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Hi,
Just pass username and password as
null.
-Hitesh
----- Original Message -----
Sent: Saturday, October 07, 2006 1:38
AM
Subject: Re: [sipxtapi-dev] NAT
Issue
Hi,
In sipXConfigEnableTurn, there are parameters for a turn username
and password. Should I register for a username and a password in order
to use the turn server?
Thank you.
Joseph
On 10/7/06, logan
<[EMAIL PROTECTED]>
wrote:
Hello,
Best Wishes,
Hitesh
----- Original Message -----
Sent: Friday, October 06, 2006 8:40
PM
Subject: Re: [sipxtapi-dev] NAT
Issue
Hi,
I have a problem with enabling turn. I don't have a turn
server. Right now I'm using stunserver.org for my
stunserver (is that a good idea?). Are there any turn servers
available for public use like stunserver.org? Do I have
to set up my own turn server? Did you use your own turn server?
Thanks,
Joseph
On 10/6/06, logan <[EMAIL PROTECTED] > wrote:
Hello,
Prsently I'm myself
struggling while making calls across NAT. My old builds of sipXtapi work
very well, but with the latest ones I'm not able to hear any kind of
audio. I have posted this message on the list but haven't got a reply.
The settings that I use are
to enable Stun, Turn, Ice, and Rport.
If this doesn't work for
you then I can send my old builds (that work very nicely for me) to you
to test out your app.
Best Wishes,
Hitesh
----- Original Message -----
Sent: Friday, October 06, 2006
7:55 PM
Subject: Re: [sipxtapi-dev] NAT
Issue
Hi,
I already changed the line in CpPhoneMediaInterface.cpp just as Mike Cohen suggested in
his post. It still doesn't seem to work. Well, it did work
a few times but most of the time I still can't hear the remote
end. I already used sipxConfigEnableStun. Is
there anything else I need to do? Thank you.
Regards,
Joseph
On 10/5/06, logan <[EMAIL PROTECTED] > wrote:
Hi Joseph,
It worked for me and if it works for
you then let me know.
Best Wishes,
Hitesh
----- Original Message -----
Sent: Thursday, October 05,
2006 7:33 PM
Subject: [sipxtapi-dev] NAT
Issue
hi,
when i make calls using my sipxezphone, i get problems in
hearing audio. when i try calling another sipxezphone that's
not on the same LAN as me, can't hear the remote end.
sometimes the remote end can't hear me. the problem seems to
be coming from the firewalls in the routers. so, i figured
it's a NAT issue. i tried using sipxConfigEnableStun. i would like to try work.
does anyone have any idea on how to solve this? thank
you.
regards,
joseph
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