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forward RTP traffic
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andrewbel
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Joined: 02 Dec 2005
Posts: 12

PostPosted: Mon Dec 03, 2007 10:51 am    Post subject: forward RTP traffic Reply with quote

1. Brekeke Product Name and version:

2. Java version:

3. OS type and the version:

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

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

6. Your problem: Is there way to config sip server to forward RTP traffic to particular UDP port?
Thanks
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voipwell.com
Partner PBX


Joined: 20 Sep 2005
Posts: 528
Location: Tannersville, Pennsylvania

PostPosted: Mon Dec 03, 2007 8:06 pm    Post subject: Reply with quote

Hi,

do you want to forward ALL rtp traffic to a port or just some calls to a particual destination? I've never seen it discussed before but there are a couple variables in the properties file that may be able to be used in the dial plan. net.rtp.port.max=10999 and net.rtp.port.min=10000 are the variables and their default values that are in the properties file. If you can set these variables in the dial plan for your particular route by setting them both to the desired port then it may work. Sorry I can't be more definitive.

Nick
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andrewbel
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Joined: 02 Dec 2005
Posts: 12

PostPosted: Tue Dec 04, 2007 12:59 pm    Post subject: Reply with quote

Hi Nick,
Thanks for reply.
I've got Linksys wireless ip-phone WIP 300 and it has NO options to configure RTP port, When the call is placed SIP server sending voice traffic to outside port 51000 on my router and I'm having a problem to forward it further. Can I configure SIP server to send media traffic to different port range, like 10000:11000 for example? I need it only for this particular UA(hardware), all other phones have an options to set up RTP port forwarding manually and it is working fine.
Andrew
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voipwell.com
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Joined: 20 Sep 2005
Posts: 528
Location: Tannersville, Pennsylvania

PostPosted: Tue Dec 04, 2007 2:23 pm    Post subject: Reply with quote

Andrew,

The sip server sends its rtp traffic out ports udp10,000-10,999. It appears your routers nat is changing it. I'm not sure from your post if the sip server is on your lan with you wireless phone or not.

If the sip server is inside your lan with the wireless phone then i would set up port forwarding on your router as follows. Let's say your sip server ip address is 192.168.1.10. Then you need to forward from the internet to your lan the following:

Port 5060 udp to 192.168.1.10
ports 10000-10999 udp to 192.168.1.10

Brekeke does nat traversal but when the sip server is on a non nat public address and the phones are behind a nat at another location. The port forwarding solves this problem.

Is this what you were talking about?

Nick
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andrewbel
Brekeke Member


Joined: 02 Dec 2005
Posts: 12

PostPosted: Tue Dec 04, 2007 2:53 pm    Post subject: Reply with quote

Sorry, Nick, I probably didn't make myself clear
My SIP server out of my LAN and on the static public IP.(something like pattern 4)
I can see that server is sent voice from port 10387 to my router's OUTSIDE port 51000. Due to specific hardware problem with my router I can NOT set up port forwarding from OUTSIDE port 51000 to inside port of Wireless IP phone.
Question: Is it possible to configure SIP server force voice traffic to be sent to port range different from 51000, for example 7000 or 8000?
Thanks again
Andrew
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voipwell.com
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Joined: 20 Sep 2005
Posts: 528
Location: Tannersville, Pennsylvania

PostPosted: Tue Dec 04, 2007 4:08 pm    Post subject: Reply with quote

Andrew,

I understand now. It is common for the router to receive the rtp on one port(from phone) and send it out on the internet on another port. That should not keep it from working since the router should know where to send the incoming packets. Brekeke's configuration only has to do with assigning it's listening port numbers 10000-10999. The port numbers that it sends back to are negotiated by your router/phone during the call setup.

Where do you perceive the problem is. Are the packets having trouble getting out or into you lan-phone? One thing you might want to make sure of is that all stun server activity is turned off on the phone. That will interfere with Brekeke's nat traversal.
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andrewbel
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Joined: 02 Dec 2005
Posts: 12

PostPosted: Wed Dec 05, 2007 12:16 pm    Post subject: Reply with quote

Hi Nick,
I have a problem receiving voice traffic inside LAN.
Again, from my router log file I can trace that RTP send out from SIP server port 10384, hit 51000 outside port of my router and then get dropped.
Andrew
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voipwell.com
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Location: Tannersville, Pennsylvania

PostPosted: Wed Dec 05, 2007 6:25 pm    Post subject: Reply with quote

Hi,

What is one of the ip addresses of a phone inside your lan?
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andrewbel
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Joined: 02 Dec 2005
Posts: 12

PostPosted: Thu Dec 06, 2007 11:01 am    Post subject: Reply with quote

192.168.2.26 my wireless ip phone private IP
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voipwell.com
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Posts: 528
Location: Tannersville, Pennsylvania

PostPosted: Thu Dec 06, 2007 7:42 pm    Post subject: Reply with quote

Andrew,

The rtp stream is being sent to port 5100 because your router is doing a nat translation. For instance a pc's browser is getting a web page from a web server using the web servers's port 80 but the nat translation sends the page back to port 5100 or 5200 because that is the port the pc is using to go thru the nat. It's not configurable. What kind of router are you using. I looked at the docs for the wip300 and it suggests you port forward which you said you can't do.
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