One or no way audio occurs when either NAT inspection fails, or the media destination is unreachable. Certain SIP signals contain SDP (Session Description Protocol) information. This information dictates the media destination IP address, port, and supported codecs. There are 3 signals that generally contain SDP. They are the INVITE, 183 Session Progress, and 200 OK. Please review "Unable to Make Outgoing Calls" and "Unable to Receive Calls" articles for normal SIP signal flows. SDP is the bottom part of these referenced signals. Sometimes dropped calls or dead air reports can be attributed to one or no way audio instead of a signaling issue. It's very important to ascertain at what point during the call did these symptoms appear.
o=- 21486763 21486763 IN IP4 22.214.171.124.
c=IN IP4 126.96.36.199.
m=audio 16402 RTP/AVP 0 100 101.
c= IN IP4 is the destination IP address for the RTP (media stream). This destination can differ from the IP address of the signaling/proxy server (server sending the signals to set up the call)
m=audio is the media description type followed by the destination port (audio type media, and port 16402 in this example)
a=rtpmap describes the codecs that this particular device is going to use. The receiving device always determines what codec will be used in the call out of all the available codecs that the calling party could send.
So, as an example, if an INVITE comes into a receiving phone with SDP that says c-IN IP4 192.168.1.5, that phone will not be able to send media to the destination address because the destination IP is a private, LAN IP address of the calling device. This is when NAT inspection fails on account of the calling party's PBX, device, or router. Below is a diagram that illustrates normal flow of SIP signaling and RTP (media) flow through Atlas. If either one of the phones can't contact one of the destinations, one or no way audio will ensue.