On 7/20/07, Jaroslav Libak <[EMAIL PROTECTED]> wrote:
> Different waiting should be implemented, so that we can wait for both
> Turn and Stun response in paralel.
I'm not too familiar with this code, but I think it could be done by
wait-on-read
model. That is we could request STUN and TURN here, a
Alexander Boreham wrote:
> Hi,
>
> Cheers for the link, it's certainly interesting. It looks to me like this
> problem has previously been found over a year ago and worked around in the
> same way as I proposed.
>
> What is curious is that the problem must have been fixed properly since that
> po
>
> After further investigation, I found out, that visual assist didn't
> correctly show me who calls OsNatDatagramSocket::markStunFailure
> therefore I thought it is never called. But it is called by
> OsNatAgentTask::markStunFailure, thus mStunState.status will be set to
> NAT_STATUS_FAILURE and
Jaroslav Libak wrote:
> Alexander Boreham wrote:
>> Hi,
>>
>> Cheers for the link, it's certainly interesting. It looks to me like this
>> problem has previously been found over a year ago and worked around in the
>> same way as I proposed.
>>
>> What is curious is that the problem must have been f
Alexander Boreham wrote:
> Hi,
>
> Cheers for the link, it's certainly interesting. It looks to me like this
> problem has previously been found over a year ago and worked around in the
> same way as I proposed.
>
> What is curious is that the problem must have been fixed properly since that
> po
Hello!
After the decision of problems with quality of a sound (now very
well), has appeared new - with transfer of calls.
Situation such: softphone_1 calls to the external_1 (asterisk,
external number), then calls to the softphone_2 and as soon as that
will respond - translates a call.
As a resu
Hi,
Cheers for the link, it's certainly interesting. It looks to me like this
problem has previously been found over a year ago and worked around in the
same way as I proposed.
What is curious is that the problem must have been fixed properly since that
post (March 06) and then broken again a few
> For me, making calls behind NAT works (with wxCommunicator and my
> provider). However, STUN doesn't work for me right now, I'm seeing icmp
> port unreachable sent from STUN server. I will have to dig into this.
Any ideas to why this maybe happening? Pointers on how to fix it?
Thanks.
Best Re
Hello,
See if this helps -
http://list.sipfoundry.org/archive/sipx-dev/msg04452.html.
Thanks.
Best Regards,
Hitesh
- Original Message -
From: "Alexander Boreham" <[EMAIL PROTECTED]>
To:
Sent: Thursday, July 19, 2007 5:09 PM
Subject: Re: [sipxtapi-dev] STUN response ignored on RTP and
Hi,
On 7/19/07, Amith Nambiar <[EMAIL PROTECTED]> wrote:
> In the following function if there is no tuple at that index,
> the code goes into the else part. And when tuple is NULL,
> tuple->remove(0), is called. Will it not SEG_FAULT ? I'm not sure whether
> i'm using the latest src'es
Hi all,
In the following function if there is no tuple at that index,
the code goes into the else part. And when tuple is NULL,
tuple->remove(0), is called. Will it not SEG_FAULT ? I'm not sure whether
i'm using the latest src'es or whether this has been fixed, if it is a bug.
UtlBool
Yup, now I'm using the latest version but with the old bridge implementation
and recording works :-)
Thanks alot!
Regards,
Daníel
> Date: Wed, 18 Jul 2007 22:18:30 +0200> From: [EMAIL PROTECTED]> To: [EMAIL
> PROTECTED]> CC: [EMAIL PROTECTED]; sipxtapi-dev@list.sipfoundry.org> Subject:
>
Hi,
I have an update on this. If I change the
CpPhoneMediaInterface::createRtpSocketPair method to pass true into the
bReadFromSocket parameter of the call to OsNatDatagramSocket::enableStun,
then the SDP is populated correctly with the STUN response.
I can't see where any listeners are set up if
Hi,
Sorry to report another problem but this is also causing issues.
The changes made in revision 9386 have caused STUN failures on the RTP and
RTCP ports. The SIP call control port still receives STUN responses
correctly. I've made many Wireshark traces and can see that the STUN
requests are cor
Hi Amith,
Thanks for the help. I agree to what you suggest. Since this is the first
step into SIP. Its always nice if there is someone to guide you through
initial steps.
Thanks again!!
Sukant
_
From: Amith Nambiar [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 19, 2007 1:33
Hi,
On 7/18/07, Kenichi Aramaki <[EMAIL PROTECTED]> wrote:
> These are several tasks that has highest priority on Windows;
> -MpMediaTask is set to THREAD_PRIORITY_HIGHEST
> (it is not very clear that this thread has highest priority)
> -dma tasks are set to THREAD_PRIORITY_TIME_CRITICAL
> Are th
Sukant,
I suggest you start with a simple (experimental ) app by
using Sipxtapi , and then build code for your project. Get familiar with the
functions SipxInitialize, SipxLineAdd, SipxLineRegister and other functions
in Sipxtapi.h
Then use the api
stack_result = sipxEventListener
Hello everyone,
I want to write a UAC, a client with minimal feature. All I want to
establish is, to register with a Gateway or UAS, Login using the user id and
password and be able to make out going calls to fixed line. No incoming
calls, no conference nothing. Following are the features I want t
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