Session Traversal Utilities for NAT overview
A SIP phone (Host, Client) operating behind a Network Address Translator (NAT) has a private IP
address and separate ports for each of the SIP signaling, RTP and RTCP ports.
When the phone first sends a data packet from its private address and some private port through
the NAT, the NAT allocates a corresponding public IP address and port (known as a binding) that
external peers can use to communicate with the phone.
Session Traversal Utilities for NAT (STUN) is a set of methods that allow the phone to discover its
public IP address and ports for SIP signaling, RTP and RTCP.
Periodic traffic, or keep-alives, must be sent from the phone’s active private ports to refresh the
NAT bindings so that the NAT will not terminate them due to inactivity.
For Avaya J100 Series IP Phones, the STUN functionality is supported only in the 3CX and
generic Open SIP environments.
Using a STUN Binding Request and Response
The phone performs STUN discovery of its public SIP signaling IP address and port at registration
time. For public RTP and RTCP address and ports, the phone performs STUN discovery every
time a call is established.
The following describes how the phone uses STUN to create a NAT binding and discover the
corresponding public IP address and the port:
• Through NAT, the phone sends a STUN Binding Request from its private SIP signaling, RTP
or RTCP source IP address and port to the STUN server. This creates a NAT binding.
• NAT overwrites the private source IP address and port of the packet IP header with the
bound public IP address and the port.
• The STUN server replies with a STUN Binding Response containing the public source IP
address and the port received from NAT.
• NAT routes the STUN Binding Response back to the phone private IP address and the port.
STUN limitations
The following are the limitations for using the STUN functionality:
• Symmetric NAT is not supported.
• NAT traversal mechanisms must support Network Address Port Translation (NAPT). See
RFC 3022
Traditional IP Network Address Translator (Traditional NAT)
for more details.
• NATs must support hairpinning to allow the phones behind the same NAT to communicate.
See RFC 4787
Network Address Translation (NAT) Behavioral Requirements for Unicast
UDP
for more details.
• Only unauthenticated STUN is supported.
• When the STUN feature is activated, SIP ALG functionality must be disabled for NAT
traversal mechanisms.
Related links
on page 63
Servers, VLAN, and IP configuration
April 2020
Installing and Administering Avaya J100 series IP Phones in an Open SIP
environment
62