ST2030s VoIP Business Phone
ST2030s SIP Administrator Guide v3.0
Page 208 of 308
NAT Keep Alive
When the phone is installed behind a gateway or in another LAN than the
Proxy, it is likely that we will face NAT traversal problems.
One solution is the automatic allocation of ports by the border router or
gateway. But this allocation is not maintained forever.
So once registered with the SIP Registrar, the phone must maintain the
channel open by sending keep-alive packets to the SIP server before the
binding expires in the NAT device.
SIP OPTIONS message (cf. RFC3261 page 67) will be used to keep the NAT
open. An example of the Option packet is as follow:
OPTIONS sip:10.0.0.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.48:5060;branch=z9hG4bK5708696430465310310-6772
From: <sip:[email protected]:5060>;tag=c0a80101-1a74
To: <sip:10.0.0.5>
Call-ID: [email protected]
CSeq: 1 OPTIONS
Max-Forwards: 70
Contact: <sip:[email protected]:5060;user=phone>
Accept: application/sdp
User-Agent: THOMSON ST2030 hw5 fw1.61 00-14-7F-E1-81-F9
Content-Length: 0
Call flow example:
Two parameters have been created:
1. The destination address for the SIP OPTIONS request. It could be either
an SIP URI (or only the host part), an ip address or a domain name.
Summary of Contents for ST 2030 SIP
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