RTP forward port ranged 10000-20000

FreePBX Configuration
1

Why do we use port ranged 10000-20000 for RTP instead of port 5060 ?

2

5060 is the SIP port, and 10000-20000 is the range for RTP, two different things.
Maybe if you try to explain what is it that you want to achieve.

3

I wrote a blurb once: SIP Port Forwarding

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4

I know that both kind of ports are different, but my instructor need more conviced answer about that, like went we have just one line peer to peer at the moment, why dont we use port 5060 instead of that range ?

5

sip signalling is on one port, commonly 5060, the conversation will open sdp ports as needed for media. In asterisk ,by default 10000 to 20000 but you can overide that in /etc,/asterisk/rtp.conf. , technically the last port needs to be ‘odd’ so 20000 is actually wrong, but most won’t need 5000 concurrent Chanel’s :wink: appease your network guy, allow perhaps as an example , 17000 to 17049 in rtp.conf, and have him be complicit . That way you get 24 concurrent calls, actually 12 as asterisk is likely bridging the calls, that are firewalled and legit

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6

Thanks for all supports, but i have a different answer in my mind. I think that the SIP server open port 5060 to tell that it’s the boss, and all SIP messages will have to send to it, and all other different ports are just for the client, which means port is defined which is SIP server, which is client to send the direct form.
Hopefully this is the right answer. :smiley:

7

Nope. That is not the answer. 5060 is the signalling port (SIP) and 10000-20000 is the range of audio ports (RTP), as explained before.

8

The thing that’s missing is that SIP is one protocol and RTP is another. It’s the difference between waving hello and saying hello - they are both greetings, but are completely different.

Also, using a wide range for RTP makes it harder for the NSA to listen to your phone calls, since the port number is unpredictable. Yes, there are newer and better ways to do this, and other protocols do this differently, but this protocol set (SIP and RTP) are old and established.

What you are describing is the way that IAX2 works, but that’s different protocol and is only supported on Asterisk and Asterisk-capable devices.

9

Okay, thanks again, i got it :smiley:

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