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=1
rtpend=2
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=1
> 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=1
> rtpend=2
>
> 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 1 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 1 and 2 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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