Define an static RTP port for a Trunk?

Hi, i captured my doorintercom hardware, in screenshot below is an indoor station, for receiving calls, for some reason the ports are hardcoded
I need to run asterisk for example on port 5060 to receive the invite coming from
Seems also the audio and video ports need to be harcoded as well,

I have setup an trunk in Asterisk to register on , that works, i also receive the invite, but how can i make sure audio coming from port 9654 and video coming frol port 9856 are being sended to 9654 (asterisk) ?

thnx for the info…


There is no hardcoding possible in Asterisk. The address information is supposed to be exchanged as part of SDP negotiation.

And what about those default RTP ports between 10000 and 20000, should I change it to like 9500-10000

Or are those not the listening ports?

Those define the ports that Asterisk listens on for RTP. They are negotiated in SDP. It should not be required to match the remote side.

Indeed I should not, bit what if the negotiate is wrong ? No way to specify them manually?

If the negotiation is wrong, then it’s fundamentally against the way the specs say it should be done. There is no way to override it.

yes,probably, thats why i was wasking for to specify the port, like 5060 , if i run asterisk on another port, invite is not possible
For instance, in linphone, i could set the listening ports as well, so i was hoping it was possible also with asterisk

It’s not. It’s chosen from within the configured range. That’s it.

Ah ok, so I should setup the range to a lower anyway then then 10000-20000 default?

Let me ask this: What is the actual problem?

Well, if you see in screenshot from first post, 0.70 is sending video in one way to my indoor station…0.72

Now I want to simulate asterisk as a trunk (inbound call) like an indoor station, I can receive the call, but I don’t see video…

If I stop asterisk, and setup linphone as a trubk for testing , I also don’t see video…
Unless I change the listening port to that static port 9654 … Then it seems to work…
So that’s why I asked how to setup asterisk on that port…

What I also don’t understand, I always thought multiple RTP ports were needed to make a single audio/video call, but if I see in the screenshot, only 1 port is needed for both video and audio? How does that work?

Multiple ports are generally needed. I don’t know the details of your door intercom hardware, so can’t comment on any of that.

Yeah, there is nothing to change there, I need Todo it on asterisk side :slight_smile:

Would it be more accurate to say that the RTP range defines the limits for the pair (2) of ports that will be negotiated. This pair of ports are only listened to for the duration of the call.

Asterisk does not listen to the range (potentially 10s of thousands) of RTP ports until a pair of ports are negotiated.


That is correct.


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