Stefan Viljoen
2019-Mar-22 06:32 UTC
[asterisk-users] Odd one-way audio problem (Mike Diehl)
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.
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190325/ddfca700/attachment.html>
On 3/25/2019 4:45 PM, Mike Diehl wrote:> > > 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. >Mike, In those pcaps, are you seeing the exact same RTP traffic between provider side and client side? And was client side captured close to the phone, past the firewall if there is one? Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190325/bc04fa0d/attachment.html>