[Asterisk-video] app_mp4 problem. please help me.

Sat Nov 29 02:40:46 CST 2008

Hi all, I have been struggling with app_mp4 to record and playback video files. And I got funny things. Originally I was trying to test the application in Linux environment using linphone but failed after many trials. You know, what ever length of video I try to record (I mean like 10 sec or 20 sec), it was creating a very big video (220 secods) as i have seen it using mp4info. So, as most people talk about x-lite, i donwloaded the windows version and tried it. Originally, it was the same thing, with the following outputs 

[Nov 28 17:12:14] NOTICE[31521]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126'
[Nov 28 17:12:14] NOTICE[31521]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126'
[Nov 28 17:12:14] NOTICE[31521]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126'

But after I selected disable hardware acceleration (i am using x-lite version 3.0 build 41150), it started to record only the audio part and doesn't record the video. I can't also playback the demo videos of sergio.

In both cases the audio and video were correctly hinted. I am using Asterisk 1.4.21. and audio codec alaw and ulaw and video codec 263 and 263+.

Please help me, I am stuck with this. I have to do other things after I test this one. Please help.

I also got the following mesasge one time:

  [Nov 29 17:54:02] NOTICE[32237]: chan_sip.c:15224 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7503

Thank you.- Zelalem S. Grahamstown, SA> Date: Fri, 28 Nov 2008 10:40:55 +0100> From: klaus.mailinglists at pernau.at> To: asterisk-video at lists.digium.com> Subject: Re: [Asterisk-video] call handshake fails> > Hi!> > Thanks for the patch. Now it also works with a Sony Ericsson V800.> > regards> Klaus> > > > Dan Julius schrieb:> > Hi, Sergio,> > > > I'm following up on the problem I've been having that some (30% - 60%, > > not sure what it depends on) calls are not connected successfully when > > dialing from Samsung 3G phone to SIP client.> > > > Turns out that increasing the retransmit delay from 20 to 2000 in > > H324CCSRLayer::GetNextPdu seems to have resolved the problem.> > > > Are these units Milliseconds?> > What do you think should be a reasonable timeout?> > > > Dan> > > > > > On Thu, May 15, 2008 at 4:42 PM, Dan Julius <dan.julius at gmail.com > > <mailto:dan.julius at gmail.com>> wrote:> > > > Hi, Sergio,> > > > I actually sent these to the list a while ago, but they bounced.> > How do we deal with private attachments while still keeping the> > discussion public?> > > > Thanks for looking into this.> > > > Dan> > > > ---------- Forwarded message ----------> > From: *Dan Julius* <dan.julius at gmail.com <mailto:dan.julius at gmail.com>>> > Date: Fri, May 9, 2008 at 2:07 PM> > Subject: Re: [Asterisk-video] call handshake fails> > To: Development discussion of video media support in Asterisk> > <asterisk-video at lists.digium.com> > <mailto:asterisk-video at lists.digium.com>>> > > > > > Hi,> > > > Attached are logs for a call that failed. After answering the call> > on the mobile device, X-Lite continues to ring and nothing happens.> > As for video in working calls - the problem is with video from H324M> > to SIP. Any ideas how to debug this?> > > > Can you provide a sample for using app_transcoder?> > > > Thanks,> > Dan> > > > > > > > On Fri, May 9, 2008 at 1:27 PM, Sergio Garcia Murillo> > <sergio.garcia at fontventa.com <mailto:sergio.garcia at fontventa.com>>> > wrote:> > > > Could you send me a file with the h245 and h223 logs? (enable> > them by h324m debug level 4)> > > > The most probable cause is that you isdn provider is doing echo> > cancelation on the line, it usually causes random problems like> > this.> > > > The problem with video from SIP->H324M is that it has to be h263> > QCIF at maximun 52 kbs, if your videophone is not able to set> > this up, you'll need to use the app_transcoder module.> > > > Best regards> > Sergio> > > > ----- Original Message -----> > From: Dan Julius [mailto:dan.julius at gmail.com> > <mailto:dan.julius at gmail.com>]> > To: asterisk-video at lists.digium.com> > <mailto:asterisk-video at lists.digium.com>> > Sent: Fri, 9 May 2008 12:25:17 +0300> > Subject: Re: [Asterisk-video] call handshake fails> > > > Further info:> > > > - In the failed calls, the mobile phone never sends a> > masterSlaveDetermination packet (according to the h223 logs)> > - Asterisk sends the terminalCapabilitiesSet,> > masterSlaveDetermination and> > then continues to send OpenLogicalChannels.> > > > Is it OK to send OpenLogicalChannel before receiving a> > masterSlaveDetermination?