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

[Mike Diehl]

Hi, and thank you for your suggestion!

As it turns out, my server didn't even HAVE an rtp.conf file...  (No, I don't know 
how that happened...)

So I created one with:

rtpstart=10000
rtpend=20000

and reloaded chan_sip.

I hope that is sufficient. Or do I need to restart asterisk completely?

Anyway, my user tested later that day and they are still having problems....

Any other ideas?

Mike.


On Friday, March 22, 2019 08:32:39 AM Stefan Viljoen wrote:
> 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.

-- 
Mike Diehl                                
Diehlnet Communications, LLC.     
Sales: (800) 254-6105               
Support: (505) 903-5700                 
Fax: (505) 903-5701                      

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