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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 <BR>
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<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT face="Times New Roman" color=#000000 size=3>[Nov 28 17:12:14] NOTICE[31521]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126'</FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT face="Times New Roman" color=#000000 size=3>[Nov 28 17:12:14] NOTICE[31521]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126'</FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT face="Times New Roman" color=#000000 size=3>[Nov 28 17:12:14] NOTICE[31521]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126'</FONT></P>
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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.<BR>
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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+.<BR>
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Please help me, I am stuck with this. I have to do other things after I test this one. Please help.<BR>
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I also got the following mesasge one time:<BR>
<BR></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><B style="mso-bidi-font-weight: normal"><FONT size=3><FONT color=#000000><FONT face="Times New Roman"> [Nov 29 17:54:02] NOTICE[32237]: chan_sip.c:15224 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7503<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /><o:p></o:p></FONT></FONT></FONT></B></P>
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Thank you.<BR><BR>- Zelalem S. <BR>Grahamstown, SA<BR><BR><BR><BR>> Date: Fri, 28 Nov 2008 10:40:55 +0100<BR>> From: klaus.mailinglists@pernau.at<BR>> To: asterisk-video@lists.digium.com<BR>> Subject: Re: [Asterisk-video] call handshake fails<BR>> <BR>> Hi!<BR>> <BR>> Thanks for the patch. Now it also works with a Sony Ericsson V800.<BR>> <BR>> regards<BR>> Klaus<BR>> <BR>> <BR>> <BR>> Dan Julius schrieb:<BR>> > Hi, Sergio,<BR>> > <BR>> > I'm following up on the problem I've been having that some (30% - 60%, <BR>> > not sure what it depends on) calls are not connected successfully when <BR>> > dialing from Samsung 3G phone to SIP client.<BR>> > <BR>> > Turns out that increasing the retransmit delay from 20 to 2000 in <BR>> > H324CCSRLayer::GetNextPdu seems to have resolved the problem.<BR>> > <BR>> > Are these units Milliseconds?<BR>> > What do you think should be a reasonable timeout?<BR>> > <BR>> > Dan<BR>> > <BR>> > <BR>> > On Thu, May 15, 2008 at 4:42 PM, Dan Julius <dan.julius@gmail.com <BR>> > <mailto:dan.julius@gmail.com>> wrote:<BR>> > <BR>> > Hi, Sergio,<BR>> > <BR>> > I actually sent these to the list a while ago, but they bounced.<BR>> > How do we deal with private attachments while still keeping the<BR>> > discussion public?<BR>> > <BR>> > Thanks for looking into this.<BR>> > <BR>> > Dan<BR>> > <BR>> > ---------- Forwarded message ----------<BR>> > From: *Dan Julius* <dan.julius@gmail.com <mailto:dan.julius@gmail.com>><BR>> > Date: Fri, May 9, 2008 at 2:07 PM<BR>> > Subject: Re: [Asterisk-video] call handshake fails<BR>> > To: Development discussion of video media support in Asterisk<BR>> > <asterisk-video@lists.digium.com<BR>> > <mailto:asterisk-video@lists.digium.com>><BR>> > <BR>> > <BR>> > Hi,<BR>> > <BR>> > Attached are logs for a call that failed. After answering the call<BR>> > on the mobile device, X-Lite continues to ring and nothing happens.<BR>> > As for video in working calls - the problem is with video from H324M<BR>> > to SIP. Any ideas how to debug this?<BR>> > <BR>> > Can you provide a sample for using app_transcoder?<BR>> > <BR>> > Thanks,<BR>> > Dan<BR>> > <BR>> > <BR>> > <BR>> > On Fri, May 9, 2008 at 1:27 PM, Sergio Garcia Murillo<BR>> > <sergio.garcia@fontventa.com <mailto:sergio.garcia@fontventa.com>><BR>> > wrote:<BR>> > <BR>> > Could you send me a file with the h245 and h223 logs? (enable<BR>> > them by h324m debug level 4)<BR>> > <BR>> > The most probable cause is that you isdn provider is doing echo<BR>> > cancelation on the line, it usually causes random problems like<BR>> > this.<BR>> > <BR>> > The problem with video from SIP->H324M is that it has to be h263<BR>> > QCIF at maximun 52 kbs, if your videophone is not able to set<BR>> > this up, you'll need to use the app_transcoder module.<BR>> > <BR>> > Best regards<BR>> > Sergio<BR>> > <BR>> > ----- Original Message -----<BR>> > From: Dan Julius [mailto:dan.julius@gmail.com<BR>> > <mailto:dan.julius@gmail.com>]<BR>> > To: asterisk-video@lists.digium.com<BR>> > <mailto:asterisk-video@lists.digium.com><BR>> > Sent: Fri, 9 May 2008 12:25:17 +0300<BR>> > Subject: Re: [Asterisk-video] call handshake fails<BR>> > <BR>> > Further info:<BR>> > <BR>> > - In the failed calls, the mobile phone never sends a<BR>> > masterSlaveDetermination packet (according to the h223 logs)<BR>> > - Asterisk sends the terminalCapabilitiesSet,<BR>> > masterSlaveDetermination and<BR>> > then continues to send OpenLogicalChannels.<BR>> > <BR>> > Is it OK to send OpenLogicalChannel before receiving a<BR>> > masterSlaveDetermination?