RTP NAT traversal

weirulong · Brekeke Junior Member Joined: 03 Dec 2007 Posts: 8

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/networkpatterns.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

voipwell.com · Posted: Fri Dec 07, 2007 10:36 am Post subject:

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 · Brekeke Junior Member Joined: 03 Dec 2007 Posts: 8

HI,
voipwell.com

set Discover Server????
I don't install STUN server,should I install STUN server in sip server?

voipwell.com · Posted: Fri Dec 07, 2007 8:59 pm Post subject:

No Stun server is needed. That setting is not required.

weirulong · Brekeke Junior Member Joined: 03 Dec 2007 Posts: 8

I tried to revise, UAs talks fail each other.

voipwell.com · Posted: Mon Dec 10, 2007 10:37 am Post subject:

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 · Brekeke Junior Member Joined: 03 Dec 2007 Posts: 8

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 · Brekeke Newbie Joined: 11 Dec 2007 Posts: 2

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 · Brekeke Talented Joined: 16 Nov 2007 Posts: 97

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 · Posted: Tue Dec 11, 2007 7:20 pm Post subject:

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 · Brekeke Newbie Joined: 11 Dec 2007 Posts: 2

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 · Brekeke Junior Member Joined: 03 Dec 2007 Posts: 8

hi,voipwell.com!

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

Thantks!

voipwell.com · Posted: Wed Dec 12, 2007 11:15 am Post subject:

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.