Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)
Hi Mike Oh yes, this: --- Note that the call ID is much longer than in the column display, e. g. a visual call ID in the sip show channelstats display may be 31f867c50ce but the full call ID is then 32d867a55cfb563b7f59da01de84d...@xxx.xxx.xxx.xxx:5060 The full ID can be obtained by typing the partial ID as displayed in the list in the Asterisk CLI and then hitting TAB on the keyboard. --- will only work if you go sip show channel 31f867c50ce and THEN press TAB on the keyboard. Apologies. Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one-way audio problem (Mike Diehl
Hi Mike No rtp.conf in /etc/asterisk?! This might be part of your problem, AFAIK rtp.conf is part of any standard installation might it be that you have a corrupted Asterisk install and / or -other- missing conf files as well? Might this not also be part of your problem? I installed Asterisk 13 by downloading the 13.22.0 version from Digium as a source .tar.gz and compiling and installig it myself, by hand. Not sure if yours comes from a distro repo or something similar which has been modified or is not a standard install. That might also be something to try, e. g. see if you can compile a "clean" instance, from canon community version source, from Digium itself directly, and see if that changes anything. Note that the 1 and 2 ranges I show below was mandated by my voice service provider, yours may differ! For example, I run 17 instances of Asterisk in South Africa with Centracom (my voice service provider) and in South Africa I have to use 1 - 2. However, we also run a branch in Colorado in the United States with Twilio LLC as voice service provider in the USA, and they -require- 1 - 65535 to be used in rtp.conf... or you get exactly what you describe, e. g. random calls have no audio in either of the two directions. For reference, your /etc/asterisk/rtp.conf file has a set format, e. g. it has to look like this (to parse, and work, I assume): /etc/asterisk/rtp.conf --- ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 ; ; Whether to enable or disable UDP checksums on RTP traffic ; ;rtpchecksums=no ; ; The amount of time a DTMF digit with no 'end' marker should be ; allowed to continue (in 'samples', 1/8000 of a second) ; ;dtmftimeout=3000 ; rtcpinterval = 5000 ; Milliseconds between rtcp reports ;(min 500, max 6, default 5000) --- As you can see, defaults (if no rtp.conf specified) is 5000 and 31000 which is probably what you have been using up to this point, since you state that there was no rtp.conf on your system. Also, you might need to do a "core reload" to make any changes to rtp.conf effective. A sip reload only re-parses sip.conf and reloads the SIP substructure and channel drivers, AFAIK it a sip reload doesn't affect RTP and doesn't parse and apply rtp.conf. As regards the problems, are you sure that (for example) the audio codecs used are fully specified on, for example, the phones on either side? With Yealink T-21 phones for example (and older ATCOM 820's and some older Polycoms) it is possible to turn off some codecs, and turn others on - I once had this kind of intermittent voice issue but only on a certain extension, and it turned out that the particular Yealink's codec mix was set too narrow (it would only accept ulaw and alaw, not G729) and this was precluding that phone from receiving incoming calls from our service provider (which only offer G729) while in-office calls to and from it worked fine as the other phones in the office were allowing running ulaw and alaw, -and- G729 so that any phone could take an incoming call (in G729) and have audio. Not sure of the dynamics (as Asterisk AFAIK should handle this situation transparently, transcoding as needed AFAIK) but it was definitively fixed by allowing G729 on the particular Yealink phone's web interface. The only remaining issue may be NATing, e. g. for a particular network path to / from a particular client, the intervening firewalls and / or routers might not allow the required UDP ports to be opened to allow RTP flow. You can set what you like in rtp.conf, but all the intervening network segments you use to get to the phone (and your VOIP trunk provider / service provider) -also- need to allow all the necessary UDP ports to be opened so RTP can flow back and forth. This might again be why it sometimes works, e. g. sometimes RTP ports are negotiated on port numbers that are too low / too high for a particular network segment or network edge device like a router or hardware / software firewall, and then there's no audio. The RTP port number is too low or too high for the intervening routers / firewalls, even though Asterisk is perfectly fine with that RTP port number, the intervening network(s) is incapable of carrying the RTP flow as the port number is invalid according to a router / firewall configuration on a device between you and the phone. If a customer is in a state of no audio, what does sip show channelstats in the Asterisk CLI, show? E. g. the "sip show channelstats" display shows RTCP (RTP control protocol) statistics, e. g. how bad the jitter and packet loss for a particular call is on a certain extension for the incoming and outgoing legs of the call. In a situation with no audio, can you check on this to see if (for example) -any- RTP packets flow at all? E. g. a particular readout
Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one-way audio problem (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=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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Re: [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=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.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
