RTP Mapping Nat users

sibernon · Post by **sibernon** » Mon Apr 30, 2007 4:52 am

1. Brekeke Product Name and version:1.5.3

2. Java version:

3. OS type and the version:windows XP

4. UA (phone), gateway or other hardware/software involved: Cisco 186

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Pattern 4

6. Your problem:

Hi, I want to configure ondo as rtp mapping as

Phone 104: 800511
Phone 204: 800512

Matching Pattern
$request=^INVITE
To=sip:(800.+)@

Deploy Pattern
$rtp=false
To=sip:%1@

Ringing Ok.
But when calle answered phone there is no voice each-side.

each users modem says "destination port unreachable"

BUT If "rtp=true" voice is ok.

as If we carriying voice. But we dont want to do this.

Is it possible doing rtp mapping which users behind NAT?

==========================
Second question;
is it possible generating fake ringtone on OnDo?

Thanx

voipwell.com · Post by **voipwell.com** » Tue May 01, 2007 6:41 am

Hi,

RTP and nat traversal are related meaning there is no nat traversal without RTP relay. There can't be since Brekeke would then not be involved in the RTP negotiation. If you want to not have rtp relay and still make things work you would need to try to set up STUN on the UA's or do port forwarding to the UA's.

Nick

sibernon · Post by **sibernon** » Wed May 02, 2007 1:14 am

Hi Nick, Thanx for your prompt response.

as you said that;
If you want to not have rtp relay and still make things work you would need to try to set up STUN on the UA's or do port forwarding to the UA's.

is there any examples wich is configurable basicly on the internet which is you tried.

And also, I asqed another question.

Is it possible that generating fake ringtone on OnDo.

Thanx

voipwell.com · Post by **voipwell.com** » Thu May 03, 2007 4:00 pm

I don't think you can fake a ring tone but I do recall Brekeke used to facke a ring tone in earlier versions but when they did I would hear someone answer the phone while the ring tone continued for a brief moment.

Port forwarding is dependent on your router so look at some web references that refer to port forwarding with your router. You want to forward the sip port usually 5060 udp and the rtp ports for example for a pap2 would be 16384-16482 udp.

Stun usually requires you to input the stun server address into the ua. Free world dialup has a stun server that they make available to the public.

sibernon · Post by **sibernon** » Fri May 04, 2007 4:58 am

Which Ondo version supported fake ringtone?

voipwell.com · Post by **voipwell.com** » Sat May 05, 2007 9:15 am

I don't remember the version but it was July 2005.