<BR>> > <BR>> > Thanks,<BR>> > Dan<BR>> > <BR>> > On Fri, May 9, 2008 at 2:25 AM, Dan Julius <dan.julius@gmail.com<BR>> > <mailto:dan.julius@gmail.com>> wrote:<BR>> > <BR>> > > Hi, Everybody,<BR>> > ><BR>> > > I'm new to this project, so I apologize if my questions<BR>> > might have<BR>> > > already been answered elsewhere.<BR>> > > I am using a X-Lite, Asterisk 1.4.19, a Digium TE122 card,<BR>> > and a Samsung<BR>> > > Z720 phone.<BR>> > ><BR>> > > So far I have been able to make SIP-h234m calls (originating<BR>> > at either<BR>> > > side) with only partial success.<BR>> > > - I only get video in one direction, from SIP to H324M. I've<BR>> > read the posts<BR>> > > stating that SIP->H324m is actually more problematic, so I'm<BR>> > quite puzzled<BR>> > > about this.<BR>> > > - About 33% of the calls fail to negotiate a video<BR>> > connection. After<BR>> > > answering the call, nothing happens until I disconnect.<BR>> > > The out-bound h223 log of a failed call is below. Does this<BR>> > log indicate<BR>> > > that Asterisk is sending terminalCapabilitySet multiple times<BR>> > until it is<BR>> > > acknowledged?<BR>> > ><BR>> > > 1 0.000000 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<BR>> > <http://2.2.2.2> H.245 terminalCapabilitySet<BR>> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<BR>> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<BR>> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<BR>> > > masterSlaveDetermination masterSlaveDetermination<BR>> > masterSlaveDetermination<BR>> > > masterSlaveDetermination masterSlaveDetermination<BR>> > masterSlaveDetermination<BR>> > > masterSlaveDetermination masterSlaveDetermination<BR>> > masterSlaveDetermination<BR>> > > masterSlaveDetermination masterSlaveDetermination<BR>> > masterSlaveDetermination<BR>> > > masterSlaveDetermination masterSlaveDetermination<BR>> > masterSlaveDetermination<BR>> > > 2 0.000001 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<BR>> > <http://2.2.2.2> H.245 openLogicalChannel<BR>> > > (generic) openLogicalChannel (generic) openLogicalChannel<BR>> > (generic)<BR>> > > openLogicalChannel (generic) openLogicalChannel (generic)<BR>> > openLogicalChannel<BR>> > > (generic) openLogicalChannel (generic) openLogicalChannel<BR>> > (generic)<BR>> > > openLogicalChannel (generic) openLogicalChannel (generic)<BR>> > openLogicalChannel<BR>> > > (generic) openLogicalChannel (generic) openLogicalChannel<BR>> > (generic)<BR>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<BR>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<BR>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<BR>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<BR>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<BR>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<BR>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<BR>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<BR>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<BR>> > > (h263VideoCapability) multiplexEntrySend multiplexEntrySend<BR>> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend<BR>> > multiplexEntrySend<BR>> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend<BR>> > multiplexEntrySend<BR>> > > multiplexEntrySend multiplexEntrySend<BR>> > > 3 0.000002 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<BR>> > <http://2.2.2.2> H.245 multiplexEntrySend<BR>> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend<BR>> > > terminalCapabilitySetAck terminalCapabilitySetAck<BR>> > terminalCapabilitySetAck<BR>> > > terminalCapabilitySetAck terminalCapabilitySetAck<BR>> > terminalCapabilitySetAck<BR>> > > terminalCapabilitySetAck terminalCapabilitySetAck<BR>> > terminalCapabilitySetAck<BR>> > > terminalCapabilitySetAck terminalCapabilitySetAck<BR>> > terminalCapabilitySetAck<BR>> > > terminalCapabilitySetAck terminalCapabilitySetAck<BR>> > terminalCapabilitySetAck<BR>> > > terminalCapabilitySetAck<BR>> > > 4 0.000003 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<BR>> > <http://2.2.2.2> H223<BR>> > > 5 0.000004 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<BR>> > <http://2.2.2.2> H223<BR>> > ><BR>> > > Any pointers on how to debug this would be much appreciated.<BR>> > ><BR>> > > Thanks,<BR>> > > Dan<BR>> > ><BR>> > > PS - This is really great work and I'm very impressed with<BR>> > the project and<BR>> > > hope that I will be able to contribute as well.<BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > <BR>> > <BR>> > _______________________________________________<BR>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> > <BR>> > asterisk-video mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > http://lists.digium.com/mailman/listinfo/asterisk-video<BR>> > <BR>> > <BR>> > <BR>> > <BR>> > <BR>> > ------------------------------------------------------------------------<BR>> > <BR>> > _______________________________________________<BR>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> > <BR>> > asterisk-video mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > http://lists.digium.com/mailman/listinfo/asterisk-video<BR>> <BR>> _______________________________________________<BR>> --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> <BR>> asterisk-video mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-video<BR><BR>
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