> > > > Thanks,> > Dan> > > > On Fri, May 9, 2008 at 2:25 AM, Dan Julius <dan.julius at gmail.com> > <mailto:dan.julius at gmail.com>> wrote:> > > > > Hi, Everybody,> > >> > > I'm new to this project, so I apologize if my questions> > might have> > > already been answered elsewhere.> > > I am using a X-Lite, Asterisk 1.4.19, a Digium TE122 card,> > and a Samsung> > > Z720 phone.> > >> > > So far I have been able to make SIP-h234m calls (originating> > at either> > > side) with only partial success.> > > - I only get video in one direction, from SIP to H324M. I've> > read the posts> > > stating that SIP->H324m is actually more problematic, so I'm> > quite puzzled> > > about this.> > > - About 33% of the calls fail to negotiate a video> > connection. After> > > answering the call, nothing happens until I disconnect.> > > The out-bound h223 log of a failed call is below. Does this> > log indicate> > > that Asterisk is sending terminalCapabilitySet multiple times> > until it is> > > acknowledged?> > >> > > 1 0.000000 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2> > <http://2.2.2.2> H.245 terminalCapabilitySet> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet> > > masterSlaveDetermination masterSlaveDetermination> > masterSlaveDetermination> > > masterSlaveDetermination masterSlaveDetermination> > masterSlaveDetermination> > > masterSlaveDetermination masterSlaveDetermination> > masterSlaveDetermination> > > masterSlaveDetermination masterSlaveDetermination> > masterSlaveDetermination> > > masterSlaveDetermination masterSlaveDetermination> > masterSlaveDetermination> > > 2 0.000001 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2> > <http://2.2.2.2> H.245 openLogicalChannel> > > (generic) openLogicalChannel (generic) openLogicalChannel> > (generic)> > > openLogicalChannel (generic) openLogicalChannel (generic)> > openLogicalChannel> > > (generic) openLogicalChannel (generic) openLogicalChannel> > (generic)> > > openLogicalChannel (generic) openLogicalChannel (generic)> > openLogicalChannel> > > (generic) openLogicalChannel (generic) openLogicalChannel> > (generic)> > > openLogicalChannel (h263VideoCapability) openLogicalChannel> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)> > > openLogicalChannel (h263VideoCapability) openLogicalChannel> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)> > > openLogicalChannel (h263VideoCapability) openLogicalChannel> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)> > > openLogicalChannel (h263VideoCapability) openLogicalChannel> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)> > > openLogicalChannel (h263VideoCapability) openLogicalChannel> > > (h263VideoCapability) multiplexEntrySend multiplexEntrySend> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend> > multiplexEntrySend> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend> > multiplexEntrySend> > > multiplexEntrySend multiplexEntrySend> > > 3 0.000002 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2> > <http://2.2.2.2> H.245 multiplexEntrySend> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend> > > terminalCapabilitySetAck terminalCapabilitySetAck> > terminalCapabilitySetAck> > > terminalCapabilitySetAck terminalCapabilitySetAck> > terminalCapabilitySetAck> > > terminalCapabilitySetAck terminalCapabilitySetAck> > terminalCapabilitySetAck> > > terminalCapabilitySetAck terminalCapabilitySetAck> > terminalCapabilitySetAck> > > terminalCapabilitySetAck terminalCapabilitySetAck> > terminalCapabilitySetAck> > > terminalCapabilitySetAck> > > 4 0.000003 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2> > <http://2.2.2.2> H223> > > 5 0.000004 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2> > <http://2.2.2.2> H223> > >> > > Any pointers on how to debug this would be much appreciated.> > >> > > Thanks,> > > Dan> > >> > > PS - This is really great work and I'm very impressed with> > the project and> > > hope that I will be able to contribute as well.> > >> > >> > >> > >> > >> > >> > >> > > > > > _______________________________________________> > --Bandwidth and Colocation Provided by http://www.api-digital.com--> > > > asterisk-video mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-video> > > > > > > > > > > > ------------------------------------------------------------------------> > > > _______________________________________________> > --Bandwidth and Colocation Provided by http://www.api-digital.com--> > > > asterisk-video mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-video> > _______________________________________________> --Bandwidth and Colocation Provided by http://www.api-digital.com--> > asterisk-video mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-video

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