gmane.comp.voip.sipxtapi.devel

Re: SIP Proxy

Meftah Tayeb — 2010-06-26T08:11:53

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Re: SIP Proxy

Sergey Malykhin — 2010-06-25T09:14:37

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Re: SIP Proxy

Meftah Tayeb — 2010-06-26T08:09:30

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SIP Proxy

Sergey Malykhin — 2010-06-25T07:42:37

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Remote_Held Call State when answering a call

Filipe Paredes — 2010-02-25T20:07:38

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Re: Building sipXtapi-media-update branch for Windows

Manuel Pimenta — 2010-02-10T19:01:00

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ACD with G.729

2010-02-10T17:37:57

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Building sipXtapi-media-update branch for Windows

Manuel Pimenta — 2010-02-09T10:06:39

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sipxAudioSetVolume() andsipxAudioSetCallInputDevice()

Marcin Głowacki — 2010-01-11T10:50:54

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Re: Presence problem second notify answered with 481

Daniel Petrie — 2010-01-07T04:23:22

Hi Jan:
I think I understand the problem.  Thank you for your explanation.  It would be very helpful if you could send me a clean sipXtapi log file with the debug level enabled.  This way I can use the transaction data in your log to test and analyze the issue.

Cheers,
Dan

--- On Wed, 1/6/10, Jan Thiemo Fricke <fanatikneo< at >gmx.de> wrote:

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Re: Presence problem second notify answered with 481

Jan Thiemo Fricke — 2010-01-06T17:19:14

Hi Dan,
I had a further look into the code of sipxtapi the last days.
Some things I figured out:

- The 481 just occurs in SipXtapi 3.3 (in 3.2 everything works fine)
- The problem just occurs if Subscription needs authentication
(subscribe, 401, subscribe, 202; If I subscribe reg-event I don't need
authentication and there is no problem)

I believe there is a problem with the dialog management with
authentication. SipXtapi does not recognize 202 after the subscribe with
authentication. It answers the first Notify with 200OK because it's
possible that the 2xx response arrives after the first notify. But after
the second notify sipxtapi decides that there is something wrong.

The point is that the second subscribe (with authentication data) does
not trigger the update of the earlyDialog. When the 202 arrives the
dialog is not updated because the earlyDialog CSeq is still 1 and the
CSeq of the 202 is already 2 because it was increased by second subscribe.

SipRefreshManager::handleMessage (Line 651) checks if

Re: New SimpleDtmfDetector

Daniel Sigurgeirsson — 2010-01-01T20:25:31

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Re: Problems with some end devices and sipxtapi

Daniel Petrie — 2009-12-25T22:45:19

Hi Jan:
For the crash problem, a stack trace of where the crash occurs  along with a sipX log of what SIp signalling events just occured would be most helpful.

For Siemens problem, a sipX log containing the SIP signalling are needed to understand the issue. 

Please post these as separate issues to the list.

Cheers,
Dan

--- On Fri, 11/13/09, Jan Thiemo Fricke <fanatikneo< at >gmx.de> wrote:

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Re: How to play the RTP packets as soon as it comesto the UDP buffer.

Daniel Petrie — 2009-12-25T22:23:59

Hi Anup:
Please provide some more context.  I am not sure you are headed down the best path for what you are trying to do.  Please explain at a high level what you are trying to accomplish and the requirements of your application related to the RTP handling.

Also have a look at 
http://sipxtapi.sipfoundry.org/wiki/SipXmediaLib_Overview
In particular look at the Flowgraph concept.  The media subsystem provides a means to handle RTP streams and send them to standard or customized devices (e.g. mic and speakers).  There are means to record audio to a buffer or file.

Cheers,
Dan

--- On Fri, 12/18/09, Anup <anup< at >rhubcom.com> wrote:

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Re: Presence problem second notify answered with 481

Daniel Petrie — 2009-12-25T22:14:28

Hi Jan:
The remote tag (To field) on the initial SUBSCRIBE request should not be set as it is initiating a new transaction as the dialog is not established yet.  Then when the 200 response comes back, it should have the remote tag (To field) set.  In updateDialog the dialog manager should match a dialog with no remote tag (early dialog) with the 200 response transaction information and then set the remote tag of the dialog in the list.  In some cases the NOTIFY request beats the SUBSCRIBE response to the client initiating the SUBSCIRBE request and dialog.  However I believe the dialog manager handles this case.

Cheers,
Dan

--- On Wed, 12/23/09, Jan Fricke <fanatikneo< at >gmx.de> wrote:

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Re: Presence problem second notify answered with 481

Jan Fricke — 2009-12-23T08:33:20

Hello again,
I figured out that sipxtapi does not find the dialog on second notify
because it seems the remote-tag of the stored dialog is empty.

I debugged SipSubscribeClient::handleNotifyRequest.

The SipDialogMgr should store the dialog (including callId, from-tag,
to-tag) on first notify. It seems to do this correct at
mpDialogMgr->updateDialog(notifyRequest, notifyDialogHandle);


But when the second notify message arrives foundDialog is false because

UtlBoolean foundDialog = mpDialogMgr->dialogExists(notifyDialogHandle);

SipDialogMgr::dialogExists iterates through all dialogs. The
corresponding (with the correct callId and localTag) has no remote-tag.
That should be the reason why sipxtapi does not recognize that the
second notify belongs to the dialog of the first notify.

I could not figure out whats going wrong at storing the dialog. Is there
anybody who can assist me with that?



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Presence problem second notify answered with 481

Jan Fricke — 2009-12-22T09:49:06

Hi members,
hope someone can help me.
My setup:
SipX 4.0.1
sipXtapi 3.3 (SVN Rev 11467) compiled with Visual Studio

I try to subscribe presence states of some phones by subscribing the
resourcelist at our sipx.
The subscription is answered with an 202 followed by a Notify message.
The second Notify message from the presence server is answered by
sipxtapi with 481 Transaction does not exist.
Below you find the subscribe and notify messages.

What does transaction does not exist mean? The second notify belongs to
the dialog initiated by the subscribe message (same call-id same from
and to header tags). The CSeq is increased by one and a new via branch
was created. So I see no problem. Same dialog, new transaction.

Thanks in advance

Jan




SUBSCRIBE sip:~~rl~F~7800< at >voip.mydomain.de SIP/2.0
From: <sip:7800< at >voip.mydomain.de>;tag=35a7b2be8686cf178af07ddbcd51b507
To: <sip:~~rl~F~7800< at >voip.mydomain.de>
Call-Id:
ssc-7ab8c84ac338b5ab85e57d567839783c< at >28b70d10c2d661262affe918ab9e2955
Cseq: 2 SUBSCRIBE
Contact: sip:7

How to play the RTP packets as soon as it comes tothe UDP buffer.

Anup — 2009-12-18T11:46:06

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Conference with one-way audio?

A S — 2009-12-07T17:01:05

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Re: [sipxtapi] QOP - Quality of Protection

Paulo Vicentini — 2009-12-01T14:15:19

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Re: PCM a/u poor quality

Paulo Vicentini — 2009-11-30T16:40:59

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