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darcy at Vex.Net
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 Posted: Sat Aug 15, 2015 10:09 am Post subject: [asterisk-users] One way audio - doesn't seem to be NAT issu |
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On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree <michael@easybitllc.com> wrote:
| Quote: | Not 100% ure, but maybe play with the canreinvite or directmedia
settings.
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Yes! That was it. Just for future searches here is what I did. I
added "directmedia = no" in sip.conf. This fixed the issue.
I believe that Asterisk was getting confused when one leg was inside
NAT and the other was outside. Perhaps there was an "OR" where there
should be an "AND". It makes sense because the other user was the one
outside NAT and he could hear me and I could not hear him no matter who
initiated the call. He could make outside calls because both he and my
provider were on public IPs.
I am not sure why this hasn't bit anyone else. Perhaps most Asterisk
systems are in one of two classes, connecting to all NAT phones or
connecting to all public phones, and I am in a minority situation where
I am talking to a mix of setups.
Thanks for that. I was going nuts trying to figure this out.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
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jcolp at digium.com
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| Posted: Sat Aug 15, 2015 10:43 am Post subject: [asterisk-users] One way audio - doesn't seem to be NAT issu |
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On Sat, Aug 15, 2015, at 12:08 PM, D'Arcy J.M. Cain wrote:
| Quote: | On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree <michael@easybitllc.com> wrote:
| Quote: | Not 100% ure, but maybe play with the canreinvite or directmedia
settings.
|
Yes! That was it. Just for future searches here is what I did. I
added "directmedia = no" in sip.conf. This fixed the issue.
I believe that Asterisk was getting confused when one leg was inside
NAT and the other was outside. Perhaps there was an "OR" where there
should be an "AND". It makes sense because the other user was the one
outside NAT and he could hear me and I could not hear him no matter who
initiated the call. He could make outside calls because both he and my
provider were on public IPs.
I am not sure why this hasn't bit anyone else. Perhaps most Asterisk
systems are in one of two classes, connecting to all NAT phones or
connecting to all public phones, and I am in a minority situation where
I am talking to a mix of setups.
|
Most people run without direct media unless they know the network
topology will allow it 100%.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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darcy at Vex.Net
Guest
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| Posted: Sat Aug 15, 2015 7:03 pm Post subject: [asterisk-users] One way audio - doesn't seem to be NAT issu |
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On Sat, 15 Aug 2015 12:42:38 -0300
Joshua Colp <jcolp@digium.com> wrote:
| Quote: | | Quote: | I am not sure why this hasn't bit anyone else. Perhaps most
Asterisk systems are in one of two classes, connecting to all NAT
phones or connecting to all public phones, and I am in a minority
situation where I am talking to a mix of setups.
|
Most people run without direct media unless they know the network
topology will allow it 100%.
|
Perhaps but the default is to run it. Perhaps the default should be
"no" to prevent these problems.
On the other hand, the documentation seemed to suggest that the default
should have worked anyway. One leg was public, the other behind a
NAT. It should recognize the latter and not try to put then in direct
contact. It's almost like it saw the public one and didn't bother
checking the other. Or, it checked both with an OR instead of an AND
as I said. That seems more likely since it didn't matter who started
the call.
I don't really care at this point. If 1% of the calls go through the
server when they didn't really need to it's no big deal.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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