[asterisk-users] Odd one-way audio problem (Mike Diehl)

Stefan Viljoen viljoens at verishare.co.za
Fri Mar 22 01:32:39 CDT 2019


Hi Mike

In rtp.conf, what are the port ranges you specify?

I had almost exactly the same problem not too long ago. People will phone, and sometimes it will work, sometimes not - one way audio would happen, then start working, then stop working.

The problem turned out to be that the port specification for RTP traffic in /etc/asterisk/rtp.conf was too wide.

It was set to

rtpstart=10000
rtpend=65535

(apparently by a previous maintainer / technician who worked on the system.)

The high port number was too high, and only after I investigated in detail with our trunk provider, were they able to determine that somtimes the Asterisk on my side was negotiating too high port numbers for RTP with their system.

I changed rtp.conf to read

rtpstart=10000
rtpend=20000

and all the random one-way audio problems have been gone for more than two months. This client now has had thousads of successful calls so far after this change was made.

I also had the issue where MOST calls in their office was fine (with rtp.conf at 10000 to 65535) though some would still fail, I'm guessing that was due to NATing not being done in the office (e. g. a wider "range" of RTP ports worked) vs. when they connected to their provider's SIP trunk on the internet to negotiate calls where it was ignoring the higher ports ("too high" ports) or their local firewall wasn't allowing some high ports to be opened that were "too high".

Restricting the RTP port range between 10000 and 20000 in this case solved their problem definitively and forever.

E. g. something similar given that you start that "most of the time" things worked fine - which is exactly the symptom I had with this client.

Just a thought...

Regards

Stefan

---

Hi all,

I have a user who is reporting one-way audio, but only when a call is made to or from particular PSTN (cell) numbers.

Their phones are behind a NAT router and my server is on the open Internet.  

Calls within their office sound fine.  Calls to/from most numbers sound fine.

When they took their phones home, those same phone numbers still had problems.

So, I don't think it's their network.  I've taken pcaps of both legs of example calls.  On the provider-side, I see 2-way audio.  On the client-side, I only hear one side.

Most of the time, though, their phones work correctly.

Any ideas where to look to fix this?

Thanks in advance.






More information about the asterisk-users mailing list