OK I've been working on the recording issue that's been bugging me. In the
meantime I took a short vacation that had been planned for some time, and
during that time I read "The Fabric of the Cosmos", which, among others, talks
a little about quantum physics. And funnily enough, the bug I've been trying to
track down turns out to be a Heisenbug.
I'm working on revision 9600 with a couple of changes (Jaroslavs recording
patch plus a few minor fixes to ensure call recorder is properly set up), and
modified the ReceiveCall example so that it starts recording as soon as a call
is started (receives the CALLSTATE_CONNECTED message). If I place a breakpoint
at the line which calls sipxCallAudioRecordFileStart, followed by a F5 when
execution stops when this breakpoint is hit, the recording will succeed.
However, if no breakpoint is present, nothing will be recorded.
Obviously it would be best if I could get the same behaviour with the latest
revision of the code, but currently I'm just trying to figure out the flow of
data. I still don't completely understand who receives the RTP traffic, and how
is the data pushed to the recorders? Somehow I think I need to understand this
better, in order to find out where the bottleneck is located. I think I
understand that the Call flow graph is essential for this, but I don't
understand how the call flow graph gets the data in the first place. Could
someone explain this?
One possible explanation would be that this is somehow connected to the
minimize problem being discussed in another thread (since the console
application running ReceiveCall.exe loses focus when a breakpoint is being hit
in Visual Studio). But obviously there is no guarantee for a connection between
these two problems.
Anyway, I will continue debugging, but I will welcome any explanations to the
flow of data within sipxtapi.
Regards,
Daníel
Date: Fri, 13 Jul 2007 12:05:59 -0400From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [sipxtapi-dev] Fwd: Recording - again...CC: [EMAIL PROTECTED]
you get back, can you do a binary search to find which revision introduced the
problem?That would help immensely.
On 7/13/07, Daniel Sigurgeirsson <[EMAIL PROTECTED]> wrote:
Hi again, just wanted to let you know that I was able to reproduce this
behaviour in my test environment (one of them), and was able to get the
recording to work using an older version of the code (rev. 9600). I haven't got
time to further track down the cause since I'm going away for a few days, but I
will get back to it right away next week. Daníel
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]; [EMAIL PROTECTED]: RE:
[sipxtapi-dev] Fwd: Recording - again...Date: Wed, 11 Jul 2007 17:51:36 +0000
Hi Alexander and Keith, this version does not seem to fix the problems I've
been having. I just tried just now, and the result is the same. I've taken the
liberty to attach a ethereal/wireshark log of a single "conversation". This log
only contains sip/rtp packets, but I can also supply a full log of the
conversation. The flow of the conversation is as follows: 1. Application plays
a greeting and asks the user to enter a digit.2. User enters "4" (in this
case)3. Application asks the user to enter another digit4. User enters 25.
Application asks the user to enter a sequence of digits (a SSN): 6. User enters
"1203735289"7. Application informs the user that recording will start after the
beep.8. Application plays a beep.9. User talks for a few seconds10. User
presses the "#" key to stop recording. 11. Application asks the user to press a
number to indicate what should be done next12. User presses "2"13. Application
says goodbye and terminates the call. All sound played by the application can
be heard by the user, but voice from user is not recorded. I'm not expecting
you guys to analyze the log file in detail :-), but does it tell us if the
problem is with a configuration, i.e. is the voice data ever received on the
computer where the application is running? I'm not that familiar with the RTP
protocol to be able to tell, but hopefully this log will shed some light to it.
Oh, and BTW: the computer running the application has IP 10.10.1.36, while the
phone station has IP 10.10.90.1. Note that I also did as someone suggested,
i.e. call sipxAudioSetCallOutputDevice and sipxAudioSetRingerOutputDevice with
"NONE" as parameter. Finally: the resulting .wav file seems to contain only
zeroes in the "data" chunk. Many thanks for your help,regards,Daníel
> Date: Wed, 11 Jul 2007 15:27:09 +0400> From: [EMAIL PROTECTED]> To: [EMAIL
> PROTECTED]> CC: [email protected]> Subject: Re: [sipxtapi-dev]
> Fwd: Recording - again...> > Hello,> > As of revision 9824 full call
> recording record locally played tones and> files. This fixes issue, described
> by Keith in previous mail. > > Daniel, could you test this revision - does it
> have problems with recording> you've described or not?> > -- > Regards,>
> Alexander Chemeris.> > SIPez LLC. > SIP VoIP, IM and Presence Consulting>
> http://www.SIPez.com> tel: +1 (617) 273-4000>
> _______________________________________________ > sipxtapi-dev mailing list>
> [email protected]> List Archive:
> http://list.sipfoundry.org/archive/sipxtapi-dev/
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