On Mon, Oct 22, 2018 at 2:53 PM, Grant Griesel
<[email protected]> wrote:
Thanks so much for your reply.
I will be honest, much of what you said required a google but I think
I have understood.
> Have you tried switching services?
>>> I have switched to Skype through my PC and I am able to make
calls without trouble. I have also used a different VOIP system
called Keyyo through an old Macbook using the same internet
connection without trouble.
No, you need to try a different SIP service with your failing device to
see whether it is linphone or your service that is failing (you have
already eliminated hardware by trying skype on the same device -
although that is not SIP). For instance, you can register with
ekiga.net (if they are still working), or add your Keyyo account to
linphone (assuming Keyyo is SIP).
> Can you use an echo-test service?
>>> I can use one and have done so without a problem.
Assuming you did the echo-test with linphone, then that indicates that
linphone is not problem. However, if the echo-test was provided by the
same SIP service that is giving you problems, then the result is more
ambiguous.
>the intermittent nature of your problem screams to me "NAT firewall
issue".
>>> I went into my router and found "NAT virtual server set up".
There was a server set up under the name Whatsapp... I removed this
and disabled NAT and I will run a few tests making calls tomorrow to
see if there is any success.
In addition I checked the settings on Linphone and within the
"settings" menu under "network settings" there are four options at
the bottom of the menu page;
'direct connection to the internet'
'Behind NAT/Firewall (specify gateway IP)'
' Behind NAT/Firewall(use STUN to resolve)'
' Behind NAT/Firewall(use ICE)'
Currently " 'direct connection to the internet' is selected and I
have not changed this. Should I change it?
That all depends on whether you have a direct connection to the
internet. :-) Does the IP address of your device look like
192.168.*.* or 172.16.*.* or 10.*.*.* ? If so, then you do not have a
direct connection. You are going to have to provide more details.
You should at least let us know what SIP service you are trying to use.
By "NAT firewall", I mean't an external firewall, like a home router.
Often, these will set up a "session" when you first connect out, that
then times out at an inopportune time. NAT is evil. IPv4 is obsolete
for 15 years.
> You just have to swear off IPv4 just like you have to delete your
Facebook
>>>In the same "network settings" section mentioned above there is an
option to tick "Allow IPv6". In the spirit of a process of
elimination I will not select this option until I have tested the
effect of changing the router, but it will be my next port of call.
It will not help you to make calls through a centralized SIP service
(like your are apparently using now). And it will not work at all
until you actually have IPv6.
The purpose of IPv6 is to allow direct peer to peer SIP calls (and
emails, and innumerable other decentralized protocols), with no NAT, so
that no centralized service is needed. If you have a fixed IPv6
address, then you can use that like a phonenumber with linphone
(assuming parties you wish to call can also be educated on peer to
peer). For instance, if your login is "griesel" and your IPv6 address
is 2001:db8::1234, then you receive peer to peer calls at
sip:grieser@[2001:db8::1234] without registering with any service or
server. These can be added to the address book just like ancient phone
numbers. (You can alse give names to your IPv6 via DNS, but that is
another topic.) Even if you still need to use a service to connect to
the 100 years obsolete telco system (or prisoners in IPv4 NAT jail),
making peer to peer calls is a valuable test to narrow down problems.
If you have other devices with peer to peer capable SIP software on the
same LAN, you can test peer to peer calls using your private IP. E.g.
call sip:[email protected] from another SIP capable device if your
PC address is 192.168.1.245. That will only work within the LAN, of
course.
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