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 ?
Keep address/port mapping
Moderator: Brekeke Support Team
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.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]
®ister.contact.nat = true
$continue = true
------------------------
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??
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.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??
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
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.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
Thanks for the support
