Keep address/port mapping

1. Brekeke Product Name and version: Brekeke SIP Server 2.4.4.8/286

2. Java version: 1.5.0_04

3. OS type and the version: XP

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : pattern 3

6. Your problem: The keep address/port mapping does not seem to work. After some time on a connection between 1xx and 2xx, the SIP port on LAN2 closes, and a BYE from 1xx cannot be received by 2xx. "keep address/port mapping" is set on with 10000 ms interval, but (Wireshark) the server does not send any dummy message on the SIP port during the link.
What else should be set for the feature to work ?

What kind of SIP client product are you using??

Let you disable NAT/STUN feature at your SIP client.
If so, the keep address/port mapping will work.

lakeview wrote:What kind of SIP client product are you using??

Let you disable NAT/STUN feature at your SIP client.
If so, the keep address/port mapping will work.

I use a UA with sipX/2.9, and its STUN feature is disabled.

If a SIP client supports a NAT traversal feature such as STUN, it should keep a port-mapping at router or firewall..


Try this DialPlan rule..
------------------------
[Matching Patterns]
$request = ^REGISTER

[Deploy Patterns]
&register.contact.nat = true
$continue = true
------------------------

lakeview wrote:If a SIP client supports a NAT traversal feature such as STUN, it should keep a port-mapping at router or firewall..


Try this DialPlan rule..
------------------------
[Matching Patterns]
$request = ^REGISTER

[Deploy Patterns]
&register.contact.nat = true
$continue = true
------------------------

OK, as I understand it this rule forces port-mapping keeping when an agent registers and regardless whether it is detected behind NAT or not? This is the result I see and it does succeeds for keeping our agent.
Is there a similar rule possible for a call-in from an unregistered UA? (as far as they are authorised, of course)

>> OK, as I understand it this rule forces port-mapping keeping when an agent registers and regardless whether it is detected behind NAT or not?

CORRECT!

>> Is there a similar rule possible for a call-in from an unregistered UA? (as far as they are authorised, of course)

how does the SIP Server get the remote IP address of unregistered UA??

lakeview wrote:>> OK, as I understand it this rule forces port-mapping keeping when an agent registers and regardless whether it is detected behind NAT or not?

CORRECT!

>> Is there a similar rule possible for a call-in from an unregistered UA? (as far as they are authorised, of course)

how does the SIP Server get the remote IP address of unregistered UA??

Sorry, I was not clear enough! The situation is: a non-registered remote UA calls in to a registered UA. I am looking whether it is possible, during the call session, to keep the port so that e.g. the registered UA can disconnect the session (BYE). Currently, after a while the port is closed on the remote NAT, and the remote UA does not receive the BYE message.

I see.

I recommend that you enable the SIP Session-Timers at UA.
It sends dummy SIP packets frequently during the session for refreshing.

Does your UA support the Session-Timers feature?
http://www.ietf.org/rfc/rfc4028.txt

lakeview wrote:I see.

I recommend that you enable the SIP Session-Timers at UA.
It sends dummy SIP packets frequently during the session for refreshing.

Does your UA support the Session-Timers feature?
http://www.ietf.org/rfc/rfc4028.txt

So far this UA does not, but we will manage to get it implemented.
Thanks for the support