Outgoing call use different RTP port than sent in INVITE

We have 2 trunks defined in one Asteris/Issabel IVR.
We use it to make automatic calls to mobiles phones, with a recorded message.

Calls made through one of the trunks works fine.

But with the other trunk, the outgoing sound is sent through a differente port than indicated in first INVITE

Any clue about where can be the problem? Trunk configuration? Range of RTP ports?

Is this incoming or outgoing? Are you behind NAT?

Which channel driver?

If you are saying the port in the m= line that Asterisk sends from is not the source port seen by the peer, when Asterisk subsequently sends media for that stream, this is almost certainly the result of a NAT router thinking that the source port number is already in use, for another call, and dynamically allocating a new port number. If so, this has to be fixed in the router.

The port assigned in INVITE

And the port used to transmint is different

That would be valid in SIP, but is not something that Asterisk can do, although it can handle the case when the other side is doing it, as long as you enable symmetric media.

I’d say this has to be a rogue router, unless you have a hardware fault.

Note that you haven’t captured any outbound RTP, and you haven’t captured the port numbers for the inbound RTP. Also, you haven’t told us whether this is the current or the deprecated driver, and which version of Asterisk.


This the RTP traffic

Asterisk versions

Where was the capture taken, and is this the same call as the one from which the INVITE was taken? What port numbers are reported by “rtp set debug on” and “pjsip set logger on”, which are the preferred sources for this information (and assuming you are using currently supported drivers).

Note that the Issabel package names and versions don’t mean a lot, although I assume the 16.7.0 on the asterisk package is the correct asterisk version (somewhat out of date).

Hi david551,
The capture has been taken in the same server where Asterisk is running, and yes is the same call. INVITE and RTP

Hi dovi5988.

It is not behind a NAT.

Problem solved. It was related with the range of port allowed in firewall. When INVITE choose one not allowed then assign other and the communication do not work.

Thanks for your comments