RTP NAT traversal

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weirulong
Posts: 8
Joined: Mon Dec 03, 2007 10:43 pm

RTP NAT traversal

Post by weirulong »

1. Brekeke Product Name and version:
SIP PROXY 2.1
2. Java version:
1.5
3. OS type and the version:
WIN2003
4. UA (phone), gateway or other hardware/software involved:
X-LITE
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
Pattern 3
6. Your problem:
The phone 201 calls phone 202, after the connection: The phone 201 can't receive the phone 202 RTP packets, the phone 202 can't receive the phone 201 RTP packets either.We used Ethereal in the UAS'S PC,it shows that the packets send to LAN IP ,not public IP address.

how to set in the SIP proxy and in detail.

I set as following:
Set port forwarding on the router which does NAT. Forward the following ports to Brekeke SIP Server's IP address.
SIP exchanger - Local Port [UDP]
Default Value: 5060
RTP Exchanger - From Minimum Port to Maximum Port [UDP]
Default value: 10000-10999
Also, set interface address on Brekeke SIP Server.
Go to Brekeke SIP Server Admintool > Config > System > Network
Set [Interface Address #] = my router's global IP Address



Thanks
Last edited by weirulong on Fri Dec 07, 2007 7:25 pm, edited 1 time in total.
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

Hi,

That's very strange. Open up the sip config on the two remote xlite phones and make sure the following fields are set under the topology tab.

Discover global ip address
Discover Server
Uncheck ICE

In the advanced tab

Register every 60 seconds
Send sip keepalives
User Rport
weirulong
Posts: 8
Joined: Mon Dec 03, 2007 10:43 pm

I don't install STUN server.

Post by weirulong »

HI,
voipwell.com

set Discover Server????
I don't install STUN server,should I install STUN server in sip server?
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

No Stun server is needed. That setting is not required.
weirulong
Posts: 8
Joined: Mon Dec 03, 2007 10:43 pm

:)

Post by weirulong »

I tried to revise, UAs talks fail each other.
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

Very strange. Is the lan ip address range for phones 201 and 202 in the private address range? Is rtp on during the conversation?
weirulong
Posts: 8
Joined: Mon Dec 03, 2007 10:43 pm

Post by weirulong »

The lan private ip address range for phones 201 and 202 is 192.168.0.0/24.
During the conversation,phones 201's RTP packets send to SIP PROXY's lan ip address(134.160.x.x),not SIP PROXY's public IP;phones 202's RTP packets send to SIP PROXY's lan ip address too.

We also found ,during the conversation phone 101 with 201,phone 101 RTP packets send to 201's public ip address,but 201 can't receive.
freeflying
Posts: 2
Joined: Tue Dec 11, 2007 1:44 am

Post by freeflying »

It's really strange. We know 134.160.X.X (used as private IP address in Lan1)is defined as public IP address used in AU. Is there any connection with 134.160.X.X ??
redroof
Posts: 97
Joined: Fri Nov 16, 2007 1:46 pm

Post by redroof »

As freeflying mentioned, 134.160.X.X is not a private IP address.
it is somebody's global IP address.
so you should not use it as your private IP address.
You need to use valid private IP address in the LAN which SIP Server is working.
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

Hi,

The string net.rtp.follow.remoteaddr = true appears to help you use a public wan address as a private lan address but it breaks the communication within the lan so instead of putting it in the properties config file put it in the dial plan so it only gets used when communication is hitting that public wan address you are using as a lan address.
freeflying
Posts: 2
Joined: Tue Dec 11, 2007 1:44 am

Post by freeflying »

Hi,VoipWell
How can I add this rule in dial plan?Please tell me in detail.I read the document,but didn't find it.
Thanks a lot!
weirulong
Posts: 8
Joined: Mon Dec 03, 2007 10:43 pm

Post by weirulong »

hi,voipwell.com!

Can you write an example to me in the dial plan?

Thantks!
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

Hi,

After rereading you post I see your problem is the reverse of the situation I experienced. I had remote ua's behind nat routers using public wan addresses. You have the actual sip server behind nat using public address space. So, I would place net.rtp.follow.remoteaddr = true in the deploy of from pbx2 and test and then in to pbx and test. Try different combinations until you find what works for you.
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