.16.0 will allow me to pin to the correct IP address? I have the externIP (i.e. public IP) configuration setting as well as the local networks defined and nothing seems to reflect a change in the SDP header. I’ve tried a few configuration changes with no success. I have two-way audio when connecting with devices attached to the PSTN. I think this all comes down to a configuration setting under Asterisk. What configuration setting or settings is going to force the SDP header to have the correct IP address?

Thanks.

2

Which SIP channel driver are you using in Asterisk? I think you are using chan_sip based on your mention of “externip” in which case you also need the localnet option set, or else Asterisk doesn’t know when it should use externip and when it shouldn’t.

3

Hi,

I have the externIP (i.e. public IP) configuration setting as well as the local networks defined and nothing seems to reflect a change in the SDP header.

  1. Which NAT type do you have? Most likely, it is “IP masquerading”, i.e.doing source NAT for the whole private LAN address space into a single public IP…But may be there is something different?
  2. Can you provide sip.conf file? You may obfuscate the external IP there.