Audio & RTP

[dancing_tux]

Hi All,

I'm new with PBX but after some research i have managed to setup a PBX. However, this only works well internally (extn <=> extn). If i do set up a call to an external phone number, the number rings but there is no audio both ways.

After research on this matter, it seems that there is a requirement to open UDP port 10000 - 20000 and forward the traffic to the PBX. As these are 10,000 ports which i will need to open its something which im not too keen on. Would like to understand where does the traffic come from. Would it be from my SIP provider?

The research online provides mixed insights. Some answers says that it is from the SIP provider and while some of them say that it is from random IP Address.


We currently have our SIP phones which is connected to the SIP trunk directly (without a PBX) which works fine for 2-way audio. Im wondering why there is a requirement for such to be done. Are the SIP phones in essence SIP extensions which is forwarded from the PBX which lies at the SIP provider.

Please advise if i am looking at a wrong angle.

 

[orsini]

Hello,

I've read both your posts regarding configuration issues. You do have to have a few ports forwarded for your PBX to function properly.

5060(UDP)
10000-20000(UDP)

Both of these should be forwarded from your public IP to your PBX's LAN IP. The first (5060) is for call setup, the second (10000-20000) is for the audio or video. I understand your concern about opening such a large number of ports. If you know the maximum amount of calls you will have at a time you can limit the number of ports used via the configuration files. Each concurrent call would require 2 ports. If you plan to have 10 concurrent calls you can edit the config file for RTP=10000-10020 and open your firewall with just those ports.