Fixing NAT and one-way audio issues

The most common production issue with self-hosted SIP — diagnosed and fixed in five minutes.

Symptoms

Call connects, both sides hear ringing, but after pickup one or both sides hear silence.
Call drops at exactly 30 seconds (RTP timeout).
SIP messages flow fine, but rtcp reports zero packets.

All three are NAT-related.

Asterisk fix

In pjsip.conf transport:

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
external_media_address = YOUR.PUBLIC.IP
external_signaling_address = YOUR.PUBLIC.IP
local_net = 192.168.0.0/16
local_net = 10.0.0.0/8

On the endpoint:

rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no

FreeSWITCH fix

In vars.xml:

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=YOUR.PUBLIC.IP"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=YOUR.PUBLIC.IP"/>

Firewall

Open UDP 10000–20000 inbound on your public IP. If you're behind a NAT router, port-forward the same range to your softswitch.

Verify

tcpdump -i any -n udp portrange 10000-20000

During a call you should see RTP packets in both directions. If you only see one direction, NAT is still mangling the other side's destination address.