Issue with no audio


#1

I have a problem with no audio, the calling party can’t hear anything from the destination.
My asterisk is behind nat, I analyzed the sip invite and noticed that “media connecion” is with NAT IP.

See below…

Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 3950 3950 IN IP4 192.168.1.251
Session Name (s): session
Connection Information ©: IN IP4 192.168.1.251
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 6660 RTP/AVP 0 8 18 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:18 G729/8000

Can you please help me?


#2

Asterisk is on a private network. The calling party is coming in from a public network. The SDP for the calilng party quotes a non-routeable address. Is that correct?

This suggests that the calling party is, itself behind NAT and is broken. It is possible that this is one of the rare cases were nat=yes helps; I don’t know if it works for RTP, as well. Even if it works for RTP, I doubt it can cope with the port number being transformed.

If nat=yes doesn’t work, you will just have to fix the calling party.